diff options
Diffstat (limited to 'thirdparties/android')
127 files changed, 17596 insertions, 0 deletions
diff --git a/thirdparties/android/include/BroadVoice16/bv16/bitpack.h b/thirdparties/android/include/BroadVoice16/bv16/bitpack.h new file mode 100644 index 0000000..032cf42 --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bv16/bitpack.h @@ -0,0 +1,35 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+
+/*****************************************************************************
+ bitpack.h: BV16 bit packing routines
+
+ $Log$
+******************************************************************************/
+
+#ifndef BITPACK_H
+#define BITPACK_H
+
+void BV16_BitPack(UWord8 * PackedStream, struct BV16_Bit_Stream * BitStruct );
+void BV16_BitUnPack(UWord8 * PackedStream, struct BV16_Bit_Stream * BitStruct );
+
+#endif
diff --git a/thirdparties/android/include/BroadVoice16/bv16/bv16.h b/thirdparties/android/include/BroadVoice16/bv16/bv16.h new file mode 100644 index 0000000..df9e5ce --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bv16/bv16.h @@ -0,0 +1,47 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ bv16.h :
+
+ $Log$
+******************************************************************************/
+
+void Reset_BV16_Encoder(
+struct BV16_Encoder_State *cs);
+
+void BV16_Encode(
+struct BV16_Bit_Stream *bs,
+struct BV16_Encoder_State *cs,
+Word16 *inx);
+
+void Reset_BV16_Decoder(
+struct BV16_Decoder_State *ds);
+
+void BV16_Decode(
+struct BV16_Bit_Stream *bs,
+struct BV16_Decoder_State *ds,
+Word16 *xq);
+
+void BV16_PLC(
+struct BV16_Decoder_State *ds,
+Word16 *x);
+
diff --git a/thirdparties/android/include/BroadVoice16/bv16/bv16cnst.h b/thirdparties/android/include/BroadVoice16/bv16/bv16cnst.h new file mode 100644 index 0000000..f6b42b5 --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bv16/bv16cnst.h @@ -0,0 +1,132 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ bv16cnst.h : BV16 constants
+
+ $Log$
+******************************************************************************/
+
+#ifndef BV16CNST_H
+#define BV16CNST_H
+
+/* ----- Basic Codec Parameters ----- */
+#define FRSZ 40 /* frame size */
+#define WINSZ 160 /* lpc analysis WINdow SiZe */
+#define MAXPP 137 /* MAXimum Pitch Period */
+#define MINPP 10 /* MINimum Pitch Period */
+#define PWSZ 120 /* Pitch analysis Window SiZe */
+#define MAXPP1 (MAXPP+1)/* MAXimum Pitch Period + 1 */
+
+/* Quantization parameters */
+#define VDIM 4 /* excitation vector dimension */
+#define CBSZ 16 /* excitation codebook size */
+#define PPCBSZ 32 /* pitch predictor codebook size */
+#define LGPORDER 8 /* Log-Gain Predictor OODER */
+#define LGPECBSZ 16 /* Log-Gain Prediction Error CodeBook SiZe */
+#define LSPPORDER 8 /* LSP MA Predictor ORDER */
+#define LSPECBSZ1 128 /* codebook size of 1st-stage LSP VQ */
+#define LSPECBSZ2 64 /* codebook size of 2nd-stage LSP VQ; 1-bit for sign */
+
+/* Excitation gain quantization */
+#define GPO 8 /* order of MA prediction */
+#define NG 18 /* number of relative gain bins */
+#define GLB -24. /* lower relative gain bound */
+#define NGC 12 /* number of gain change bins */
+#define GCLB -8. /* lower gain change bound */
+#define Minlg 0. /* minimum log-gain */
+#define TMinlg 1. /* minimum linear gain */
+#define LGCBSZ 16 /* size of codebook */
+
+/* Definitions for periodicity to gain scaling mapping */
+#define ScPLCGmin 1639 /* 0.1 Q14 */
+#define ScPLCGmax 14746 /* 0.9 Q14 */
+#define ScPLCG_b -32768 /* -2.0 in Q14 */
+#define ScPLCG_a 31129 /* 1.9 in Q14 */
+#define HoldPLCG 8
+#define AttnPLCG 50
+#define AttnFacPLCG 20971
+
+/* Level Estimation */
+#define estl_alpha 32760
+#define estl_alpha1 32640
+#define estl_beta 32704
+#define estl_beta1 64
+#define estl_a 32640
+#define estl_a1 128
+#define estl_TH 6554
+#define Nfdm 100 /* Max number of frames with fast decay of Lmin */
+
+/* Log-Gain Limitation */
+#define LGLB -24 /* Log-Gain Lower Bound */
+#define LGCLB -8 /* Log-Gain Change Lower Bound */
+#define NGB 18 /* Number of Gain Bins */
+#define NGCB 12 /* Number of Gain Change Bins */
+
+/* Buffer offsets and sizes */
+#define XOFF MAXPP1 /* offset for x() frame */
+#define LX (XOFF+FRSZ) /* Length of x() buffer */
+#define XQOFF (MAXPP1) /* xq() offset before current subframe */
+#define LXQ (XQOFF+FRSZ) /* Length of xq() buffer */
+#define LTMOFF (MAXPP1) /* Long-Term filter Memory OFFset */
+
+/* Long-term postfilter */
+#define DPPQNS 4 /* Delta pitch period for search */
+#define NINT 20 /* length of filter interpolation */
+#define ATHLD1 18022 /* 0.55 Q15 threshold on normalized pitch correlation */
+#define ATHLD2 26214 /* 0.80 Q15 threshold on normalized pitch correlation */
+#define ScLTPF 9830 /* 0.3 Q15 scaling of LTPF coefficient */
+
+/* Coarse pitch search */
+#define MAX_NPEAKS 7
+#define TH1 23921 /* first threshold for cor*cor/energy */
+#define TH2 13107 /* second threshold for cor*cor/energy */
+#define LPTH1 25887 /* Last Pitch cor*cor/energy THreshold 1 */
+#define LPTH2 14090 /* Last Pitch cor*cor/energy THreshold 2 */
+#define MPDTH 2130 /* Multiple Pitch Deviation THreshold */
+#define SMDTH 3113 /* Sub-Multiple pitch Deviation THreshold 0.125 */
+#define MPTH4 9830
+
+/* Decimation parameters */
+#define DECF 4 /* DECimation Factor for coarse pitch period search */
+#define FRSZD (FRSZ/DECF) /* FRame SiZe in DECF:1 lowband domain */
+#define MAXPPD (MAXPP/DECF) /* MAX Pitch in DECF:1, */
+#define MINPPD ((int)(MINPP/DECF)) /* MINimum Pitch Period in DECF:1 */
+#define PWSZD (PWSZ/DECF) /* Pitch ana Window SiZe in DECF:1 domain */
+#define DFO 4
+#define MAXPPD1 (MAXPPD+1)
+#define LXD (MAXPPD1+PWSZD)
+#define XDOFF (LXD-FRSZD)
+#define HMAXPPD (MAXPPD/2)
+#define M1 (MINPPD-1)
+#define M2 MAXPPD1
+#define HDECF (DECF/2)
+
+/* Front-end 150 Hz highpass filter */
+#define HPO 2 /* High-pass filter order */
+
+/* LPC weighting filter */
+#define LTWF 4096 /* 0.5 in Q13 perceptual Weighting Factor Lowband */
+
+/* pole-zero NFC shaping filter */
+#define NSTORDER 8
+
+#endif /* BV16CNST_H */
diff --git a/thirdparties/android/include/BroadVoice16/bv16/bv16externs.h b/thirdparties/android/include/BroadVoice16/bv16/bv16externs.h new file mode 100644 index 0000000..dc61fb5 --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bv16/bv16externs.h @@ -0,0 +1,157 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ bv16externs.c : BV16 Fixed-Point externs
+
+ $Log$
+******************************************************************************/
+
+/* POINTERS */
+extern Word16 winl[WINSZ];
+extern Word16 sstwinl_h[];
+extern Word16 sstwinl_l[];
+extern Word16 gfsz[];
+extern Word16 gfsp[];
+extern Word16 idxord[];
+extern Word16 hpfa[];
+extern Word16 hpfb[];
+extern Word16 adf_h[];
+extern Word16 adf_l[];
+extern Word16 bdf[];
+extern Word16 x[];
+extern Word16 x2[];
+extern Word16 MPTH[];
+
+/* LSP Quantization */
+extern Word16 lspecb1[];
+extern Word16 lspecb2[];
+extern Word16 lspmean[];
+extern Word16 lspp[];
+
+/* Log-Gain Quantization */
+extern Word16 lgpecb[];
+extern Word16 lgp[];
+extern Word16 lgmean;
+
+/* Log-Gain Limitation */
+extern Word16 lgclimit[];
+
+/* Excitation Codebook */
+extern Word16 cccb[];
+
+/* Function Prototypes */
+Word32 estlevel(
+Word32 lg,
+Word32 *level,
+Word32 *lmax,
+Word32 *lmin,
+Word32 *lmean,
+Word32 *x1,
+Word16 ngfae,
+Word16 nggalgc,
+Word16 *estl_alpha_min); /* Q15 */
+
+void excdec_w_synth(
+Word16 *xq, /* (o) Q0 quantized signal vector */
+Word16 *ltsym, /* (i/o) Q16 quantized excitation signal vector */
+Word16 *stsym, /* (i/o) Q0 short-term predictor memory */
+Word16 *idx, /* (o) quantizer codebook index for uq[] vector */
+Word16 *b, /* (i) Q15 coefficient of 3-tap pitch predictor */
+Word16 *cb, /* (i) Q0 codebook */
+Word16 pp, /* pitch period (# of 8 kHz samples) */
+Word16 *aq, /* (i) Q12 short-term predictor coefficients */
+Word16 gain_exp, /* gain_exp of current sub-frame */
+Word32 *EE
+);
+
+Word32 gaindec(
+Word32 *lgq, /* Q25 */
+Word16 gidx,
+Word16 *lgpm, /* Q11 */
+Word32 *prevlg, /* Q25 */
+Word32 level, /* Q25 */
+Word16 *nggalgc,
+Word32 *lg_el);
+
+void gainplc(Word32 E, Word16 *lgeqm, Word32 *lgqm);
+
+void lspdec(
+Word16 *lspq, /* Q15 */
+Word16 *lspidx,
+Word16 *lsppm, /* Q15 */
+Word16 *lspqlast);
+
+void lspdecplc(
+Word16 *lspq, /* Q15 */
+Word16 *lsppm); /* Q15 */
+
+Word16 coarsepitch(
+Word16 *xw, /* (i) Q1 weighted low-band signal frame */
+struct BV16_Encoder_State *c); /* (i/o) coder state */
+
+Word16 refinepitch(
+Word16 *x,
+Word16 cpp,
+Word16 *ppt);
+
+Word16 pitchtapquan(
+Word16 *x,
+Word16 pp,
+Word16 *b,
+Word32 *re);
+
+void excquan(
+Word16 *idx, /* quantizer codebook index for uq[] vector */
+Word16 *s, /* (i) Q0 input signal vector */
+Word16 *aq, /* (i) Q12 noise feedback filter coefficient array */
+Word16 *fsz, /* (i) Q12 short-term noise feedback filter - numerator */
+Word16 *fsp, /* (i) Q12 short-term noise feedback filter - denominator */
+Word16 *b, /* (i) Q15 coefficient of 3-tap pitch predictor */
+Word16 beta, /* (i) Q13 coefficient of pitch feedback filter */
+Word16 *stsym, /* (i/o) Q0 filter memory */
+Word16 *ltsym, /* (i/0) Q0 long-term synthesis filter memory */
+Word16 *ltnfm, /* (i/o) Q0 long-term noise feedback filter memory */
+Word16 *stnfz, /* (i/o) Q0 filter memory */
+Word16 *stnfp, /* (i/o) Q0 filter memory */
+Word16 *cb, /* (i) scalar quantizer codebook - normalized by gain_exp */
+Word16 pp, /* pitch period (# of 8 kHz samples) */
+Word16 gain_exp
+);
+
+Word16 gainquan(
+Word32 *gainq, /* Q18 */
+Word32 *ee, /* Q3 */
+Word16 *lgpm, /* Q11 */
+Word32 *prevlg, /* Q25 */
+Word32 level); /* Q25 */
+
+void lspquan(
+Word16 *lspq,
+Word16 *lspidx,
+Word16 *lsp,
+Word16 *lsppm);
+
+void preprocess(
+struct BV16_Encoder_State *cs,
+Word16 *output, /* (o) Q0 output signal, less factor 1.5 */
+Word16 *input, /* (i) Q0 input signal */
+Word16 N); /* length of signal */
diff --git a/thirdparties/android/include/BroadVoice16/bv16/bv16strct.h b/thirdparties/android/include/BroadVoice16/bv16/bv16strct.h new file mode 100644 index 0000000..ee6895f --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bv16/bv16strct.h @@ -0,0 +1,96 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ bv16strct.h : BV16 data structures
+
+ $Log$
+******************************************************************************/
+
+#ifndef BV16STRCT_H
+#define BV16STRCT_H
+
+struct BV16_Decoder_State {
+Word16 stsym[LPCO];
+Word16 ltsym[LTMOFF];
+Word16 xq[XQOFF];
+Word16 lsppm[LPCO*LSPPORDER];
+Word16 lgpm[LGPORDER];
+Word16 lsplast[LPCO];
+Word32 prevlg[2];
+Word32 lmax;
+Word32 lmin;
+Word32 lmean;
+Word32 x1;
+Word32 level;
+Word16 pp_last;
+Word16 cfecount;
+Word16 ngfae;
+Word16 bq_last[3];
+Word16 nggalgc;
+Word16 estl_alpha_min;
+UWord32 idum;
+Word16 per; /* Q15 */
+Word32 E;
+Word16 atplc[LPCO+1];
+Word16 ma_a;
+Word16 b_prv[2];
+Word16 pp_prv;
+};
+
+struct BV16_Encoder_State {
+Word32 prevlg[2];
+Word32 lmax;
+Word32 lmin;
+Word32 lmean;
+Word32 x1;
+Word32 level;
+Word16 x[XOFF]; /* Signal memory */
+Word16 xwd[XDOFF]; /* Memory of DECF:1 decimated version of xw() */
+Word16 xwd_exp; /* or block floating-point in coarptch.c */
+Word16 dq[XOFF]; /* Q0 - Quantized short-term pred error */
+Word16 dfm_h[DFO]; /* Decimated xwd() filter memory */
+Word16 dfm_l[DFO];
+Word16 stwpm[LPCO]; /* Q0 - Short-term weighting all-pole filter memory */
+Word16 stsym[LPCO]; /* Q0 - Short-term synthesis filter memory */
+Word16 stnfz[NSTORDER]; /* Q0 - Short-term noise feedback filter memory - zero section */
+Word16 stnfp[NSTORDER]; /* Q0 - Short-term noise feedback filter memory - pole section */
+Word16 ltnfm[MAXPP1]; /* Q0 - Long-term noise feedback filter memory */
+Word16 lsplast[LPCO];
+Word16 lsppm[LPCO*LSPPORDER]; /* Q15 - LSP Predictor Memory */
+Word16 lgpm[LGPORDER]; /* Q11 - Log-Gain Predictor Memory */
+Word16 cpplast; /* Pitch period pf the previous frame */
+Word16 hpfzm[HPO];
+Word16 hpfpm[2*HPO];
+Word16 old_A[1+LPCO]; /* Q12 - LPC of previous frame */
+};
+
+struct BV16_Bit_Stream {
+Word16 lspidx[2];
+Word16 ppidx;
+Word16 bqidx;
+Word16 gidx;
+Word16 qvidx[FRSZ/VDIM];
+};
+
+#endif /* BV16STRCT_H */
+
+
diff --git a/thirdparties/android/include/BroadVoice16/bv16/g192.h b/thirdparties/android/include/BroadVoice16/bv16/g192.h new file mode 100644 index 0000000..26e3c8e --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bv16/g192.h @@ -0,0 +1,30 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ g192.h : BV16 G.192
+
+ $Log$
+******************************************************************************/
+
+/* prototypes for G.192 bit-stream functions */
+void bv16_fwrite_g192bitstrm(struct BV16_Bit_Stream *bs, FILE *fo);
+short bv16_fread_g192bitstrm(struct BV16_Bit_Stream *bs, FILE *fi);
diff --git a/thirdparties/android/include/BroadVoice16/bv16/postfilt.h b/thirdparties/android/include/BroadVoice16/bv16/postfilt.h new file mode 100644 index 0000000..76db5e0 --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bv16/postfilt.h @@ -0,0 +1,34 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ postfilt.h : BV16 Post Filter
+
+ $Log$
+******************************************************************************/
+
+void postfilter(
+Word16 *s, /* input : quantized speech signal */
+Word16 pp, /* input : pitch period */
+Word16 *ma_a,
+Word16 *b_prv,
+Word16 *pp_prv,
+Word16 *e); /* output: enhanced speech signal */
diff --git a/thirdparties/android/include/BroadVoice16/bvcommon/basop32.h b/thirdparties/android/include/BroadVoice16/bvcommon/basop32.h new file mode 100644 index 0000000..f3c47d8 --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bvcommon/basop32.h @@ -0,0 +1,22 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+#include "../itug191lib/basop32.h"
+#include "../itug729ilib/oper_32b.h"
diff --git a/thirdparties/android/include/BroadVoice16/bvcommon/bvcommon.h b/thirdparties/android/include/BroadVoice16/bvcommon/bvcommon.h new file mode 100644 index 0000000..b8accf1 --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bvcommon/bvcommon.h @@ -0,0 +1,130 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ bvcommon.h : Common Fixed-Point Library: common #defines and prototypes
+
+ $Log$
+******************************************************************************/
+
+#ifndef BVCOMMON_H
+#define BVCOMMON_H
+
+/* ----- Basic Codec Parameters ----- */
+
+#define LPCO 8 /* LPC Order for 8 kHz sampled lowband signal */
+#define Ngrd 60
+
+#define LSPMIN 49 /* 0.00150 minimum lsp frequency */
+#define LSPMAX 32694 /* 0.99775 maximum lsp frequency */
+#define DLSPMIN 410 /* 0.01250 minimum lsp spacing */
+#define STBLDIM 3 /* dimension of stability enforcement */
+
+/* LPC bandwidth expansion */
+extern Word16 bwel[];
+
+/* LPC to lsp Conversion */
+extern Word16 grid[];
+
+/* LPC WEIGHTING FILTER */
+extern Word16 STWAL[];
+
+/* Coarse Pitch Search */
+extern Word16 invk[];
+
+/* Pitch tap codebook - actually content different for BV16 and BV32 */
+extern Word16 pp9cb[];
+
+/* Function prototypes */
+
+void azfilter(
+ Word16 a[], /* (i) Q12 : prediction coefficients */
+ Word16 m, /* (i) : LPC order */
+ Word16 x[], /* (i) Q0 : input signal samples, incl. past */
+ Word16 y[], /* (o) Q0 : filtered output signal */
+ Word16 lg /* (i) : size of filtering */
+ );
+
+void apfilter(
+ Word16 a[], /* (i) Q12 : prediction coefficients */
+ Word16 m, /* (i) : LPC order */
+ Word16 x[], /* (i) Q0 : input signal */
+ Word16 y[], /* (o) Q0 : output signal */
+ Word16 lg, /* (i) : size of filtering */
+ Word16 mem[], /* (i/o) Q0: filter memory */
+ Word16 update /* (i) : memory update flag */
+ );
+
+void lsp2a(
+Word16 lsp[], /* (i) Q15 : line spectral frequencies */
+Word16 a[]); /* (o) Q12 : predictor coefficients (order = 10) */
+
+void stblz_lsp(
+Word16 *lsp, /* Q15 */
+Word16 order);
+
+Word16 stblchck(
+Word16 *x,
+Word16 vdim);
+
+void a2lsp(
+Word16 a[], /* (i) Q12 : predictor coefficients */
+Word16 lsp[], /* (o) Q15 : line spectral frequencies */
+Word16 old_lsp[]); /* (i) : old lsp[] (in case not found 10 roots) */
+
+void Autocorr(
+ Word32 r[], /* (o) : Autocorrelations */
+ Word16 x[], /* (i) : Input signal */
+ Word16 window[],/* (i) : LPC Analysis window */
+ Word16 l_window,/* (i) : window length */
+ Word16 m); /* (i) : LPC order */
+
+void Spectral_Smoothing(
+Word16 m, /* (i) : LPC order */
+Word32 rl[], /* (i/o) : Autocorrelations lags */
+Word16 lag_h[], /* (i) : SST coefficients (msb) */
+Word16 lag_l[]); /* (i) : SST coefficients (lsb) */
+
+void Levinson(
+ Word32 r32[], /* (i) : r32[] double precision vector of autocorrelation coefficients */
+ Word16 a[], /* (o) : a[] in Q12 - LPC coefficients */
+ Word16 old_a[], /* (i/o): old_a[] in Q12 - previous LPC coefficients */
+ Word16 m); /* (i) : LPC order */
+
+void pp3dec(
+Word16 idx,
+Word16 *b);
+
+void vqdec(
+Word16 *xq,
+Word16 idx,
+Word16 *cb,
+Word16 vdim);
+
+#ifdef _MSC_VER
+#include "math.h"
+static __inline double round(double val)
+{
+ return floor(val + 0.5);
+}
+#endif
+
+#endif /* BVCOMMON_H */
diff --git a/thirdparties/android/include/BroadVoice16/bvcommon/mathutil.h b/thirdparties/android/include/BroadVoice16/bvcommon/mathutil.h new file mode 100644 index 0000000..e54a74c --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bvcommon/mathutil.h @@ -0,0 +1,47 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ fixmath.h : Common Fixed-Point Library:
+
+ $Log$
+******************************************************************************/
+
+Word32 Pow2( /* Q0 output */
+ Word16 int_comp, /* Q0 Integer part */
+ Word16 frac_comp /* Q15 frac_compal part */
+ );
+
+void Log2(
+ Word32 x, /* (i) input */
+ Word16 *int_comp, /* Q0 integer part */
+ Word16 *frac_comp /* Q15 fractional part */
+ );
+
+void sqrt_i(Word16 x_man, Word16 x_exp, Word16 *y_man, Word16 *y_exp);
+Word16 sqrts(Word16 x);
+
+extern Word16 tabsqrt[];
+extern Word16 tablog[];
+extern Word16 tabpow[];
+extern Word16 costable[];
+extern Word16 acosslope[];
+
diff --git a/thirdparties/android/include/BroadVoice16/bvcommon/memutil.h b/thirdparties/android/include/BroadVoice16/bvcommon/memutil.h new file mode 100644 index 0000000..4c67f5f --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bvcommon/memutil.h @@ -0,0 +1,31 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ memutil.h : Common Fixed-Point Library: memory utilities
+
+ $Log$
+******************************************************************************/
+
+Word16 *allocWord16(long nl, long nh);
+void deallocWord16(Word16 *v, long nl, long nh);
+Word32 *allocWord32(long nl, long nh);
+void deallocWord32(Word32 *v, long nl, long nh);
diff --git a/thirdparties/android/include/BroadVoice16/bvcommon/typedef.h b/thirdparties/android/include/BroadVoice16/bvcommon/typedef.h new file mode 100644 index 0000000..68f5e17 --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bvcommon/typedef.h @@ -0,0 +1,36 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ typedef.h : Common Fixed-Point Library:
+
+ $Log$
+******************************************************************************/
+
+#ifndef __TYPEDEF__
+#define __TYPEDEF__
+typedef unsigned char UWord8;
+typedef short Word16;
+typedef int Word32;
+typedef unsigned int UWord32;
+typedef int Flag;
+#endif
+
diff --git a/thirdparties/android/include/BroadVoice16/bvcommon/utility.h b/thirdparties/android/include/BroadVoice16/bvcommon/utility.h new file mode 100644 index 0000000..ea4202e --- /dev/null +++ b/thirdparties/android/include/BroadVoice16/bvcommon/utility.h @@ -0,0 +1,31 @@ +/*****************************************************************************/
+/* BroadVoice(R)16 (BV16) Fixed-Point ANSI-C Source Code */
+/* Revision Date: November 13, 2009 */
+/* Version 1.1 */
+/*****************************************************************************/
+
+/*****************************************************************************/
+/* Copyright 2000-2009 Broadcom Corporation */
+/* */
+/* This software is provided under the GNU Lesser General Public License, */
+/* version 2.1, as published by the Free Software Foundation ("LGPL"). */
+/* This program is distributed in the hope that it will be useful, but */
+/* WITHOUT ANY SUPPORT OR WARRANTY; without even the implied warranty of */
+/* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the LGPL for */
+/* more details. A copy of the LGPL is available at */
+/* http://www.broadcom.com/licenses/LGPLv2.1.php, */
+/* or by writing to the Free Software Foundation, Inc., */
+/* 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */
+/*****************************************************************************/
+
+
+/*****************************************************************************
+ utility.h : Common Fixed-Point Library:
+
+ $Log$
+******************************************************************************/
+
+void W16copy(Word16 *y, Word16 *x, int size);
+void W16zero(Word16 *x, int size);
+void W32copy(Word32 *y, Word32 *x, int size);
+
diff --git a/thirdparties/android/include/g729a/BASIC_OP.H b/thirdparties/android/include/g729a/BASIC_OP.H new file mode 100644 index 0000000..fa88f01 --- /dev/null +++ b/thirdparties/android/include/g729a/BASIC_OP.H @@ -0,0 +1,64 @@ +
+/*___________________________________________________________________________
+ | |
+ | Constants and Globals |
+ |___________________________________________________________________________|
+*/
+extern Flag Overflow;
+extern Flag Carry;
+
+#define MAX_32 (Word32)0x7fffffffL
+#define MIN_32 (Word32)0x80000000L
+
+#define MAX_16 (Word16)0x7fff
+#define MIN_16 (Word16)0x8000
+
+
+/*___________________________________________________________________________
+ | |
+ | Operators prototypes |
+ |___________________________________________________________________________|
+*/
+
+Word16 sature(Word32 L_var1); /* Limit to 16 bits, 1 */
+Word16 add(Word16 var1, Word16 var2); /* Short add, 1 */
+Word16 sub(Word16 var1, Word16 var2); /* Short sub, 1 */
+Word16 abs_s(Word16 var1); /* Short abs, 1 */
+Word16 shl(Word16 var1, Word16 var2); /* Short shift left, 1 */
+Word16 shr(Word16 var1, Word16 var2); /* Short shift right, 1 */
+Word16 mult(Word16 var1, Word16 var2); /* Short mult, 1 */
+Word32 L_mult(Word16 var1, Word16 var2); /* Long mult, 1 */
+Word16 negate(Word16 var1); /* Short negate, 1 */
+Word16 extract_h(Word32 L_var1); /* Extract high, 1 */
+Word16 extract_l(Word32 L_var1); /* Extract low, 1 */
+Word16 round(Word32 L_var1); /* Round, 1 */
+Word32 L_mac(Word32 L_var3, Word16 var1, Word16 var2); /* Mac, 1 */
+Word32 L_msu(Word32 L_var3, Word16 var1, Word16 var2); /* Msu, 1 */
+Word32 L_macNs(Word32 L_var3, Word16 var1, Word16 var2);/* Mac without sat, 1*/
+Word32 L_msuNs(Word32 L_var3, Word16 var1, Word16 var2);/* Msu without sat, 1*/
+
+Word32 L_add(Word32 L_var1, Word32 L_var2); /* Long add, 2 */
+Word32 L_sub(Word32 L_var1, Word32 L_var2); /* Long sub, 2 */
+Word32 L_add_c(Word32 L_var1, Word32 L_var2); /*Long add with c, 2 */
+Word32 L_sub_c(Word32 L_var1, Word32 L_var2); /*Long sub with c, 2 */
+Word32 L_negate(Word32 L_var1); /* Long negate, 2 */
+Word16 mult_r(Word16 var1, Word16 var2); /* Mult with round, 2 */
+Word32 L_shl(Word32 L_var1, Word16 var2); /* Long shift left, 2 */
+Word32 L_shr(Word32 L_var1, Word16 var2); /* Long shift right, 2 */
+Word16 shr_r(Word16 var1, Word16 var2);/* Shift right with round, 2 */
+Word16 mac_r(Word32 L_var3, Word16 var1, Word16 var2);/* Mac with rounding, 2*/
+Word16 msu_r(Word32 L_var3, Word16 var1, Word16 var2);/* Msu with rounding, 2*/
+Word32 L_deposit_h(Word16 var1); /* 16 bit var1 -> MSB, 2 */
+Word32 L_deposit_l(Word16 var1); /* 16 bit var1 -> LSB, 2 */
+
+Word32 L_shr_r(Word32 L_var1, Word16 var2);/* Long shift right with round, 3*/
+Word32 L_abs(Word32 L_var1); /* Long abs, 3 */
+
+Word32 L_sat(Word32 L_var1); /* Long saturation, 4 */
+
+Word16 norm_s(Word16 var1); /* Short norm, 15 */
+
+Word16 div_s(Word16 var1, Word16 var2); /* Short division, 18 */
+
+Word16 norm_l(Word32 L_var1); /* Long norm, 30 */
+
diff --git a/thirdparties/android/include/g729a/LD8A.H b/thirdparties/android/include/g729a/LD8A.H new file mode 100644 index 0000000..be12c34 --- /dev/null +++ b/thirdparties/android/include/g729a/LD8A.H @@ -0,0 +1,623 @@ +/*
+ ITU-T G.729A Speech Coder ANSI-C Source Code
+ Version 1.1 Last modified: September 1996
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke
+ All rights reserved.
+*/
+
+/*---------------------------------------------------------------*
+ * LD8A.H *
+ * ~~~~~~ *
+ * Function prototypes and constants use for G.729A 8kb/s coder. *
+ * *
+ *---------------------------------------------------------------*/
+
+/*--------------------------------------------------------------------------*
+ * Codec constant parameters (coder, decoder, and postfilter) *
+ *--------------------------------------------------------------------------*/
+
+#define L_TOTAL 240 /* Total size of speech buffer. */
+#define L_WINDOW 240 /* Window size in LP analysis. */
+#define L_NEXT 40 /* Lookahead in LP analysis. */
+#define L_FRAME 80 /* Frame size. */
+#define L_SUBFR 40 /* Subframe size. */
+#define M 10 /* Order of LP filter. */
+#define MP1 (M+1) /* Order of LP filter + 1 */
+#define PIT_MIN 20 /* Minimum pitch lag. */
+#define PIT_MAX 143 /* Maximum pitch lag. */
+#define L_INTERPOL (10+1) /* Length of filter for interpolation. */
+#define GAMMA1 24576 /* Bandwitdh factor = 0.75 in Q15 */
+
+#define PRM_SIZE 11 /* Size of vector of analysis parameters. */
+#define SERIAL_SIZE (80+2) /* bfi+ number of speech bits */
+
+#define SHARPMAX 13017 /* Maximum value of pitch sharpening 0.8 Q14 */
+#define SHARPMIN 3277 /* Minimum value of pitch sharpening 0.2 Q14 */
+
+
+/*-------------------------------*
+ * Mathematic functions. *
+ *-------------------------------*/
+
+Word32 Inv_sqrt( /* (o) Q30 : output value (range: 0<=val<1) */
+ Word32 L_x /* (i) Q0 : input value (range: 0<=val<=7fffffff) */
+);
+
+void Log2(
+ Word32 L_x, /* (i) Q0 : input value */
+ Word16 *exponent, /* (o) Q0 : Integer part of Log2. (range: 0<=val<=30) */
+ Word16 *fraction /* (o) Q15: Fractionnal part of Log2. (range: 0<=val<1) */
+);
+
+Word32 Pow2( /* (o) Q0 : result (range: 0<=val<=0x7fffffff) */
+ Word16 exponent, /* (i) Q0 : Integer part. (range: 0<=val<=30) */
+ Word16 fraction /* (i) Q15 : Fractionnal part. (range: 0.0<=val<1.0) */
+);
+
+/*-------------------------------*
+ * Pre and post-process. *
+ *-------------------------------*/
+
+void Init_Pre_Process(void);
+void Init_Post_Process(void);
+
+void Pre_Process(
+ Word16 signal[], /* Input/output signal */
+ Word16 lg /* Length of signal */
+);
+
+void Post_Process(
+ Word16 signal[], /* Input/output signal */
+ Word16 lg /* Length of signal */
+);
+
+/*----------------------------------*
+ * Main coder and decoder functions *
+ *----------------------------------*/
+
+void Init_Coder_ld8a(void);
+
+void Coder_ld8a(
+ Word16 ana[] /* output : Analysis parameters */
+);
+
+void Init_Decod_ld8a(void);
+
+void Decod_ld8a(
+ Word16 parm[], /* (i) : vector of synthesis parameters
+ parm[0] = bad frame indicator (bfi) */
+ Word16 synth[], /* (o) : synthesis speech */
+ Word16 A_t[], /* (o) : decoded LP filter in 2 subframes */
+ Word16 *T2 /* (o) : decoded pitch lag in 2 subframes */
+);
+
+/*-------------------------------*
+ * LPC analysis and filtering. *
+ *-------------------------------*/
+
+void Autocorr(
+ Word16 x[], /* (i) : Input signal */
+ Word16 m, /* (i) : LPC order */
+ Word16 r_h[], /* (o) : Autocorrelations (msb) */
+ Word16 r_l[] /* (o) : Autocorrelations (lsb) */
+);
+
+void Lag_window(
+ Word16 m, /* (i) : LPC order */
+ Word16 r_h[], /* (i/o) : Autocorrelations (msb) */
+ Word16 r_l[] /* (i/o) : Autocorrelations (lsb) */
+);
+
+void Levinson(
+ Word16 Rh[], /* (i) : Rh[m+1] Vector of autocorrelations (msb) */
+ Word16 Rl[], /* (i) : Rl[m+1] Vector of autocorrelations (lsb) */
+ Word16 A[], /* (o) Q12 : A[m] LPC coefficients (m = 10) */
+ Word16 rc[] /* (o) Q15 : rc[M] Relection coefficients. */
+);
+
+void Az_lsp(
+ Word16 a[], /* (i) Q12 : predictor coefficients */
+ Word16 lsp[], /* (o) Q15 : line spectral pairs */
+ Word16 old_lsp[] /* (i) : old lsp[] (in case not found 10 roots) */
+);
+
+void Lsp_Az(
+ Word16 lsp[], /* (i) Q15 : line spectral frequencies */
+ Word16 a[] /* (o) Q12 : predictor coefficients (order = 10) */
+);
+
+void Lsf_lsp(
+ Word16 lsf[], /* (i) Q15 : lsf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 lsp[], /* (o) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Lsp_lsf(
+ Word16 lsp[], /* (i) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 lsf[], /* (o) Q15 : lsf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Int_qlpc(
+ Word16 lsp_old[], /* input : LSP vector of past frame */
+ Word16 lsp_new[], /* input : LSP vector of present frame */
+ Word16 Az[] /* output: interpolated Az() for the 2 subframes */
+);
+
+void Weight_Az(
+ Word16 a[], /* (i) Q12 : a[m+1] LPC coefficients */
+ Word16 gamma, /* (i) Q15 : Spectral expansion factor. */
+ Word16 m, /* (i) : LPC order. */
+ Word16 ap[] /* (o) Q12 : Spectral expanded LPC coefficients */
+);
+
+void Residu(
+ Word16 a[], /* (i) Q12 : prediction coefficients */
+ Word16 x[], /* (i) : speech (values x[-m..-1] are needed (m=10) */
+ Word16 y[], /* (o) : residual signal */
+ Word16 lg /* (i) : size of filtering */
+);
+
+void Syn_filt(
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients (m=10) */
+ Word16 x[], /* (i) : input signal */
+ Word16 y[], /* (o) : output signal */
+ Word16 lg, /* (i) : size of filtering */
+ Word16 mem[], /* (i/o) : memory associated with this filtering. */
+ Word16 update /* (i) : 0=no update, 1=update of memory. */
+);
+
+void Convolve(
+ Word16 x[], /* (i) : input vector */
+ Word16 h[], /* (i) Q12 : impulse response */
+ Word16 y[], /* (o) : output vector */
+ Word16 L /* (i) : vector size */
+);
+
+/*--------------------------------------------------------------------------*
+ * LTP constant parameters *
+ *--------------------------------------------------------------------------*/
+
+#define UP_SAMP 3
+#define L_INTER10 10
+#define FIR_SIZE_SYN (UP_SAMP*L_INTER10+1)
+
+/*-----------------------*
+ * Pitch functions. *
+ *-----------------------*/
+
+Word16 Pitch_ol_fast( /* output: open loop pitch lag */
+ Word16 signal[], /* input : signal used to compute the open loop pitch */
+ /* signal[-pit_max] to signal[-1] should be known */
+ Word16 pit_max, /* input : maximum pitch lag */
+ Word16 L_frame /* input : length of frame to compute pitch */
+);
+
+Word16 Pitch_fr3_fast(/* (o) : pitch period. */
+ Word16 exc[], /* (i) : excitation buffer */
+ Word16 xn[], /* (i) : target vector */
+ Word16 h[], /* (i) Q12 : impulse response of filters. */
+ Word16 L_subfr, /* (i) : Length of subframe */
+ Word16 t0_min, /* (i) : minimum value in the searched range. */
+ Word16 t0_max, /* (i) : maximum value in the searched range. */
+ Word16 i_subfr, /* (i) : indicator for first subframe. */
+ Word16 *pit_frac /* (o) : chosen fraction. */
+);
+
+Word16 G_pitch( /* (o) Q14 : Gain of pitch lag saturated to 1.2 */
+ Word16 xn[], /* (i) : Pitch target. */
+ Word16 y1[], /* (i) : Filtered adaptive codebook. */
+ Word16 g_coeff[], /* (i) : Correlations need for gain quantization. */
+ Word16 L_subfr /* (i) : Length of subframe. */
+);
+
+Word16 Enc_lag3( /* output: Return index of encoding */
+ Word16 T0, /* input : Pitch delay */
+ Word16 T0_frac, /* input : Fractional pitch delay */
+ Word16 *T0_min, /* in/out: Minimum search delay */
+ Word16 *T0_max, /* in/out: Maximum search delay */
+ Word16 pit_min, /* input : Minimum pitch delay */
+ Word16 pit_max, /* input : Maximum pitch delay */
+ Word16 pit_flag /* input : Flag for 1st subframe */
+);
+
+void Dec_lag3( /* output: return integer pitch lag */
+ Word16 index, /* input : received pitch index */
+ Word16 pit_min, /* input : minimum pitch lag */
+ Word16 pit_max, /* input : maximum pitch lag */
+ Word16 i_subfr, /* input : subframe flag */
+ Word16 *T0, /* output: integer part of pitch lag */
+ Word16 *T0_frac /* output: fractional part of pitch lag */
+);
+
+Word16 Interpol_3( /* (o) : interpolated value */
+ Word16 *x, /* (i) : input vector */
+ Word16 frac /* (i) : fraction */
+);
+
+void Pred_lt_3(
+ Word16 exc[], /* in/out: excitation buffer */
+ Word16 T0, /* input : integer pitch lag */
+ Word16 frac, /* input : fraction of lag */
+ Word16 L_subfr /* input : subframe size */
+);
+
+Word16 Parity_Pitch( /* output: parity bit (XOR of 6 MSB bits) */
+ Word16 pitch_index /* input : index for which parity to compute */
+);
+
+Word16 Check_Parity_Pitch( /* output: 0 = no error, 1= error */
+ Word16 pitch_index, /* input : index of parameter */
+ Word16 parity /* input : parity bit */
+);
+
+void Cor_h_X(
+ Word16 h[], /* (i) Q12 :Impulse response of filters */
+ Word16 X[], /* (i) :Target vector */
+ Word16 D[] /* (o) :Correlations between h[] and D[] */
+ /* Normalized to 13 bits */
+);
+
+/*-----------------------*
+ * Innovative codebook. *
+ *-----------------------*/
+
+#define DIM_RR 616 /* size of correlation matrix */
+#define NB_POS 8 /* Number of positions for each pulse */
+#define STEP 5 /* Step betweem position of the same pulse. */
+#define MSIZE 64 /* Size of vectors for cross-correlation between 2 pulses*/
+
+/* The following constants are Q15 fractions.
+ These fractions is used to keep maximum precision on "alp" sum */
+
+#define _1_2 (Word16)(16384)
+#define _1_4 (Word16)( 8192)
+#define _1_8 (Word16)( 4096)
+#define _1_16 (Word16)( 2048)
+
+Word16 ACELP_Code_A( /* (o) :index of pulses positions */
+ Word16 x[], /* (i) :Target vector */
+ Word16 h[], /* (i) Q12 :Inpulse response of filters */
+ Word16 T0, /* (i) :Pitch lag */
+ Word16 pitch_sharp, /* (i) Q14 :Last quantized pitch gain */
+ Word16 code[], /* (o) Q13 :Innovative codebook */
+ Word16 y[], /* (o) Q12 :Filtered innovative codebook */
+ Word16 *sign /* (o) :Signs of 4 pulses */
+);
+
+void Decod_ACELP(
+ Word16 sign, /* (i) : signs of 4 pulses. */
+ Word16 index, /* (i) : Positions of the 4 pulses. */
+ Word16 cod[] /* (o) Q13 : algebraic (fixed) codebook excitation */
+);
+/*--------------------------------------------------------------------------*
+ * LSP constant parameters *
+ *--------------------------------------------------------------------------*/
+
+#define NC 5 /* NC = M/2 */
+#define MA_NP 4 /* MA prediction order for LSP */
+#define MODE 2 /* number of modes for MA prediction */
+#define NC0_B 7 /* number of first stage bits */
+#define NC1_B 5 /* number of second stage bits */
+#define NC0 (1<<NC0_B)
+#define NC1 (1<<NC1_B)
+
+#define L_LIMIT 40 /* Q13:0.005 */
+#define M_LIMIT 25681 /* Q13:3.135 */
+
+#define GAP1 10 /* Q13 */
+#define GAP2 5 /* Q13 */
+#define GAP3 321 /* Q13 */
+#define GRID_POINTS 50
+
+#define PI04 ((Word16)1029) /* Q13 pi*0.04 */
+#define PI92 ((Word16)23677) /* Q13 pi*0.92 */
+#define CONST10 ((Word16)10*(1<<11)) /* Q11 10.0 */
+#define CONST12 ((Word16)19661) /* Q14 1.2 */
+
+/*-------------------------------*
+ * LSP VQ functions. *
+ *-------------------------------*/
+
+void Lsf_lsp2(
+ Word16 lsf[], /* (i) Q13 : lsf[m] (range: 0.0<=val<PI) */
+ Word16 lsp[], /* (o) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Lsp_lsf2(
+ Word16 lsp[], /* (i) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 lsf[], /* (o) Q13 : lsf[m] (range: 0.0<=val<PI) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Qua_lsp(
+ Word16 lsp[], /* (i) Q15 : Unquantized LSP */
+ Word16 lsp_q[], /* (o) Q15 : Quantized LSP */
+ Word16 ana[] /* (o) : indexes */
+);
+
+void Get_wegt(
+ Word16 flsp[], /* Q13 */
+ Word16 wegt[] /* Q11 -> normalized */
+);
+
+void Lsp_encw_reset(
+ void
+);
+
+void Lsp_qua_cs(
+ Word16 flsp_in[M], /* Q13 */
+ Word16 lspq_out[M], /* Q13 */
+ Word16 *code
+);
+
+void Lsp_expand_1(
+ Word16 buf[], /* Q13 */
+ Word16 gap /* Q13 */
+);
+
+void Lsp_expand_2(
+ Word16 buf[], /* Q13 */
+ Word16 gap /* Q13 */
+);
+
+void Lsp_expand_1_2(
+ Word16 buf[], /* Q13 */
+ Word16 gap /* Q13 */
+);
+
+void Lsp_get_quant(
+ Word16 lspcb1[][M], /* Q13 */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 code0,
+ Word16 code1,
+ Word16 code2,
+ Word16 fg[][M], /* Q15 */
+ Word16 freq_prev[][M], /* Q13 */
+ Word16 lspq[], /* Q13 */
+ Word16 fg_sum[] /* Q15 */
+);
+
+void Lsp_get_tdist(
+ Word16 wegt[], /* normalized */
+ Word16 buf[], /* Q13 */
+ Word32 *L_tdist, /* Q27 */
+ Word16 rbuf[], /* Q13 */
+ Word16 fg_sum[] /* Q15 */
+);
+
+void Lsp_last_select(
+ Word32 L_tdist[], /* Q27 */
+ Word16 *mode_index
+);
+
+void Lsp_pre_select(
+ Word16 rbuf[], /* Q13 */
+ Word16 lspcb1[][M], /* Q13 */
+ Word16 *cand
+);
+
+void Lsp_select_1(
+ Word16 rbuf[], /* Q13 */
+ Word16 lspcb1[], /* Q13 */
+ Word16 wegt[], /* normalized */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 *index
+);
+
+void Lsp_select_2(
+ Word16 rbuf[], /* Q13 */
+ Word16 lspcb1[], /* Q13 */
+ Word16 wegt[], /* normalized */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 *index
+);
+
+void Lsp_stability(
+ Word16 buf[] /* Q13 */
+);
+
+void Relspwed(
+ Word16 lsp[], /* Q13 */
+ Word16 wegt[], /* normalized */
+ Word16 lspq[], /* Q13 */
+ Word16 lspcb1[][M], /* Q13 */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 fg[MODE][MA_NP][M], /* Q15 */
+ Word16 freq_prev[MA_NP][M], /* Q13 */
+ Word16 fg_sum[MODE][M], /* Q15 */
+ Word16 fg_sum_inv[MODE][M], /* Q12 */
+ Word16 code_ana[]
+);
+
+void D_lsp(
+ Word16 prm[], /* (i) : indexes of the selected LSP */
+ Word16 lsp_q[], /* (o) Q15 : Quantized LSP parameters */
+ Word16 erase /* (i) : frame erase information */
+);
+
+void Lsp_decw_reset(
+ void
+);
+
+void Lsp_iqua_cs(
+ Word16 prm[], /* input : codes of the selected LSP*/
+ Word16 lsp_q[], /* output: Quantized LSP parameters*/
+ Word16 erase /* input : frame erase information */
+);
+
+void Lsp_prev_compose(
+ Word16 lsp_ele[], /* Q13 */
+ Word16 lsp[], /* Q13 */
+ Word16 fg[][M], /* Q15 */
+ Word16 freq_prev[][M], /* Q13 */
+ Word16 fg_sum[] /* Q15 */
+);
+
+void Lsp_prev_extract(
+ Word16 lsp[M], /* Q13 */
+ Word16 lsp_ele[M], /* Q13 */
+ Word16 fg[MA_NP][M], /* Q15 */
+ Word16 freq_prev[MA_NP][M], /* Q13 */
+ Word16 fg_sum_inv[M] /* Q12 */
+);
+
+void Lsp_prev_update(
+ Word16 lsp_ele[M], /* Q13 */
+ Word16 freq_prev[MA_NP][M] /* Q13 */
+);
+
+/*-------------------------------*
+ * gain VQ constants. *
+ *-------------------------------*/
+
+#define NCODE1_B 3 /* number of Codebook-bit */
+#define NCODE2_B 4 /* number of Codebook-bit */
+#define NCODE1 (1<<NCODE1_B) /* Codebook 1 size */
+#define NCODE2 (1<<NCODE2_B) /* Codebook 2 size */
+#define NCAN1 4 /* Pre-selecting order for #1 */
+#define NCAN2 8 /* Pre-selecting order for #2 */
+#define INV_COEF -17103 /* Q19 */
+
+/*--------------------------------------------------------------------------*
+ * gain VQ functions. *
+ *--------------------------------------------------------------------------*/
+
+Word16 Qua_gain(
+ Word16 code[], /* (i) Q13 : Innovative vector. */
+ Word16 g_coeff[], /* (i) : Correlations <xn y1> -2<y1 y1> */
+ /* <y2,y2>, -2<xn,y2>, 2<y1,y2> */
+ Word16 exp_coeff[],/* (i) : Q-Format g_coeff[] */
+ Word16 L_subfr, /* (i) : Subframe length. */
+ Word16 *gain_pit, /* (o) Q14 : Pitch gain. */
+ Word16 *gain_cod, /* (o) Q1 : Code gain. */
+ Word16 tameflag /* (i) : flag set to 1 if taming is needed */
+);
+
+void Dec_gain(
+ Word16 index, /* (i) : Index of quantization. */
+ Word16 code[], /* (i) Q13 : Innovative vector. */
+ Word16 L_subfr, /* (i) : Subframe length. */
+ Word16 bfi, /* (i) : Bad frame indicator */
+ Word16 *gain_pit, /* (o) Q14 : Pitch gain. */
+ Word16 *gain_cod /* (o) Q1 : Code gain. */
+);
+
+void Gain_predict(
+ Word16 past_qua_en[],/* (i) Q10 :Past quantized energies */
+ Word16 code[], /* (i) Q13 : Innovative vector. */
+ Word16 L_subfr, /* (i) : Subframe length. */
+ Word16 *gcode0, /* (o) Qxx : Predicted codebook gain */
+ Word16 *exp_gcode0 /* (o) : Q-Format(gcode0) */
+);
+
+void Gain_update(
+ Word16 past_qua_en[],/* (i) Q10 :Past quantized energies */
+ Word32 L_gbk12 /* (i) Q13 : gbk1[indice1][1]+gbk2[indice2][1] */
+);
+
+void Gain_update_erasure(
+ Word16 past_qua_en[]/* (i) Q10 :Past quantized energies */
+);
+
+void Corr_xy2(
+ Word16 xn[], /* (i) Q0 :Target vector. */
+ Word16 y1[], /* (i) Q0 :Adaptive codebook. */
+ Word16 y2[], /* (i) Q12 :Filtered innovative vector. */
+ Word16 g_coeff[], /* (o) Q[exp]:Correlations between xn,y1,y2 */
+ Word16 exp_g_coeff[] /* (o) :Q-format of g_coeff[] */
+);
+
+/*-----------------------*
+ * Bitstream function *
+ *-----------------------*/
+
+void prm2bits_ld8k(Word16 prm[], Word16 bits[]);
+void bits2prm_ld8k(Word16 bits[], Word16 prm[]);
+#define BIT_0 (short)0x007f /* definition of zero-bit in bit-stream */
+#define BIT_1 (short)0x0081 /* definition of one-bit in bit-stream */
+#define SYNC_WORD (short)0x6b21 /* definition of frame erasure flag */
+#define SIZE_WORD (short)80 /* number of speech bits */
+
+
+/*-----------------------------------*
+ * Post-filter functions. *
+ *-----------------------------------*/
+
+#define L_H 22 /* size of truncated impulse response of A(z/g1)/A(z/g2) */
+
+#define GAMMAP 16384 /* 0.5 (Q15) */
+#define INV_GAMMAP 21845 /* 1/(1+GAMMAP) (Q15) */
+#define GAMMAP_2 10923 /* GAMMAP/(1+GAMMAP) (Q15) */
+
+#define GAMMA2_PST 18022 /* Formant postfilt factor (numerator) 0.55 Q15 */
+#define GAMMA1_PST 22938 /* Formant postfilt factor (denominator) 0.70 Q15 */
+
+#define MU 26214 /* Factor for tilt compensation filter 0.8 Q15 */
+#define AGC_FAC 29491 /* Factor for automatic gain control 0.9 Q15 */
+#define AGC_FAC1 (Word16)(32767 - AGC_FAC) /* 1-AGC_FAC in Q15 */
+
+
+void Init_Post_Filter(void);
+
+void Post_Filter(
+ Word16 *syn, /* in/out: synthesis speech (postfiltered is output) */
+ Word16 *Az_4, /* input : interpolated LPC parameters in all subframes */
+ Word16 *T /* input : decoded pitch lags in all subframes */
+);
+
+void pit_pst_filt(
+ Word16 *signal, /* (i) : input signal */
+ Word16 *scal_sig, /* (i) : input signal (scaled, divided by 4) */
+ Word16 t0_min, /* (i) : minimum value in the searched range */
+ Word16 t0_max, /* (i) : maximum value in the searched range */
+ Word16 L_subfr, /* (i) : size of filtering */
+ Word16 *signal_pst /* (o) : harmonically postfiltered signal */
+);
+
+void preemphasis(
+ Word16 *signal, /* (i/o) : input signal overwritten by the output */
+ Word16 g, /* (i) Q15 : preemphasis coefficient */
+ Word16 L /* (i) : size of filtering */
+);
+
+void agc(
+ Word16 *sig_in, /* (i) : postfilter input signal */
+ Word16 *sig_out, /* (i/o) : postfilter output signal */
+ Word16 l_trm /* (i) : subframe size */
+);
+
+/*--------------------------------------------------------------------------*
+ * Constants and prototypes for taming procedure. *
+ *--------------------------------------------------------------------------*/
+
+#define GPCLIP 15564 /* Maximum pitch gain if taming is needed Q14*/
+#define GPCLIP2 481 /* Maximum pitch gain if taming is needed Q9 */
+#define GP0999 16383 /* Maximum pitch gain if taming is needed */
+#define L_THRESH_ERR 983040000L /* Error threshold taming 16384. * 60000. */
+
+void Init_exc_err(void);
+void update_exc_err(Word16 gain_pit, Word16 t0);
+Word16 test_err(Word16 t0, Word16 t0_frac);
+
+/*--------------------------------------------------------------------------*
+ * Prototypes for auxiliary functions. *
+ *--------------------------------------------------------------------------*/
+
+void Copy(
+ Word16 x[], /* (i) : input vector */
+ Word16 y[], /* (o) : output vector */
+ Word16 L /* (i) : vector length */
+);
+
+void Set_zero(
+ Word16 x[], /* (o) : vector to clear */
+ Word16 L /* (i) : length of vector */
+);
+
+Word16 Random(void);
+
+
diff --git a/thirdparties/android/include/g729a/OPER_32B.H b/thirdparties/android/include/g729a/OPER_32B.H new file mode 100644 index 0000000..e8f5da1 --- /dev/null +++ b/thirdparties/android/include/g729a/OPER_32B.H @@ -0,0 +1,18 @@ +/*
+ ITU-T G.729A Speech Coder ANSI-C Source Code
+ Version 1.1 Last modified: September 1996
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke
+ All rights reserved.
+*/
+
+/* Double precision operations */
+
+void L_Extract(Word32 L_32, Word16 *hi, Word16 *lo);
+Word32 L_Comp(Word16 hi, Word16 lo);
+Word32 Mpy_32(Word16 hi1, Word16 lo1, Word16 hi2, Word16 lo2);
+Word32 Mpy_32_16(Word16 hi, Word16 lo, Word16 n);
+Word32 Div_32(Word32 L_num, Word16 denom_hi, Word16 denom_lo);
+
+
diff --git a/thirdparties/android/include/g729a/TAB_LD8A.H b/thirdparties/android/include/g729a/TAB_LD8A.H new file mode 100644 index 0000000..930de49 --- /dev/null +++ b/thirdparties/android/include/g729a/TAB_LD8A.H @@ -0,0 +1,45 @@ +/*
+ ITU-T G.729A Speech Coder ANSI-C Source Code
+ Version 1.1 Last modified: September 1996
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke
+ All rights reserved.
+*/
+
+extern Word16 hamwindow[L_WINDOW];
+extern Word16 lag_h[M];
+extern Word16 lag_l[M];
+extern Word16 table[65];
+extern Word16 slope[64];
+extern Word16 table2[64];
+extern Word16 slope_cos[64];
+extern Word16 slope_acos[64];
+extern Word16 lspcb1[NC0][M];
+extern Word16 lspcb2[NC1][M];
+extern Word16 fg[2][MA_NP][M];
+extern Word16 fg_sum[2][M];
+extern Word16 fg_sum_inv[2][M];
+extern Word16 grid[GRID_POINTS+1];
+extern Word16 inter_3l[FIR_SIZE_SYN];
+extern Word16 pred[4];
+extern Word16 gbk1[NCODE1][2];
+extern Word16 gbk2[NCODE2][2];
+extern Word16 map1[NCODE1];
+extern Word16 map2[NCODE2];
+extern Word16 coef[2][2];
+extern Word32 L_coef[2][2];
+extern Word16 thr1[NCODE1-NCAN1];
+extern Word16 thr2[NCODE2-NCAN2];
+extern Word16 imap1[NCODE1];
+extern Word16 imap2[NCODE2];
+extern Word16 b100[3];
+extern Word16 a100[3];
+extern Word16 b140[3];
+extern Word16 a140[3];
+extern Word16 bitsno[PRM_SIZE];
+extern Word16 tabpow[33];
+extern Word16 tablog[33];
+extern Word16 tabsqr[49];
+extern Word16 tab_zone[PIT_MAX+L_INTERPOL-1];
+
diff --git a/thirdparties/android/include/g729a/TYPEDEF.H b/thirdparties/android/include/g729a/TYPEDEF.H new file mode 100644 index 0000000..ca864e5 --- /dev/null +++ b/thirdparties/android/include/g729a/TYPEDEF.H @@ -0,0 +1,36 @@ +/*
+ ITU-T G.729A Speech Coder ANSI-C Source Code
+ Version 1.1 Last modified: September 1996
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke
+ All rights reserved.
+*/
+
+/* WARNING: Make sure that the proper flags are defined for your system */
+
+/*
+ Types definitions
+*/
+
+#if defined(__BORLANDC__) || defined (__WATCOMC__) || defined(_MSC_VER) || defined(__ZTC__) || defined(__HIGHC__) || defined(_TURBOC_)
+typedef long int Word32 ;
+typedef short int Word16 ;
+typedef short int Flag ;
+#elif defined( __sun)
+typedef short Word16;
+typedef long Word32;
+typedef int Flag;
+#elif defined(__unix__) || defined(__unix)
+typedef short Word16;
+typedef int Word32;
+typedef int Flag;
+#elif defined(VMS) || defined(__VMS)
+typedef short Word16;
+typedef long Word32;
+typedef int Flag;
+#else
+#error COMPILER NOT TESTED typedef.h needs to be updated, see readme
+#endif
+
+
diff --git a/thirdparties/android/include/g729b/basic_op.h b/thirdparties/android/include/g729b/basic_op.h new file mode 100644 index 0000000..2bdbac2 --- /dev/null +++ b/thirdparties/android/include/g729b/basic_op.h @@ -0,0 +1,63 @@ +/*___________________________________________________________________________
+ | |
+ | Constants and Globals |
+ |___________________________________________________________________________|
+*/
+extern Flag Overflow;
+extern Flag Carry;
+
+#define MAX_32 (Word32)0x7fffffffL
+#define MIN_32 (Word32)0x80000000L
+
+#define MAX_16 (Word16)0x7fff
+#define MIN_16 (Word16)0x8000
+
+
+/*___________________________________________________________________________
+ | |
+ | Operators prototypes |
+ |___________________________________________________________________________|
+*/
+
+Word16 sature(Word32 L_var1); /* Limit to 16 bits, 1 */
+Word16 add(Word16 var1, Word16 var2); /* Short add, 1 */
+Word16 sub(Word16 var1, Word16 var2); /* Short sub, 1 */
+Word16 abs_s(Word16 var1); /* Short abs, 1 */
+Word16 shl(Word16 var1, Word16 var2); /* Short shift left, 1 */
+Word16 shr(Word16 var1, Word16 var2); /* Short shift right, 1 */
+Word16 mult(Word16 var1, Word16 var2); /* Short mult, 1 */
+Word32 L_mult(Word16 var1, Word16 var2); /* Long mult, 1 */
+Word16 negate(Word16 var1); /* Short negate, 1 */
+Word16 extract_h(Word32 L_var1); /* Extract high, 1 */
+Word16 extract_l(Word32 L_var1); /* Extract low, 1 */
+Word16 round(Word32 L_var1); /* Round, 1 */
+Word32 L_mac(Word32 L_var3, Word16 var1, Word16 var2); /* Mac, 1 */
+Word32 L_msu(Word32 L_var3, Word16 var1, Word16 var2); /* Msu, 1 */
+Word32 L_macNs(Word32 L_var3, Word16 var1, Word16 var2);/* Mac without sat, 1*/
+Word32 L_msuNs(Word32 L_var3, Word16 var1, Word16 var2);/* Msu without sat, 1*/
+
+Word32 L_add(Word32 L_var1, Word32 L_var2); /* Long add, 2 */
+Word32 L_sub(Word32 L_var1, Word32 L_var2); /* Long sub, 2 */
+Word32 L_add_c(Word32 L_var1, Word32 L_var2); /*Long add with c, 2 */
+Word32 L_sub_c(Word32 L_var1, Word32 L_var2); /*Long sub with c, 2 */
+Word32 L_negate(Word32 L_var1); /* Long negate, 2 */
+Word16 mult_r(Word16 var1, Word16 var2); /* Mult with round, 2 */
+Word32 L_shl(Word32 L_var1, Word16 var2); /* Long shift left, 2 */
+Word32 L_shr(Word32 L_var1, Word16 var2); /* Long shift right, 2 */
+Word16 shr_r(Word16 var1, Word16 var2);/* Shift right with round, 2 */
+Word16 mac_r(Word32 L_var3, Word16 var1, Word16 var2);/* Mac with rounding, 2*/
+Word16 msu_r(Word32 L_var3, Word16 var1, Word16 var2);/* Msu with rounding, 2*/
+Word32 L_deposit_h(Word16 var1); /* 16 bit var1 -> MSB, 2 */
+Word32 L_deposit_l(Word16 var1); /* 16 bit var1 -> LSB, 2 */
+
+Word32 L_shr_r(Word32 L_var1, Word16 var2);/* Long shift right with round, 3*/
+Word32 L_abs(Word32 L_var1); /* Long abs, 3 */
+
+Word32 L_sat(Word32 L_var1); /* Long saturation, 4 */
+
+Word16 norm_s(Word16 var1); /* Short norm, 15 */
+
+Word16 div_s(Word16 var1, Word16 var2); /* Short division, 18 */
+
+Word16 norm_l(Word32 L_var1); /* Long norm, 30 */
+
diff --git a/thirdparties/android/include/g729b/dtx.h b/thirdparties/android/include/g729b/dtx.h new file mode 100644 index 0000000..7b15017 --- /dev/null +++ b/thirdparties/android/include/g729b/dtx.h @@ -0,0 +1,99 @@ +/*
+ ITU-T G.729A Annex B ANSI-C Source Code
+ Version 1.3 Last modified: August 1997
+ Copyright (c) 1996, France Telecom, Rockwell International,
+ Universite de Sherbrooke.
+ All rights reserved.
+*/
+
+/*--------------------------------------------------------------------------*
+ * Prototypes for DTX/CNG *
+ *--------------------------------------------------------------------------*/
+
+/* Encoder DTX/CNG functions */
+void Init_Cod_cng(void);
+void Cod_cng(
+ Word16 *exc, /* (i/o) : excitation array */
+ Word16 pastVad, /* (i) : previous VAD decision */
+ Word16 *lsp_old_q, /* (i/o) : previous quantized lsp */
+ Word16 *Aq, /* (o) : set of interpolated LPC coefficients */
+ Word16 *ana, /* (o) : coded SID parameters */
+ Word16 freq_prev[MA_NP][M],
+ /* (i/o) : previous LPS for quantization */
+ Word16 *seed /* (i/o) : random generator seed */
+);
+void Update_cng(
+ Word16 *r_h, /* (i) : MSB of frame autocorrelation */
+ Word16 exp_r, /* (i) : scaling factor associated */
+ Word16 Vad /* (i) : current Vad decision */
+);
+
+/* SID gain Quantization */
+void Qua_Sidgain(
+ Word16 *ener, /* (i) array of energies */
+ Word16 *sh_ener, /* (i) corresponding scaling factors */
+ Word16 nb_ener, /* (i) number of energies or */
+ Word16 *enerq, /* (o) decoded energies in dB */
+ Word16 *idx /* (o) SID gain quantization index */
+);
+
+/* CNG excitation generation */
+void Calc_exc_rand(
+ Word16 cur_gain, /* (i) : target sample gain */
+ Word16 *exc, /* (i/o) : excitation array */
+ Word16 *seed, /* (i) : current Vad decision */
+ Flag flag_cod /* (i) : encoder/decoder flag */
+);
+
+/* SID LSP Quantization */
+void Get_freq_prev(Word16 x[MA_NP][M]);
+void Update_freq_prev(Word16 x[MA_NP][M]);
+void Get_decfreq_prev(Word16 x[MA_NP][M]);
+void Update_decfreq_prev(Word16 x[MA_NP][M]);
+
+/* Decoder CNG generation */
+void Init_Dec_cng(void);
+void Dec_cng(
+ Word16 past_ftyp, /* (i) : past frame type */
+ Word16 sid_sav, /* (i) : energy to recover SID gain */
+ Word16 sh_sid_sav, /* (i) : corresponding scaling factor */
+ Word16 *parm, /* (i) : coded SID parameters */
+ Word16 *exc, /* (i/o) : excitation array */
+ Word16 *lsp_old, /* (i/o) : previous lsp */
+ Word16 *A_t, /* (o) : set of interpolated LPC coefficients */
+ Word16 *seed, /* (i/o) : random generator seed */
+ Word16 freq_prev[MA_NP][M]
+ /* (i/o) : previous LPS for quantization */
+);
+Word16 read_frame(FILE *f_serial, Word16 *parm);
+
+/*--------------------------------------------------------------------------*
+ * Constants for DTX/CNG *
+ *--------------------------------------------------------------------------*/
+
+/* DTX constants */
+#define FLAG_COD (Flag)1
+#define FLAG_DEC (Flag)0
+#define INIT_SEED 11111
+#define FR_SID_MIN 3
+#define NB_SUMACF 3
+#define NB_CURACF 2
+#define NB_GAIN 2
+#define FRAC_THRESH1 4855
+#define FRAC_THRESH2 3161
+#define A_GAIN0 28672
+
+#define SIZ_SUMACF (NB_SUMACF * MP1)
+#define SIZ_ACF (NB_CURACF * MP1)
+#define A_GAIN1 4096 /* 32768L - A_GAIN0 */
+
+#define RATE_8000 80 /* Full rate (8000 bit/s) */
+#define RATE_SID 15 /* SID */
+#define RATE_0 0 /* 0 bit/s rate */
+
+/* CNG excitation generation constant */
+ /* alpha = 0.5 */
+#define FRAC1 19043 /* (sqrt(40)xalpha/2 - 1) * 32768 */
+#define K0 24576 /* (1 - alpha ** 2) in Q15 */
+#define G_MAX 5000
+
diff --git a/thirdparties/android/include/g729b/ld8a.h b/thirdparties/android/include/g729b/ld8a.h new file mode 100644 index 0000000..784d88d --- /dev/null +++ b/thirdparties/android/include/g729b/ld8a.h @@ -0,0 +1,629 @@ +/*
+ ITU-T G.729A Speech Coder with Annex B ANSI-C Source Code
+ Version 1.3 Last modified: August 1997
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke, Lucent Technologies,
+ Rockwell International
+ All rights reserved.
+*/
+
+/*---------------------------------------------------------------*
+ * LD8A.H *
+ * ~~~~~~ *
+ * Function prototypes and constants use for G.729A 8kb/s coder. *
+ * *
+ *---------------------------------------------------------------*/
+
+/*--------------------------------------------------------------------------*
+ * Codec constant parameters (coder, decoder, and postfilter) *
+ *--------------------------------------------------------------------------*/
+
+#define L_TOTAL 240 /* Total size of speech buffer. */
+#define L_WINDOW 240 /* Window size in LP analysis. */
+#define L_NEXT 40 /* Lookahead in LP analysis. */
+#define L_FRAME 80 /* Frame size. */
+#define L_SUBFR 40 /* Subframe size. */
+#define M 10 /* Order of LP filter. */
+#define MP1 (M+1) /* Order of LP filter + 1 */
+#define PIT_MIN 20 /* Minimum pitch lag. */
+#define PIT_MAX 143 /* Maximum pitch lag. */
+#define L_INTERPOL (10+1) /* Length of filter for interpolation. */
+#define GAMMA1 24576 /* Bandwitdh factor = 0.75 in Q15 */
+
+#define PRM_SIZE 11 /* Size of vector of analysis parameters. */
+#define SERIAL_SIZE (80+2) /* bfi+ number of speech bits */
+
+#define SHARPMAX 13017 /* Maximum value of pitch sharpening 0.8 Q14 */
+#define SHARPMIN 3277 /* Minimum value of pitch sharpening 0.2 Q14 */
+
+
+/*-------------------------------*
+ * Mathematic functions. *
+ *-------------------------------*/
+
+Word32 Inv_sqrt( /* (o) Q30 : output value (range: 0<=val<1) */
+ Word32 L_x /* (i) Q0 : input value (range: 0<=val<=7fffffff) */
+);
+
+void Log2(
+ Word32 L_x, /* (i) Q0 : input value */
+ Word16 *exponent, /* (o) Q0 : Integer part of Log2. (range: 0<=val<=30) */
+ Word16 *fraction /* (o) Q15: Fractionnal part of Log2. (range: 0<=val<1) */
+);
+
+Word32 Pow2( /* (o) Q0 : result (range: 0<=val<=0x7fffffff) */
+ Word16 exponent, /* (i) Q0 : Integer part. (range: 0<=val<=30) */
+ Word16 fraction /* (i) Q15 : Fractionnal part. (range: 0.0<=val<1.0) */
+);
+
+/*-------------------------------*
+ * Pre and post-process. *
+ *-------------------------------*/
+
+void Init_Pre_Process(void);
+void Init_Post_Process(void);
+
+void Pre_Process(
+ Word16 signal[], /* Input/output signal */
+ Word16 lg /* Length of signal */
+);
+
+void Post_Process(
+ Word16 signal[], /* Input/output signal */
+ Word16 lg /* Length of signal */
+);
+
+/*----------------------------------*
+ * Main coder and decoder functions *
+ *----------------------------------*/
+
+void Init_Coder_ld8a(void);
+
+void Coder_ld8a(
+ Word16 ana[], /* output : Analysis parameters */
+ Word16 frame,
+ Word16 vad_enable
+);
+
+void Init_Decod_ld8a(void);
+
+void Decod_ld8a(
+ Word16 parm[], /* (i) : vector of synthesis parameters
+ parm[0] = bad frame indicator (bfi) */
+ Word16 synth[], /* (o) : synthesis speech */
+ Word16 A_t[], /* (o) : decoded LP filter in 2 subframes */
+ Word16 *T2, /* (o) : decoded pitch lag in 2 subframes */
+ Word16 *Vad /* (o) : VAD */
+);
+
+/*-------------------------------*
+ * LPC analysis and filtering. *
+ *-------------------------------*/
+
+void Autocorr(
+ Word16 x[], /* (i) : Input signal */
+ Word16 m, /* (i) : LPC order */
+ Word16 r_h[], /* (o) : Autocorrelations (msb) */
+ Word16 r_l[], /* (o) : Autocorrelations (lsb) */
+ Word16 *exp_R0
+);
+
+void Lag_window(
+ Word16 m, /* (i) : LPC order */
+ Word16 r_h[], /* (i/o) : Autocorrelations (msb) */
+ Word16 r_l[] /* (i/o) : Autocorrelations (lsb) */
+);
+
+void Levinson(
+ Word16 Rh[], /* (i) : Rh[m+1] Vector of autocorrelations (msb) */
+ Word16 Rl[], /* (i) : Rl[m+1] Vector of autocorrelations (lsb) */
+ Word16 A[], /* (o) Q12 : A[m] LPC coefficients (m = 10) */
+ Word16 rc[], /* (o) Q15 : rc[M] Relection coefficients. */
+ Word16 *Err /* (o) : Residual energy */
+);
+
+void Az_lsp(
+ Word16 a[], /* (i) Q12 : predictor coefficients */
+ Word16 lsp[], /* (o) Q15 : line spectral pairs */
+ Word16 old_lsp[] /* (i) : old lsp[] (in case not found 10 roots) */
+);
+
+void Lsp_Az(
+ Word16 lsp[], /* (i) Q15 : line spectral frequencies */
+ Word16 a[] /* (o) Q12 : predictor coefficients (order = 10) */
+);
+
+void Lsf_lsp(
+ Word16 lsf[], /* (i) Q15 : lsf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 lsp[], /* (o) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Lsp_lsf(
+ Word16 lsp[], /* (i) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 lsf[], /* (o) Q15 : lsf[m] normalized (range: 0.0<=val<=0.5) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Int_qlpc(
+ Word16 lsp_old[], /* input : LSP vector of past frame */
+ Word16 lsp_new[], /* input : LSP vector of present frame */
+ Word16 Az[] /* output: interpolated Az() for the 2 subframes */
+);
+
+void Weight_Az(
+ Word16 a[], /* (i) Q12 : a[m+1] LPC coefficients */
+ Word16 gamma, /* (i) Q15 : Spectral expansion factor. */
+ Word16 m, /* (i) : LPC order. */
+ Word16 ap[] /* (o) Q12 : Spectral expanded LPC coefficients */
+);
+
+void Residu(
+ Word16 a[], /* (i) Q12 : prediction coefficients */
+ Word16 x[], /* (i) : speech (values x[-m..-1] are needed (m=10) */
+ Word16 y[], /* (o) : residual signal */
+ Word16 lg /* (i) : size of filtering */
+);
+
+void Syn_filt(
+ Word16 a[], /* (i) Q12 : a[m+1] prediction coefficients (m=10) */
+ Word16 x[], /* (i) : input signal */
+ Word16 y[], /* (o) : output signal */
+ Word16 lg, /* (i) : size of filtering */
+ Word16 mem[], /* (i/o) : memory associated with this filtering. */
+ Word16 update /* (i) : 0=no update, 1=update of memory. */
+);
+
+void Convolve(
+ Word16 x[], /* (i) : input vector */
+ Word16 h[], /* (i) Q12 : impulse response */
+ Word16 y[], /* (o) : output vector */
+ Word16 L /* (i) : vector size */
+);
+
+/*--------------------------------------------------------------------------*
+ * LTP constant parameters *
+ *--------------------------------------------------------------------------*/
+
+#define UP_SAMP 3
+#define L_INTER10 10
+#define FIR_SIZE_SYN (UP_SAMP*L_INTER10+1)
+
+/*-----------------------*
+ * Pitch functions. *
+ *-----------------------*/
+
+Word16 Pitch_ol_fast( /* output: open loop pitch lag */
+ Word16 signal[], /* input : signal used to compute the open loop pitch */
+ /* signal[-pit_max] to signal[-1] should be known */
+ Word16 pit_max, /* input : maximum pitch lag */
+ Word16 L_frame /* input : length of frame to compute pitch */
+);
+
+Word16 Pitch_fr3_fast(/* (o) : pitch period. */
+ Word16 exc[], /* (i) : excitation buffer */
+ Word16 xn[], /* (i) : target vector */
+ Word16 h[], /* (i) Q12 : impulse response of filters. */
+ Word16 L_subfr, /* (i) : Length of subframe */
+ Word16 t0_min, /* (i) : minimum value in the searched range. */
+ Word16 t0_max, /* (i) : maximum value in the searched range. */
+ Word16 i_subfr, /* (i) : indicator for first subframe. */
+ Word16 *pit_frac /* (o) : chosen fraction. */
+);
+
+Word16 G_pitch( /* (o) Q14 : Gain of pitch lag saturated to 1.2 */
+ Word16 xn[], /* (i) : Pitch target. */
+ Word16 y1[], /* (i) : Filtered adaptive codebook. */
+ Word16 g_coeff[], /* (i) : Correlations need for gain quantization. */
+ Word16 L_subfr /* (i) : Length of subframe. */
+);
+
+Word16 Enc_lag3( /* output: Return index of encoding */
+ Word16 T0, /* input : Pitch delay */
+ Word16 T0_frac, /* input : Fractional pitch delay */
+ Word16 *T0_min, /* in/out: Minimum search delay */
+ Word16 *T0_max, /* in/out: Maximum search delay */
+ Word16 pit_min, /* input : Minimum pitch delay */
+ Word16 pit_max, /* input : Maximum pitch delay */
+ Word16 pit_flag /* input : Flag for 1st subframe */
+);
+
+void Dec_lag3( /* output: return integer pitch lag */
+ Word16 index, /* input : received pitch index */
+ Word16 pit_min, /* input : minimum pitch lag */
+ Word16 pit_max, /* input : maximum pitch lag */
+ Word16 i_subfr, /* input : subframe flag */
+ Word16 *T0, /* output: integer part of pitch lag */
+ Word16 *T0_frac /* output: fractional part of pitch lag */
+);
+
+Word16 Interpol_3( /* (o) : interpolated value */
+ Word16 *x, /* (i) : input vector */
+ Word16 frac /* (i) : fraction */
+);
+
+void Pred_lt_3(
+ Word16 exc[], /* in/out: excitation buffer */
+ Word16 T0, /* input : integer pitch lag */
+ Word16 frac, /* input : fraction of lag */
+ Word16 L_subfr /* input : subframe size */
+);
+
+Word16 Parity_Pitch( /* output: parity bit (XOR of 6 MSB bits) */
+ Word16 pitch_index /* input : index for which parity to compute */
+);
+
+Word16 Check_Parity_Pitch( /* output: 0 = no error, 1= error */
+ Word16 pitch_index, /* input : index of parameter */
+ Word16 parity /* input : parity bit */
+);
+
+void Cor_h_X(
+ Word16 h[], /* (i) Q12 :Impulse response of filters */
+ Word16 X[], /* (i) :Target vector */
+ Word16 D[] /* (o) :Correlations between h[] and D[] */
+ /* Normalized to 13 bits */
+);
+
+/*-----------------------*
+ * Innovative codebook. *
+ *-----------------------*/
+
+#define DIM_RR 616 /* size of correlation matrix */
+#define NB_POS 8 /* Number of positions for each pulse */
+#define STEP 5 /* Step betweem position of the same pulse. */
+#define MSIZE 64 /* Size of vectors for cross-correlation between 2 pulses*/
+
+/* The following constants are Q15 fractions.
+ These fractions is used to keep maximum precision on "alp" sum */
+
+#define _1_2 (Word16)(16384)
+#define _1_4 (Word16)( 8192)
+#define _1_8 (Word16)( 4096)
+#define _1_16 (Word16)( 2048)
+
+Word16 ACELP_Code_A( /* (o) :index of pulses positions */
+ Word16 x[], /* (i) :Target vector */
+ Word16 h[], /* (i) Q12 :Inpulse response of filters */
+ Word16 T0, /* (i) :Pitch lag */
+ Word16 pitch_sharp, /* (i) Q14 :Last quantized pitch gain */
+ Word16 code[], /* (o) Q13 :Innovative codebook */
+ Word16 y[], /* (o) Q12 :Filtered innovative codebook */
+ Word16 *sign /* (o) :Signs of 4 pulses */
+);
+
+void Decod_ACELP(
+ Word16 sign, /* (i) : signs of 4 pulses. */
+ Word16 index, /* (i) : Positions of the 4 pulses. */
+ Word16 cod[] /* (o) Q13 : algebraic (fixed) codebook excitation */
+);
+/*--------------------------------------------------------------------------*
+ * LSP constant parameters *
+ *--------------------------------------------------------------------------*/
+
+#define NC 5 /* NC = M/2 */
+#define MA_NP 4 /* MA prediction order for LSP */
+#define MODE 2 /* number of modes for MA prediction */
+#define NC0_B 7 /* number of first stage bits */
+#define NC1_B 5 /* number of second stage bits */
+#define NC0 (1<<NC0_B)
+#define NC1 (1<<NC1_B)
+
+#define L_LIMIT 40 /* Q13:0.005 */
+#define M_LIMIT 25681 /* Q13:3.135 */
+
+#define GAP1 10 /* Q13 */
+#define GAP2 5 /* Q13 */
+#define GAP3 321 /* Q13 */
+#define GRID_POINTS 50
+
+#define PI04 ((Word16)1029) /* Q13 pi*0.04 */
+#define PI92 ((Word16)23677) /* Q13 pi*0.92 */
+#define CONST10 ((Word16)10*(1<<11)) /* Q11 10.0 */
+#define CONST12 ((Word16)19661) /* Q14 1.2 */
+
+/*-------------------------------*
+ * LSP VQ functions. *
+ *-------------------------------*/
+
+void Lsf_lsp2(
+ Word16 lsf[], /* (i) Q13 : lsf[m] (range: 0.0<=val<PI) */
+ Word16 lsp[], /* (o) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Lsp_lsf2(
+ Word16 lsp[], /* (i) Q15 : lsp[m] (range: -1<=val<1) */
+ Word16 lsf[], /* (o) Q13 : lsf[m] (range: 0.0<=val<PI) */
+ Word16 m /* (i) : LPC order */
+);
+
+void Qua_lsp(
+ Word16 lsp[], /* (i) Q15 : Unquantized LSP */
+ Word16 lsp_q[], /* (o) Q15 : Quantized LSP */
+ Word16 ana[] /* (o) : indexes */
+);
+
+void Get_wegt(
+ Word16 flsp[], /* Q13 */
+ Word16 wegt[] /* Q11 -> normalized */
+);
+
+void Lsp_encw_reset(
+ void
+);
+
+void Lsp_qua_cs(
+ Word16 flsp_in[M], /* Q13 */
+ Word16 lspq_out[M], /* Q13 */
+ Word16 *code
+);
+
+void Lsp_expand_1(
+ Word16 buf[], /* Q13 */
+ Word16 gap /* Q13 */
+);
+
+void Lsp_expand_2(
+ Word16 buf[], /* Q13 */
+ Word16 gap /* Q13 */
+);
+
+void Lsp_expand_1_2(
+ Word16 buf[], /* Q13 */
+ Word16 gap /* Q13 */
+);
+
+void Lsp_get_quant(
+ Word16 lspcb1[][M], /* Q13 */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 code0,
+ Word16 code1,
+ Word16 code2,
+ Word16 fg[][M], /* Q15 */
+ Word16 freq_prev[][M], /* Q13 */
+ Word16 lspq[], /* Q13 */
+ Word16 fg_sum[] /* Q15 */
+);
+
+void Lsp_get_tdist(
+ Word16 wegt[], /* normalized */
+ Word16 buf[], /* Q13 */
+ Word32 *L_tdist, /* Q27 */
+ Word16 rbuf[], /* Q13 */
+ Word16 fg_sum[] /* Q15 */
+);
+
+void Lsp_last_select(
+ Word32 L_tdist[], /* Q27 */
+ Word16 *mode_index
+);
+
+void Lsp_pre_select(
+ Word16 rbuf[], /* Q13 */
+ Word16 lspcb1[][M], /* Q13 */
+ Word16 *cand
+);
+
+void Lsp_select_1(
+ Word16 rbuf[], /* Q13 */
+ Word16 lspcb1[], /* Q13 */
+ Word16 wegt[], /* normalized */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 *index
+);
+
+void Lsp_select_2(
+ Word16 rbuf[], /* Q13 */
+ Word16 lspcb1[], /* Q13 */
+ Word16 wegt[], /* normalized */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 *index
+);
+
+void Lsp_stability(
+ Word16 buf[] /* Q13 */
+);
+
+void Relspwed(
+ Word16 lsp[], /* Q13 */
+ Word16 wegt[], /* normalized */
+ Word16 lspq[], /* Q13 */
+ Word16 lspcb1[][M], /* Q13 */
+ Word16 lspcb2[][M], /* Q13 */
+ Word16 fg[MODE][MA_NP][M], /* Q15 */
+ Word16 freq_prev[MA_NP][M], /* Q13 */
+ Word16 fg_sum[MODE][M], /* Q15 */
+ Word16 fg_sum_inv[MODE][M], /* Q12 */
+ Word16 code_ana[]
+);
+
+void D_lsp(
+ Word16 prm[], /* (i) : indexes of the selected LSP */
+ Word16 lsp_q[], /* (o) Q15 : Quantized LSP parameters */
+ Word16 erase /* (i) : frame erase information */
+);
+
+void Lsp_decw_reset(
+ void
+);
+
+void Lsp_iqua_cs(
+ Word16 prm[], /* input : codes of the selected LSP*/
+ Word16 lsp_q[], /* output: Quantized LSP parameters*/
+ Word16 erase /* input : frame erase information */
+);
+
+void Lsp_prev_compose(
+ Word16 lsp_ele[], /* Q13 */
+ Word16 lsp[], /* Q13 */
+ Word16 fg[][M], /* Q15 */
+ Word16 freq_prev[][M], /* Q13 */
+ Word16 fg_sum[] /* Q15 */
+);
+
+void Lsp_prev_extract(
+ Word16 lsp[M], /* Q13 */
+ Word16 lsp_ele[M], /* Q13 */
+ Word16 fg[MA_NP][M], /* Q15 */
+ Word16 freq_prev[MA_NP][M], /* Q13 */
+ Word16 fg_sum_inv[M] /* Q12 */
+);
+
+void Lsp_prev_update(
+ Word16 lsp_ele[M], /* Q13 */
+ Word16 freq_prev[MA_NP][M] /* Q13 */
+);
+
+/*-------------------------------*
+ * gain VQ constants. *
+ *-------------------------------*/
+
+#define NCODE1_B 3 /* number of Codebook-bit */
+#define NCODE2_B 4 /* number of Codebook-bit */
+#define NCODE1 (1<<NCODE1_B) /* Codebook 1 size */
+#define NCODE2 (1<<NCODE2_B) /* Codebook 2 size */
+#define NCAN1 4 /* Pre-selecting order for #1 */
+#define NCAN2 8 /* Pre-selecting order for #2 */
+#define INV_COEF -17103 /* Q19 */
+
+/*--------------------------------------------------------------------------*
+ * gain VQ functions. *
+ *--------------------------------------------------------------------------*/
+
+Word16 Qua_gain(
+ Word16 code[], /* (i) Q13 : Innovative vector. */
+ Word16 g_coeff[], /* (i) : Correlations <xn y1> -2<y1 y1> */
+ /* <y2,y2>, -2<xn,y2>, 2<y1,y2> */
+ Word16 exp_coeff[],/* (i) : Q-Format g_coeff[] */
+ Word16 L_subfr, /* (i) : Subframe length. */
+ Word16 *gain_pit, /* (o) Q14 : Pitch gain. */
+ Word16 *gain_cod, /* (o) Q1 : Code gain. */
+ Word16 tameflag /* (i) : flag set to 1 if taming is needed */
+);
+
+void Dec_gain(
+ Word16 index, /* (i) : Index of quantization. */
+ Word16 code[], /* (i) Q13 : Innovative vector. */
+ Word16 L_subfr, /* (i) : Subframe length. */
+ Word16 bfi, /* (i) : Bad frame indicator */
+ Word16 *gain_pit, /* (o) Q14 : Pitch gain. */
+ Word16 *gain_cod /* (o) Q1 : Code gain. */
+);
+
+void Gain_predict(
+ Word16 past_qua_en[],/* (i) Q10 :Past quantized energies */
+ Word16 code[], /* (i) Q13 : Innovative vector. */
+ Word16 L_subfr, /* (i) : Subframe length. */
+ Word16 *gcode0, /* (o) Qxx : Predicted codebook gain */
+ Word16 *exp_gcode0 /* (o) : Q-Format(gcode0) */
+);
+
+void Gain_update(
+ Word16 past_qua_en[],/* (i) Q10 :Past quantized energies */
+ Word32 L_gbk12 /* (i) Q13 : gbk1[indice1][1]+gbk2[indice2][1] */
+);
+
+void Gain_update_erasure(
+ Word16 past_qua_en[]/* (i) Q10 :Past quantized energies */
+);
+
+void Corr_xy2(
+ Word16 xn[], /* (i) Q0 :Target vector. */
+ Word16 y1[], /* (i) Q0 :Adaptive codebook. */
+ Word16 y2[], /* (i) Q12 :Filtered innovative vector. */
+ Word16 g_coeff[], /* (o) Q[exp]:Correlations between xn,y1,y2 */
+ Word16 exp_g_coeff[] /* (o) :Q-format of g_coeff[] */
+);
+
+/*-----------------------*
+ * Bitstream function *
+ *-----------------------*/
+
+void prm2bits_ld8k(Word16 prm[], Word16 bits[]);
+void bits2prm_ld8k(Word16 bits[], Word16 prm[]);
+#define BIT_0 (short)0x007f /* definition of zero-bit in bit-stream */
+#define BIT_1 (short)0x0081 /* definition of one-bit in bit-stream */
+#define SYNC_WORD (short)0x6b21 /* definition of frame erasure flag */
+#define SIZE_WORD (short)80 /* number of speech bits */
+
+
+/*-----------------------------------*
+ * Post-filter functions. *
+ *-----------------------------------*/
+
+#define L_H 22 /* size of truncated impulse response of A(z/g1)/A(z/g2) */
+
+#define GAMMAP 16384 /* 0.5 (Q15) */
+#define INV_GAMMAP 21845 /* 1/(1+GAMMAP) (Q15) */
+#define GAMMAP_2 10923 /* GAMMAP/(1+GAMMAP) (Q15) */
+
+#define GAMMA2_PST 18022 /* Formant postfilt factor (numerator) 0.55 Q15 */
+#define GAMMA1_PST 22938 /* Formant postfilt factor (denominator) 0.70 Q15 */
+
+#define MU 26214 /* Factor for tilt compensation filter 0.8 Q15 */
+#define AGC_FAC 29491 /* Factor for automatic gain control 0.9 Q15 */
+#define AGC_FAC1 (Word16)(32767 - AGC_FAC) /* 1-AGC_FAC in Q15 */
+
+
+void Init_Post_Filter(void);
+
+void Post_Filter(
+ Word16 *syn, /* in/out: synthesis speech (postfiltered is output) */
+ Word16 *Az_4, /* input : interpolated LPC parameters in all subframes */
+ Word16 *T, /* input : decoded pitch lags in all subframes */
+ Word16 Vad
+);
+
+void pit_pst_filt(
+ Word16 *signal, /* (i) : input signal */
+ Word16 *scal_sig, /* (i) : input signal (scaled, divided by 4) */
+ Word16 t0_min, /* (i) : minimum value in the searched range */
+ Word16 t0_max, /* (i) : maximum value in the searched range */
+ Word16 L_subfr, /* (i) : size of filtering */
+ Word16 *signal_pst /* (o) : harmonically postfiltered signal */
+);
+
+void preemphasis(
+ Word16 *signal, /* (i/o) : input signal overwritten by the output */
+ Word16 g, /* (i) Q15 : preemphasis coefficient */
+ Word16 L /* (i) : size of filtering */
+);
+
+void agc(
+ Word16 *sig_in, /* (i) : postfilter input signal */
+ Word16 *sig_out, /* (i/o) : postfilter output signal */
+ Word16 l_trm /* (i) : subframe size */
+);
+
+/*--------------------------------------------------------------------------*
+ * Constants and prototypes for taming procedure. *
+ *--------------------------------------------------------------------------*/
+
+#define GPCLIP 15564 /* Maximum pitch gain if taming is needed Q14*/
+#define GPCLIP2 481 /* Maximum pitch gain if taming is needed Q9 */
+#define GP0999 16383 /* Maximum pitch gain if taming is needed */
+#define L_THRESH_ERR 983040000L /* Error threshold taming 16384. * 60000. */
+
+void Init_exc_err(void);
+void update_exc_err(Word16 gain_pit, Word16 t0);
+Word16 test_err(Word16 t0, Word16 t0_frac);
+
+/*--------------------------------------------------------------------------*
+ * Prototypes for auxiliary functions. *
+ *--------------------------------------------------------------------------*/
+
+void Copy(
+ Word16 x[], /* (i) : input vector */
+ Word16 y[], /* (o) : output vector */
+ Word16 L /* (i) : vector length */
+);
+
+void Set_zero(
+ Word16 x[], /* (o) : vector to clear */
+ Word16 L /* (i) : length of vector */
+);
+
+Word16 Random(Word16 *seed);
+
diff --git a/thirdparties/android/include/g729b/octet.h b/thirdparties/android/include/g729b/octet.h new file mode 100644 index 0000000..a96d173 --- /dev/null +++ b/thirdparties/android/include/g729b/octet.h @@ -0,0 +1,20 @@ +/*
+ ITU-T G.729A Annex B ANSI-C Source Code
+ Version 1.3 Last modified: August 1997
+ Copyright (c) 1996, France Telecom, Rockwell International,
+ Universite de Sherbrooke.
+ All rights reserved.
+*/
+
+/* Definition for Octet Transmission mode */
+/* When Annex B is used for transmission systems that operate on octet boundary,
+ an extra bit (with value zero) will be packed at the end of a SID bit stream.
+ This will change the number of bits in a SID bit stream from 15 bits to
+ 16 bits (i.e., 2 bytes).
+*/
+
+#define OCTET_TX_MODE
+#define RATE_SID_OCTET 16 /* number of bits in Octet Transmission mode */
+
+
+
diff --git a/thirdparties/android/include/g729b/oper_32b.h b/thirdparties/android/include/g729b/oper_32b.h new file mode 100644 index 0000000..dbb8ebe --- /dev/null +++ b/thirdparties/android/include/g729b/oper_32b.h @@ -0,0 +1,17 @@ +/*
+ ITU-T G.729A Speech Coder ANSI-C Source Code
+ Version 1.1 Last modified: September 1996
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke, Lucent Technologies
+ All rights reserved.
+*/
+
+/* Double precision operations */
+
+void L_Extract(Word32 L_32, Word16 *hi, Word16 *lo);
+Word32 L_Comp(Word16 hi, Word16 lo);
+Word32 Mpy_32(Word16 hi1, Word16 lo1, Word16 hi2, Word16 lo2);
+Word32 Mpy_32_16(Word16 hi, Word16 lo, Word16 n);
+Word32 Div_32(Word32 L_num, Word16 denom_hi, Word16 denom_lo);
+
diff --git a/thirdparties/android/include/g729b/sid.h b/thirdparties/android/include/g729b/sid.h new file mode 100644 index 0000000..477e1ea --- /dev/null +++ b/thirdparties/android/include/g729b/sid.h @@ -0,0 +1,23 @@ +/*
+ ITU-T G.729A Annex B ANSI-C Source Code
+ Version 1.3 Last modified: August 1997
+ Copyright (c) 1996, France Telecom, Rockwell International,
+ Universite de Sherbrooke.
+ All rights reserved.
+*/
+
+#define TRUE 1
+#define FALSE 0
+#define sqr(a) ((a)*(a))
+#define R_LSFQ 10
+
+void Init_lsfq_noise(void);
+void lsfq_noise(Word16 *lsp_new, Word16 *lspq,
+ Word16 freq_prev[MA_NP][M], Word16 *idx);
+void sid_lsfq_decode(Word16 *index, Word16 *lspq,
+ Word16 freq_prev[MA_NP][M]);
+
+
+
+
+
diff --git a/thirdparties/android/include/g729b/tab_dtx.h b/thirdparties/android/include/g729b/tab_dtx.h new file mode 100644 index 0000000..08711fc --- /dev/null +++ b/thirdparties/android/include/g729b/tab_dtx.h @@ -0,0 +1,28 @@ +/*
+ ITU-T G.729A Annex B ANSI-C Source Code
+ Version 1.3 Last modified: August 1997
+ Copyright (c) 1996, France Telecom, Rockwell International,
+ Universite de Sherbrooke.
+ All rights reserved.
+*/
+
+/* VAD constants */
+extern Word16 lbf_corr[NP+1];
+extern Word16 shift_fx[33];
+extern Word16 factor_fx[33];
+
+/* SID LSF quantization */
+extern Word16 noise_fg[MODE][MA_NP][M];
+extern Word16 noise_fg_sum[MODE][M];
+extern Word16 noise_fg_sum_inv[MODE][M];
+extern Word16 PtrTab_1[32];
+extern Word16 PtrTab_2[2][16];
+extern Word16 Mp[MODE];
+
+/* SID gain quantization */
+extern Word16 fact[NB_GAIN+1];
+extern Word16 marg[NB_GAIN+1];
+extern Word16 tab_Sidgain[32];
+
+
+
diff --git a/thirdparties/android/include/g729b/tab_ld8a.h b/thirdparties/android/include/g729b/tab_ld8a.h new file mode 100644 index 0000000..c2d9cc6 --- /dev/null +++ b/thirdparties/android/include/g729b/tab_ld8a.h @@ -0,0 +1,52 @@ +/*
+ ITU-T G.729A Speech Coder with Annex B ANSI-C Source Code
+ Version 1.3 Last modified: August 1997
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke, Lucent Technologies,
+ Rockwell International
+ All rights reserved.
+*/
+
+extern Word16 hamwindow[L_WINDOW];
+extern Word16 lag_h[M+2];
+extern Word16 lag_l[M+2];
+extern Word16 table[65];
+extern Word16 slope[64];
+extern Word16 table2[64];
+extern Word16 slope_cos[64];
+extern Word16 slope_acos[64];
+extern Word16 lspcb1[NC0][M];
+extern Word16 lspcb2[NC1][M];
+extern Word16 fg[2][MA_NP][M];
+extern Word16 fg_sum[2][M];
+extern Word16 fg_sum_inv[2][M];
+extern Word16 grid[GRID_POINTS+1];
+extern Word16 freq_prev_reset[M];
+extern Word16 inter_3l[FIR_SIZE_SYN];
+extern Word16 pred[4];
+extern Word16 gbk1[NCODE1][2];
+extern Word16 gbk2[NCODE2][2];
+extern Word16 map1[NCODE1];
+extern Word16 map2[NCODE2];
+extern Word16 coef[2][2];
+extern Word32 L_coef[2][2];
+extern Word16 thr1[NCODE1-NCAN1];
+extern Word16 thr2[NCODE2-NCAN2];
+extern Word16 imap1[NCODE1];
+extern Word16 imap2[NCODE2];
+extern Word16 b100[3];
+extern Word16 a100[3];
+extern Word16 b140[3];
+extern Word16 a140[3];
+extern Word16 bitsno[PRM_SIZE];
+extern Word16 bitsno2[4];
+extern Word16 tabpow[33];
+extern Word16 tablog[33];
+extern Word16 tabsqr[49];
+extern Word16 tab_zone[PIT_MAX+L_INTERPOL-1];
+
+
+
+
+
diff --git a/thirdparties/android/include/g729b/typedef.h b/thirdparties/android/include/g729b/typedef.h new file mode 100644 index 0000000..7408f8f --- /dev/null +++ b/thirdparties/android/include/g729b/typedef.h @@ -0,0 +1,35 @@ +/*
+ ITU-T G.729A Speech Coder ANSI-C Source Code
+ Version 1.1 Last modified: September 1996
+
+ Copyright (c) 1996,
+ AT&T, France Telecom, NTT, Universite de Sherbrooke, Lucent Technologies
+ All rights reserved.
+*/
+
+/* WARNING: Make sure that the proper flags are defined for your system */
+
+/*
+ Types definitions
+*/
+
+#if defined(__BORLANDC__) || defined (__WATCOMC__) || defined(_MSC_VER) || defined(__ZTC__) || defined(__HIGHC__)
+typedef long int Word32 ;
+typedef short int Word16 ;
+typedef short int Flag ;
+#elif defined( __sun)
+typedef short Word16;
+typedef long Word32;
+typedef int Flag;
+#elif defined(__unix__) || defined(__unix)
+typedef short Word16;
+typedef int Word32;
+typedef int Flag;
+#elif defined(VMS) || defined(__VMS)
+typedef short Word16;
+typedef long Word32;
+typedef int Flag;
+#else
+#error COMPILER NOT TESTED typedef.h needs to be updated, see readme
+#endif
+
diff --git a/thirdparties/android/include/g729b/vad.h b/thirdparties/android/include/g729b/vad.h new file mode 100644 index 0000000..c598e9b --- /dev/null +++ b/thirdparties/android/include/g729b/vad.h @@ -0,0 +1,30 @@ +/*
+ ITU-T G.729A Annex B ANSI-C Source Code
+ Version 1.3 Last modified: August 1997
+ Copyright (c) 1996, France Telecom, Rockwell International,
+ Universite de Sherbrooke.
+ All rights reserved.
+*/
+
+#define NP 12 /* Increased LPC order */
+#define NOISE 0
+#define VOICE 1
+#define INIT_FRAME 32
+#define INIT_COUNT 20
+#define ZC_START 120
+#define ZC_END 200
+
+void vad_init(void);
+
+void vad(
+ Word16 rc,
+ Word16 *lsf,
+ Word16 *r_h,
+ Word16 *r_l,
+ Word16 exp_R0,
+ Word16 *sigpp,
+ Word16 frm_count,
+ Word16 prev_marker,
+ Word16 pprev_marker,
+ Word16 *marker);
+
diff --git a/thirdparties/android/include/gsm.h b/thirdparties/android/include/gsm.h new file mode 100644 index 0000000..4714ab6 --- /dev/null +++ b/thirdparties/android/include/gsm.h @@ -0,0 +1,71 @@ +/* + * Copyright 1992 by Jutta Degener and Carsten Bormann, Technische + * Universitaet Berlin. See the accompanying file "COPYRIGHT" for + * details. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE. + */ + +/*$Header: /home/kbs/jutta/src/gsm/gsm-1.0/inc/RCS/gsm.h,v 1.11 1996/07/05 18:02:56 jutta Exp $*/ + +#ifndef GSM_H +#define GSM_H + +#ifdef __cplusplus +# define NeedFunctionPrototypes 1 +#endif + +#if __STDC__ +# define NeedFunctionPrototypes 1 +#endif + +#ifdef _NO_PROTO +# undef NeedFunctionPrototypes +#endif + +#ifdef NeedFunctionPrototypes +# include <stdio.h> /* for FILE * */ +#endif + +#undef GSM_P +#if NeedFunctionPrototypes +# define GSM_P( protos ) protos +#else +# define GSM_P( protos ) ( /* protos */ ) +#endif + +/* + * Interface + */ + +typedef struct gsm_state * gsm; +typedef short gsm_signal; /* signed 16 bit */ +typedef unsigned char gsm_byte; +typedef gsm_byte gsm_frame[33]; /* 33 * 8 bits */ + +#define GSM_MAGIC 0xD /* 13 kbit/s RPE-LTP */ + +#define GSM_PATCHLEVEL 10 +#define GSM_MINOR 0 +#define GSM_MAJOR 1 + +#define GSM_OPT_VERBOSE 1 +#define GSM_OPT_FAST 2 +#define GSM_OPT_LTP_CUT 3 +#define GSM_OPT_WAV49 4 +#define GSM_OPT_FRAME_INDEX 5 +#define GSM_OPT_FRAME_CHAIN 6 + +extern gsm gsm_create GSM_P((void)); +extern void gsm_destroy GSM_P((gsm)); + +extern int gsm_print GSM_P((FILE *, gsm, gsm_byte *)); +extern int gsm_option GSM_P((gsm, int, int *)); + +extern void gsm_encode GSM_P((gsm, gsm_signal *, gsm_byte *)); +extern int gsm_decode GSM_P((gsm, gsm_byte *, gsm_signal *)); + +extern int gsm_explode GSM_P((gsm, gsm_byte *, gsm_signal *)); +extern void gsm_implode GSM_P((gsm, gsm_signal *, gsm_byte *)); + +#undef GSM_P + +#endif /* GSM_H */ diff --git a/thirdparties/android/include/ilbc/FrameClassify.h b/thirdparties/android/include/ilbc/FrameClassify.h new file mode 100644 index 0000000..8445094 --- /dev/null +++ b/thirdparties/android/include/ilbc/FrameClassify.h @@ -0,0 +1,27 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ FrameClassify.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_FRAMECLASSIFY_H
+ #define __iLBC_FRAMECLASSIFY_H
+
+ int FrameClassify( /* index to the max-energy sub-frame */
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i/o) the encoder state structure */
+ float *residual /* (i) lpc residual signal */
+ );
+
+
+
+
+
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/LPCdecode.h b/thirdparties/android/include/ilbc/LPCdecode.h new file mode 100644 index 0000000..7f59984 --- /dev/null +++ b/thirdparties/android/include/ilbc/LPCdecode.h @@ -0,0 +1,52 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ LPC_decode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_LPC_DECODE_H
+ #define __iLBC_LPC_DECODE_H
+
+ void LSFinterpolate2a_dec(
+ float *a, /* (o) lpc coefficients for a sub-frame */
+ float *lsf1, /* (i) first lsf coefficient vector */
+ float *lsf2, /* (i) second lsf coefficient vector */
+ float coef, /* (i) interpolation weight */
+ int length /* (i) length of lsf vectors */
+ );
+
+ void SimplelsfDEQ(
+ float *lsfdeq, /* (o) dequantized lsf coefficients */
+ int *index, /* (i) quantization index */
+ int lpc_n /* (i) number of LPCs */
+ );
+
+ void DecoderInterpolateLSF(
+ float *syntdenum, /* (o) synthesis filter coefficients */
+ float *weightdenum, /* (o) weighting denumerator
+ coefficients */
+ float *lsfdeq, /* (i) dequantized lsf coefficients */
+ int length, /* (i) length of lsf coefficient vector */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i) the decoder state structure */
+ );
+
+ #endif
+
+
+
+
+
+
+
+
+
+
+
+
+
diff --git a/thirdparties/android/include/ilbc/LPCencode.h b/thirdparties/android/include/ilbc/LPCencode.h new file mode 100644 index 0000000..7bee69a --- /dev/null +++ b/thirdparties/android/include/ilbc/LPCencode.h @@ -0,0 +1,27 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ LPCencode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_LPCENCOD_H
+ #define __iLBC_LPCENCOD_H
+
+ void LPCencode(
+ float *syntdenum, /* (i/o) synthesis filter coefficients
+ before/after encoding */
+ float *weightdenum, /* (i/o) weighting denumerator coefficients
+ before/after encoding */
+ int *lsf_index, /* (o) lsf quantization index */
+ float *data, /* (i) lsf coefficients to quantize */
+ iLBC_Enc_Inst_t *iLBCenc_inst
+ /* (i/o) the encoder state structure */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/StateConstructW.h b/thirdparties/android/include/ilbc/StateConstructW.h new file mode 100644 index 0000000..010ab3a --- /dev/null +++ b/thirdparties/android/include/ilbc/StateConstructW.h @@ -0,0 +1,31 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ StateConstructW.h
+
+
+
+
+
+
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_STATECONSTRUCTW_H
+ #define __iLBC_STATECONSTRUCTW_H
+
+ void StateConstructW(
+ int idxForMax, /* (i) 6-bit index for the quantization of
+ max amplitude */
+ int *idxVec, /* (i) vector of quantization indexes */
+ float *syntDenum, /* (i) synthesis filter denumerator */
+ float *out, /* (o) the decoded state vector */
+ int len /* (i) length of a state vector */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/StateSearchW.h b/thirdparties/android/include/ilbc/StateSearchW.h new file mode 100644 index 0000000..f74f8e9 --- /dev/null +++ b/thirdparties/android/include/ilbc/StateSearchW.h @@ -0,0 +1,52 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ StateSearchW.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_STATESEARCHW_H
+ #define __iLBC_STATESEARCHW_H
+
+ void AbsQuantW(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ float *in, /* (i) vector to encode */
+ float *syntDenum, /* (i) denominator of synthesis filter */
+ float *weightDenum, /* (i) denominator of weighting filter */
+ int *out, /* (o) vector of quantizer indexes */
+ int len, /* (i) length of vector to encode and
+ vector of quantizer indexes */
+ int state_first /* (i) position of start state in the
+ 80 vec */
+ );
+
+ void StateSearchW(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) Encoder instance */
+ float *residual,/* (i) target residual vector */
+ float *syntDenum, /* (i) lpc synthesis filter */
+ float *weightDenum, /* (i) weighting filter denuminator */
+ int *idxForMax, /* (o) quantizer index for maximum
+ amplitude */
+ int *idxVec, /* (o) vector of quantization indexes */
+ int len, /* (i) length of all vectors */
+ int state_first /* (i) position of start state in the
+ 80 vec */
+ );
+
+
+ #endif
+
+
+
+
+
+
+
+
+
diff --git a/thirdparties/android/include/ilbc/anaFilter.h b/thirdparties/android/include/ilbc/anaFilter.h new file mode 100644 index 0000000..ae9c4e5 --- /dev/null +++ b/thirdparties/android/include/ilbc/anaFilter.h @@ -0,0 +1,30 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ anaFilter.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ANAFILTER_H
+ #define __iLBC_ANAFILTER_H
+
+ void anaFilter(
+
+
+
+
+
+
+ float *In, /* (i) Signal to be filtered */
+ float *a, /* (i) LP parameters */
+ int len,/* (i) Length of signal */
+ float *Out, /* (o) Filtered signal */
+ float *mem /* (i/o) Filter state */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/constants.h b/thirdparties/android/include/ilbc/constants.h new file mode 100644 index 0000000..459410b --- /dev/null +++ b/thirdparties/android/include/ilbc/constants.h @@ -0,0 +1,81 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_CONSTANTS_H
+ #define __iLBC_CONSTANTS_H
+
+ #include "iLBC_define.h"
+
+
+ /* ULP bit allocation */
+
+
+
+
+
+
+
+ extern const iLBC_ULP_Inst_t ULP_20msTbl;
+ extern const iLBC_ULP_Inst_t ULP_30msTbl;
+
+ /* high pass filters */
+
+ extern float hpi_zero_coefsTbl[];
+ extern float hpi_pole_coefsTbl[];
+ extern float hpo_zero_coefsTbl[];
+ extern float hpo_pole_coefsTbl[];
+
+ /* low pass filters */
+ extern float lpFilt_coefsTbl[];
+
+ /* LPC analysis and quantization */
+
+ extern float lpc_winTbl[];
+ extern float lpc_asymwinTbl[];
+ extern float lpc_lagwinTbl[];
+ extern float lsfCbTbl[];
+ extern float lsfmeanTbl[];
+ extern int dim_lsfCbTbl[];
+ extern int size_lsfCbTbl[];
+ extern float lsf_weightTbl_30ms[];
+ extern float lsf_weightTbl_20ms[];
+
+ /* state quantization tables */
+
+ extern float state_sq3Tbl[];
+ extern float state_frgqTbl[];
+
+ /* gain quantization tables */
+
+ extern float gain_sq3Tbl[];
+ extern float gain_sq4Tbl[];
+ extern float gain_sq5Tbl[];
+
+ /* adaptive codebook definitions */
+
+ extern int search_rangeTbl[5][CB_NSTAGES];
+ extern int memLfTbl[];
+ extern int stMemLTbl;
+ extern float cbfiltersTbl[CB_FILTERLEN];
+
+ /* enhancer definitions */
+
+ extern float polyphaserTbl[];
+ extern float enh_plocsTbl[];
+
+
+
+
+
+
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/createCB.h b/thirdparties/android/include/ilbc/createCB.h new file mode 100644 index 0000000..1f7ac65 --- /dev/null +++ b/thirdparties/android/include/ilbc/createCB.h @@ -0,0 +1,58 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ createCB.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_CREATECB_H
+ #define __iLBC_CREATECB_H
+
+ void filteredCBvecs(
+ float *cbvectors, /* (o) Codebook vector for the
+ higher section */
+
+
+
+
+
+
+ float *mem, /* (i) Buffer to create codebook
+ vectors from */
+ int lMem /* (i) Length of buffer */
+ );
+
+ void searchAugmentedCB(
+ int low, /* (i) Start index for the search */
+ int high, /* (i) End index for the search */
+ int stage, /* (i) Current stage */
+ int startIndex, /* (i) CB index for the first
+ augmented vector */
+ float *target, /* (i) Target vector for encoding */
+ float *buffer, /* (i) Pointer to the end of the
+ buffer for augmented codebook
+ construction */
+ float *max_measure, /* (i/o) Currently maximum measure */
+ int *best_index,/* (o) Currently the best index */
+ float *gain, /* (o) Currently the best gain */
+ float *energy, /* (o) Energy of augmented
+ codebook vectors */
+ float *invenergy/* (o) Inv energy of aug codebook
+ vectors */
+ );
+
+ void createAugmentedVec(
+ int index, /* (i) Index for the aug vector
+ to be created */
+ float *buffer, /* (i) Pointer to the end of the
+ buffer for augmented codebook
+ construction */
+ float *cbVec /* (o) The construced codebook vector */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/doCPLC.h b/thirdparties/android/include/ilbc/doCPLC.h new file mode 100644 index 0000000..0c1d028 --- /dev/null +++ b/thirdparties/android/include/ilbc/doCPLC.h @@ -0,0 +1,28 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ doCPLC.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_DOLPC_H
+ #define __iLBC_DOLPC_H
+
+ void doThePLC(
+ float *PLCresidual, /* (o) concealed residual */
+ float *PLClpc, /* (o) concealed LP parameters */
+ int PLI, /* (i) packet loss indicator
+ 0 - no PL, 1 = PL */
+ float *decresidual, /* (i) decoded residual */
+ float *lpc, /* (i) decoded LPC (only used for no PL) */
+ int inlag, /* (i) pitch lag */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i/o) decoder instance */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/enhancer.h b/thirdparties/android/include/ilbc/enhancer.h new file mode 100644 index 0000000..3632e72 --- /dev/null +++ b/thirdparties/android/include/ilbc/enhancer.h @@ -0,0 +1,36 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ enhancer.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+
+
+
+
+
+ ******************************************************************/
+
+ #ifndef __ENHANCER_H
+ #define __ENHANCER_H
+
+ #include "iLBC_define.h"
+
+ float xCorrCoef(
+ float *target, /* (i) first array */
+ float *regressor, /* (i) second array */
+ int subl /* (i) dimension arrays */
+ );
+
+ int enhancerInterface(
+ float *out, /* (o) the enhanced recidual signal */
+ float *in, /* (i) the recidual signal to enhance */
+ iLBC_Dec_Inst_t *iLBCdec_inst
+ /* (i/o) the decoder state structure */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/filter.h b/thirdparties/android/include/ilbc/filter.h new file mode 100644 index 0000000..c49ddff --- /dev/null +++ b/thirdparties/android/include/ilbc/filter.h @@ -0,0 +1,81 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ filter.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+
+
+
+
+
+
+ #ifndef __iLBC_FILTER_H
+ #define __iLBC_FILTER_H
+
+ void AllPoleFilter(
+ float *InOut, /* (i/o) on entrance InOut[-orderCoef] to
+ InOut[-1] contain the state of the
+ filter (delayed samples). InOut[0] to
+ InOut[lengthInOut-1] contain the filter
+ input, on en exit InOut[-orderCoef] to
+ InOut[-1] is unchanged and InOut[0] to
+ InOut[lengthInOut-1] contain filtered
+ samples */
+ float *Coef,/* (i) filter coefficients, Coef[0] is assumed
+ to be 1.0 */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef /* (i) number of filter coefficients */
+ );
+
+ void AllZeroFilter(
+ float *In, /* (i) In[0] to In[lengthInOut-1] contain
+ filter input samples */
+ float *Coef,/* (i) filter coefficients (Coef[0] is assumed
+ to be 1.0) */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef, /* (i) number of filter coefficients */
+ float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
+ contain the filter state, on exit Out[0]
+ to Out[lengthInOut-1] contain filtered
+ samples */
+ );
+
+ void ZeroPoleFilter(
+ float *In, /* (i) In[0] to In[lengthInOut-1] contain filter
+ input samples In[-orderCoef] to In[-1]
+ contain state of all-zero section */
+ float *ZeroCoef,/* (i) filter coefficients for all-zero
+ section (ZeroCoef[0] is assumed to
+ be 1.0) */
+ float *PoleCoef,/* (i) filter coefficients for all-pole section
+ (ZeroCoef[0] is assumed to be 1.0) */
+ int lengthInOut,/* (i) number of input/output samples */
+ int orderCoef, /* (i) number of filter coefficients */
+ float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
+ contain state of all-pole section. On
+ exit Out[0] to Out[lengthInOut-1]
+ contain filtered samples */
+ );
+
+
+
+
+
+
+
+ void DownSample (
+ float *In, /* (i) input samples */
+ float *Coef, /* (i) filter coefficients */
+ int lengthIn, /* (i) number of input samples */
+ float *state, /* (i) filter state */
+ float *Out /* (o) downsampled output */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/gainquant.h b/thirdparties/android/include/ilbc/gainquant.h new file mode 100644 index 0000000..5abe60a --- /dev/null +++ b/thirdparties/android/include/ilbc/gainquant.h @@ -0,0 +1,29 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ gainquant.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_GAINQUANT_H
+ #define __iLBC_GAINQUANT_H
+
+ float gainquant(/* (o) quantized gain value */
+ float in, /* (i) gain value */
+ float maxIn,/* (i) maximum of gain value */
+ int cblen, /* (i) number of quantization indices */
+ int *index /* (o) quantization index */
+ );
+
+ float gaindequant( /* (o) quantized gain value */
+ int index, /* (i) quantization index */
+ float maxIn,/* (i) maximum of unquantized gain */
+ int cblen /* (i) number of quantization indices */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/getCBvec.h b/thirdparties/android/include/ilbc/getCBvec.h new file mode 100644 index 0000000..a92eb46 --- /dev/null +++ b/thirdparties/android/include/ilbc/getCBvec.h @@ -0,0 +1,24 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ getCBvec.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_GETCBVEC_H
+ #define __iLBC_GETCBVEC_H
+
+ void getCBvec(
+ float *cbvec, /* (o) Constructed codebook vector */
+ float *mem, /* (i) Codebook buffer */
+ int index, /* (i) Codebook index */
+ int lMem, /* (i) Length of codebook buffer */
+ int cbveclen/* (i) Codebook vector length */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/helpfun.h b/thirdparties/android/include/ilbc/helpfun.h new file mode 100644 index 0000000..16b7a2f --- /dev/null +++ b/thirdparties/android/include/ilbc/helpfun.h @@ -0,0 +1,107 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ helpfun.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_HELPFUN_H
+ #define __iLBC_HELPFUN_H
+
+ void autocorr(
+ float *r, /* (o) autocorrelation vector */
+ const float *x, /* (i) data vector */
+ int N, /* (i) length of data vector */
+ int order /* largest lag for calculated
+ autocorrelations */
+ );
+
+ void window(
+ float *z, /* (o) the windowed data */
+ const float *x, /* (i) the original data vector */
+ const float *y, /* (i) the window */
+ int N /* (i) length of all vectors */
+ );
+
+ void levdurb(
+ float *a, /* (o) lpc coefficient vector starting
+ with 1.0 */
+ float *k, /* (o) reflection coefficients */
+ float *r, /* (i) autocorrelation vector */
+ int order /* (i) order of lpc filter */
+ );
+
+ void interpolate(
+
+
+
+
+
+
+ float *out, /* (o) the interpolated vector */
+ float *in1, /* (i) the first vector for the
+ interpolation */
+ float *in2, /* (i) the second vector for the
+ interpolation */
+ float coef, /* (i) interpolation weights */
+ int length /* (i) length of all vectors */
+ );
+
+ void bwexpand(
+ float *out, /* (o) the bandwidth expanded lpc
+ coefficients */
+ float *in, /* (i) the lpc coefficients before bandwidth
+ expansion */
+ float coef, /* (i) the bandwidth expansion factor */
+ int length /* (i) the length of lpc coefficient vectors */
+ );
+
+ void vq(
+ float *Xq, /* (o) the quantized vector */
+ int *index, /* (o) the quantization index */
+ const float *CB,/* (i) the vector quantization codebook */
+ float *X, /* (i) the vector to quantize */
+ int n_cb, /* (i) the number of vectors in the codebook */
+ int dim /* (i) the dimension of all vectors */
+ );
+
+ void SplitVQ(
+ float *qX, /* (o) the quantized vector */
+ int *index, /* (o) a vector of indexes for all vector
+ codebooks in the split */
+ float *X, /* (i) the vector to quantize */
+ const float *CB,/* (i) the quantizer codebook */
+ int nsplit, /* the number of vector splits */
+ const int *dim, /* the dimension of X and qX */
+ const int *cbsize /* the number of vectors in the codebook */
+ );
+
+
+ void sort_sq(
+ float *xq, /* (o) the quantized value */
+ int *index, /* (o) the quantization index */
+ float x, /* (i) the value to quantize */
+ const float *cb,/* (i) the quantization codebook */
+ int cb_size /* (i) the size of the quantization codebook */
+ );
+
+ int LSF_check( /* (o) 1 for stable lsf vectors and 0 for
+
+
+
+
+
+
+ nonstable ones */
+ float *lsf, /* (i) a table of lsf vectors */
+ int dim, /* (i) the dimension of each lsf vector */
+ int NoAn /* (i) the number of lsf vectors in the
+ table */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/hpInput.h b/thirdparties/android/include/ilbc/hpInput.h new file mode 100644 index 0000000..0ba5c1e --- /dev/null +++ b/thirdparties/android/include/ilbc/hpInput.h @@ -0,0 +1,23 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ hpInput.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_HPINPUT_H
+ #define __iLBC_HPINPUT_H
+
+ void hpInput(
+ float *In, /* (i) vector to filter */
+ int len, /* (i) length of vector to filter */
+ float *Out, /* (o) the resulting filtered vector */
+ float *mem /* (i/o) the filter state */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/hpOutput.h b/thirdparties/android/include/ilbc/hpOutput.h new file mode 100644 index 0000000..63c18e9 --- /dev/null +++ b/thirdparties/android/include/ilbc/hpOutput.h @@ -0,0 +1,23 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ hpOutput.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_HPOUTPUT_H
+ #define __iLBC_HPOUTPUT_H
+
+ void hpOutput(
+ float *In, /* (i) vector to filter */
+ int len,/* (i) length of vector to filter */
+ float *Out, /* (o) the resulting filtered vector */
+ float *mem /* (i/o) the filter state */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/iCBConstruct.h b/thirdparties/android/include/ilbc/iCBConstruct.h new file mode 100644 index 0000000..ac5dfa9 --- /dev/null +++ b/thirdparties/android/include/ilbc/iCBConstruct.h @@ -0,0 +1,40 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iCBConstruct.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+
+
+
+
+
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ICBCONSTRUCT_H
+ #define __iLBC_ICBCONSTRUCT_H
+
+ void index_conv_enc(
+ int *index /* (i/o) Codebook indexes */
+ );
+
+ void index_conv_dec(
+ int *index /* (i/o) Codebook indexes */
+ );
+
+ void iCBConstruct(
+ float *decvector, /* (o) Decoded vector */
+ int *index, /* (i) Codebook indices */
+ int *gain_index,/* (i) Gain quantization indices */
+ float *mem, /* (i) Buffer for codevector construction */
+ int lMem, /* (i) Length of buffer */
+ int veclen, /* (i) Length of vector */
+ int nStages /* (i) Number of codebook stages */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/iCBSearch.h b/thirdparties/android/include/ilbc/iCBSearch.h new file mode 100644 index 0000000..1c78689 --- /dev/null +++ b/thirdparties/android/include/ilbc/iCBSearch.h @@ -0,0 +1,37 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iCBSearch.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ICBSEARCH_H
+ #define __iLBC_ICBSEARCH_H
+
+
+
+
+
+
+
+ void iCBSearch(
+ iLBC_Enc_Inst_t *iLBCenc_inst,
+ /* (i) the encoder state structure */
+ int *index, /* (o) Codebook indices */
+ int *gain_index,/* (o) Gain quantization indices */
+ float *intarget,/* (i) Target vector for encoding */
+ float *mem, /* (i) Buffer for codebook construction */
+ int lMem, /* (i) Length of buffer */
+ int lTarget, /* (i) Length of vector */
+ int nStages, /* (i) Number of codebook stages */
+ float *weightDenum, /* (i) weighting filter coefficients */
+ float *weightState, /* (i) weighting filter state */
+ int block /* (i) the sub-block number */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/iLBC_decode.h b/thirdparties/android/include/ilbc/iLBC_decode.h new file mode 100644 index 0000000..3626199 --- /dev/null +++ b/thirdparties/android/include/ilbc/iLBC_decode.h @@ -0,0 +1,42 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_decode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ILBCDECODE_H
+ #define __iLBC_ILBCDECODE_H
+
+ #include "iLBC_define.h"
+
+ short initDecode( /* (o) Number of decoded
+ samples */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
+ int mode, /* (i) frame size mode */
+ int use_enhancer /* (i) 1 to use enhancer
+ 0 to run without
+ enhancer */
+ );
+
+ void iLBC_decode(
+ float *decblock, /* (o) decoded signal block */
+ unsigned char *bytes, /* (i) encoded signal bits */
+ iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
+ structure */
+ int mode /* (i) 0: bad packet, PLC,
+ 1: normal */
+
+
+
+
+
+
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/iLBC_define.h b/thirdparties/android/include/ilbc/iLBC_define.h new file mode 100644 index 0000000..f6a863d --- /dev/null +++ b/thirdparties/android/include/ilbc/iLBC_define.h @@ -0,0 +1,220 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_define.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+ #include <string.h>
+
+ #ifndef __iLBC_ILBCDEFINE_H
+ #define __iLBC_ILBCDEFINE_H
+
+ /* general codec settings */
+
+ #define FS (float)8000.0
+ #define BLOCKL_20MS 160
+ #define BLOCKL_30MS 240
+ #define BLOCKL_MAX 240
+ #define NSUB_20MS 4
+ #define NSUB_30MS 6
+ #define NSUB_MAX 6
+ #define NASUB_20MS 2
+
+
+
+
+
+
+ #define NASUB_30MS 4
+ #define NASUB_MAX 4
+ #define SUBL 40
+ #define STATE_LEN 80
+ #define STATE_SHORT_LEN_30MS 58
+ #define STATE_SHORT_LEN_20MS 57
+
+ /* LPC settings */
+
+ #define LPC_FILTERORDER 10
+ #define LPC_CHIRP_SYNTDENUM (float)0.9025
+ #define LPC_CHIRP_WEIGHTDENUM (float)0.4222
+ #define LPC_LOOKBACK 60
+ #define LPC_N_20MS 1
+ #define LPC_N_30MS 2
+ #define LPC_N_MAX 2
+ #define LPC_ASYMDIFF 20
+ #define LPC_BW (float)60.0
+ #define LPC_WN (float)1.0001
+ #define LSF_NSPLIT 3
+ #define LSF_NUMBER_OF_STEPS 4
+ #define LPC_HALFORDER (LPC_FILTERORDER/2)
+
+ /* cb settings */
+
+ #define CB_NSTAGES 3
+ #define CB_EXPAND 2
+ #define CB_MEML 147
+ #define CB_FILTERLEN 2*4
+ #define CB_HALFFILTERLEN 4
+ #define CB_RESRANGE 34
+ #define CB_MAXGAIN (float)1.3
+
+ /* enhancer */
+
+ #define ENH_BLOCKL 80 /* block length */
+ #define ENH_BLOCKL_HALF (ENH_BLOCKL/2)
+ #define ENH_HL 3 /* 2*ENH_HL+1 is number blocks
+ in said second sequence */
+ #define ENH_SLOP 2 /* max difference estimated and
+ correct pitch period */
+ #define ENH_PLOCSL 20 /* pitch-estimates and pitch-
+ locations buffer length */
+ #define ENH_OVERHANG 2
+ #define ENH_UPS0 4 /* upsampling rate */
+ #define ENH_FL0 3 /* 2*FLO+1 is the length of
+ each filter */
+ #define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0)
+
+
+
+
+
+
+ #define ENH_CORRDIM (2*ENH_SLOP+1)
+ #define ENH_NBLOCKS (BLOCKL_MAX/ENH_BLOCKL)
+ #define ENH_NBLOCKS_EXTRA 5
+ #define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS +
+ ENH_NBLOCKS_EXTRA */
+ #define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL
+ #define ENH_ALPHA0 (float)0.05
+
+ /* Down sampling */
+
+ #define FILTERORDER_DS 7
+ #define DELAY_DS 3
+ #define FACTOR_DS 2
+
+ /* bit stream defs */
+
+ #define NO_OF_BYTES_20MS 38
+ #define NO_OF_BYTES_30MS 50
+ #define NO_OF_WORDS_20MS 19
+ #define NO_OF_WORDS_30MS 25
+ #define STATE_BITS 3
+ #define BYTE_LEN 8
+ #define ULP_CLASSES 3
+
+ /* help parameters */
+
+ #define FLOAT_MAX (float)1.0e37
+ #define EPS (float)2.220446049250313e-016
+ #define PI (float)3.14159265358979323846
+ #define MIN_SAMPLE -32768
+ #define MAX_SAMPLE 32767
+ #define TWO_PI (float)6.283185307
+ #define PI2 (float)0.159154943
+
+ /* type definition encoder instance */
+ typedef struct iLBC_ULP_Inst_t_ {
+ int lsf_bits[6][ULP_CLASSES+2];
+ int start_bits[ULP_CLASSES+2];
+ int startfirst_bits[ULP_CLASSES+2];
+ int scale_bits[ULP_CLASSES+2];
+ int state_bits[ULP_CLASSES+2];
+ int extra_cb_index[CB_NSTAGES][ULP_CLASSES+2];
+ int extra_cb_gain[CB_NSTAGES][ULP_CLASSES+2];
+ int cb_index[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
+ int cb_gain[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
+ } iLBC_ULP_Inst_t;
+
+ /* type definition encoder instance */
+
+
+
+
+
+
+ typedef struct iLBC_Enc_Inst_t_ {
+
+ /* flag for frame size mode */
+ int mode;
+
+ /* basic parameters for different frame sizes */
+ int blockl;
+ int nsub;
+ int nasub;
+ int no_of_bytes, no_of_words;
+ int lpc_n;
+ int state_short_len;
+ const iLBC_ULP_Inst_t *ULP_inst;
+
+ /* analysis filter state */
+ float anaMem[LPC_FILTERORDER];
+
+ /* old lsf parameters for interpolation */
+ float lsfold[LPC_FILTERORDER];
+ float lsfdeqold[LPC_FILTERORDER];
+
+ /* signal buffer for LP analysis */
+ float lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
+
+ /* state of input HP filter */
+ float hpimem[4];
+
+ } iLBC_Enc_Inst_t;
+
+ /* type definition decoder instance */
+ typedef struct iLBC_Dec_Inst_t_ {
+
+ /* flag for frame size mode */
+ int mode;
+
+ /* basic parameters for different frame sizes */
+ int blockl;
+ int nsub;
+ int nasub;
+ int no_of_bytes, no_of_words;
+ int lpc_n;
+ int state_short_len;
+ const iLBC_ULP_Inst_t *ULP_inst;
+
+ /* synthesis filter state */
+ float syntMem[LPC_FILTERORDER];
+
+ /* old LSF for interpolation */
+
+
+
+
+
+
+ float lsfdeqold[LPC_FILTERORDER];
+
+ /* pitch lag estimated in enhancer and used in PLC */
+ int last_lag;
+
+ /* PLC state information */
+ int prevLag, consPLICount, prevPLI, prev_enh_pl;
+ float prevLpc[LPC_FILTERORDER+1];
+ float prevResidual[NSUB_MAX*SUBL];
+ float per;
+ unsigned long seed;
+
+ /* previous synthesis filter parameters */
+ float old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+
+ /* state of output HP filter */
+ float hpomem[4];
+
+ /* enhancer state information */
+ int use_enhancer;
+ float enh_buf[ENH_BUFL];
+ float enh_period[ENH_NBLOCKS_TOT];
+
+ } iLBC_Dec_Inst_t;
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/iLBC_encode.h b/thirdparties/android/include/ilbc/iLBC_encode.h new file mode 100644 index 0000000..b8bcd14 --- /dev/null +++ b/thirdparties/android/include/ilbc/iLBC_encode.h @@ -0,0 +1,39 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBC_encode.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_ILBCENCODE_H
+ #define __iLBC_ILBCENCODE_H
+
+ #include "iLBC_define.h"
+
+ short initEncode( /* (o) Number of bytes
+ encoded */
+ iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
+ int mode /* (i) frame size mode */
+ );
+
+ void iLBC_encode(
+
+ unsigned char *bytes, /* (o) encoded data bits iLBC */
+ float *block, /* (o) speech vector to
+ encode */
+ iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
+ state */
+ );
+
+ #endif
+
+
+
+
+
+
+
diff --git a/thirdparties/android/include/ilbc/lsf.h b/thirdparties/android/include/ilbc/lsf.h new file mode 100644 index 0000000..e220a7b --- /dev/null +++ b/thirdparties/android/include/ilbc/lsf.h @@ -0,0 +1,26 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ lsf.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_LSF_H
+ #define __iLBC_LSF_H
+
+ void a2lsf(
+ float *freq,/* (o) lsf coefficients */
+ float *a /* (i) lpc coefficients */
+ );
+
+ void lsf2a(
+ float *a_coef, /* (o) lpc coefficients */
+ float *freq /* (i) lsf coefficients */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/packing.h b/thirdparties/android/include/ilbc/packing.h new file mode 100644 index 0000000..6472376 --- /dev/null +++ b/thirdparties/android/include/ilbc/packing.h @@ -0,0 +1,68 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ packing.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __PACKING_H
+ #define __PACKING_H
+
+ void packsplit(
+ int *index, /* (i) the value to split */
+ int *firstpart, /* (o) the value specified by most
+ significant bits */
+ int *rest, /* (o) the value specified by least
+ significant bits */
+ int bitno_firstpart, /* (i) number of bits in most
+ significant part */
+ int bitno_total /* (i) number of bits in full range
+ of value */
+ );
+
+ void packcombine(
+ int *index, /* (i/o) the msb value in the
+ combined value out */
+ int rest, /* (i) the lsb value */
+ int bitno_rest /* (i) the number of bits in the
+ lsb part */
+ );
+
+ void dopack(
+ unsigned char **bitstream, /* (i/o) on entrance pointer to
+ place in bitstream to pack
+ new data, on exit pointer
+ to place in bitstream to
+ pack future data */
+ int index, /* (i) the value to pack */
+ int bitno, /* (i) the number of bits that the
+ value will fit within */
+ int *pos /* (i/o) write position in the
+ current byte */
+ );
+
+
+
+
+
+
+ void unpack(
+ unsigned char **bitstream, /* (i/o) on entrance pointer to
+ place in bitstream to
+ unpack new data from, on
+ exit pointer to place in
+ bitstream to unpack future
+ data from */
+ int *index, /* (o) resulting value */
+ int bitno, /* (i) number of bits used to
+ represent the value */
+ int *pos /* (i/o) read position in the
+ current byte */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/ilbc/syntFilter.h b/thirdparties/android/include/ilbc/syntFilter.h new file mode 100644 index 0000000..7b4a9f0 --- /dev/null +++ b/thirdparties/android/include/ilbc/syntFilter.h @@ -0,0 +1,23 @@ + /******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ syntFilter.h
+
+ Copyright (C) The Internet Society (2004).
+ All Rights Reserved.
+
+ ******************************************************************/
+
+ #ifndef __iLBC_SYNTFILTER_H
+ #define __iLBC_SYNTFILTER_H
+
+ void syntFilter(
+ float *Out, /* (i/o) Signal to be filtered */
+ float *a, /* (i) LP parameters */
+ int len, /* (i) Length of signal */
+ float *mem /* (i/o) Filter state */
+ );
+
+ #endif
+
diff --git a/thirdparties/android/include/libavcodec/avcodec.h b/thirdparties/android/include/libavcodec/avcodec.h new file mode 100644 index 0000000..0c3774b --- /dev/null +++ b/thirdparties/android/include/libavcodec/avcodec.h @@ -0,0 +1,4045 @@ +/*
+ * copyright (c) 2001 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_AVCODEC_H
+#define AVCODEC_AVCODEC_H
+
+/**
+ * @file
+ * external API header
+ */
+
+#include <errno.h>
+#include "libavutil/avutil.h"
+
+#define LIBAVCODEC_VERSION_MAJOR 52
+#define LIBAVCODEC_VERSION_MINOR 84
+#define LIBAVCODEC_VERSION_MICRO 2
+
+#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
+ LIBAVCODEC_VERSION_MINOR, \
+ LIBAVCODEC_VERSION_MICRO)
+#define LIBAVCODEC_VERSION AV_VERSION(LIBAVCODEC_VERSION_MAJOR, \
+ LIBAVCODEC_VERSION_MINOR, \
+ LIBAVCODEC_VERSION_MICRO)
+#define LIBAVCODEC_BUILD LIBAVCODEC_VERSION_INT
+
+#define LIBAVCODEC_IDENT "Lavc" AV_STRINGIFY(LIBAVCODEC_VERSION)
+
+#define AV_NOPTS_VALUE INT64_C(0x8000000000000000)
+#define AV_TIME_BASE 1000000
+#define AV_TIME_BASE_Q (AVRational){1, AV_TIME_BASE}
+
+/**
+ * Identify the syntax and semantics of the bitstream.
+ * The principle is roughly:
+ * Two decoders with the same ID can decode the same streams.
+ * Two encoders with the same ID can encode compatible streams.
+ * There may be slight deviations from the principle due to implementation
+ * details.
+ *
+ * If you add a codec ID to this list, add it so that
+ * 1. no value of a existing codec ID changes (that would break ABI),
+ * 2. it is as close as possible to similar codecs.
+ */
+enum CodecID {
+ CODEC_ID_NONE,
+
+ /* video codecs */
+ CODEC_ID_MPEG1VIDEO,
+ CODEC_ID_MPEG2VIDEO, ///< preferred ID for MPEG-1/2 video decoding
+ CODEC_ID_MPEG2VIDEO_XVMC,
+ CODEC_ID_H261,
+ CODEC_ID_H263,
+ CODEC_ID_RV10,
+ CODEC_ID_RV20,
+ CODEC_ID_MJPEG,
+ CODEC_ID_MJPEGB,
+ CODEC_ID_LJPEG,
+ CODEC_ID_SP5X,
+ CODEC_ID_JPEGLS,
+ CODEC_ID_MPEG4,
+ CODEC_ID_RAWVIDEO,
+ CODEC_ID_MSMPEG4V1,
+ CODEC_ID_MSMPEG4V2,
+ CODEC_ID_MSMPEG4V3,
+ CODEC_ID_WMV1,
+ CODEC_ID_WMV2,
+ CODEC_ID_H263P,
+ CODEC_ID_H263I,
+ CODEC_ID_FLV1,
+ CODEC_ID_SVQ1,
+ CODEC_ID_SVQ3,
+ CODEC_ID_DVVIDEO,
+ CODEC_ID_HUFFYUV,
+ CODEC_ID_CYUV,
+ CODEC_ID_H264,
+ CODEC_ID_INDEO3,
+ CODEC_ID_VP3,
+ CODEC_ID_THEORA,
+ CODEC_ID_ASV1,
+ CODEC_ID_ASV2,
+ CODEC_ID_FFV1,
+ CODEC_ID_4XM,
+ CODEC_ID_VCR1,
+ CODEC_ID_CLJR,
+ CODEC_ID_MDEC,
+ CODEC_ID_ROQ,
+ CODEC_ID_INTERPLAY_VIDEO,
+ CODEC_ID_XAN_WC3,
+ CODEC_ID_XAN_WC4,
+ CODEC_ID_RPZA,
+ CODEC_ID_CINEPAK,
+ CODEC_ID_WS_VQA,
+ CODEC_ID_MSRLE,
+ CODEC_ID_MSVIDEO1,
+ CODEC_ID_IDCIN,
+ CODEC_ID_8BPS,
+ CODEC_ID_SMC,
+ CODEC_ID_FLIC,
+ CODEC_ID_TRUEMOTION1,
+ CODEC_ID_VMDVIDEO,
+ CODEC_ID_MSZH,
+ CODEC_ID_ZLIB,
+ CODEC_ID_QTRLE,
+ CODEC_ID_SNOW,
+ CODEC_ID_TSCC,
+ CODEC_ID_ULTI,
+ CODEC_ID_QDRAW,
+ CODEC_ID_VIXL,
+ CODEC_ID_QPEG,
+#if LIBAVCODEC_VERSION_MAJOR < 53
+ CODEC_ID_XVID,
+#endif
+ CODEC_ID_PNG,
+ CODEC_ID_PPM,
+ CODEC_ID_PBM,
+ CODEC_ID_PGM,
+ CODEC_ID_PGMYUV,
+ CODEC_ID_PAM,
+ CODEC_ID_FFVHUFF,
+ CODEC_ID_RV30,
+ CODEC_ID_RV40,
+ CODEC_ID_VC1,
+ CODEC_ID_WMV3,
+ CODEC_ID_LOCO,
+ CODEC_ID_WNV1,
+ CODEC_ID_AASC,
+ CODEC_ID_INDEO2,
+ CODEC_ID_FRAPS,
+ CODEC_ID_TRUEMOTION2,
+ CODEC_ID_BMP,
+ CODEC_ID_CSCD,
+ CODEC_ID_MMVIDEO,
+ CODEC_ID_ZMBV,
+ CODEC_ID_AVS,
+ CODEC_ID_SMACKVIDEO,
+ CODEC_ID_NUV,
+ CODEC_ID_KMVC,
+ CODEC_ID_FLASHSV,
+ CODEC_ID_CAVS,
+ CODEC_ID_JPEG2000,
+ CODEC_ID_VMNC,
+ CODEC_ID_VP5,
+ CODEC_ID_VP6,
+ CODEC_ID_VP6F,
+ CODEC_ID_TARGA,
+ CODEC_ID_DSICINVIDEO,
+ CODEC_ID_TIERTEXSEQVIDEO,
+ CODEC_ID_TIFF,
+ CODEC_ID_GIF,
+ CODEC_ID_FFH264,
+ CODEC_ID_DXA,
+ CODEC_ID_DNXHD,
+ CODEC_ID_THP,
+ CODEC_ID_SGI,
+ CODEC_ID_C93,
+ CODEC_ID_BETHSOFTVID,
+ CODEC_ID_PTX,
+ CODEC_ID_TXD,
+ CODEC_ID_VP6A,
+ CODEC_ID_AMV,
+ CODEC_ID_VB,
+ CODEC_ID_PCX,
+ CODEC_ID_SUNRAST,
+ CODEC_ID_INDEO4,
+ CODEC_ID_INDEO5,
+ CODEC_ID_MIMIC,
+ CODEC_ID_RL2,
+ CODEC_ID_8SVX_EXP,
+ CODEC_ID_8SVX_FIB,
+ CODEC_ID_ESCAPE124,
+ CODEC_ID_DIRAC,
+ CODEC_ID_BFI,
+ CODEC_ID_CMV,
+ CODEC_ID_MOTIONPIXELS,
+ CODEC_ID_TGV,
+ CODEC_ID_TGQ,
+ CODEC_ID_TQI,
+ CODEC_ID_AURA,
+ CODEC_ID_AURA2,
+ CODEC_ID_V210X,
+ CODEC_ID_TMV,
+ CODEC_ID_V210,
+ CODEC_ID_DPX,
+ CODEC_ID_MAD,
+ CODEC_ID_FRWU,
+ CODEC_ID_FLASHSV2,
+ CODEC_ID_CDGRAPHICS,
+ CODEC_ID_R210,
+ CODEC_ID_ANM,
+ CODEC_ID_BINKVIDEO,
+ CODEC_ID_IFF_ILBM,
+ CODEC_ID_IFF_BYTERUN1,
+ CODEC_ID_KGV1,
+ CODEC_ID_YOP,
+ CODEC_ID_VP8,
+ CODEC_ID_PICTOR,
+ CODEC_ID_ANSI,
+
+ /* various PCM "codecs" */
+ CODEC_ID_PCM_S16LE= 0x10000,
+ CODEC_ID_PCM_S16BE,
+ CODEC_ID_PCM_U16LE,
+ CODEC_ID_PCM_U16BE,
+ CODEC_ID_PCM_S8,
+ CODEC_ID_PCM_U8,
+ CODEC_ID_PCM_MULAW,
+ CODEC_ID_PCM_ALAW,
+ CODEC_ID_PCM_S32LE,
+ CODEC_ID_PCM_S32BE,
+ CODEC_ID_PCM_U32LE,
+ CODEC_ID_PCM_U32BE,
+ CODEC_ID_PCM_S24LE,
+ CODEC_ID_PCM_S24BE,
+ CODEC_ID_PCM_U24LE,
+ CODEC_ID_PCM_U24BE,
+ CODEC_ID_PCM_S24DAUD,
+ CODEC_ID_PCM_ZORK,
+ CODEC_ID_PCM_S16LE_PLANAR,
+ CODEC_ID_PCM_DVD,
+ CODEC_ID_PCM_F32BE,
+ CODEC_ID_PCM_F32LE,
+ CODEC_ID_PCM_F64BE,
+ CODEC_ID_PCM_F64LE,
+ CODEC_ID_PCM_BLURAY,
+
+ /* various ADPCM codecs */
+ CODEC_ID_ADPCM_IMA_QT= 0x11000,
+ CODEC_ID_ADPCM_IMA_WAV,
+ CODEC_ID_ADPCM_IMA_DK3,
+ CODEC_ID_ADPCM_IMA_DK4,
+ CODEC_ID_ADPCM_IMA_WS,
+ CODEC_ID_ADPCM_IMA_SMJPEG,
+ CODEC_ID_ADPCM_MS,
+ CODEC_ID_ADPCM_4XM,
+ CODEC_ID_ADPCM_XA,
+ CODEC_ID_ADPCM_ADX,
+ CODEC_ID_ADPCM_EA,
+ CODEC_ID_ADPCM_G726,
+ CODEC_ID_ADPCM_CT,
+ CODEC_ID_ADPCM_SWF,
+ CODEC_ID_ADPCM_YAMAHA,
+ CODEC_ID_ADPCM_SBPRO_4,
+ CODEC_ID_ADPCM_SBPRO_3,
+ CODEC_ID_ADPCM_SBPRO_2,
+ CODEC_ID_ADPCM_THP,
+ CODEC_ID_ADPCM_IMA_AMV,
+ CODEC_ID_ADPCM_EA_R1,
+ CODEC_ID_ADPCM_EA_R3,
+ CODEC_ID_ADPCM_EA_R2,
+ CODEC_ID_ADPCM_IMA_EA_SEAD,
+ CODEC_ID_ADPCM_IMA_EA_EACS,
+ CODEC_ID_ADPCM_EA_XAS,
+ CODEC_ID_ADPCM_EA_MAXIS_XA,
+ CODEC_ID_ADPCM_IMA_ISS,
+
+ /* AMR */
+ CODEC_ID_AMR_NB= 0x12000,
+ CODEC_ID_AMR_WB,
+
+ /* RealAudio codecs*/
+ CODEC_ID_RA_144= 0x13000,
+ CODEC_ID_RA_288,
+
+ /* various DPCM codecs */
+ CODEC_ID_ROQ_DPCM= 0x14000,
+ CODEC_ID_INTERPLAY_DPCM,
+ CODEC_ID_XAN_DPCM,
+ CODEC_ID_SOL_DPCM,
+
+ /* audio codecs */
+ CODEC_ID_MP2= 0x15000,
+ CODEC_ID_MP3, ///< preferred ID for decoding MPEG audio layer 1, 2 or 3
+ CODEC_ID_AAC,
+ CODEC_ID_AC3,
+ CODEC_ID_DTS,
+ CODEC_ID_VORBIS,
+ CODEC_ID_DVAUDIO,
+ CODEC_ID_WMAV1,
+ CODEC_ID_WMAV2,
+ CODEC_ID_MACE3,
+ CODEC_ID_MACE6,
+ CODEC_ID_VMDAUDIO,
+ CODEC_ID_SONIC,
+ CODEC_ID_SONIC_LS,
+ CODEC_ID_FLAC,
+ CODEC_ID_MP3ADU,
+ CODEC_ID_MP3ON4,
+ CODEC_ID_SHORTEN,
+ CODEC_ID_ALAC,
+ CODEC_ID_WESTWOOD_SND1,
+ CODEC_ID_GSM, ///< as in Berlin toast format
+ CODEC_ID_QDM2,
+ CODEC_ID_COOK,
+ CODEC_ID_TRUESPEECH,
+ CODEC_ID_TTA,
+ CODEC_ID_SMACKAUDIO,
+ CODEC_ID_QCELP,
+ CODEC_ID_WAVPACK,
+ CODEC_ID_DSICINAUDIO,
+ CODEC_ID_IMC,
+ CODEC_ID_MUSEPACK7,
+ CODEC_ID_MLP,
+ CODEC_ID_GSM_MS, /* as found in WAV */
+ CODEC_ID_ATRAC3,
+ CODEC_ID_VOXWARE,
+ CODEC_ID_APE,
+ CODEC_ID_NELLYMOSER,
+ CODEC_ID_MUSEPACK8,
+ CODEC_ID_SPEEX,
+ CODEC_ID_WMAVOICE,
+ CODEC_ID_WMAPRO,
+ CODEC_ID_WMALOSSLESS,
+ CODEC_ID_ATRAC3P,
+ CODEC_ID_EAC3,
+ CODEC_ID_SIPR,
+ CODEC_ID_MP1,
+ CODEC_ID_TWINVQ,
+ CODEC_ID_TRUEHD,
+ CODEC_ID_MP4ALS,
+ CODEC_ID_ATRAC1,
+ CODEC_ID_BINKAUDIO_RDFT,
+ CODEC_ID_BINKAUDIO_DCT,
+
+ /* subtitle codecs */
+ CODEC_ID_DVD_SUBTITLE= 0x17000,
+ CODEC_ID_DVB_SUBTITLE,
+ CODEC_ID_TEXT, ///< raw UTF-8 text
+ CODEC_ID_XSUB,
+ CODEC_ID_SSA,
+ CODEC_ID_MOV_TEXT,
+ CODEC_ID_HDMV_PGS_SUBTITLE,
+ CODEC_ID_DVB_TELETEXT,
+ CODEC_ID_SRT,
+
+ /* other specific kind of codecs (generally used for attachments) */
+ CODEC_ID_TTF= 0x18000,
+
+ CODEC_ID_PROBE= 0x19000, ///< codec_id is not known (like CODEC_ID_NONE) but lavf should attempt to identify it
+
+ CODEC_ID_MPEG2TS= 0x20000, /**< _FAKE_ codec to indicate a raw MPEG-2 TS
+ * stream (only used by libavformat) */
+};
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+#define CodecType AVMediaType
+
+#define CODEC_TYPE_UNKNOWN AVMEDIA_TYPE_UNKNOWN
+#define CODEC_TYPE_VIDEO AVMEDIA_TYPE_VIDEO
+#define CODEC_TYPE_AUDIO AVMEDIA_TYPE_AUDIO
+#define CODEC_TYPE_DATA AVMEDIA_TYPE_DATA
+#define CODEC_TYPE_SUBTITLE AVMEDIA_TYPE_SUBTITLE
+#define CODEC_TYPE_ATTACHMENT AVMEDIA_TYPE_ATTACHMENT
+#define CODEC_TYPE_NB AVMEDIA_TYPE_NB
+#endif
+
+/**
+ * all in native-endian format
+ */
+enum SampleFormat {
+ SAMPLE_FMT_NONE = -1,
+ SAMPLE_FMT_U8, ///< unsigned 8 bits
+ SAMPLE_FMT_S16, ///< signed 16 bits
+ SAMPLE_FMT_S32, ///< signed 32 bits
+ SAMPLE_FMT_FLT, ///< float
+ SAMPLE_FMT_DBL, ///< double
+ SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if dynamically linking to libavcodec
+};
+
+/* Audio channel masks */
+#define CH_FRONT_LEFT 0x00000001
+#define CH_FRONT_RIGHT 0x00000002
+#define CH_FRONT_CENTER 0x00000004
+#define CH_LOW_FREQUENCY 0x00000008
+#define CH_BACK_LEFT 0x00000010
+#define CH_BACK_RIGHT 0x00000020
+#define CH_FRONT_LEFT_OF_CENTER 0x00000040
+#define CH_FRONT_RIGHT_OF_CENTER 0x00000080
+#define CH_BACK_CENTER 0x00000100
+#define CH_SIDE_LEFT 0x00000200
+#define CH_SIDE_RIGHT 0x00000400
+#define CH_TOP_CENTER 0x00000800
+#define CH_TOP_FRONT_LEFT 0x00001000
+#define CH_TOP_FRONT_CENTER 0x00002000
+#define CH_TOP_FRONT_RIGHT 0x00004000
+#define CH_TOP_BACK_LEFT 0x00008000
+#define CH_TOP_BACK_CENTER 0x00010000
+#define CH_TOP_BACK_RIGHT 0x00020000
+#define CH_STEREO_LEFT 0x20000000 ///< Stereo downmix.
+#define CH_STEREO_RIGHT 0x40000000 ///< See CH_STEREO_LEFT.
+
+/** Channel mask value used for AVCodecContext.request_channel_layout
+ to indicate that the user requests the channel order of the decoder output
+ to be the native codec channel order. */
+#define CH_LAYOUT_NATIVE 0x8000000000000000LL
+
+/* Audio channel convenience macros */
+#define CH_LAYOUT_MONO (CH_FRONT_CENTER)
+#define CH_LAYOUT_STEREO (CH_FRONT_LEFT|CH_FRONT_RIGHT)
+#define CH_LAYOUT_2_1 (CH_LAYOUT_STEREO|CH_BACK_CENTER)
+#define CH_LAYOUT_SURROUND (CH_LAYOUT_STEREO|CH_FRONT_CENTER)
+#define CH_LAYOUT_4POINT0 (CH_LAYOUT_SURROUND|CH_BACK_CENTER)
+#define CH_LAYOUT_2_2 (CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT)
+#define CH_LAYOUT_QUAD (CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT)
+#define CH_LAYOUT_5POINT0 (CH_LAYOUT_SURROUND|CH_SIDE_LEFT|CH_SIDE_RIGHT)
+#define CH_LAYOUT_5POINT1 (CH_LAYOUT_5POINT0|CH_LOW_FREQUENCY)
+#define CH_LAYOUT_5POINT0_BACK (CH_LAYOUT_SURROUND|CH_BACK_LEFT|CH_BACK_RIGHT)
+#define CH_LAYOUT_5POINT1_BACK (CH_LAYOUT_5POINT0_BACK|CH_LOW_FREQUENCY)
+#define CH_LAYOUT_7POINT0 (CH_LAYOUT_5POINT0|CH_BACK_LEFT|CH_BACK_RIGHT)
+#define CH_LAYOUT_7POINT1 (CH_LAYOUT_5POINT1|CH_BACK_LEFT|CH_BACK_RIGHT)
+#define CH_LAYOUT_7POINT1_WIDE (CH_LAYOUT_5POINT1_BACK|\
+ CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER)
+#define CH_LAYOUT_STEREO_DOWNMIX (CH_STEREO_LEFT|CH_STEREO_RIGHT)
+
+/* in bytes */
+#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
+
+/**
+ * Required number of additionally allocated bytes at the end of the input bitstream for decoding.
+ * This is mainly needed because some optimized bitstream readers read
+ * 32 or 64 bit at once and could read over the end.<br>
+ * Note: If the first 23 bits of the additional bytes are not 0, then damaged
+ * MPEG bitstreams could cause overread and segfault.
+ */
+#define FF_INPUT_BUFFER_PADDING_SIZE 8
+
+/**
+ * minimum encoding buffer size
+ * Used to avoid some checks during header writing.
+ */
+#define FF_MIN_BUFFER_SIZE 16384
+
+
+/**
+ * motion estimation type.
+ */
+enum Motion_Est_ID {
+ ME_ZERO = 1, ///< no search, that is use 0,0 vector whenever one is needed
+ ME_FULL,
+ ME_LOG,
+ ME_PHODS,
+ ME_EPZS, ///< enhanced predictive zonal search
+ ME_X1, ///< reserved for experiments
+ ME_HEX, ///< hexagon based search
+ ME_UMH, ///< uneven multi-hexagon search
+ ME_ITER, ///< iterative search
+ ME_TESA, ///< transformed exhaustive search algorithm
+};
+
+enum AVDiscard{
+ /* We leave some space between them for extensions (drop some
+ * keyframes for intra-only or drop just some bidir frames). */
+ AVDISCARD_NONE =-16, ///< discard nothing
+ AVDISCARD_DEFAULT= 0, ///< discard useless packets like 0 size packets in avi
+ AVDISCARD_NONREF = 8, ///< discard all non reference
+ AVDISCARD_BIDIR = 16, ///< discard all bidirectional frames
+ AVDISCARD_NONKEY = 32, ///< discard all frames except keyframes
+ AVDISCARD_ALL = 48, ///< discard all
+};
+
+enum AVColorPrimaries{
+ AVCOL_PRI_BT709 =1, ///< also ITU-R BT1361 / IEC 61966-2-4 / SMPTE RP177 Annex B
+ AVCOL_PRI_UNSPECIFIED=2,
+ AVCOL_PRI_BT470M =4,
+ AVCOL_PRI_BT470BG =5, ///< also ITU-R BT601-6 625 / ITU-R BT1358 625 / ITU-R BT1700 625 PAL & SECAM
+ AVCOL_PRI_SMPTE170M =6, ///< also ITU-R BT601-6 525 / ITU-R BT1358 525 / ITU-R BT1700 NTSC
+ AVCOL_PRI_SMPTE240M =7, ///< functionally identical to above
+ AVCOL_PRI_FILM =8,
+ AVCOL_PRI_NB , ///< Not part of ABI
+};
+
+enum AVColorTransferCharacteristic{
+ AVCOL_TRC_BT709 =1, ///< also ITU-R BT1361
+ AVCOL_TRC_UNSPECIFIED=2,
+ AVCOL_TRC_GAMMA22 =4, ///< also ITU-R BT470M / ITU-R BT1700 625 PAL & SECAM
+ AVCOL_TRC_GAMMA28 =5, ///< also ITU-R BT470BG
+ AVCOL_TRC_NB , ///< Not part of ABI
+};
+
+enum AVColorSpace{
+ AVCOL_SPC_RGB =0,
+ AVCOL_SPC_BT709 =1, ///< also ITU-R BT1361 / IEC 61966-2-4 xvYCC709 / SMPTE RP177 Annex B
+ AVCOL_SPC_UNSPECIFIED=2,
+ AVCOL_SPC_FCC =4,
+ AVCOL_SPC_BT470BG =5, ///< also ITU-R BT601-6 625 / ITU-R BT1358 625 / ITU-R BT1700 625 PAL & SECAM / IEC 61966-2-4 xvYCC601
+ AVCOL_SPC_SMPTE170M =6, ///< also ITU-R BT601-6 525 / ITU-R BT1358 525 / ITU-R BT1700 NTSC / functionally identical to above
+ AVCOL_SPC_SMPTE240M =7,
+ AVCOL_SPC_NB , ///< Not part of ABI
+};
+
+enum AVColorRange{
+ AVCOL_RANGE_UNSPECIFIED=0,
+ AVCOL_RANGE_MPEG =1, ///< the normal 219*2^(n-8) "MPEG" YUV ranges
+ AVCOL_RANGE_JPEG =2, ///< the normal 2^n-1 "JPEG" YUV ranges
+ AVCOL_RANGE_NB , ///< Not part of ABI
+};
+
+/**
+ * X X 3 4 X X are luma samples,
+ * 1 2 1-6 are possible chroma positions
+ * X X 5 6 X 0 is undefined/unknown position
+ */
+enum AVChromaLocation{
+ AVCHROMA_LOC_UNSPECIFIED=0,
+ AVCHROMA_LOC_LEFT =1, ///< mpeg2/4, h264 default
+ AVCHROMA_LOC_CENTER =2, ///< mpeg1, jpeg, h263
+ AVCHROMA_LOC_TOPLEFT =3, ///< DV
+ AVCHROMA_LOC_TOP =4,
+ AVCHROMA_LOC_BOTTOMLEFT =5,
+ AVCHROMA_LOC_BOTTOM =6,
+ AVCHROMA_LOC_NB , ///< Not part of ABI
+};
+
+/**
+ * LPC analysis type
+ */
+enum AVLPCType {
+ AV_LPC_TYPE_DEFAULT = -1, ///< use the codec default LPC type
+ AV_LPC_TYPE_NONE = 0, ///< do not use LPC prediction or use all zero coefficients
+ AV_LPC_TYPE_FIXED = 1, ///< fixed LPC coefficients
+ AV_LPC_TYPE_LEVINSON = 2, ///< Levinson-Durbin recursion
+ AV_LPC_TYPE_CHOLESKY = 3, ///< Cholesky factorization
+ AV_LPC_TYPE_NB , ///< Not part of ABI
+};
+
+typedef struct RcOverride{
+ int start_frame;
+ int end_frame;
+ int qscale; // If this is 0 then quality_factor will be used instead.
+ float quality_factor;
+} RcOverride;
+
+#define FF_MAX_B_FRAMES 16
+
+/* encoding support
+ These flags can be passed in AVCodecContext.flags before initialization.
+ Note: Not everything is supported yet.
+*/
+
+#define CODEC_FLAG_QSCALE 0x0002 ///< Use fixed qscale.
+#define CODEC_FLAG_4MV 0x0004 ///< 4 MV per MB allowed / advanced prediction for H.263.
+#define CODEC_FLAG_QPEL 0x0010 ///< Use qpel MC.
+#define CODEC_FLAG_GMC 0x0020 ///< Use GMC.
+#define CODEC_FLAG_MV0 0x0040 ///< Always try a MB with MV=<0,0>.
+#define CODEC_FLAG_PART 0x0080 ///< Use data partitioning.
+/**
+ * The parent program guarantees that the input for B-frames containing
+ * streams is not written to for at least s->max_b_frames+1 frames, if
+ * this is not set the input will be copied.
+ */
+#define CODEC_FLAG_INPUT_PRESERVED 0x0100
+#define CODEC_FLAG_PASS1 0x0200 ///< Use internal 2pass ratecontrol in first pass mode.
+#define CODEC_FLAG_PASS2 0x0400 ///< Use internal 2pass ratecontrol in second pass mode.
+#define CODEC_FLAG_EXTERN_HUFF 0x1000 ///< Use external Huffman table (for MJPEG).
+#define CODEC_FLAG_GRAY 0x2000 ///< Only decode/encode grayscale.
+#define CODEC_FLAG_EMU_EDGE 0x4000 ///< Don't draw edges.
+#define CODEC_FLAG_PSNR 0x8000 ///< error[?] variables will be set during encoding.
+#define CODEC_FLAG_TRUNCATED 0x00010000 /** Input bitstream might be truncated at a random
+ location instead of only at frame boundaries. */
+#define CODEC_FLAG_NORMALIZE_AQP 0x00020000 ///< Normalize adaptive quantization.
+#define CODEC_FLAG_INTERLACED_DCT 0x00040000 ///< Use interlaced DCT.
+#define CODEC_FLAG_LOW_DELAY 0x00080000 ///< Force low delay.
+#define CODEC_FLAG_ALT_SCAN 0x00100000 ///< Use alternate scan.
+#define CODEC_FLAG_GLOBAL_HEADER 0x00400000 ///< Place global headers in extradata instead of every keyframe.
+#define CODEC_FLAG_BITEXACT 0x00800000 ///< Use only bitexact stuff (except (I)DCT).
+/* Fx : Flag for h263+ extra options */
+#define CODEC_FLAG_AC_PRED 0x01000000 ///< H.263 advanced intra coding / MPEG-4 AC prediction
+#define CODEC_FLAG_H263P_UMV 0x02000000 ///< unlimited motion vector
+#define CODEC_FLAG_CBP_RD 0x04000000 ///< Use rate distortion optimization for cbp.
+#define CODEC_FLAG_QP_RD 0x08000000 ///< Use rate distortion optimization for qp selectioon.
+#define CODEC_FLAG_H263P_AIV 0x00000008 ///< H.263 alternative inter VLC
+#define CODEC_FLAG_OBMC 0x00000001 ///< OBMC
+#define CODEC_FLAG_LOOP_FILTER 0x00000800 ///< loop filter
+#define CODEC_FLAG_H263P_SLICE_STRUCT 0x10000000
+#define CODEC_FLAG_INTERLACED_ME 0x20000000 ///< interlaced motion estimation
+#define CODEC_FLAG_SVCD_SCAN_OFFSET 0x40000000 ///< Will reserve space for SVCD scan offset user data.
+#define CODEC_FLAG_CLOSED_GOP 0x80000000
+#define CODEC_FLAG2_FAST 0x00000001 ///< Allow non spec compliant speedup tricks.
+#define CODEC_FLAG2_STRICT_GOP 0x00000002 ///< Strictly enforce GOP size.
+#define CODEC_FLAG2_NO_OUTPUT 0x00000004 ///< Skip bitstream encoding.
+#define CODEC_FLAG2_LOCAL_HEADER 0x00000008 ///< Place global headers at every keyframe instead of in extradata.
+#define CODEC_FLAG2_BPYRAMID 0x00000010 ///< H.264 allow B-frames to be used as references.
+#define CODEC_FLAG2_WPRED 0x00000020 ///< H.264 weighted biprediction for B-frames
+#define CODEC_FLAG2_MIXED_REFS 0x00000040 ///< H.264 one reference per partition, as opposed to one reference per macroblock
+#define CODEC_FLAG2_8X8DCT 0x00000080 ///< H.264 high profile 8x8 transform
+#define CODEC_FLAG2_FASTPSKIP 0x00000100 ///< H.264 fast pskip
+#define CODEC_FLAG2_AUD 0x00000200 ///< H.264 access unit delimiters
+#define CODEC_FLAG2_BRDO 0x00000400 ///< B-frame rate-distortion optimization
+#define CODEC_FLAG2_INTRA_VLC 0x00000800 ///< Use MPEG-2 intra VLC table.
+#define CODEC_FLAG2_MEMC_ONLY 0x00001000 ///< Only do ME/MC (I frames -> ref, P frame -> ME+MC).
+#define CODEC_FLAG2_DROP_FRAME_TIMECODE 0x00002000 ///< timecode is in drop frame format.
+#define CODEC_FLAG2_SKIP_RD 0x00004000 ///< RD optimal MB level residual skipping
+#define CODEC_FLAG2_CHUNKS 0x00008000 ///< Input bitstream might be truncated at a packet boundaries instead of only at frame boundaries.
+#define CODEC_FLAG2_NON_LINEAR_QUANT 0x00010000 ///< Use MPEG-2 nonlinear quantizer.
+#define CODEC_FLAG2_BIT_RESERVOIR 0x00020000 ///< Use a bit reservoir when encoding if possible
+#define CODEC_FLAG2_MBTREE 0x00040000 ///< Use macroblock tree ratecontrol (x264 only)
+#define CODEC_FLAG2_PSY 0x00080000 ///< Use psycho visual optimizations.
+#define CODEC_FLAG2_SSIM 0x00100000 ///< Compute SSIM during encoding, error[] values are undefined.
+#define CODEC_FLAG2_INTRA_REFRESH 0x00200000 ///< Use periodic insertion of intra blocks instead of keyframes.
+
+/* Unsupported options :
+ * Syntax Arithmetic coding (SAC)
+ * Reference Picture Selection
+ * Independent Segment Decoding */
+/* /Fx */
+/* codec capabilities */
+
+#define CODEC_CAP_DRAW_HORIZ_BAND 0x0001 ///< Decoder can use draw_horiz_band callback.
+/**
+ * Codec uses get_buffer() for allocating buffers and supports custom allocators.
+ * If not set, it might not use get_buffer() at all or use operations that
+ * assume the buffer was allocated by avcodec_default_get_buffer.
+ */
+#define CODEC_CAP_DR1 0x0002
+/* If 'parse_only' field is true, then avcodec_parse_frame() can be used. */
+#define CODEC_CAP_PARSE_ONLY 0x0004
+#define CODEC_CAP_TRUNCATED 0x0008
+/* Codec can export data for HW decoding (XvMC). */
+#define CODEC_CAP_HWACCEL 0x0010
+/**
+ * Codec has a nonzero delay and needs to be fed with NULL at the end to get the delayed data.
+ * If this is not set, the codec is guaranteed to never be fed with NULL data.
+ */
+#define CODEC_CAP_DELAY 0x0020
+/**
+ * Codec can be fed a final frame with a smaller size.
+ * This can be used to prevent truncation of the last audio samples.
+ */
+#define CODEC_CAP_SMALL_LAST_FRAME 0x0040
+/**
+ * Codec can export data for HW decoding (VDPAU).
+ */
+#define CODEC_CAP_HWACCEL_VDPAU 0x0080
+/**
+ * Codec can output multiple frames per AVPacket
+ * Normally demuxers return one frame at a time, demuxers which do not do
+ * are connected to a parser to split what they return into proper frames.
+ * This flag is reserved to the very rare category of codecs which have a
+ * bitstream that cannot be split into frames without timeconsuming
+ * operations like full decoding. Demuxers carring such bitstreams thus
+ * may return multiple frames in a packet. This has many disadvantages like
+ * prohibiting stream copy in many cases thus it should only be considered
+ * as a last resort.
+ */
+#define CODEC_CAP_SUBFRAMES 0x0100
+/**
+ * Codec is experimental and is thus avoided in favor of non experimental
+ * encoders
+ */
+#define CODEC_CAP_EXPERIMENTAL 0x0200
+
+//The following defines may change, don't expect compatibility if you use them.
+#define MB_TYPE_INTRA4x4 0x0001
+#define MB_TYPE_INTRA16x16 0x0002 //FIXME H.264-specific
+#define MB_TYPE_INTRA_PCM 0x0004 //FIXME H.264-specific
+#define MB_TYPE_16x16 0x0008
+#define MB_TYPE_16x8 0x0010
+#define MB_TYPE_8x16 0x0020
+#define MB_TYPE_8x8 0x0040
+#define MB_TYPE_INTERLACED 0x0080
+#define MB_TYPE_DIRECT2 0x0100 //FIXME
+#define MB_TYPE_ACPRED 0x0200
+#define MB_TYPE_GMC 0x0400
+#define MB_TYPE_SKIP 0x0800
+#define MB_TYPE_P0L0 0x1000
+#define MB_TYPE_P1L0 0x2000
+#define MB_TYPE_P0L1 0x4000
+#define MB_TYPE_P1L1 0x8000
+#define MB_TYPE_L0 (MB_TYPE_P0L0 | MB_TYPE_P1L0)
+#define MB_TYPE_L1 (MB_TYPE_P0L1 | MB_TYPE_P1L1)
+#define MB_TYPE_L0L1 (MB_TYPE_L0 | MB_TYPE_L1)
+#define MB_TYPE_QUANT 0x00010000
+#define MB_TYPE_CBP 0x00020000
+//Note bits 24-31 are reserved for codec specific use (h264 ref0, mpeg1 0mv, ...)
+
+/**
+ * Pan Scan area.
+ * This specifies the area which should be displayed.
+ * Note there may be multiple such areas for one frame.
+ */
+typedef struct AVPanScan{
+ /**
+ * id
+ * - encoding: Set by user.
+ * - decoding: Set by libavcodec.
+ */
+ int id;
+
+ /**
+ * width and height in 1/16 pel
+ * - encoding: Set by user.
+ * - decoding: Set by libavcodec.
+ */
+ int width;
+ int height;
+
+ /**
+ * position of the top left corner in 1/16 pel for up to 3 fields/frames
+ * - encoding: Set by user.
+ * - decoding: Set by libavcodec.
+ */
+ int16_t position[3][2];
+}AVPanScan;
+
+#define FF_COMMON_FRAME \
+ /**\
+ * pointer to the picture planes.\
+ * This might be different from the first allocated byte\
+ * - encoding: \
+ * - decoding: \
+ */\
+ uint8_t *data[4];\
+ int linesize[4];\
+ /**\
+ * pointer to the first allocated byte of the picture. Can be used in get_buffer/release_buffer.\
+ * This isn't used by libavcodec unless the default get/release_buffer() is used.\
+ * - encoding: \
+ * - decoding: \
+ */\
+ uint8_t *base[4];\
+ /**\
+ * 1 -> keyframe, 0-> not\
+ * - encoding: Set by libavcodec.\
+ * - decoding: Set by libavcodec.\
+ */\
+ int key_frame;\
+\
+ /**\
+ * Picture type of the frame, see ?_TYPE below.\
+ * - encoding: Set by libavcodec. for coded_picture (and set by user for input).\
+ * - decoding: Set by libavcodec.\
+ */\
+ int pict_type;\
+\
+ /**\
+ * presentation timestamp in time_base units (time when frame should be shown to user)\
+ * If AV_NOPTS_VALUE then frame_rate = 1/time_base will be assumed.\
+ * - encoding: MUST be set by user.\
+ * - decoding: Set by libavcodec.\
+ */\
+ int64_t pts;\
+\
+ /**\
+ * picture number in bitstream order\
+ * - encoding: set by\
+ * - decoding: Set by libavcodec.\
+ */\
+ int coded_picture_number;\
+ /**\
+ * picture number in display order\
+ * - encoding: set by\
+ * - decoding: Set by libavcodec.\
+ */\
+ int display_picture_number;\
+\
+ /**\
+ * quality (between 1 (good) and FF_LAMBDA_MAX (bad)) \
+ * - encoding: Set by libavcodec. for coded_picture (and set by user for input).\
+ * - decoding: Set by libavcodec.\
+ */\
+ int quality; \
+\
+ /**\
+ * buffer age (1->was last buffer and dint change, 2->..., ...).\
+ * Set to INT_MAX if the buffer has not been used yet.\
+ * - encoding: unused\
+ * - decoding: MUST be set by get_buffer().\
+ */\
+ int age;\
+\
+ /**\
+ * is this picture used as reference\
+ * The values for this are the same as the MpegEncContext.picture_structure\
+ * variable, that is 1->top field, 2->bottom field, 3->frame/both fields.\
+ * Set to 4 for delayed, non-reference frames.\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec. (before get_buffer() call)).\
+ */\
+ int reference;\
+\
+ /**\
+ * QP table\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec.\
+ */\
+ int8_t *qscale_table;\
+ /**\
+ * QP store stride\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec.\
+ */\
+ int qstride;\
+\
+ /**\
+ * mbskip_table[mb]>=1 if MB didn't change\
+ * stride= mb_width = (width+15)>>4\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec.\
+ */\
+ uint8_t *mbskip_table;\
+\
+ /**\
+ * motion vector table\
+ * @code\
+ * example:\
+ * int mv_sample_log2= 4 - motion_subsample_log2;\
+ * int mb_width= (width+15)>>4;\
+ * int mv_stride= (mb_width << mv_sample_log2) + 1;\
+ * motion_val[direction][x + y*mv_stride][0->mv_x, 1->mv_y];\
+ * @endcode\
+ * - encoding: Set by user.\
+ * - decoding: Set by libavcodec.\
+ */\
+ int16_t (*motion_val[2])[2];\
+\
+ /**\
+ * macroblock type table\
+ * mb_type_base + mb_width + 2\
+ * - encoding: Set by user.\
+ * - decoding: Set by libavcodec.\
+ */\
+ uint32_t *mb_type;\
+\
+ /**\
+ * log2 of the size of the block which a single vector in motion_val represents: \
+ * (4->16x16, 3->8x8, 2-> 4x4, 1-> 2x2)\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec.\
+ */\
+ uint8_t motion_subsample_log2;\
+\
+ /**\
+ * for some private data of the user\
+ * - encoding: unused\
+ * - decoding: Set by user.\
+ */\
+ void *opaque;\
+\
+ /**\
+ * error\
+ * - encoding: Set by libavcodec. if flags&CODEC_FLAG_PSNR.\
+ * - decoding: unused\
+ */\
+ uint64_t error[4];\
+\
+ /**\
+ * type of the buffer (to keep track of who has to deallocate data[*])\
+ * - encoding: Set by the one who allocates it.\
+ * - decoding: Set by the one who allocates it.\
+ * Note: User allocated (direct rendering) & internal buffers cannot coexist currently.\
+ */\
+ int type;\
+ \
+ /**\
+ * When decoding, this signals how much the picture must be delayed.\
+ * extra_delay = repeat_pict / (2*fps)\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec.\
+ */\
+ int repeat_pict;\
+ \
+ /**\
+ * \
+ */\
+ int qscale_type;\
+ \
+ /**\
+ * The content of the picture is interlaced.\
+ * - encoding: Set by user.\
+ * - decoding: Set by libavcodec. (default 0)\
+ */\
+ int interlaced_frame;\
+ \
+ /**\
+ * If the content is interlaced, is top field displayed first.\
+ * - encoding: Set by user.\
+ * - decoding: Set by libavcodec.\
+ */\
+ int top_field_first;\
+ \
+ /**\
+ * Pan scan.\
+ * - encoding: Set by user.\
+ * - decoding: Set by libavcodec.\
+ */\
+ AVPanScan *pan_scan;\
+ \
+ /**\
+ * Tell user application that palette has changed from previous frame.\
+ * - encoding: ??? (no palette-enabled encoder yet)\
+ * - decoding: Set by libavcodec. (default 0).\
+ */\
+ int palette_has_changed;\
+ \
+ /**\
+ * codec suggestion on buffer type if != 0\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec. (before get_buffer() call)).\
+ */\
+ int buffer_hints;\
+\
+ /**\
+ * DCT coefficients\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec.\
+ */\
+ short *dct_coeff;\
+\
+ /**\
+ * motion reference frame index\
+ * the order in which these are stored can depend on the codec.\
+ * - encoding: Set by user.\
+ * - decoding: Set by libavcodec.\
+ */\
+ int8_t *ref_index[2];\
+\
+ /**\
+ * reordered opaque 64bit number (generally a PTS) from AVCodecContext.reordered_opaque\
+ * output in AVFrame.reordered_opaque\
+ * - encoding: unused\
+ * - decoding: Read by user.\
+ */\
+ int64_t reordered_opaque;\
+\
+ /**\
+ * hardware accelerator private data (FFmpeg allocated)\
+ * - encoding: unused\
+ * - decoding: Set by libavcodec\
+ */\
+ void *hwaccel_picture_private;\
+
+
+#define FF_QSCALE_TYPE_MPEG1 0
+#define FF_QSCALE_TYPE_MPEG2 1
+#define FF_QSCALE_TYPE_H264 2
+#define FF_QSCALE_TYPE_VP56 3
+
+#define FF_BUFFER_TYPE_INTERNAL 1
+#define FF_BUFFER_TYPE_USER 2 ///< direct rendering buffers (image is (de)allocated by user)
+#define FF_BUFFER_TYPE_SHARED 4 ///< Buffer from somewhere else; don't deallocate image (data/base), all other tables are not shared.
+#define FF_BUFFER_TYPE_COPY 8 ///< Just a (modified) copy of some other buffer, don't deallocate anything.
+
+
+#define FF_I_TYPE 1 ///< Intra
+#define FF_P_TYPE 2 ///< Predicted
+#define FF_B_TYPE 3 ///< Bi-dir predicted
+#define FF_S_TYPE 4 ///< S(GMC)-VOP MPEG4
+#define FF_SI_TYPE 5 ///< Switching Intra
+#define FF_SP_TYPE 6 ///< Switching Predicted
+#define FF_BI_TYPE 7
+
+#define FF_BUFFER_HINTS_VALID 0x01 // Buffer hints value is meaningful (if 0 ignore).
+#define FF_BUFFER_HINTS_READABLE 0x02 // Codec will read from buffer.
+#define FF_BUFFER_HINTS_PRESERVE 0x04 // User must not alter buffer content.
+#define FF_BUFFER_HINTS_REUSABLE 0x08 // Codec will reuse the buffer (update).
+
+typedef struct AVPacket {
+ /**
+ * Presentation timestamp in AVStream->time_base units; the time at which
+ * the decompressed packet will be presented to the user.
+ * Can be AV_NOPTS_VALUE if it is not stored in the file.
+ * pts MUST be larger or equal to dts as presentation cannot happen before
+ * decompression, unless one wants to view hex dumps. Some formats misuse
+ * the terms dts and pts/cts to mean something different. Such timestamps
+ * must be converted to true pts/dts before they are stored in AVPacket.
+ */
+ int64_t pts;
+ /**
+ * Decompression timestamp in AVStream->time_base units; the time at which
+ * the packet is decompressed.
+ * Can be AV_NOPTS_VALUE if it is not stored in the file.
+ */
+ int64_t dts;
+ uint8_t *data;
+ int size;
+ int stream_index;
+ int flags;
+ /**
+ * Duration of this packet in AVStream->time_base units, 0 if unknown.
+ * Equals next_pts - this_pts in presentation order.
+ */
+ int duration;
+ void (*destruct)(struct AVPacket *);
+ void *priv;
+ int64_t pos; ///< byte position in stream, -1 if unknown
+
+ /**
+ * Time difference in AVStream->time_base units from the pts of this
+ * packet to the point at which the output from the decoder has converged
+ * independent from the availability of previous frames. That is, the
+ * frames are virtually identical no matter if decoding started from
+ * the very first frame or from this keyframe.
+ * Is AV_NOPTS_VALUE if unknown.
+ * This field is not the display duration of the current packet.
+ * This field has no meaning if the packet does not have AV_PKT_FLAG_KEY
+ * set.
+ *
+ * The purpose of this field is to allow seeking in streams that have no
+ * keyframes in the conventional sense. It corresponds to the
+ * recovery point SEI in H.264 and match_time_delta in NUT. It is also
+ * essential for some types of subtitle streams to ensure that all
+ * subtitles are correctly displayed after seeking.
+ */
+ int64_t convergence_duration;
+} AVPacket;
+#define AV_PKT_FLAG_KEY 0x0001
+#if LIBAVCODEC_VERSION_MAJOR < 53
+#define PKT_FLAG_KEY AV_PKT_FLAG_KEY
+#endif
+
+/**
+ * Audio Video Frame.
+ * New fields can be added to the end of FF_COMMON_FRAME with minor version
+ * bumps.
+ * Removal, reordering and changes to existing fields require a major
+ * version bump. No fields should be added into AVFrame before or after
+ * FF_COMMON_FRAME!
+ * sizeof(AVFrame) must not be used outside libav*.
+ */
+typedef struct AVFrame {
+ FF_COMMON_FRAME
+} AVFrame;
+
+/**
+ * main external API structure.
+ * New fields can be added to the end with minor version bumps.
+ * Removal, reordering and changes to existing fields require a major
+ * version bump.
+ * sizeof(AVCodecContext) must not be used outside libav*.
+ */
+typedef struct AVCodecContext {
+ /**
+ * information on struct for av_log
+ * - set by avcodec_alloc_context
+ */
+ const AVClass *av_class;
+ /**
+ * the average bitrate
+ * - encoding: Set by user; unused for constant quantizer encoding.
+ * - decoding: Set by libavcodec. 0 or some bitrate if this info is available in the stream.
+ */
+ int bit_rate;
+
+ /**
+ * number of bits the bitstream is allowed to diverge from the reference.
+ * the reference can be CBR (for CBR pass1) or VBR (for pass2)
+ * - encoding: Set by user; unused for constant quantizer encoding.
+ * - decoding: unused
+ */
+ int bit_rate_tolerance;
+
+ /**
+ * CODEC_FLAG_*.
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ */
+ int flags;
+
+ /**
+ * Some codecs need additional format info. It is stored here.
+ * If any muxer uses this then ALL demuxers/parsers AND encoders for the
+ * specific codec MUST set it correctly otherwise stream copy breaks.
+ * In general use of this field by muxers is not recommanded.
+ * - encoding: Set by libavcodec.
+ * - decoding: Set by libavcodec. (FIXME: Is this OK?)
+ */
+ int sub_id;
+
+ /**
+ * Motion estimation algorithm used for video coding.
+ * 1 (zero), 2 (full), 3 (log), 4 (phods), 5 (epzs), 6 (x1), 7 (hex),
+ * 8 (umh), 9 (iter), 10 (tesa) [7, 8, 10 are x264 specific, 9 is snow specific]
+ * - encoding: MUST be set by user.
+ * - decoding: unused
+ */
+ int me_method;
+
+ /**
+ * some codecs need / can use extradata like Huffman tables.
+ * mjpeg: Huffman tables
+ * rv10: additional flags
+ * mpeg4: global headers (they can be in the bitstream or here)
+ * The allocated memory should be FF_INPUT_BUFFER_PADDING_SIZE bytes larger
+ * than extradata_size to avoid prolems if it is read with the bitstream reader.
+ * The bytewise contents of extradata must not depend on the architecture or CPU endianness.
+ * - encoding: Set/allocated/freed by libavcodec.
+ * - decoding: Set/allocated/freed by user.
+ */
+ uint8_t *extradata;
+ int extradata_size;
+
+ /**
+ * This is the fundamental unit of time (in seconds) in terms
+ * of which frame timestamps are represented. For fixed-fps content,
+ * timebase should be 1/framerate and timestamp increments should be
+ * identically 1.
+ * - encoding: MUST be set by user.
+ * - decoding: Set by libavcodec.
+ */
+ AVRational time_base;
+
+ /* video only */
+ /**
+ * picture width / height.
+ * - encoding: MUST be set by user.
+ * - decoding: Set by libavcodec.
+ * Note: For compatibility it is possible to set this instead of
+ * coded_width/height before decoding.
+ */
+ int width, height;
+
+#define FF_ASPECT_EXTENDED 15
+
+ /**
+ * the number of pictures in a group of pictures, or 0 for intra_only
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int gop_size;
+
+ /**
+ * Pixel format, see PIX_FMT_xxx.
+ * May be set by the demuxer if known from headers.
+ * May be overriden by the decoder if it knows better.
+ * - encoding: Set by user.
+ * - decoding: Set by user if known, overridden by libavcodec if known
+ */
+ enum PixelFormat pix_fmt;
+
+ /**
+ * Frame rate emulation. If not zero, the lower layer (i.e. format handler)
+ * has to read frames at native frame rate.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int rate_emu;
+
+ /**
+ * If non NULL, 'draw_horiz_band' is called by the libavcodec
+ * decoder to draw a horizontal band. It improves cache usage. Not
+ * all codecs can do that. You must check the codec capabilities
+ * beforehand.
+ * The function is also used by hardware acceleration APIs.
+ * It is called at least once during frame decoding to pass
+ * the data needed for hardware render.
+ * In that mode instead of pixel data, AVFrame points to
+ * a structure specific to the acceleration API. The application
+ * reads the structure and can change some fields to indicate progress
+ * or mark state.
+ * - encoding: unused
+ * - decoding: Set by user.
+ * @param height the height of the slice
+ * @param y the y position of the slice
+ * @param type 1->top field, 2->bottom field, 3->frame
+ * @param offset offset into the AVFrame.data from which the slice should be read
+ */
+ void (*draw_horiz_band)(struct AVCodecContext *s,
+ const AVFrame *src, int offset[4],
+ int y, int type, int height);
+
+ /* audio only */
+ int sample_rate; ///< samples per second
+ int channels; ///< number of audio channels
+
+ /**
+ * audio sample format
+ * - encoding: Set by user.
+ * - decoding: Set by libavcodec.
+ */
+ enum SampleFormat sample_fmt; ///< sample format
+
+ /* The following data should not be initialized. */
+ /**
+ * Samples per packet, initialized when calling 'init'.
+ */
+ int frame_size;
+ int frame_number; ///< audio or video frame number
+#if LIBAVCODEC_VERSION_MAJOR < 53
+ int real_pict_num; ///< Returns the real picture number of previous encoded frame.
+#endif
+
+ /**
+ * Number of frames the decoded output will be delayed relative to
+ * the encoded input.
+ * - encoding: Set by libavcodec.
+ * - decoding: unused
+ */
+ int delay;
+
+ /* - encoding parameters */
+ float qcompress; ///< amount of qscale change between easy & hard scenes (0.0-1.0)
+ float qblur; ///< amount of qscale smoothing over time (0.0-1.0)
+
+ /**
+ * minimum quantizer
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int qmin;
+
+ /**
+ * maximum quantizer
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int qmax;
+
+ /**
+ * maximum quantizer difference between frames
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int max_qdiff;
+
+ /**
+ * maximum number of B-frames between non-B-frames
+ * Note: The output will be delayed by max_b_frames+1 relative to the input.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int max_b_frames;
+
+ /**
+ * qscale factor between IP and B-frames
+ * If > 0 then the last P-frame quantizer will be used (q= lastp_q*factor+offset).
+ * If < 0 then normal ratecontrol will be done (q= -normal_q*factor+offset).
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float b_quant_factor;
+
+ /** obsolete FIXME remove */
+ int rc_strategy;
+#define FF_RC_STRATEGY_XVID 1
+
+ int b_frame_strategy;
+
+ /**
+ * hurry up amount
+ * - encoding: unused
+ * - decoding: Set by user. 1-> Skip B-frames, 2-> Skip IDCT/dequant too, 5-> Skip everything except header
+ * @deprecated Deprecated in favor of skip_idct and skip_frame.
+ */
+ int hurry_up;
+
+ struct AVCodec *codec;
+
+ void *priv_data;
+
+ int rtp_payload_size; /* The size of the RTP payload: the coder will */
+ /* do its best to deliver a chunk with size */
+ /* below rtp_payload_size, the chunk will start */
+ /* with a start code on some codecs like H.263. */
+ /* This doesn't take account of any particular */
+ /* headers inside the transmitted RTP payload. */
+
+
+ /* The RTP callback: This function is called */
+ /* every time the encoder has a packet to send. */
+ /* It depends on the encoder if the data starts */
+ /* with a Start Code (it should). H.263 does. */
+ /* mb_nb contains the number of macroblocks */
+ /* encoded in the RTP payload. */
+ void (*rtp_callback)(struct AVCodecContext *avctx, void *data, int size, int mb_nb);
+
+ /* statistics, used for 2-pass encoding */
+ int mv_bits;
+ int header_bits;
+ int i_tex_bits;
+ int p_tex_bits;
+ int i_count;
+ int p_count;
+ int skip_count;
+ int misc_bits;
+
+ /**
+ * number of bits used for the previously encoded frame
+ * - encoding: Set by libavcodec.
+ * - decoding: unused
+ */
+ int frame_bits;
+
+ /**
+ * Private data of the user, can be used to carry app specific stuff.
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ */
+ void *opaque;
+
+ char codec_name[32];
+ enum AVMediaType codec_type; /* see AVMEDIA_TYPE_xxx */
+ enum CodecID codec_id; /* see CODEC_ID_xxx */
+
+ /**
+ * fourcc (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
+ * This is used to work around some encoder bugs.
+ * A demuxer should set this to what is stored in the field used to identify the codec.
+ * If there are multiple such fields in a container then the demuxer should choose the one
+ * which maximizes the information about the used codec.
+ * If the codec tag field in a container is larger then 32 bits then the demuxer should
+ * remap the longer ID to 32 bits with a table or other structure. Alternatively a new
+ * extra_codec_tag + size could be added but for this a clear advantage must be demonstrated
+ * first.
+ * - encoding: Set by user, if not then the default based on codec_id will be used.
+ * - decoding: Set by user, will be converted to uppercase by libavcodec during init.
+ */
+ unsigned int codec_tag;
+
+ /**
+ * Work around bugs in encoders which sometimes cannot be detected automatically.
+ * - encoding: Set by user
+ * - decoding: Set by user
+ */
+ int workaround_bugs;
+#define FF_BUG_AUTODETECT 1 ///< autodetection
+#define FF_BUG_OLD_MSMPEG4 2
+#define FF_BUG_XVID_ILACE 4
+#define FF_BUG_UMP4 8
+#define FF_BUG_NO_PADDING 16
+#define FF_BUG_AMV 32
+#define FF_BUG_AC_VLC 0 ///< Will be removed, libavcodec can now handle these non-compliant files by default.
+#define FF_BUG_QPEL_CHROMA 64
+#define FF_BUG_STD_QPEL 128
+#define FF_BUG_QPEL_CHROMA2 256
+#define FF_BUG_DIRECT_BLOCKSIZE 512
+#define FF_BUG_EDGE 1024
+#define FF_BUG_HPEL_CHROMA 2048
+#define FF_BUG_DC_CLIP 4096
+#define FF_BUG_MS 8192 ///< Work around various bugs in Microsoft's broken decoders.
+#define FF_BUG_TRUNCATED 16384
+//#define FF_BUG_FAKE_SCALABILITY 16 //Autodetection should work 100%.
+
+ /**
+ * luma single coefficient elimination threshold
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int luma_elim_threshold;
+
+ /**
+ * chroma single coeff elimination threshold
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int chroma_elim_threshold;
+
+ /**
+ * strictly follow the standard (MPEG4, ...).
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ * Setting this to STRICT or higher means the encoder and decoder will
+ * generally do stupid things, whereas setting it to unofficial or lower
+ * will mean the encoder might produce output that is not supported by all
+ * spec-compliant decoders. Decoders don't differentiate between normal,
+ * unofficial and experimental (that is, they always try to decode things
+ * when they can) unless they are explicitly asked to behave stupidly
+ * (=strictly conform to the specs)
+ */
+ int strict_std_compliance;
+#define FF_COMPLIANCE_VERY_STRICT 2 ///< Strictly conform to an older more strict version of the spec or reference software.
+#define FF_COMPLIANCE_STRICT 1 ///< Strictly conform to all the things in the spec no matter what consequences.
+#define FF_COMPLIANCE_NORMAL 0
+#if LIBAVCODEC_VERSION_MAJOR < 53
+#define FF_COMPLIANCE_INOFFICIAL -1 ///< Allow inofficial extensions (deprecated - use FF_COMPLIANCE_UNOFFICIAL instead).
+#endif
+#define FF_COMPLIANCE_UNOFFICIAL -1 ///< Allow unofficial extensions
+#define FF_COMPLIANCE_EXPERIMENTAL -2 ///< Allow nonstandardized experimental things.
+
+ /**
+ * qscale offset between IP and B-frames
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float b_quant_offset;
+
+ /**
+ * Error recognization; higher values will detect more errors but may
+ * misdetect some more or less valid parts as errors.
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int error_recognition;
+#define FF_ER_CAREFUL 1
+#define FF_ER_COMPLIANT 2
+#define FF_ER_AGGRESSIVE 3
+#define FF_ER_VERY_AGGRESSIVE 4
+
+ /**
+ * Called at the beginning of each frame to get a buffer for it.
+ * If pic.reference is set then the frame will be read later by libavcodec.
+ * avcodec_align_dimensions2() should be used to find the required width and
+ * height, as they normally need to be rounded up to the next multiple of 16.
+ * if CODEC_CAP_DR1 is not set then get_buffer() must call
+ * avcodec_default_get_buffer() instead of providing buffers allocated by
+ * some other means.
+ * - encoding: unused
+ * - decoding: Set by libavcodec, user can override.
+ */
+ int (*get_buffer)(struct AVCodecContext *c, AVFrame *pic);
+
+ /**
+ * Called to release buffers which were allocated with get_buffer.
+ * A released buffer can be reused in get_buffer().
+ * pic.data[*] must be set to NULL.
+ * - encoding: unused
+ * - decoding: Set by libavcodec, user can override.
+ */
+ void (*release_buffer)(struct AVCodecContext *c, AVFrame *pic);
+
+ /**
+ * Size of the frame reordering buffer in the decoder.
+ * For MPEG-2 it is 1 IPB or 0 low delay IP.
+ * - encoding: Set by libavcodec.
+ * - decoding: Set by libavcodec.
+ */
+ int has_b_frames;
+
+ /**
+ * number of bytes per packet if constant and known or 0
+ * Used by some WAV based audio codecs.
+ */
+ int block_align;
+
+ int parse_only; /* - decoding only: If true, only parsing is done
+ (function avcodec_parse_frame()). The frame
+ data is returned. Only MPEG codecs support this now. */
+
+ /**
+ * 0-> h263 quant 1-> mpeg quant
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int mpeg_quant;
+
+ /**
+ * pass1 encoding statistics output buffer
+ * - encoding: Set by libavcodec.
+ * - decoding: unused
+ */
+ char *stats_out;
+
+ /**
+ * pass2 encoding statistics input buffer
+ * Concatenated stuff from stats_out of pass1 should be placed here.
+ * - encoding: Allocated/set/freed by user.
+ * - decoding: unused
+ */
+ char *stats_in;
+
+ /**
+ * ratecontrol qmin qmax limiting method
+ * 0-> clipping, 1-> use a nice continous function to limit qscale wthin qmin/qmax.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float rc_qsquish;
+
+ float rc_qmod_amp;
+ int rc_qmod_freq;
+
+ /**
+ * ratecontrol override, see RcOverride
+ * - encoding: Allocated/set/freed by user.
+ * - decoding: unused
+ */
+ RcOverride *rc_override;
+ int rc_override_count;
+
+ /**
+ * rate control equation
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ const char *rc_eq;
+
+ /**
+ * maximum bitrate
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int rc_max_rate;
+
+ /**
+ * minimum bitrate
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int rc_min_rate;
+
+ /**
+ * decoder bitstream buffer size
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int rc_buffer_size;
+ float rc_buffer_aggressivity;
+
+ /**
+ * qscale factor between P and I-frames
+ * If > 0 then the last p frame quantizer will be used (q= lastp_q*factor+offset).
+ * If < 0 then normal ratecontrol will be done (q= -normal_q*factor+offset).
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float i_quant_factor;
+
+ /**
+ * qscale offset between P and I-frames
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float i_quant_offset;
+
+ /**
+ * initial complexity for pass1 ratecontrol
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float rc_initial_cplx;
+
+ /**
+ * DCT algorithm, see FF_DCT_* below
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int dct_algo;
+#define FF_DCT_AUTO 0
+#define FF_DCT_FASTINT 1
+#define FF_DCT_INT 2
+#define FF_DCT_MMX 3
+#define FF_DCT_MLIB 4
+#define FF_DCT_ALTIVEC 5
+#define FF_DCT_FAAN 6
+
+ /**
+ * luminance masking (0-> disabled)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float lumi_masking;
+
+ /**
+ * temporary complexity masking (0-> disabled)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float temporal_cplx_masking;
+
+ /**
+ * spatial complexity masking (0-> disabled)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float spatial_cplx_masking;
+
+ /**
+ * p block masking (0-> disabled)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float p_masking;
+
+ /**
+ * darkness masking (0-> disabled)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float dark_masking;
+
+ /**
+ * IDCT algorithm, see FF_IDCT_* below.
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ */
+ int idct_algo;
+#define FF_IDCT_AUTO 0
+#define FF_IDCT_INT 1
+#define FF_IDCT_SIMPLE 2
+#define FF_IDCT_SIMPLEMMX 3
+#define FF_IDCT_LIBMPEG2MMX 4
+#define FF_IDCT_PS2 5
+#define FF_IDCT_MLIB 6
+#define FF_IDCT_ARM 7
+#define FF_IDCT_ALTIVEC 8
+#define FF_IDCT_SH4 9
+#define FF_IDCT_SIMPLEARM 10
+#define FF_IDCT_H264 11
+#define FF_IDCT_VP3 12
+#define FF_IDCT_IPP 13
+#define FF_IDCT_XVIDMMX 14
+#define FF_IDCT_CAVS 15
+#define FF_IDCT_SIMPLEARMV5TE 16
+#define FF_IDCT_SIMPLEARMV6 17
+#define FF_IDCT_SIMPLEVIS 18
+#define FF_IDCT_WMV2 19
+#define FF_IDCT_FAAN 20
+#define FF_IDCT_EA 21
+#define FF_IDCT_SIMPLENEON 22
+#define FF_IDCT_SIMPLEALPHA 23
+#define FF_IDCT_BINK 24
+
+ /**
+ * slice count
+ * - encoding: Set by libavcodec.
+ * - decoding: Set by user (or 0).
+ */
+ int slice_count;
+ /**
+ * slice offsets in the frame in bytes
+ * - encoding: Set/allocated by libavcodec.
+ * - decoding: Set/allocated by user (or NULL).
+ */
+ int *slice_offset;
+
+ /**
+ * error concealment flags
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int error_concealment;
+#define FF_EC_GUESS_MVS 1
+#define FF_EC_DEBLOCK 2
+
+ /**
+ * dsp_mask could be add used to disable unwanted CPU features
+ * CPU features (i.e. MMX, SSE. ...)
+ *
+ * With the FORCE flag you may instead enable given CPU features.
+ * (Dangerous: Usable in case of misdetection, improper usage however will
+ * result into program crash.)
+ */
+ unsigned dsp_mask;
+#define FF_MM_FORCE 0x80000000 /* Force usage of selected flags (OR) */
+ /* lower 16 bits - CPU features */
+#define FF_MM_MMX 0x0001 ///< standard MMX
+#define FF_MM_3DNOW 0x0004 ///< AMD 3DNOW
+#if LIBAVCODEC_VERSION_MAJOR < 53
+#define FF_MM_MMXEXT 0x0002 ///< SSE integer functions or AMD MMX ext
+#endif
+#define FF_MM_MMX2 0x0002 ///< SSE integer functions or AMD MMX ext
+#define FF_MM_SSE 0x0008 ///< SSE functions
+#define FF_MM_SSE2 0x0010 ///< PIV SSE2 functions
+#define FF_MM_SSE2SLOW 0x40000000 ///< SSE2 supported, but usually not faster
+ ///< than regular MMX/SSE (e.g. Core1)
+#define FF_MM_3DNOWEXT 0x0020 ///< AMD 3DNowExt
+#define FF_MM_SSE3 0x0040 ///< Prescott SSE3 functions
+#define FF_MM_SSE3SLOW 0x20000000 ///< SSE3 supported, but usually not faster
+ ///< than regular MMX/SSE (e.g. Core1)
+#define FF_MM_SSSE3 0x0080 ///< Conroe SSSE3 functions
+#define FF_MM_SSE4 0x0100 ///< Penryn SSE4.1 functions
+#define FF_MM_SSE42 0x0200 ///< Nehalem SSE4.2 functions
+#define FF_MM_IWMMXT 0x0100 ///< XScale IWMMXT
+#define FF_MM_ALTIVEC 0x0001 ///< standard AltiVec
+
+ /**
+ * bits per sample/pixel from the demuxer (needed for huffyuv).
+ * - encoding: Set by libavcodec.
+ * - decoding: Set by user.
+ */
+ int bits_per_coded_sample;
+
+ /**
+ * prediction method (needed for huffyuv)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int prediction_method;
+#define FF_PRED_LEFT 0
+#define FF_PRED_PLANE 1
+#define FF_PRED_MEDIAN 2
+
+ /**
+ * sample aspect ratio (0 if unknown)
+ * That is the width of a pixel divided by the height of the pixel.
+ * Numerator and denominator must be relatively prime and smaller than 256 for some video standards.
+ * - encoding: Set by user.
+ * - decoding: Set by libavcodec.
+ */
+ AVRational sample_aspect_ratio;
+
+ /**
+ * the picture in the bitstream
+ * - encoding: Set by libavcodec.
+ * - decoding: Set by libavcodec.
+ */
+ AVFrame *coded_frame;
+
+ /**
+ * debug
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ */
+ int debug;
+#define FF_DEBUG_PICT_INFO 1
+#define FF_DEBUG_RC 2
+#define FF_DEBUG_BITSTREAM 4
+#define FF_DEBUG_MB_TYPE 8
+#define FF_DEBUG_QP 16
+#define FF_DEBUG_MV 32
+#define FF_DEBUG_DCT_COEFF 0x00000040
+#define FF_DEBUG_SKIP 0x00000080
+#define FF_DEBUG_STARTCODE 0x00000100
+#define FF_DEBUG_PTS 0x00000200
+#define FF_DEBUG_ER 0x00000400
+#define FF_DEBUG_MMCO 0x00000800
+#define FF_DEBUG_BUGS 0x00001000
+#define FF_DEBUG_VIS_QP 0x00002000
+#define FF_DEBUG_VIS_MB_TYPE 0x00004000
+#define FF_DEBUG_BUFFERS 0x00008000
+
+ /**
+ * debug
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ */
+ int debug_mv;
+#define FF_DEBUG_VIS_MV_P_FOR 0x00000001 //visualize forward predicted MVs of P frames
+#define FF_DEBUG_VIS_MV_B_FOR 0x00000002 //visualize forward predicted MVs of B frames
+#define FF_DEBUG_VIS_MV_B_BACK 0x00000004 //visualize backward predicted MVs of B frames
+
+ /**
+ * error
+ * - encoding: Set by libavcodec if flags&CODEC_FLAG_PSNR.
+ * - decoding: unused
+ */
+ uint64_t error[4];
+
+ /**
+ * minimum MB quantizer
+ * - encoding: unused
+ * - decoding: unused
+ */
+ int mb_qmin;
+
+ /**
+ * maximum MB quantizer
+ * - encoding: unused
+ * - decoding: unused
+ */
+ int mb_qmax;
+
+ /**
+ * motion estimation comparison function
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int me_cmp;
+ /**
+ * subpixel motion estimation comparison function
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int me_sub_cmp;
+ /**
+ * macroblock comparison function (not supported yet)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int mb_cmp;
+ /**
+ * interlaced DCT comparison function
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int ildct_cmp;
+#define FF_CMP_SAD 0
+#define FF_CMP_SSE 1
+#define FF_CMP_SATD 2
+#define FF_CMP_DCT 3
+#define FF_CMP_PSNR 4
+#define FF_CMP_BIT 5
+#define FF_CMP_RD 6
+#define FF_CMP_ZERO 7
+#define FF_CMP_VSAD 8
+#define FF_CMP_VSSE 9
+#define FF_CMP_NSSE 10
+#define FF_CMP_W53 11
+#define FF_CMP_W97 12
+#define FF_CMP_DCTMAX 13
+#define FF_CMP_DCT264 14
+#define FF_CMP_CHROMA 256
+
+ /**
+ * ME diamond size & shape
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int dia_size;
+
+ /**
+ * amount of previous MV predictors (2a+1 x 2a+1 square)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int last_predictor_count;
+
+ /**
+ * prepass for motion estimation
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int pre_me;
+
+ /**
+ * motion estimation prepass comparison function
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int me_pre_cmp;
+
+ /**
+ * ME prepass diamond size & shape
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int pre_dia_size;
+
+ /**
+ * subpel ME quality
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int me_subpel_quality;
+
+ /**
+ * callback to negotiate the pixelFormat
+ * @param fmt is the list of formats which are supported by the codec,
+ * it is terminated by -1 as 0 is a valid format, the formats are ordered by quality.
+ * The first is always the native one.
+ * @return the chosen format
+ * - encoding: unused
+ * - decoding: Set by user, if not set the native format will be chosen.
+ */
+ enum PixelFormat (*get_format)(struct AVCodecContext *s, const enum PixelFormat * fmt);
+
+ /**
+ * DTG active format information (additional aspect ratio
+ * information only used in DVB MPEG-2 transport streams)
+ * 0 if not set.
+ *
+ * - encoding: unused
+ * - decoding: Set by decoder.
+ */
+ int dtg_active_format;
+#define FF_DTG_AFD_SAME 8
+#define FF_DTG_AFD_4_3 9
+#define FF_DTG_AFD_16_9 10
+#define FF_DTG_AFD_14_9 11
+#define FF_DTG_AFD_4_3_SP_14_9 13
+#define FF_DTG_AFD_16_9_SP_14_9 14
+#define FF_DTG_AFD_SP_4_3 15
+
+ /**
+ * maximum motion estimation search range in subpel units
+ * If 0 then no limit.
+ *
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int me_range;
+
+ /**
+ * intra quantizer bias
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int intra_quant_bias;
+#define FF_DEFAULT_QUANT_BIAS 999999
+
+ /**
+ * inter quantizer bias
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int inter_quant_bias;
+
+ /**
+ * color table ID
+ * - encoding: unused
+ * - decoding: Which clrtable should be used for 8bit RGB images.
+ * Tables have to be stored somewhere. FIXME
+ */
+ int color_table_id;
+
+ /**
+ * internal_buffer count
+ * Don't touch, used by libavcodec default_get_buffer().
+ */
+ int internal_buffer_count;
+
+ /**
+ * internal_buffers
+ * Don't touch, used by libavcodec default_get_buffer().
+ */
+ void *internal_buffer;
+
+#define FF_LAMBDA_SHIFT 7
+#define FF_LAMBDA_SCALE (1<<FF_LAMBDA_SHIFT)
+#define FF_QP2LAMBDA 118 ///< factor to convert from H.263 QP to lambda
+#define FF_LAMBDA_MAX (256*128-1)
+
+#define FF_QUALITY_SCALE FF_LAMBDA_SCALE //FIXME maybe remove
+ /**
+ * Global quality for codecs which cannot change it per frame.
+ * This should be proportional to MPEG-1/2/4 qscale.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int global_quality;
+
+#define FF_CODER_TYPE_VLC 0
+#define FF_CODER_TYPE_AC 1
+#define FF_CODER_TYPE_RAW 2
+#define FF_CODER_TYPE_RLE 3
+#define FF_CODER_TYPE_DEFLATE 4
+ /**
+ * coder type
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int coder_type;
+
+ /**
+ * context model
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int context_model;
+#if 0
+ /**
+ *
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ uint8_t * (*realloc)(struct AVCodecContext *s, uint8_t *buf, int buf_size);
+#endif
+
+ /**
+ * slice flags
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int slice_flags;
+#define SLICE_FLAG_CODED_ORDER 0x0001 ///< draw_horiz_band() is called in coded order instead of display
+#define SLICE_FLAG_ALLOW_FIELD 0x0002 ///< allow draw_horiz_band() with field slices (MPEG2 field pics)
+#define SLICE_FLAG_ALLOW_PLANE 0x0004 ///< allow draw_horiz_band() with 1 component at a time (SVQ1)
+
+ /**
+ * XVideo Motion Acceleration
+ * - encoding: forbidden
+ * - decoding: set by decoder
+ */
+ int xvmc_acceleration;
+
+ /**
+ * macroblock decision mode
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int mb_decision;
+#define FF_MB_DECISION_SIMPLE 0 ///< uses mb_cmp
+#define FF_MB_DECISION_BITS 1 ///< chooses the one which needs the fewest bits
+#define FF_MB_DECISION_RD 2 ///< rate distortion
+
+ /**
+ * custom intra quantization matrix
+ * - encoding: Set by user, can be NULL.
+ * - decoding: Set by libavcodec.
+ */
+ uint16_t *intra_matrix;
+
+ /**
+ * custom inter quantization matrix
+ * - encoding: Set by user, can be NULL.
+ * - decoding: Set by libavcodec.
+ */
+ uint16_t *inter_matrix;
+
+ /**
+ * fourcc from the AVI stream header (LSB first, so "ABCD" -> ('D'<<24) + ('C'<<16) + ('B'<<8) + 'A').
+ * This is used to work around some encoder bugs.
+ * - encoding: unused
+ * - decoding: Set by user, will be converted to uppercase by libavcodec during init.
+ */
+ unsigned int stream_codec_tag;
+
+ /**
+ * scene change detection threshold
+ * 0 is default, larger means fewer detected scene changes.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int scenechange_threshold;
+
+ /**
+ * minimum Lagrange multipler
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int lmin;
+
+ /**
+ * maximum Lagrange multipler
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int lmax;
+
+ /**
+ * palette control structure
+ * - encoding: ??? (no palette-enabled encoder yet)
+ * - decoding: Set by user.
+ */
+ struct AVPaletteControl *palctrl;
+
+ /**
+ * noise reduction strength
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int noise_reduction;
+
+ /**
+ * Called at the beginning of a frame to get cr buffer for it.
+ * Buffer type (size, hints) must be the same. libavcodec won't check it.
+ * libavcodec will pass previous buffer in pic, function should return
+ * same buffer or new buffer with old frame "painted" into it.
+ * If pic.data[0] == NULL must behave like get_buffer().
+ * if CODEC_CAP_DR1 is not set then reget_buffer() must call
+ * avcodec_default_reget_buffer() instead of providing buffers allocated by
+ * some other means.
+ * - encoding: unused
+ * - decoding: Set by libavcodec, user can override.
+ */
+ int (*reget_buffer)(struct AVCodecContext *c, AVFrame *pic);
+
+ /**
+ * Number of bits which should be loaded into the rc buffer before decoding starts.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int rc_initial_buffer_occupancy;
+
+ /**
+ *
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int inter_threshold;
+
+ /**
+ * CODEC_FLAG2_*
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ */
+ int flags2;
+
+ /**
+ * Simulates errors in the bitstream to test error concealment.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int error_rate;
+
+ /**
+ * MP3 antialias algorithm, see FF_AA_* below.
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int antialias_algo;
+#define FF_AA_AUTO 0
+#define FF_AA_FASTINT 1 //not implemented yet
+#define FF_AA_INT 2
+#define FF_AA_FLOAT 3
+ /**
+ * quantizer noise shaping
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int quantizer_noise_shaping;
+
+ /**
+ * thread count
+ * is used to decide how many independent tasks should be passed to execute()
+ * - encoding: Set by user.
+ * - decoding: Set by user.
+ */
+ int thread_count;
+
+ /**
+ * The codec may call this to execute several independent things.
+ * It will return only after finishing all tasks.
+ * The user may replace this with some multithreaded implementation,
+ * the default implementation will execute the parts serially.
+ * @param count the number of things to execute
+ * - encoding: Set by libavcodec, user can override.
+ * - decoding: Set by libavcodec, user can override.
+ */
+ int (*execute)(struct AVCodecContext *c, int (*func)(struct AVCodecContext *c2, void *arg), void *arg2, int *ret, int count, int size);
+
+ /**
+ * thread opaque
+ * Can be used by execute() to store some per AVCodecContext stuff.
+ * - encoding: set by execute()
+ * - decoding: set by execute()
+ */
+ void *thread_opaque;
+
+ /**
+ * Motion estimation threshold below which no motion estimation is
+ * performed, but instead the user specified motion vectors are used.
+ *
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int me_threshold;
+
+ /**
+ * Macroblock threshold below which the user specified macroblock types will be used.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int mb_threshold;
+
+ /**
+ * precision of the intra DC coefficient - 8
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int intra_dc_precision;
+
+ /**
+ * noise vs. sse weight for the nsse comparsion function
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int nsse_weight;
+
+ /**
+ * Number of macroblock rows at the top which are skipped.
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int skip_top;
+
+ /**
+ * Number of macroblock rows at the bottom which are skipped.
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int skip_bottom;
+
+ /**
+ * profile
+ * - encoding: Set by user.
+ * - decoding: Set by libavcodec.
+ */
+ int profile;
+#define FF_PROFILE_UNKNOWN -99
+
+#define FF_PROFILE_AAC_MAIN 0
+#define FF_PROFILE_AAC_LOW 1
+#define FF_PROFILE_AAC_SSR 2
+#define FF_PROFILE_AAC_LTP 3
+
+#define FF_PROFILE_H264_BASELINE 66
+#define FF_PROFILE_H264_MAIN 77
+#define FF_PROFILE_H264_EXTENDED 88
+#define FF_PROFILE_H264_HIGH 100
+#define FF_PROFILE_H264_HIGH_10 110
+#define FF_PROFILE_H264_HIGH_422 122
+#define FF_PROFILE_H264_HIGH_444 244
+#define FF_PROFILE_H264_CAVLC_444 44
+
+ /**
+ * level
+ * - encoding: Set by user.
+ * - decoding: Set by libavcodec.
+ */
+ int level;
+#define FF_LEVEL_UNKNOWN -99
+
+ /**
+ * low resolution decoding, 1-> 1/2 size, 2->1/4 size
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int lowres;
+
+ /**
+ * Bitstream width / height, may be different from width/height if lowres
+ * or other things are used.
+ * - encoding: unused
+ * - decoding: Set by user before init if known. Codec should override / dynamically change if needed.
+ */
+ int coded_width, coded_height;
+
+ /**
+ * frame skip threshold
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int frame_skip_threshold;
+
+ /**
+ * frame skip factor
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int frame_skip_factor;
+
+ /**
+ * frame skip exponent
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int frame_skip_exp;
+
+ /**
+ * frame skip comparison function
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int frame_skip_cmp;
+
+ /**
+ * Border processing masking, raises the quantizer for mbs on the borders
+ * of the picture.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float border_masking;
+
+ /**
+ * minimum MB lagrange multipler
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int mb_lmin;
+
+ /**
+ * maximum MB lagrange multipler
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int mb_lmax;
+
+ /**
+ *
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int me_penalty_compensation;
+
+ /**
+ *
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ enum AVDiscard skip_loop_filter;
+
+ /**
+ *
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ enum AVDiscard skip_idct;
+
+ /**
+ *
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ enum AVDiscard skip_frame;
+
+ /**
+ *
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int bidir_refine;
+
+ /**
+ *
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int brd_scale;
+
+ /**
+ * constant rate factor - quality-based VBR - values ~correspond to qps
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float crf;
+
+ /**
+ * constant quantization parameter rate control method
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int cqp;
+
+ /**
+ * minimum GOP size
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int keyint_min;
+
+ /**
+ * number of reference frames
+ * - encoding: Set by user.
+ * - decoding: Set by lavc.
+ */
+ int refs;
+
+ /**
+ * chroma qp offset from luma
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int chromaoffset;
+
+ /**
+ * Influences how often B-frames are used.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int bframebias;
+
+ /**
+ * trellis RD quantization
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int trellis;
+
+ /**
+ * Reduce fluctuations in qp (before curve compression).
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float complexityblur;
+
+ /**
+ * in-loop deblocking filter alphac0 parameter
+ * alpha is in the range -6...6
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int deblockalpha;
+
+ /**
+ * in-loop deblocking filter beta parameter
+ * beta is in the range -6...6
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int deblockbeta;
+
+ /**
+ * macroblock subpartition sizes to consider - p8x8, p4x4, b8x8, i8x8, i4x4
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int partitions;
+#define X264_PART_I4X4 0x001 /* Analyze i4x4 */
+#define X264_PART_I8X8 0x002 /* Analyze i8x8 (requires 8x8 transform) */
+#define X264_PART_P8X8 0x010 /* Analyze p16x8, p8x16 and p8x8 */
+#define X264_PART_P4X4 0x020 /* Analyze p8x4, p4x8, p4x4 */
+#define X264_PART_B8X8 0x100 /* Analyze b16x8, b8x16 and b8x8 */
+
+ /**
+ * direct MV prediction mode - 0 (none), 1 (spatial), 2 (temporal), 3 (auto)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int directpred;
+
+ /**
+ * Audio cutoff bandwidth (0 means "automatic")
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int cutoff;
+
+ /**
+ * Multiplied by qscale for each frame and added to scene_change_score.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int scenechange_factor;
+
+ /**
+ *
+ * Note: Value depends upon the compare function used for fullpel ME.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int mv0_threshold;
+
+ /**
+ * Adjusts sensitivity of b_frame_strategy 1.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int b_sensitivity;
+
+ /**
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int compression_level;
+#define FF_COMPRESSION_DEFAULT -1
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+ /**
+ * Sets whether to use LPC mode - used by FLAC encoder.
+ * - encoding: Set by user.
+ * - decoding: unused
+ * @deprecated Deprecated in favor of lpc_type and lpc_passes.
+ */
+ int use_lpc;
+#endif
+
+ /**
+ * LPC coefficient precision - used by FLAC encoder
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int lpc_coeff_precision;
+
+ /**
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int min_prediction_order;
+
+ /**
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int max_prediction_order;
+
+ /**
+ * search method for selecting prediction order
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int prediction_order_method;
+
+ /**
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int min_partition_order;
+
+ /**
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int max_partition_order;
+
+ /**
+ * GOP timecode frame start number, in non drop frame format
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int64_t timecode_frame_start;
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+ /**
+ * Decoder should decode to this many channels if it can (0 for default)
+ * - encoding: unused
+ * - decoding: Set by user.
+ * @deprecated Deprecated in favor of request_channel_layout.
+ */
+ int request_channels;
+#endif
+
+ /**
+ * Percentage of dynamic range compression to be applied by the decoder.
+ * The default value is 1.0, corresponding to full compression.
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ float drc_scale;
+
+ /**
+ * opaque 64bit number (generally a PTS) that will be reordered and
+ * output in AVFrame.reordered_opaque
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int64_t reordered_opaque;
+
+ /**
+ * Bits per sample/pixel of internal libavcodec pixel/sample format.
+ * This field is applicable only when sample_fmt is SAMPLE_FMT_S32.
+ * - encoding: set by user.
+ * - decoding: set by libavcodec.
+ */
+ int bits_per_raw_sample;
+
+ /**
+ * Audio channel layout.
+ * - encoding: set by user.
+ * - decoding: set by libavcodec.
+ */
+ int64_t channel_layout;
+
+ /**
+ * Request decoder to use this channel layout if it can (0 for default)
+ * - encoding: unused
+ * - decoding: Set by user.
+ */
+ int64_t request_channel_layout;
+
+ /**
+ * Ratecontrol attempt to use, at maximum, <value> of what can be used without an underflow.
+ * - encoding: Set by user.
+ * - decoding: unused.
+ */
+ float rc_max_available_vbv_use;
+
+ /**
+ * Ratecontrol attempt to use, at least, <value> times the amount needed to prevent a vbv overflow.
+ * - encoding: Set by user.
+ * - decoding: unused.
+ */
+ float rc_min_vbv_overflow_use;
+
+ /**
+ * Hardware accelerator in use
+ * - encoding: unused.
+ * - decoding: Set by libavcodec
+ */
+ struct AVHWAccel *hwaccel;
+
+ /**
+ * For some codecs, the time base is closer to the field rate than the frame rate.
+ * Most notably, H.264 and MPEG-2 specify time_base as half of frame duration
+ * if no telecine is used ...
+ *
+ * Set to time_base ticks per frame. Default 1, e.g., H.264/MPEG-2 set it to 2.
+ */
+ int ticks_per_frame;
+
+ /**
+ * Hardware accelerator context.
+ * For some hardware accelerators, a global context needs to be
+ * provided by the user. In that case, this holds display-dependent
+ * data FFmpeg cannot instantiate itself. Please refer to the
+ * FFmpeg HW accelerator documentation to know how to fill this
+ * is. e.g. for VA API, this is a struct vaapi_context.
+ * - encoding: unused
+ * - decoding: Set by user
+ */
+ void *hwaccel_context;
+
+ /**
+ * Chromaticity coordinates of the source primaries.
+ * - encoding: Set by user
+ * - decoding: Set by libavcodec
+ */
+ enum AVColorPrimaries color_primaries;
+
+ /**
+ * Color Transfer Characteristic.
+ * - encoding: Set by user
+ * - decoding: Set by libavcodec
+ */
+ enum AVColorTransferCharacteristic color_trc;
+
+ /**
+ * YUV colorspace type.
+ * - encoding: Set by user
+ * - decoding: Set by libavcodec
+ */
+ enum AVColorSpace colorspace;
+
+ /**
+ * MPEG vs JPEG YUV range.
+ * - encoding: Set by user
+ * - decoding: Set by libavcodec
+ */
+ enum AVColorRange color_range;
+
+ /**
+ * This defines the location of chroma samples.
+ * - encoding: Set by user
+ * - decoding: Set by libavcodec
+ */
+ enum AVChromaLocation chroma_sample_location;
+
+ /**
+ * The codec may call this to execute several independent things.
+ * It will return only after finishing all tasks.
+ * The user may replace this with some multithreaded implementation,
+ * the default implementation will execute the parts serially.
+ * Also see avcodec_thread_init and e.g. the --enable-pthread configure option.
+ * @param c context passed also to func
+ * @param count the number of things to execute
+ * @param arg2 argument passed unchanged to func
+ * @param ret return values of executed functions, must have space for "count" values. May be NULL.
+ * @param func function that will be called count times, with jobnr from 0 to count-1.
+ * threadnr will be in the range 0 to c->thread_count-1 < MAX_THREADS and so that no
+ * two instances of func executing at the same time will have the same threadnr.
+ * @return always 0 currently, but code should handle a future improvement where when any call to func
+ * returns < 0 no further calls to func may be done and < 0 is returned.
+ * - encoding: Set by libavcodec, user can override.
+ * - decoding: Set by libavcodec, user can override.
+ */
+ int (*execute2)(struct AVCodecContext *c, int (*func)(struct AVCodecContext *c2, void *arg, int jobnr, int threadnr), void *arg2, int *ret, int count);
+
+ /**
+ * explicit P-frame weighted prediction analysis method
+ * 0: off
+ * 1: fast blind weighting (one reference duplicate with -1 offset)
+ * 2: smart weighting (full fade detection analysis)
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ int weighted_p_pred;
+
+ /**
+ * AQ mode
+ * 0: Disabled
+ * 1: Variance AQ (complexity mask)
+ * 2: Auto-variance AQ (experimental)
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ int aq_mode;
+
+ /**
+ * AQ strength
+ * Reduces blocking and blurring in flat and textured areas.
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ float aq_strength;
+
+ /**
+ * PSY RD
+ * Strength of psychovisual optimization
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ float psy_rd;
+
+ /**
+ * PSY trellis
+ * Strength of psychovisual optimization
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ float psy_trellis;
+
+ /**
+ * RC lookahead
+ * Number of frames for frametype and ratecontrol lookahead
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ int rc_lookahead;
+
+ /**
+ * Constant rate factor maximum
+ * With CRF encoding mode and VBV restrictions enabled, prevents quality from being worse
+ * than crf_max, even if doing so would violate VBV restrictions.
+ * - encoding: Set by user.
+ * - decoding: unused
+ */
+ float crf_max;
+
+ int log_level_offset;
+
+ /**
+ * Determines which LPC analysis algorithm to use.
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ enum AVLPCType lpc_type;
+
+ /**
+ * Number of passes to use for Cholesky factorization during LPC analysis
+ * - encoding: Set by user
+ * - decoding: unused
+ */
+ int lpc_passes;
+} AVCodecContext;
+
+/**
+ * AVCodec.
+ */
+typedef struct AVCodec {
+ /**
+ * Name of the codec implementation.
+ * The name is globally unique among encoders and among decoders (but an
+ * encoder and a decoder can share the same name).
+ * This is the primary way to find a codec from the user perspective.
+ */
+ const char *name;
+ enum AVMediaType type;
+ enum CodecID id;
+ int priv_data_size;
+ int (*init)(AVCodecContext *);
+ int (*encode)(AVCodecContext *, uint8_t *buf, int buf_size, void *data);
+ int (*close)(AVCodecContext *);
+ int (*decode)(AVCodecContext *, void *outdata, int *outdata_size, AVPacket *avpkt);
+ /**
+ * Codec capabilities.
+ * see CODEC_CAP_*
+ */
+ int capabilities;
+ struct AVCodec *next;
+ /**
+ * Flush buffers.
+ * Will be called when seeking
+ */
+ void (*flush)(AVCodecContext *);
+ const AVRational *supported_framerates; ///< array of supported framerates, or NULL if any, array is terminated by {0,0}
+ const enum PixelFormat *pix_fmts; ///< array of supported pixel formats, or NULL if unknown, array is terminated by -1
+ /**
+ * Descriptive name for the codec, meant to be more human readable than name.
+ * You should use the NULL_IF_CONFIG_SMALL() macro to define it.
+ */
+ const char *long_name;
+ const int *supported_samplerates; ///< array of supported audio samplerates, or NULL if unknown, array is terminated by 0
+ const enum SampleFormat *sample_fmts; ///< array of supported sample formats, or NULL if unknown, array is terminated by -1
+ const int64_t *channel_layouts; ///< array of support channel layouts, or NULL if unknown. array is terminated by 0
+ uint8_t max_lowres; ///< maximum value for lowres supported by the decoder
+} AVCodec;
+
+/**
+ * AVHWAccel.
+ */
+typedef struct AVHWAccel {
+ /**
+ * Name of the hardware accelerated codec.
+ * The name is globally unique among encoders and among decoders (but an
+ * encoder and a decoder can share the same name).
+ */
+ const char *name;
+
+ /**
+ * Type of codec implemented by the hardware accelerator.
+ *
+ * See AVMEDIA_TYPE_xxx
+ */
+ enum AVMediaType type;
+
+ /**
+ * Codec implemented by the hardware accelerator.
+ *
+ * See CODEC_ID_xxx
+ */
+ enum CodecID id;
+
+ /**
+ * Supported pixel format.
+ *
+ * Only hardware accelerated formats are supported here.
+ */
+ enum PixelFormat pix_fmt;
+
+ /**
+ * Hardware accelerated codec capabilities.
+ * see FF_HWACCEL_CODEC_CAP_*
+ */
+ int capabilities;
+
+ struct AVHWAccel *next;
+
+ /**
+ * Called at the beginning of each frame or field picture.
+ *
+ * Meaningful frame information (codec specific) is guaranteed to
+ * be parsed at this point. This function is mandatory.
+ *
+ * Note that buf can be NULL along with buf_size set to 0.
+ * Otherwise, this means the whole frame is available at this point.
+ *
+ * @param avctx the codec context
+ * @param buf the frame data buffer base
+ * @param buf_size the size of the frame in bytes
+ * @return zero if successful, a negative value otherwise
+ */
+ int (*start_frame)(AVCodecContext *avctx, const uint8_t *buf, uint32_t buf_size);
+
+ /**
+ * Callback for each slice.
+ *
+ * Meaningful slice information (codec specific) is guaranteed to
+ * be parsed at this point. This function is mandatory.
+ *
+ * @param avctx the codec context
+ * @param buf the slice data buffer base
+ * @param buf_size the size of the slice in bytes
+ * @return zero if successful, a negative value otherwise
+ */
+ int (*decode_slice)(AVCodecContext *avctx, const uint8_t *buf, uint32_t buf_size);
+
+ /**
+ * Called at the end of each frame or field picture.
+ *
+ * The whole picture is parsed at this point and can now be sent
+ * to the hardware accelerator. This function is mandatory.
+ *
+ * @param avctx the codec context
+ * @return zero if successful, a negative value otherwise
+ */
+ int (*end_frame)(AVCodecContext *avctx);
+
+ /**
+ * Size of HW accelerator private data.
+ *
+ * Private data is allocated with av_mallocz() before
+ * AVCodecContext.get_buffer() and deallocated after
+ * AVCodecContext.release_buffer().
+ */
+ int priv_data_size;
+} AVHWAccel;
+
+/**
+ * four components are given, that's all.
+ * the last component is alpha
+ */
+typedef struct AVPicture {
+ uint8_t *data[4];
+ int linesize[4]; ///< number of bytes per line
+} AVPicture;
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * AVPaletteControl
+ * This structure defines a method for communicating palette changes
+ * between and demuxer and a decoder.
+ *
+ * @deprecated Use AVPacket to send palette changes instead.
+ * This is totally broken.
+ */
+#define AVPALETTE_SIZE 1024
+#define AVPALETTE_COUNT 256
+typedef struct AVPaletteControl {
+
+ /* Demuxer sets this to 1 to indicate the palette has changed;
+ * decoder resets to 0. */
+ int palette_changed;
+
+ /* 4-byte ARGB palette entries, stored in native byte order; note that
+ * the individual palette components should be on a 8-bit scale; if
+ * the palette data comes from an IBM VGA native format, the component
+ * data is probably 6 bits in size and needs to be scaled. */
+ unsigned int palette[AVPALETTE_COUNT];
+
+} AVPaletteControl attribute_deprecated;
+#endif
+
+enum AVSubtitleType {
+ SUBTITLE_NONE,
+
+ SUBTITLE_BITMAP, ///< A bitmap, pict will be set
+
+ /**
+ * Plain text, the text field must be set by the decoder and is
+ * authoritative. ass and pict fields may contain approximations.
+ */
+ SUBTITLE_TEXT,
+
+ /**
+ * Formatted text, the ass field must be set by the decoder and is
+ * authoritative. pict and text fields may contain approximations.
+ */
+ SUBTITLE_ASS,
+};
+
+typedef struct AVSubtitleRect {
+ int x; ///< top left corner of pict, undefined when pict is not set
+ int y; ///< top left corner of pict, undefined when pict is not set
+ int w; ///< width of pict, undefined when pict is not set
+ int h; ///< height of pict, undefined when pict is not set
+ int nb_colors; ///< number of colors in pict, undefined when pict is not set
+
+ /**
+ * data+linesize for the bitmap of this subtitle.
+ * can be set for text/ass as well once they where rendered
+ */
+ AVPicture pict;
+ enum AVSubtitleType type;
+
+ char *text; ///< 0 terminated plain UTF-8 text
+
+ /**
+ * 0 terminated ASS/SSA compatible event line.
+ * The pressentation of this is unaffected by the other values in this
+ * struct.
+ */
+ char *ass;
+} AVSubtitleRect;
+
+typedef struct AVSubtitle {
+ uint16_t format; /* 0 = graphics */
+ uint32_t start_display_time; /* relative to packet pts, in ms */
+ uint32_t end_display_time; /* relative to packet pts, in ms */
+ unsigned num_rects;
+ AVSubtitleRect **rects;
+ int64_t pts; ///< Same as packet pts, in AV_TIME_BASE
+} AVSubtitle;
+
+/* packet functions */
+
+/**
+ * @deprecated use NULL instead
+ */
+attribute_deprecated void av_destruct_packet_nofree(AVPacket *pkt);
+
+/**
+ * Default packet destructor.
+ */
+void av_destruct_packet(AVPacket *pkt);
+
+/**
+ * Initialize optional fields of a packet with default values.
+ *
+ * @param pkt packet
+ */
+void av_init_packet(AVPacket *pkt);
+
+/**
+ * Allocate the payload of a packet and initialize its fields with
+ * default values.
+ *
+ * @param pkt packet
+ * @param size wanted payload size
+ * @return 0 if OK, AVERROR_xxx otherwise
+ */
+int av_new_packet(AVPacket *pkt, int size);
+
+/**
+ * Reduce packet size, correctly zeroing padding
+ *
+ * @param pkt packet
+ * @param size new size
+ */
+void av_shrink_packet(AVPacket *pkt, int size);
+
+/**
+ * @warning This is a hack - the packet memory allocation stuff is broken. The
+ * packet is allocated if it was not really allocated.
+ */
+int av_dup_packet(AVPacket *pkt);
+
+/**
+ * Free a packet.
+ *
+ * @param pkt packet to free
+ */
+void av_free_packet(AVPacket *pkt);
+
+/* resample.c */
+
+struct ReSampleContext;
+struct AVResampleContext;
+
+typedef struct ReSampleContext ReSampleContext;
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * @deprecated Use av_audio_resample_init() instead.
+ */
+attribute_deprecated ReSampleContext *audio_resample_init(int output_channels, int input_channels,
+ int output_rate, int input_rate);
+#endif
+/**
+ * Initialize audio resampling context
+ *
+ * @param output_channels number of output channels
+ * @param input_channels number of input channels
+ * @param output_rate output sample rate
+ * @param input_rate input sample rate
+ * @param sample_fmt_out requested output sample format
+ * @param sample_fmt_in input sample format
+ * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
+ * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
+ * @param linear If 1 then the used FIR filter will be linearly interpolated
+ between the 2 closest, if 0 the closest will be used
+ * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
+ * @return allocated ReSampleContext, NULL if error occured
+ */
+ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
+ int output_rate, int input_rate,
+ enum SampleFormat sample_fmt_out,
+ enum SampleFormat sample_fmt_in,
+ int filter_length, int log2_phase_count,
+ int linear, double cutoff);
+
+int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
+void audio_resample_close(ReSampleContext *s);
+
+
+/**
+ * Initialize an audio resampler.
+ * Note, if either rate is not an integer then simply scale both rates up so they are.
+ * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
+ * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
+ * @param linear If 1 then the used FIR filter will be linearly interpolated
+ between the 2 closest, if 0 the closest will be used
+ * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
+ */
+struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff);
+
+/**
+ * Resample an array of samples using a previously configured context.
+ * @param src an array of unconsumed samples
+ * @param consumed the number of samples of src which have been consumed are returned here
+ * @param src_size the number of unconsumed samples available
+ * @param dst_size the amount of space in samples available in dst
+ * @param update_ctx If this is 0 then the context will not be modified, that way several channels can be resampled with the same context.
+ * @return the number of samples written in dst or -1 if an error occurred
+ */
+int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
+
+
+/**
+ * Compensate samplerate/timestamp drift. The compensation is done by changing
+ * the resampler parameters, so no audible clicks or similar distortions occur
+ * @param compensation_distance distance in output samples over which the compensation should be performed
+ * @param sample_delta number of output samples which should be output less
+ *
+ * example: av_resample_compensate(c, 10, 500)
+ * here instead of 510 samples only 500 samples would be output
+ *
+ * note, due to rounding the actual compensation might be slightly different,
+ * especially if the compensation_distance is large and the in_rate used during init is small
+ */
+void av_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
+void av_resample_close(struct AVResampleContext *c);
+
+/**
+ * Allocate memory for a picture. Call avpicture_free to free it.
+ *
+ * @param picture the picture to be filled in
+ * @param pix_fmt the format of the picture
+ * @param width the width of the picture
+ * @param height the height of the picture
+ * @return zero if successful, a negative value if not
+ */
+int avpicture_alloc(AVPicture *picture, enum PixelFormat pix_fmt, int width, int height);
+
+/**
+ * Free a picture previously allocated by avpicture_alloc().
+ *
+ * @param picture the AVPicture to be freed
+ */
+void avpicture_free(AVPicture *picture);
+
+/**
+ * Fill in the AVPicture fields.
+ * The fields of the given AVPicture are filled in by using the 'ptr' address
+ * which points to the image data buffer. Depending on the specified picture
+ * format, one or multiple image data pointers and line sizes will be set.
+ * If a planar format is specified, several pointers will be set pointing to
+ * the different picture planes and the line sizes of the different planes
+ * will be stored in the lines_sizes array.
+ * Call with ptr == NULL to get the required size for the ptr buffer.
+ *
+ * @param picture AVPicture whose fields are to be filled in
+ * @param ptr Buffer which will contain or contains the actual image data
+ * @param pix_fmt The format in which the picture data is stored.
+ * @param width the width of the image in pixels
+ * @param height the height of the image in pixels
+ * @return size of the image data in bytes
+ */
+int avpicture_fill(AVPicture *picture, uint8_t *ptr,
+ enum PixelFormat pix_fmt, int width, int height);
+int avpicture_layout(const AVPicture* src, enum PixelFormat pix_fmt, int width, int height,
+ unsigned char *dest, int dest_size);
+
+/**
+ * Calculate the size in bytes that a picture of the given width and height
+ * would occupy if stored in the given picture format.
+ * Note that this returns the size of a compact representation as generated
+ * by avpicture_layout, which can be smaller than the size required for e.g.
+ * avpicture_fill.
+ *
+ * @param pix_fmt the given picture format
+ * @param width the width of the image
+ * @param height the height of the image
+ * @return Image data size in bytes or -1 on error (e.g. too large dimensions).
+ */
+int avpicture_get_size(enum PixelFormat pix_fmt, int width, int height);
+void avcodec_get_chroma_sub_sample(enum PixelFormat pix_fmt, int *h_shift, int *v_shift);
+const char *avcodec_get_pix_fmt_name(enum PixelFormat pix_fmt);
+void avcodec_set_dimensions(AVCodecContext *s, int width, int height);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * Return the pixel format corresponding to the name name.
+ *
+ * If there is no pixel format with name name, then look for a
+ * pixel format with the name corresponding to the native endian
+ * format of name.
+ * For example in a little-endian system, first look for "gray16",
+ * then for "gray16le".
+ *
+ * Finally if no pixel format has been found, return PIX_FMT_NONE.
+ *
+ * @deprecated Deprecated in favor of av_get_pix_fmt().
+ */
+attribute_deprecated enum PixelFormat avcodec_get_pix_fmt(const char* name);
+#endif
+
+/**
+ * Return a value representing the fourCC code associated to the
+ * pixel format pix_fmt, or 0 if no associated fourCC code can be
+ * found.
+ */
+unsigned int avcodec_pix_fmt_to_codec_tag(enum PixelFormat pix_fmt);
+
+/**
+ * Put a string representing the codec tag codec_tag in buf.
+ *
+ * @param buf_size size in bytes of buf
+ * @return the length of the string that would have been generated if
+ * enough space had been available, excluding the trailing null
+ */
+size_t av_get_codec_tag_string(char *buf, size_t buf_size, unsigned int codec_tag);
+
+#define FF_LOSS_RESOLUTION 0x0001 /**< loss due to resolution change */
+#define FF_LOSS_DEPTH 0x0002 /**< loss due to color depth change */
+#define FF_LOSS_COLORSPACE 0x0004 /**< loss due to color space conversion */
+#define FF_LOSS_ALPHA 0x0008 /**< loss of alpha bits */
+#define FF_LOSS_COLORQUANT 0x0010 /**< loss due to color quantization */
+#define FF_LOSS_CHROMA 0x0020 /**< loss of chroma (e.g. RGB to gray conversion) */
+
+/**
+ * Compute what kind of losses will occur when converting from one specific
+ * pixel format to another.
+ * When converting from one pixel format to another, information loss may occur.
+ * For example, when converting from RGB24 to GRAY, the color information will
+ * be lost. Similarly, other losses occur when converting from some formats to
+ * other formats. These losses can involve loss of chroma, but also loss of
+ * resolution, loss of color depth, loss due to the color space conversion, loss
+ * of the alpha bits or loss due to color quantization.
+ * avcodec_get_fix_fmt_loss() informs you about the various types of losses
+ * which will occur when converting from one pixel format to another.
+ *
+ * @param[in] dst_pix_fmt destination pixel format
+ * @param[in] src_pix_fmt source pixel format
+ * @param[in] has_alpha Whether the source pixel format alpha channel is used.
+ * @return Combination of flags informing you what kind of losses will occur.
+ */
+int avcodec_get_pix_fmt_loss(enum PixelFormat dst_pix_fmt, enum PixelFormat src_pix_fmt,
+ int has_alpha);
+
+/**
+ * Find the best pixel format to convert to given a certain source pixel
+ * format. When converting from one pixel format to another, information loss
+ * may occur. For example, when converting from RGB24 to GRAY, the color
+ * information will be lost. Similarly, other losses occur when converting from
+ * some formats to other formats. avcodec_find_best_pix_fmt() searches which of
+ * the given pixel formats should be used to suffer the least amount of loss.
+ * The pixel formats from which it chooses one, are determined by the
+ * pix_fmt_mask parameter.
+ *
+ * @code
+ * src_pix_fmt = PIX_FMT_YUV420P;
+ * pix_fmt_mask = (1 << PIX_FMT_YUV422P) || (1 << PIX_FMT_RGB24);
+ * dst_pix_fmt = avcodec_find_best_pix_fmt(pix_fmt_mask, src_pix_fmt, alpha, &loss);
+ * @endcode
+ *
+ * @param[in] pix_fmt_mask bitmask determining which pixel format to choose from
+ * @param[in] src_pix_fmt source pixel format
+ * @param[in] has_alpha Whether the source pixel format alpha channel is used.
+ * @param[out] loss_ptr Combination of flags informing you what kind of losses will occur.
+ * @return The best pixel format to convert to or -1 if none was found.
+ */
+enum PixelFormat avcodec_find_best_pix_fmt(int64_t pix_fmt_mask, enum PixelFormat src_pix_fmt,
+ int has_alpha, int *loss_ptr);
+
+
+/**
+ * Print in buf the string corresponding to the pixel format with
+ * number pix_fmt, or an header if pix_fmt is negative.
+ *
+ * @param[in] buf the buffer where to write the string
+ * @param[in] buf_size the size of buf
+ * @param[in] pix_fmt the number of the pixel format to print the corresponding info string, or
+ * a negative value to print the corresponding header.
+ * Meaningful values for obtaining a pixel format info vary from 0 to PIX_FMT_NB -1.
+ */
+void avcodec_pix_fmt_string (char *buf, int buf_size, enum PixelFormat pix_fmt);
+
+#define FF_ALPHA_TRANSP 0x0001 /* image has some totally transparent pixels */
+#define FF_ALPHA_SEMI_TRANSP 0x0002 /* image has some transparent pixels */
+
+/**
+ * Tell if an image really has transparent alpha values.
+ * @return ored mask of FF_ALPHA_xxx constants
+ */
+int img_get_alpha_info(const AVPicture *src,
+ enum PixelFormat pix_fmt, int width, int height);
+
+/* deinterlace a picture */
+/* deinterlace - if not supported return -1 */
+int avpicture_deinterlace(AVPicture *dst, const AVPicture *src,
+ enum PixelFormat pix_fmt, int width, int height);
+
+/* external high level API */
+
+/**
+ * If c is NULL, returns the first registered codec,
+ * if c is non-NULL, returns the next registered codec after c,
+ * or NULL if c is the last one.
+ */
+AVCodec *av_codec_next(AVCodec *c);
+
+/**
+ * Return the LIBAVCODEC_VERSION_INT constant.
+ */
+unsigned avcodec_version(void);
+
+/**
+ * Return the libavcodec build-time configuration.
+ */
+const char *avcodec_configuration(void);
+
+/**
+ * Return the libavcodec license.
+ */
+const char *avcodec_license(void);
+
+/**
+ * Initialize libavcodec.
+ *
+ * @warning This function must be called before any other libavcodec
+ * function.
+ */
+void avcodec_init(void);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * @deprecated Deprecated in favor of avcodec_register().
+ */
+attribute_deprecated void register_avcodec(AVCodec *codec);
+#endif
+
+/**
+ * Register the codec codec and initialize libavcodec.
+ *
+ * @see avcodec_init()
+ */
+void avcodec_register(AVCodec *codec);
+
+/**
+ * Find a registered encoder with a matching codec ID.
+ *
+ * @param id CodecID of the requested encoder
+ * @return An encoder if one was found, NULL otherwise.
+ */
+AVCodec *avcodec_find_encoder(enum CodecID id);
+
+/**
+ * Find a registered encoder with the specified name.
+ *
+ * @param name name of the requested encoder
+ * @return An encoder if one was found, NULL otherwise.
+ */
+AVCodec *avcodec_find_encoder_by_name(const char *name);
+
+/**
+ * Find a registered decoder with a matching codec ID.
+ *
+ * @param id CodecID of the requested decoder
+ * @return A decoder if one was found, NULL otherwise.
+ */
+AVCodec *avcodec_find_decoder(enum CodecID id);
+
+/**
+ * Find a registered decoder with the specified name.
+ *
+ * @param name name of the requested decoder
+ * @return A decoder if one was found, NULL otherwise.
+ */
+AVCodec *avcodec_find_decoder_by_name(const char *name);
+void avcodec_string(char *buf, int buf_size, AVCodecContext *enc, int encode);
+
+/**
+ * Set the fields of the given AVCodecContext to default values.
+ *
+ * @param s The AVCodecContext of which the fields should be set to default values.
+ */
+void avcodec_get_context_defaults(AVCodecContext *s);
+
+/** THIS FUNCTION IS NOT YET PART OF THE PUBLIC API!
+ * we WILL change its arguments and name a few times! */
+void avcodec_get_context_defaults2(AVCodecContext *s, enum AVMediaType);
+
+/**
+ * Allocate an AVCodecContext and set its fields to default values. The
+ * resulting struct can be deallocated by simply calling av_free().
+ *
+ * @return An AVCodecContext filled with default values or NULL on failure.
+ * @see avcodec_get_context_defaults
+ */
+AVCodecContext *avcodec_alloc_context(void);
+
+/** THIS FUNCTION IS NOT YET PART OF THE PUBLIC API!
+ * we WILL change its arguments and name a few times! */
+AVCodecContext *avcodec_alloc_context2(enum AVMediaType);
+
+/**
+ * Copy the settings of the source AVCodecContext into the destination
+ * AVCodecContext. The resulting destination codec context will be
+ * unopened, i.e. you are required to call avcodec_open() before you
+ * can use this AVCodecContext to decode/encode video/audio data.
+ *
+ * @param dest target codec context, should be initialized with
+ * avcodec_alloc_context(), but otherwise uninitialized
+ * @param src source codec context
+ * @return AVERROR() on error (e.g. memory allocation error), 0 on success
+ */
+int avcodec_copy_context(AVCodecContext *dest, const AVCodecContext *src);
+
+/**
+ * Set the fields of the given AVFrame to default values.
+ *
+ * @param pic The AVFrame of which the fields should be set to default values.
+ */
+void avcodec_get_frame_defaults(AVFrame *pic);
+
+/**
+ * Allocate an AVFrame and set its fields to default values. The resulting
+ * struct can be deallocated by simply calling av_free().
+ *
+ * @return An AVFrame filled with default values or NULL on failure.
+ * @see avcodec_get_frame_defaults
+ */
+AVFrame *avcodec_alloc_frame(void);
+
+int avcodec_default_get_buffer(AVCodecContext *s, AVFrame *pic);
+void avcodec_default_release_buffer(AVCodecContext *s, AVFrame *pic);
+int avcodec_default_reget_buffer(AVCodecContext *s, AVFrame *pic);
+
+/**
+ * Return the amount of padding in pixels which the get_buffer callback must
+ * provide around the edge of the image for codecs which do not have the
+ * CODEC_FLAG_EMU_EDGE flag.
+ *
+ * @return Required padding in pixels.
+ */
+unsigned avcodec_get_edge_width(void);
+/**
+ * Modify width and height values so that they will result in a memory
+ * buffer that is acceptable for the codec if you do not use any horizontal
+ * padding.
+ *
+ * May only be used if a codec with CODEC_CAP_DR1 has been opened.
+ * If CODEC_FLAG_EMU_EDGE is not set, the dimensions must have been increased
+ * according to avcodec_get_edge_width() before.
+ */
+void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height);
+/**
+ * Modify width and height values so that they will result in a memory
+ * buffer that is acceptable for the codec if you also ensure that all
+ * line sizes are a multiple of the respective linesize_align[i].
+ *
+ * May only be used if a codec with CODEC_CAP_DR1 has been opened.
+ * If CODEC_FLAG_EMU_EDGE is not set, the dimensions must have been increased
+ * according to avcodec_get_edge_width() before.
+ */
+void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height,
+ int linesize_align[4]);
+
+/**
+ * Check if the given dimension of a picture is valid, meaning that all
+ * bytes of the picture can be addressed with a signed int.
+ *
+ * @param[in] w Width of the picture.
+ * @param[in] h Height of the picture.
+ * @return Zero if valid, a negative value if invalid.
+ */
+int avcodec_check_dimensions(void *av_log_ctx, unsigned int w, unsigned int h);
+enum PixelFormat avcodec_default_get_format(struct AVCodecContext *s, const enum PixelFormat * fmt);
+
+int avcodec_thread_init(AVCodecContext *s, int thread_count);
+void avcodec_thread_free(AVCodecContext *s);
+int avcodec_default_execute(AVCodecContext *c, int (*func)(AVCodecContext *c2, void *arg2),void *arg, int *ret, int count, int size);
+int avcodec_default_execute2(AVCodecContext *c, int (*func)(AVCodecContext *c2, void *arg2, int, int),void *arg, int *ret, int count);
+//FIXME func typedef
+
+/**
+ * Initialize the AVCodecContext to use the given AVCodec. Prior to using this
+ * function the context has to be allocated.
+ *
+ * The functions avcodec_find_decoder_by_name(), avcodec_find_encoder_by_name(),
+ * avcodec_find_decoder() and avcodec_find_encoder() provide an easy way for
+ * retrieving a codec.
+ *
+ * @warning This function is not thread safe!
+ *
+ * @code
+ * avcodec_register_all();
+ * codec = avcodec_find_decoder(CODEC_ID_H264);
+ * if (!codec)
+ * exit(1);
+ *
+ * context = avcodec_alloc_context();
+ *
+ * if (avcodec_open(context, codec) < 0)
+ * exit(1);
+ * @endcode
+ *
+ * @param avctx The context which will be set up to use the given codec.
+ * @param codec The codec to use within the context.
+ * @return zero on success, a negative value on error
+ * @see avcodec_alloc_context, avcodec_find_decoder, avcodec_find_encoder
+ */
+int avcodec_open(AVCodecContext *avctx, AVCodec *codec);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * Decode an audio frame from buf into samples.
+ * Wrapper function which calls avcodec_decode_audio3.
+ *
+ * @deprecated Use avcodec_decode_audio3 instead.
+ * @param avctx the codec context
+ * @param[out] samples the output buffer
+ * @param[in,out] frame_size_ptr the output buffer size in bytes
+ * @param[in] buf the input buffer
+ * @param[in] buf_size the input buffer size in bytes
+ * @return On error a negative value is returned, otherwise the number of bytes
+ * used or zero if no frame could be decompressed.
+ */
+attribute_deprecated int avcodec_decode_audio2(AVCodecContext *avctx, int16_t *samples,
+ int *frame_size_ptr,
+ const uint8_t *buf, int buf_size);
+#endif
+
+/**
+ * Decode the audio frame of size avpkt->size from avpkt->data into samples.
+ * Some decoders may support multiple frames in a single AVPacket, such
+ * decoders would then just decode the first frame. In this case,
+ * avcodec_decode_audio3 has to be called again with an AVPacket that contains
+ * the remaining data in order to decode the second frame etc.
+ * If no frame
+ * could be outputted, frame_size_ptr is zero. Otherwise, it is the
+ * decompressed frame size in bytes.
+ *
+ * @warning You must set frame_size_ptr to the allocated size of the
+ * output buffer before calling avcodec_decode_audio3().
+ *
+ * @warning The input buffer must be FF_INPUT_BUFFER_PADDING_SIZE larger than
+ * the actual read bytes because some optimized bitstream readers read 32 or 64
+ * bits at once and could read over the end.
+ *
+ * @warning The end of the input buffer avpkt->data should be set to 0 to ensure that
+ * no overreading happens for damaged MPEG streams.
+ *
+ * @note You might have to align the input buffer avpkt->data and output buffer
+ * samples. The alignment requirements depend on the CPU: On some CPUs it isn't
+ * necessary at all, on others it won't work at all if not aligned and on others
+ * it will work but it will have an impact on performance.
+ *
+ * In practice, avpkt->data should have 4 byte alignment at minimum and
+ * samples should be 16 byte aligned unless the CPU doesn't need it
+ * (AltiVec and SSE do).
+ *
+ * @param avctx the codec context
+ * @param[out] samples the output buffer, sample type in avctx->sample_fmt
+ * @param[in,out] frame_size_ptr the output buffer size in bytes
+ * @param[in] avpkt The input AVPacket containing the input buffer.
+ * You can create such packet with av_init_packet() and by then setting
+ * data and size, some decoders might in addition need other fields.
+ * All decoders are designed to use the least fields possible though.
+ * @return On error a negative value is returned, otherwise the number of bytes
+ * used or zero if no frame data was decompressed (used) from the input AVPacket.
+ */
+int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
+ int *frame_size_ptr,
+ AVPacket *avpkt);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * Decode a video frame from buf into picture.
+ * Wrapper function which calls avcodec_decode_video2.
+ *
+ * @deprecated Use avcodec_decode_video2 instead.
+ * @param avctx the codec context
+ * @param[out] picture The AVFrame in which the decoded video frame will be stored.
+ * @param[in] buf the input buffer
+ * @param[in] buf_size the size of the input buffer in bytes
+ * @param[in,out] got_picture_ptr Zero if no frame could be decompressed, otherwise, it is nonzero.
+ * @return On error a negative value is returned, otherwise the number of bytes
+ * used or zero if no frame could be decompressed.
+ */
+attribute_deprecated int avcodec_decode_video(AVCodecContext *avctx, AVFrame *picture,
+ int *got_picture_ptr,
+ const uint8_t *buf, int buf_size);
+#endif
+
+/**
+ * Decode the video frame of size avpkt->size from avpkt->data into picture.
+ * Some decoders may support multiple frames in a single AVPacket, such
+ * decoders would then just decode the first frame.
+ *
+ * @warning The input buffer must be FF_INPUT_BUFFER_PADDING_SIZE larger than
+ * the actual read bytes because some optimized bitstream readers read 32 or 64
+ * bits at once and could read over the end.
+ *
+ * @warning The end of the input buffer buf should be set to 0 to ensure that
+ * no overreading happens for damaged MPEG streams.
+ *
+ * @note You might have to align the input buffer avpkt->data.
+ * The alignment requirements depend on the CPU: on some CPUs it isn't
+ * necessary at all, on others it won't work at all if not aligned and on others
+ * it will work but it will have an impact on performance.
+ *
+ * In practice, avpkt->data should have 4 byte alignment at minimum.
+ *
+ * @note Some codecs have a delay between input and output, these need to be
+ * fed with avpkt->data=NULL, avpkt->size=0 at the end to return the remaining frames.
+ *
+ * @param avctx the codec context
+ * @param[out] picture The AVFrame in which the decoded video frame will be stored.
+ * Use avcodec_alloc_frame to get an AVFrame, the codec will
+ * allocate memory for the actual bitmap.
+ * with default get/release_buffer(), the decoder frees/reuses the bitmap as it sees fit.
+ * with overridden get/release_buffer() (needs CODEC_CAP_DR1) the user decides into what buffer the decoder
+ * decodes and the decoder tells the user once it does not need the data anymore,
+ * the user app can at this point free/reuse/keep the memory as it sees fit.
+ *
+ * @param[in] avpkt The input AVpacket containing the input buffer.
+ * You can create such packet with av_init_packet() and by then setting
+ * data and size, some decoders might in addition need other fields like
+ * flags&AV_PKT_FLAG_KEY. All decoders are designed to use the least
+ * fields possible.
+ * @param[in,out] got_picture_ptr Zero if no frame could be decompressed, otherwise, it is nonzero.
+ * @return On error a negative value is returned, otherwise the number of bytes
+ * used or zero if no frame could be decompressed.
+ */
+int avcodec_decode_video2(AVCodecContext *avctx, AVFrame *picture,
+ int *got_picture_ptr,
+ AVPacket *avpkt);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/* Decode a subtitle message. Return -1 if error, otherwise return the
+ * number of bytes used. If no subtitle could be decompressed,
+ * got_sub_ptr is zero. Otherwise, the subtitle is stored in *sub. */
+attribute_deprecated int avcodec_decode_subtitle(AVCodecContext *avctx, AVSubtitle *sub,
+ int *got_sub_ptr,
+ const uint8_t *buf, int buf_size);
+#endif
+
+/**
+ * Decode a subtitle message.
+ * Return a negative value on error, otherwise return the number of bytes used.
+ * If no subtitle could be decompressed, got_sub_ptr is zero.
+ * Otherwise, the subtitle is stored in *sub.
+ * Note that CODEC_CAP_DR1 is not available for subtitle codecs. This is for
+ * simplicity, because the performance difference is expect to be negligible
+ * and reusing a get_buffer written for video codecs would probably perform badly
+ * due to a potentially very different allocation pattern.
+ *
+ * @param avctx the codec context
+ * @param[out] sub The AVSubtitle in which the decoded subtitle will be stored, must be
+ freed with avsubtitle_free if *got_sub_ptr is set.
+ * @param[in,out] got_sub_ptr Zero if no subtitle could be decompressed, otherwise, it is nonzero.
+ * @param[in] avpkt The input AVPacket containing the input buffer.
+ */
+int avcodec_decode_subtitle2(AVCodecContext *avctx, AVSubtitle *sub,
+ int *got_sub_ptr,
+ AVPacket *avpkt);
+
+/**
+ * Frees all allocated data in the given subtitle struct.
+ *
+ * @param sub AVSubtitle to free.
+ */
+void avsubtitle_free(AVSubtitle *sub);
+
+int avcodec_parse_frame(AVCodecContext *avctx, uint8_t **pdata,
+ int *data_size_ptr,
+ uint8_t *buf, int buf_size);
+
+/**
+ * Encode an audio frame from samples into buf.
+ *
+ * @note The output buffer should be at least FF_MIN_BUFFER_SIZE bytes large.
+ * However, for PCM audio the user will know how much space is needed
+ * because it depends on the value passed in buf_size as described
+ * below. In that case a lower value can be used.
+ *
+ * @param avctx the codec context
+ * @param[out] buf the output buffer
+ * @param[in] buf_size the output buffer size
+ * @param[in] samples the input buffer containing the samples
+ * The number of samples read from this buffer is frame_size*channels,
+ * both of which are defined in avctx.
+ * For PCM audio the number of samples read from samples is equal to
+ * buf_size * input_sample_size / output_sample_size.
+ * @return On error a negative value is returned, on success zero or the number
+ * of bytes used to encode the data read from the input buffer.
+ */
+int avcodec_encode_audio(AVCodecContext *avctx, uint8_t *buf, int buf_size,
+ const short *samples);
+
+/**
+ * Encode a video frame from pict into buf.
+ * The input picture should be
+ * stored using a specific format, namely avctx.pix_fmt.
+ *
+ * @param avctx the codec context
+ * @param[out] buf the output buffer for the bitstream of encoded frame
+ * @param[in] buf_size the size of the output buffer in bytes
+ * @param[in] pict the input picture to encode
+ * @return On error a negative value is returned, on success zero or the number
+ * of bytes used from the output buffer.
+ */
+int avcodec_encode_video(AVCodecContext *avctx, uint8_t *buf, int buf_size,
+ const AVFrame *pict);
+int avcodec_encode_subtitle(AVCodecContext *avctx, uint8_t *buf, int buf_size,
+ const AVSubtitle *sub);
+
+int avcodec_close(AVCodecContext *avctx);
+
+/**
+ * Register all the codecs, parsers and bitstream filters which were enabled at
+ * configuration time. If you do not call this function you can select exactly
+ * which formats you want to support, by using the individual registration
+ * functions.
+ *
+ * @see avcodec_register
+ * @see av_register_codec_parser
+ * @see av_register_bitstream_filter
+ */
+void avcodec_register_all(void);
+
+/**
+ * Flush buffers, should be called when seeking or when switching to a different stream.
+ */
+void avcodec_flush_buffers(AVCodecContext *avctx);
+
+void avcodec_default_free_buffers(AVCodecContext *s);
+
+/* misc useful functions */
+
+/**
+ * Return a single letter to describe the given picture type pict_type.
+ *
+ * @param[in] pict_type the picture type
+ * @return A single character representing the picture type.
+ */
+char av_get_pict_type_char(int pict_type);
+
+/**
+ * Return codec bits per sample.
+ *
+ * @param[in] codec_id the codec
+ * @return Number of bits per sample or zero if unknown for the given codec.
+ */
+int av_get_bits_per_sample(enum CodecID codec_id);
+
+/**
+ * Return sample format bits per sample.
+ *
+ * @param[in] sample_fmt the sample format
+ * @return Number of bits per sample or zero if unknown for the given sample format.
+ */
+int av_get_bits_per_sample_format(enum SampleFormat sample_fmt);
+
+/* frame parsing */
+typedef struct AVCodecParserContext {
+ void *priv_data;
+ struct AVCodecParser *parser;
+ int64_t frame_offset; /* offset of the current frame */
+ int64_t cur_offset; /* current offset
+ (incremented by each av_parser_parse()) */
+ int64_t next_frame_offset; /* offset of the next frame */
+ /* video info */
+ int pict_type; /* XXX: Put it back in AVCodecContext. */
+ /**
+ * This field is used for proper frame duration computation in lavf.
+ * It signals, how much longer the frame duration of the current frame
+ * is compared to normal frame duration.
+ *
+ * frame_duration = (1 + repeat_pict) * time_base
+ *
+ * It is used by codecs like H.264 to display telecined material.
+ */
+ int repeat_pict; /* XXX: Put it back in AVCodecContext. */
+ int64_t pts; /* pts of the current frame */
+ int64_t dts; /* dts of the current frame */
+
+ /* private data */
+ int64_t last_pts;
+ int64_t last_dts;
+ int fetch_timestamp;
+
+#define AV_PARSER_PTS_NB 4
+ int cur_frame_start_index;
+ int64_t cur_frame_offset[AV_PARSER_PTS_NB];
+ int64_t cur_frame_pts[AV_PARSER_PTS_NB];
+ int64_t cur_frame_dts[AV_PARSER_PTS_NB];
+
+ int flags;
+#define PARSER_FLAG_COMPLETE_FRAMES 0x0001
+#define PARSER_FLAG_ONCE 0x0002
+
+ int64_t offset; ///< byte offset from starting packet start
+ int64_t cur_frame_end[AV_PARSER_PTS_NB];
+
+ /*!
+ * Set by parser to 1 for key frames and 0 for non-key frames.
+ * It is initialized to -1, so if the parser doesn't set this flag,
+ * old-style fallback using FF_I_TYPE picture type as key frames
+ * will be used.
+ */
+ int key_frame;
+
+ /**
+ * Time difference in stream time base units from the pts of this
+ * packet to the point at which the output from the decoder has converged
+ * independent from the availability of previous frames. That is, the
+ * frames are virtually identical no matter if decoding started from
+ * the very first frame or from this keyframe.
+ * Is AV_NOPTS_VALUE if unknown.
+ * This field is not the display duration of the current frame.
+ * This field has no meaning if the packet does not have AV_PKT_FLAG_KEY
+ * set.
+ *
+ * The purpose of this field is to allow seeking in streams that have no
+ * keyframes in the conventional sense. It corresponds to the
+ * recovery point SEI in H.264 and match_time_delta in NUT. It is also
+ * essential for some types of subtitle streams to ensure that all
+ * subtitles are correctly displayed after seeking.
+ */
+ int64_t convergence_duration;
+
+ // Timestamp generation support:
+ /**
+ * Synchronization point for start of timestamp generation.
+ *
+ * Set to >0 for sync point, 0 for no sync point and <0 for undefined
+ * (default).
+ *
+ * For example, this corresponds to presence of H.264 buffering period
+ * SEI message.
+ */
+ int dts_sync_point;
+
+ /**
+ * Offset of the current timestamp against last timestamp sync point in
+ * units of AVCodecContext.time_base.
+ *
+ * Set to INT_MIN when dts_sync_point unused. Otherwise, it must
+ * contain a valid timestamp offset.
+ *
+ * Note that the timestamp of sync point has usually a nonzero
+ * dts_ref_dts_delta, which refers to the previous sync point. Offset of
+ * the next frame after timestamp sync point will be usually 1.
+ *
+ * For example, this corresponds to H.264 cpb_removal_delay.
+ */
+ int dts_ref_dts_delta;
+
+ /**
+ * Presentation delay of current frame in units of AVCodecContext.time_base.
+ *
+ * Set to INT_MIN when dts_sync_point unused. Otherwise, it must
+ * contain valid non-negative timestamp delta (presentation time of a frame
+ * must not lie in the past).
+ *
+ * This delay represents the difference between decoding and presentation
+ * time of the frame.
+ *
+ * For example, this corresponds to H.264 dpb_output_delay.
+ */
+ int pts_dts_delta;
+
+ /**
+ * Position of the packet in file.
+ *
+ * Analogous to cur_frame_pts/dts
+ */
+ int64_t cur_frame_pos[AV_PARSER_PTS_NB];
+
+ /**
+ * Byte position of currently parsed frame in stream.
+ */
+ int64_t pos;
+
+ /**
+ * Previous frame byte position.
+ */
+ int64_t last_pos;
+} AVCodecParserContext;
+
+typedef struct AVCodecParser {
+ int codec_ids[5]; /* several codec IDs are permitted */
+ int priv_data_size;
+ int (*parser_init)(AVCodecParserContext *s);
+ int (*parser_parse)(AVCodecParserContext *s,
+ AVCodecContext *avctx,
+ const uint8_t **poutbuf, int *poutbuf_size,
+ const uint8_t *buf, int buf_size);
+ void (*parser_close)(AVCodecParserContext *s);
+ int (*split)(AVCodecContext *avctx, const uint8_t *buf, int buf_size);
+ struct AVCodecParser *next;
+} AVCodecParser;
+
+AVCodecParser *av_parser_next(AVCodecParser *c);
+
+void av_register_codec_parser(AVCodecParser *parser);
+AVCodecParserContext *av_parser_init(int codec_id);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+attribute_deprecated
+int av_parser_parse(AVCodecParserContext *s,
+ AVCodecContext *avctx,
+ uint8_t **poutbuf, int *poutbuf_size,
+ const uint8_t *buf, int buf_size,
+ int64_t pts, int64_t dts);
+#endif
+
+/**
+ * Parse a packet.
+ *
+ * @param s parser context.
+ * @param avctx codec context.
+ * @param poutbuf set to pointer to parsed buffer or NULL if not yet finished.
+ * @param poutbuf_size set to size of parsed buffer or zero if not yet finished.
+ * @param buf input buffer.
+ * @param buf_size input length, to signal EOF, this should be 0 (so that the last frame can be output).
+ * @param pts input presentation timestamp.
+ * @param dts input decoding timestamp.
+ * @param pos input byte position in stream.
+ * @return the number of bytes of the input bitstream used.
+ *
+ * Example:
+ * @code
+ * while(in_len){
+ * len = av_parser_parse2(myparser, AVCodecContext, &data, &size,
+ * in_data, in_len,
+ * pts, dts, pos);
+ * in_data += len;
+ * in_len -= len;
+ *
+ * if(size)
+ * decode_frame(data, size);
+ * }
+ * @endcode
+ */
+int av_parser_parse2(AVCodecParserContext *s,
+ AVCodecContext *avctx,
+ uint8_t **poutbuf, int *poutbuf_size,
+ const uint8_t *buf, int buf_size,
+ int64_t pts, int64_t dts,
+ int64_t pos);
+
+int av_parser_change(AVCodecParserContext *s,
+ AVCodecContext *avctx,
+ uint8_t **poutbuf, int *poutbuf_size,
+ const uint8_t *buf, int buf_size, int keyframe);
+void av_parser_close(AVCodecParserContext *s);
+
+
+typedef struct AVBitStreamFilterContext {
+ void *priv_data;
+ struct AVBitStreamFilter *filter;
+ AVCodecParserContext *parser;
+ struct AVBitStreamFilterContext *next;
+} AVBitStreamFilterContext;
+
+
+typedef struct AVBitStreamFilter {
+ const char *name;
+ int priv_data_size;
+ int (*filter)(AVBitStreamFilterContext *bsfc,
+ AVCodecContext *avctx, const char *args,
+ uint8_t **poutbuf, int *poutbuf_size,
+ const uint8_t *buf, int buf_size, int keyframe);
+ void (*close)(AVBitStreamFilterContext *bsfc);
+ struct AVBitStreamFilter *next;
+} AVBitStreamFilter;
+
+void av_register_bitstream_filter(AVBitStreamFilter *bsf);
+AVBitStreamFilterContext *av_bitstream_filter_init(const char *name);
+int av_bitstream_filter_filter(AVBitStreamFilterContext *bsfc,
+ AVCodecContext *avctx, const char *args,
+ uint8_t **poutbuf, int *poutbuf_size,
+ const uint8_t *buf, int buf_size, int keyframe);
+void av_bitstream_filter_close(AVBitStreamFilterContext *bsf);
+
+AVBitStreamFilter *av_bitstream_filter_next(AVBitStreamFilter *f);
+
+/* memory */
+
+/**
+ * Reallocate the given block if it is not large enough, otherwise do nothing.
+ *
+ * @see av_realloc
+ */
+void *av_fast_realloc(void *ptr, unsigned int *size, unsigned int min_size);
+
+/**
+ * Allocate a buffer, reusing the given one if large enough.
+ *
+ * Contrary to av_fast_realloc the current buffer contents might not be
+ * preserved and on error the old buffer is freed, thus no special
+ * handling to avoid memleaks is necessary.
+ *
+ * @param ptr pointer to pointer to already allocated buffer, overwritten with pointer to new buffer
+ * @param size size of the buffer *ptr points to
+ * @param min_size minimum size of *ptr buffer after returning, *ptr will be NULL and
+ * *size 0 if an error occurred.
+ */
+void av_fast_malloc(void *ptr, unsigned int *size, unsigned int min_size);
+
+/**
+ * Copy image 'src' to 'dst'.
+ */
+void av_picture_copy(AVPicture *dst, const AVPicture *src,
+ enum PixelFormat pix_fmt, int width, int height);
+
+/**
+ * Crop image top and left side.
+ */
+int av_picture_crop(AVPicture *dst, const AVPicture *src,
+ enum PixelFormat pix_fmt, int top_band, int left_band);
+
+/**
+ * Pad image.
+ */
+int av_picture_pad(AVPicture *dst, const AVPicture *src, int height, int width, enum PixelFormat pix_fmt,
+ int padtop, int padbottom, int padleft, int padright, int *color);
+
+/**
+ * Encode extradata length to a buffer. Used by xiph codecs.
+ *
+ * @param s buffer to write to; must be at least (v/255+1) bytes long
+ * @param v size of extradata in bytes
+ * @return number of bytes written to the buffer.
+ */
+unsigned int av_xiphlacing(unsigned char *s, unsigned int v);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * Parse str and put in width_ptr and height_ptr the detected values.
+ *
+ * @deprecated Deprecated in favor of av_parse_video_size().
+ */
+attribute_deprecated int av_parse_video_frame_size(int *width_ptr, int *height_ptr, const char *str);
+
+/**
+ * Parse str and store the detected values in *frame_rate.
+ *
+ * @deprecated Deprecated in favor of av_parse_video_rate().
+ */
+attribute_deprecated int av_parse_video_frame_rate(AVRational *frame_rate, const char *str);
+#endif
+
+/**
+ * Logs a generic warning message about a missing feature. This function is
+ * intended to be used internally by FFmpeg (libavcodec, libavformat, etc.)
+ * only, and would normally not be used by applications.
+ * @param[in] avc a pointer to an arbitrary struct of which the first field is
+ * a pointer to an AVClass struct
+ * @param[in] feature string containing the name of the missing feature
+ * @param[in] want_sample indicates if samples are wanted which exhibit this feature.
+ * If want_sample is non-zero, additional verbage will be added to the log
+ * message which tells the user how to report samples to the development
+ * mailing list.
+ */
+void av_log_missing_feature(void *avc, const char *feature, int want_sample);
+
+/**
+ * Log a generic warning message asking for a sample. This function is
+ * intended to be used internally by FFmpeg (libavcodec, libavformat, etc.)
+ * only, and would normally not be used by applications.
+ * @param[in] avc a pointer to an arbitrary struct of which the first field is
+ * a pointer to an AVClass struct
+ * @param[in] msg string containing an optional message, or NULL if no message
+ */
+void av_log_ask_for_sample(void *avc, const char *msg);
+
+/**
+ * Register the hardware accelerator hwaccel.
+ */
+void av_register_hwaccel(AVHWAccel *hwaccel);
+
+/**
+ * If hwaccel is NULL, returns the first registered hardware accelerator,
+ * if hwaccel is non-NULL, returns the next registered hardware accelerator
+ * after hwaccel, or NULL if hwaccel is the last one.
+ */
+AVHWAccel *av_hwaccel_next(AVHWAccel *hwaccel);
+
+
+/**
+ * Lock operation used by lockmgr
+ */
+enum AVLockOp {
+ AV_LOCK_CREATE, ///< Create a mutex
+ AV_LOCK_OBTAIN, ///< Lock the mutex
+ AV_LOCK_RELEASE, ///< Unlock the mutex
+ AV_LOCK_DESTROY, ///< Free mutex resources
+};
+
+/**
+ * Register a user provided lock manager supporting the operations
+ * specified by AVLockOp. mutex points to a (void *) where the
+ * lockmgr should store/get a pointer to a user allocated mutex. It's
+ * NULL upon AV_LOCK_CREATE and != NULL for all other ops.
+ *
+ * @param cb User defined callback. Note: FFmpeg may invoke calls to this
+ * callback during the call to av_lockmgr_register().
+ * Thus, the application must be prepared to handle that.
+ * If cb is set to NULL the lockmgr will be unregistered.
+ * Also note that during unregistration the previously registered
+ * lockmgr callback may also be invoked.
+ */
+int av_lockmgr_register(int (*cb)(void **mutex, enum AVLockOp op));
+
+#endif /* AVCODEC_AVCODEC_H */
diff --git a/thirdparties/android/include/libavcodec/avfft.h b/thirdparties/android/include/libavcodec/avfft.h new file mode 100644 index 0000000..be2d9c7 --- /dev/null +++ b/thirdparties/android/include/libavcodec/avfft.h @@ -0,0 +1,99 @@ +/* + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_AVFFT_H +#define AVCODEC_AVFFT_H + +typedef float FFTSample; + +typedef struct FFTComplex { + FFTSample re, im; +} FFTComplex; + +typedef struct FFTContext FFTContext; + +/** + * Set up a complex FFT. + * @param nbits log2 of the length of the input array + * @param inverse if 0 perform the forward transform, if 1 perform the inverse + */ +FFTContext *av_fft_init(int nbits, int inverse); + +/** + * Do the permutation needed BEFORE calling ff_fft_calc(). + */ +void av_fft_permute(FFTContext *s, FFTComplex *z); + +/** + * Do a complex FFT with the parameters defined in av_fft_init(). The + * input data must be permuted before. No 1.0/sqrt(n) normalization is done. + */ +void av_fft_calc(FFTContext *s, FFTComplex *z); + +void av_fft_end(FFTContext *s); + +FFTContext *av_mdct_init(int nbits, int inverse, double scale); +void av_imdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input); +void av_imdct_half(FFTContext *s, FFTSample *output, const FFTSample *input); +void av_mdct_calc(FFTContext *s, FFTSample *output, const FFTSample *input); +void av_mdct_end(FFTContext *s); + +/* Real Discrete Fourier Transform */ + +enum RDFTransformType { + DFT_R2C, + IDFT_C2R, + IDFT_R2C, + DFT_C2R, +}; + +typedef struct RDFTContext RDFTContext; + +/** + * Set up a real FFT. + * @param nbits log2 of the length of the input array + * @param trans the type of transform + */ +RDFTContext *av_rdft_init(int nbits, enum RDFTransformType trans); +void av_rdft_calc(RDFTContext *s, FFTSample *data); +void av_rdft_end(RDFTContext *s); + +/* Discrete Cosine Transform */ + +typedef struct DCTContext DCTContext; + +enum DCTTransformType { + DCT_II = 0, + DCT_III, + DCT_I, + DST_I, +}; + +/** + * Set up DCT. + * @param nbits size of the input array: + * (1 << nbits) for DCT-II, DCT-III and DST-I + * (1 << nbits) + 1 for DCT-I + * + * @note the first element of the input of DST-I is ignored + */ +DCTContext *av_dct_init(int nbits, enum DCTTransformType type); +void av_dct_calc(DCTContext *s, FFTSample *data); +void av_dct_end (DCTContext *s); + +#endif /* AVCODEC_AVFFT_H */ diff --git a/thirdparties/android/include/libavcodec/dxva2.h b/thirdparties/android/include/libavcodec/dxva2.h new file mode 100644 index 0000000..5c5fe21 --- /dev/null +++ b/thirdparties/android/include/libavcodec/dxva2.h @@ -0,0 +1,68 @@ +/* + * DXVA2 HW acceleration + * + * copyright (c) 2009 Laurent Aimar + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_DXVA_H +#define AVCODEC_DXVA_H + +#include <stdint.h> + +#include <dxva2api.h> + +/** + * This structure is used to provides the necessary configurations and data + * to the DXVA2 FFmpeg HWAccel implementation. + * + * The application must make it available as AVCodecContext.hwaccel_context. + */ +struct dxva_context { + /** + * DXVA2 decoder object + */ + IDirectXVideoDecoder *decoder; + + /** + * DXVA2 configuration used to create the decoder + */ + const DXVA2_ConfigPictureDecode *cfg; + + /** + * The number of surface in the surface array + */ + unsigned surface_count; + + /** + * The array of Direct3D surfaces used to create the decoder + */ + LPDIRECT3DSURFACE9 *surface; + + /** + * A bit field configuring the workarounds needed for using the decoder + */ + uint64_t workaround; + + /** + * Private to the FFmpeg AVHWAccel implementation + */ + unsigned report_id; +}; + +#endif /* AVCODEC_DXVA_H */ diff --git a/thirdparties/android/include/libavcodec/opt.h b/thirdparties/android/include/libavcodec/opt.h new file mode 100644 index 0000000..6721469 --- /dev/null +++ b/thirdparties/android/include/libavcodec/opt.h @@ -0,0 +1,211 @@ +/*
+ * AVOptions
+ * copyright (c) 2005 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPT_H
+#define AVCODEC_OPT_H
+
+/**
+ * @file
+ * AVOptions
+ */
+
+#include "libavutil/rational.h"
+#include "avcodec.h"
+
+enum AVOptionType{
+ FF_OPT_TYPE_FLAGS,
+ FF_OPT_TYPE_INT,
+ FF_OPT_TYPE_INT64,
+ FF_OPT_TYPE_DOUBLE,
+ FF_OPT_TYPE_FLOAT,
+ FF_OPT_TYPE_STRING,
+ FF_OPT_TYPE_RATIONAL,
+ FF_OPT_TYPE_BINARY, ///< offset must point to a pointer immediately followed by an int for the length
+ FF_OPT_TYPE_CONST=128,
+};
+
+/**
+ * AVOption
+ */
+typedef struct AVOption {
+ const char *name;
+
+ /**
+ * short English help text
+ * @todo What about other languages?
+ */
+ const char *help;
+
+ /**
+ * The offset relative to the context structure where the option
+ * value is stored. It should be 0 for named constants.
+ */
+ int offset;
+ enum AVOptionType type;
+
+ /**
+ * the default value for scalar options
+ */
+ double default_val;
+ double min; ///< minimum valid value for the option
+ double max; ///< maximum valid value for the option
+
+ int flags;
+#define AV_OPT_FLAG_ENCODING_PARAM 1 ///< a generic parameter which can be set by the user for muxing or encoding
+#define AV_OPT_FLAG_DECODING_PARAM 2 ///< a generic parameter which can be set by the user for demuxing or decoding
+#define AV_OPT_FLAG_METADATA 4 ///< some data extracted or inserted into the file like title, comment, ...
+#define AV_OPT_FLAG_AUDIO_PARAM 8
+#define AV_OPT_FLAG_VIDEO_PARAM 16
+#define AV_OPT_FLAG_SUBTITLE_PARAM 32
+//FIXME think about enc-audio, ... style flags
+
+ /**
+ * The logical unit to which the option belongs. Non-constant
+ * options and corresponding named constants share the same
+ * unit. May be NULL.
+ */
+ const char *unit;
+} AVOption;
+
+/**
+ * AVOption2.
+ * THIS IS NOT PART OF THE API/ABI YET!
+ * This is identical to AVOption except that default_val was replaced by
+ * an union, it should be compatible with AVOption on normal platforms.
+ */
+typedef struct AVOption2 {
+ const char *name;
+
+ /**
+ * short English help text
+ * @todo What about other languages?
+ */
+ const char *help;
+
+ /**
+ * The offset relative to the context structure where the option
+ * value is stored. It should be 0 for named constants.
+ */
+ int offset;
+ enum AVOptionType type;
+
+ /**
+ * the default value for scalar options
+ */
+ union {
+ double dbl;
+ const char *str;
+ } default_val;
+
+ double min; ///< minimum valid value for the option
+ double max; ///< maximum valid value for the option
+
+ int flags;
+/*
+#define AV_OPT_FLAG_ENCODING_PARAM 1 ///< a generic parameter which can be set by the user for muxing or encoding
+#define AV_OPT_FLAG_DECODING_PARAM 2 ///< a generic parameter which can be set by the user for demuxing or decoding
+#define AV_OPT_FLAG_METADATA 4 ///< some data extracted or inserted into the file like title, comment, ...
+#define AV_OPT_FLAG_AUDIO_PARAM 8
+#define AV_OPT_FLAG_VIDEO_PARAM 16
+#define AV_OPT_FLAG_SUBTITLE_PARAM 32
+*/
+//FIXME think about enc-audio, ... style flags
+
+ /**
+ * The logical unit to which the option belongs. Non-constant
+ * options and corresponding named constants share the same
+ * unit. May be NULL.
+ */
+ const char *unit;
+} AVOption2;
+
+
+/**
+ * Look for an option in obj. Look only for the options which
+ * have the flags set as specified in mask and flags (that is,
+ * for which it is the case that opt->flags & mask == flags).
+ *
+ * @param[in] obj a pointer to a struct whose first element is a
+ * pointer to an AVClass
+ * @param[in] name the name of the option to look for
+ * @param[in] unit the unit of the option to look for, or any if NULL
+ * @return a pointer to the option found, or NULL if no option
+ * has been found
+ */
+const AVOption *av_find_opt(void *obj, const char *name, const char *unit, int mask, int flags);
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+/**
+ * @see av_set_string2()
+ */
+attribute_deprecated const AVOption *av_set_string(void *obj, const char *name, const char *val);
+
+/**
+ * @return a pointer to the AVOption corresponding to the field set or
+ * NULL if no matching AVOption exists, or if the value val is not
+ * valid
+ * @see av_set_string3()
+ */
+attribute_deprecated const AVOption *av_set_string2(void *obj, const char *name, const char *val, int alloc);
+#endif
+
+/**
+ * Set the field of obj with the given name to value.
+ *
+ * @param[in] obj A struct whose first element is a pointer to an
+ * AVClass.
+ * @param[in] name the name of the field to set
+ * @param[in] val The value to set. If the field is not of a string
+ * type, then the given string is parsed.
+ * SI postfixes and some named scalars are supported.
+ * If the field is of a numeric type, it has to be a numeric or named
+ * scalar. Behavior with more than one scalar and +- infix operators
+ * is undefined.
+ * If the field is of a flags type, it has to be a sequence of numeric
+ * scalars or named flags separated by '+' or '-'. Prefixing a flag
+ * with '+' causes it to be set without affecting the other flags;
+ * similarly, '-' unsets a flag.
+ * @param[out] o_out if non-NULL put here a pointer to the AVOption
+ * found
+ * @param alloc when 1 then the old value will be av_freed() and the
+ * new av_strduped()
+ * when 0 then no av_free() nor av_strdup() will be used
+ * @return 0 if the value has been set, or an AVERROR code in case of
+ * error:
+ * AVERROR(ENOENT) if no matching option exists
+ * AVERROR(ERANGE) if the value is out of range
+ * AVERROR(EINVAL) if the value is not valid
+ */
+int av_set_string3(void *obj, const char *name, const char *val, int alloc, const AVOption **o_out);
+
+const AVOption *av_set_double(void *obj, const char *name, double n);
+const AVOption *av_set_q(void *obj, const char *name, AVRational n);
+const AVOption *av_set_int(void *obj, const char *name, int64_t n);
+double av_get_double(void *obj, const char *name, const AVOption **o_out);
+AVRational av_get_q(void *obj, const char *name, const AVOption **o_out);
+int64_t av_get_int(void *obj, const char *name, const AVOption **o_out);
+const char *av_get_string(void *obj, const char *name, const AVOption **o_out, char *buf, int buf_len);
+const AVOption *av_next_option(void *obj, const AVOption *last);
+int av_opt_show(void *obj, void *av_log_obj);
+void av_opt_set_defaults(void *s);
+void av_opt_set_defaults2(void *s, int mask, int flags);
+
+#endif /* AVCODEC_OPT_H */
diff --git a/thirdparties/android/include/libavcodec/vaapi.h b/thirdparties/android/include/libavcodec/vaapi.h new file mode 100644 index 0000000..07568a4 --- /dev/null +++ b/thirdparties/android/include/libavcodec/vaapi.h @@ -0,0 +1,167 @@ +/* + * Video Acceleration API (shared data between FFmpeg and the video player) + * HW decode acceleration for MPEG-2, MPEG-4, H.264 and VC-1 + * + * Copyright (C) 2008-2009 Splitted-Desktop Systems + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_VAAPI_H +#define AVCODEC_VAAPI_H + +#include <stdint.h> + +/** + * \defgroup VAAPI_Decoding VA API Decoding + * \ingroup Decoder + * @{ + */ + +/** + * This structure is used to share data between the FFmpeg library and + * the client video application. + * This shall be zero-allocated and available as + * AVCodecContext.hwaccel_context. All user members can be set once + * during initialization or through each AVCodecContext.get_buffer() + * function call. In any case, they must be valid prior to calling + * decoding functions. + */ +struct vaapi_context { + /** + * Window system dependent data + * + * - encoding: unused + * - decoding: Set by user + */ + void *display; + + /** + * Configuration ID + * + * - encoding: unused + * - decoding: Set by user + */ + uint32_t config_id; + + /** + * Context ID (video decode pipeline) + * + * - encoding: unused + * - decoding: Set by user + */ + uint32_t context_id; + + /** + * VAPictureParameterBuffer ID + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + uint32_t pic_param_buf_id; + + /** + * VAIQMatrixBuffer ID + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + uint32_t iq_matrix_buf_id; + + /** + * VABitPlaneBuffer ID (for VC-1 decoding) + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + uint32_t bitplane_buf_id; + + /** + * Slice parameter/data buffer IDs + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + uint32_t *slice_buf_ids; + + /** + * Number of effective slice buffer IDs to send to the HW + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + unsigned int n_slice_buf_ids; + + /** + * Size of pre-allocated slice_buf_ids + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + unsigned int slice_buf_ids_alloc; + + /** + * Pointer to VASliceParameterBuffers + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + void *slice_params; + + /** + * Size of a VASliceParameterBuffer element + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + unsigned int slice_param_size; + + /** + * Size of pre-allocated slice_params + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + unsigned int slice_params_alloc; + + /** + * Number of slices currently filled in + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + unsigned int slice_count; + + /** + * Pointer to slice data buffer base + * - encoding: unused + * - decoding: Set by libavcodec + */ + const uint8_t *slice_data; + + /** + * Current size of slice data + * + * - encoding: unused + * - decoding: Set by libavcodec + */ + uint32_t slice_data_size; +}; + +/* @} */ + +#endif /* AVCODEC_VAAPI_H */ diff --git a/thirdparties/android/include/libavcodec/vdpau.h b/thirdparties/android/include/libavcodec/vdpau.h new file mode 100644 index 0000000..ab5f682 --- /dev/null +++ b/thirdparties/android/include/libavcodec/vdpau.h @@ -0,0 +1,88 @@ +/* + * The Video Decode and Presentation API for UNIX (VDPAU) is used for + * hardware-accelerated decoding of MPEG-1/2, H.264 and VC-1. + * + * Copyright (C) 2008 NVIDIA + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_VDPAU_H +#define AVCODEC_VDPAU_H + +/** + * \defgroup Decoder VDPAU Decoder and Renderer + * + * VDPAU hardware acceleration has two modules + * - VDPAU decoding + * - VDPAU presentation + * + * The VDPAU decoding module parses all headers using FFmpeg + * parsing mechanisms and uses VDPAU for the actual decoding. + * + * As per the current implementation, the actual decoding + * and rendering (API calls) are done as part of the VDPAU + * presentation (vo_vdpau.c) module. + * + * \defgroup VDPAU_Decoding VDPAU Decoding + * \ingroup Decoder + * @{ + */ + +#include <vdpau/vdpau.h> +#include <vdpau/vdpau_x11.h> + +/** \brief The videoSurface is used for rendering. */ +#define FF_VDPAU_STATE_USED_FOR_RENDER 1 + +/** + * \brief The videoSurface is needed for reference/prediction. + * The codec manipulates this. + */ +#define FF_VDPAU_STATE_USED_FOR_REFERENCE 2 + +/** + * \brief This structure is used as a callback between the FFmpeg + * decoder (vd_) and presentation (vo_) module. + * This is used for defining a video frame containing surface, + * picture parameter, bitstream information etc which are passed + * between the FFmpeg decoder and its clients. + */ +struct vdpau_render_state { + VdpVideoSurface surface; ///< Used as rendered surface, never changed. + + int state; ///< Holds FF_VDPAU_STATE_* values. + + /** picture parameter information for all supported codecs */ + union VdpPictureInfo { + VdpPictureInfoH264 h264; + VdpPictureInfoMPEG1Or2 mpeg; + VdpPictureInfoVC1 vc1; + VdpPictureInfoMPEG4Part2 mpeg4; + } info; + + /** Describe size/location of the compressed video data. + Set to 0 when freeing bitstream_buffers. */ + int bitstream_buffers_allocated; + int bitstream_buffers_used; + /** The user is responsible for freeing this buffer using av_freep(). */ + VdpBitstreamBuffer *bitstream_buffers; +}; + +/* @}*/ + +#endif /* AVCODEC_VDPAU_H */ diff --git a/thirdparties/android/include/libavcodec/xvmc.h b/thirdparties/android/include/libavcodec/xvmc.h new file mode 100644 index 0000000..32ec866 --- /dev/null +++ b/thirdparties/android/include/libavcodec/xvmc.h @@ -0,0 +1,172 @@ +/*
+ * Copyright (C) 2003 Ivan Kalvachev
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_XVMC_H
+#define AVCODEC_XVMC_H
+
+#include <X11/extensions/XvMC.h>
+
+#include "avcodec.h"
+
+#if LIBAVCODEC_VERSION_MAJOR < 53
+#define AV_XVMC_STATE_DISPLAY_PENDING 1 /** the surface should be shown, the video driver manipulates this */
+#define AV_XVMC_STATE_PREDICTION 2 /** the surface is needed for prediction, the codec manipulates this */
+#define AV_XVMC_STATE_OSD_SOURCE 4 /** the surface is needed for subpicture rendering */
+#endif
+#define AV_XVMC_ID 0x1DC711C0 /**< special value to ensure that regular pixel routines haven't corrupted the struct
+ the number is 1337 speak for the letters IDCT MCo (motion compensation) */
+
+struct xvmc_pix_fmt {
+ /** The field contains the special constant value AV_XVMC_ID.
+ It is used as a test that the application correctly uses the API,
+ and that there is no corruption caused by pixel routines.
+ - application - set during initialization
+ - libavcodec - unchanged
+ */
+ int xvmc_id;
+
+ /** Pointer to the block array allocated by XvMCCreateBlocks().
+ The array has to be freed by XvMCDestroyBlocks().
+ Each group of 64 values represents one data block of differential
+ pixel information (in MoCo mode) or coefficients for IDCT.
+ - application - set the pointer during initialization
+ - libavcodec - fills coefficients/pixel data into the array
+ */
+ short* data_blocks;
+
+ /** Pointer to the macroblock description array allocated by
+ XvMCCreateMacroBlocks() and freed by XvMCDestroyMacroBlocks().
+ - application - set the pointer during initialization
+ - libavcodec - fills description data into the array
+ */
+ XvMCMacroBlock* mv_blocks;
+
+ /** Number of macroblock descriptions that can be stored in the mv_blocks
+ array.
+ - application - set during initialization
+ - libavcodec - unchanged
+ */
+ int allocated_mv_blocks;
+
+ /** Number of blocks that can be stored at once in the data_blocks array.
+ - application - set during initialization
+ - libavcodec - unchanged
+ */
+ int allocated_data_blocks;
+
+ /** Indicate that the hardware would interpret data_blocks as IDCT
+ coefficients and perform IDCT on them.
+ - application - set during initialization
+ - libavcodec - unchanged
+ */
+ int idct;
+
+ /** In MoCo mode it indicates that intra macroblocks are assumed to be in
+ unsigned format; same as the XVMC_INTRA_UNSIGNED flag.
+ - application - set during initialization
+ - libavcodec - unchanged
+ */
+ int unsigned_intra;
+
+ /** Pointer to the surface allocated by XvMCCreateSurface().
+ It has to be freed by XvMCDestroySurface() on application exit.
+ It identifies the frame and its state on the video hardware.
+ - application - set during initialization
+ - libavcodec - unchanged
+ */
+ XvMCSurface* p_surface;
+
+/** Set by the decoder before calling ff_draw_horiz_band(),
+ needed by the XvMCRenderSurface function. */
+//@{
+ /** Pointer to the surface used as past reference
+ - application - unchanged
+ - libavcodec - set
+ */
+ XvMCSurface* p_past_surface;
+
+ /** Pointer to the surface used as future reference
+ - application - unchanged
+ - libavcodec - set
+ */
+ XvMCSurface* p_future_surface;
+
+ /** top/bottom field or frame
+ - application - unchanged
+ - libavcodec - set
+ */
+ unsigned int picture_structure;
+
+ /** XVMC_SECOND_FIELD - 1st or 2nd field in the sequence
+ - application - unchanged
+ - libavcodec - set
+ */
+ unsigned int flags;
+//}@
+
+ /** Number of macroblock descriptions in the mv_blocks array
+ that have already been passed to the hardware.
+ - application - zeroes it on get_buffer().
+ A successful ff_draw_horiz_band() may increment it
+ with filled_mb_block_num or zero both.
+ - libavcodec - unchanged
+ */
+ int start_mv_blocks_num;
+
+ /** Number of new macroblock descriptions in the mv_blocks array (after
+ start_mv_blocks_num) that are filled by libavcodec and have to be
+ passed to the hardware.
+ - application - zeroes it on get_buffer() or after successful
+ ff_draw_horiz_band().
+ - libavcodec - increment with one of each stored MB
+ */
+ int filled_mv_blocks_num;
+
+ /** Number of the the next free data block; one data block consists of
+ 64 short values in the data_blocks array.
+ All blocks before this one have already been claimed by placing their
+ position into the corresponding block description structure field,
+ that are part of the mv_blocks array.
+ - application - zeroes it on get_buffer().
+ A successful ff_draw_horiz_band() may zero it together
+ with start_mb_blocks_num.
+ - libavcodec - each decoded macroblock increases it by the number
+ of coded blocks it contains.
+ */
+ int next_free_data_block_num;
+
+/** extensions may be placed here */
+#if LIBAVCODEC_VERSION_MAJOR < 53
+//@{
+ /** State flags used to work around limitations in the MPlayer video system.
+ 0 - Surface is not used.
+ 1 - Surface is still held in application to be displayed or is
+ still visible.
+ 2 - Surface is still held in libavcodec buffer for prediction.
+ */
+ int state;
+
+ /** pointer to the surface where the subpicture is rendered */
+ void* p_osd_target_surface_render;
+//}@
+#endif
+};
+
+#endif /* AVCODEC_XVMC_H */
diff --git a/thirdparties/android/include/libavcore/avcore.h b/thirdparties/android/include/libavcore/avcore.h new file mode 100644 index 0000000..5ddb616 --- /dev/null +++ b/thirdparties/android/include/libavcore/avcore.h @@ -0,0 +1,58 @@ +/* + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCORE_AVCORE_H +#define AVCORE_AVCORE_H + +/** + * @file + * shared media utilities for the libav* libraries + */ + +#include <libavutil/avutil.h> + +#define LIBAVCORE_VERSION_MAJOR 0 +#define LIBAVCORE_VERSION_MINOR 2 +#define LIBAVCORE_VERSION_MICRO 0 + +#define LIBAVCORE_VERSION_INT AV_VERSION_INT(LIBAVCORE_VERSION_MAJOR, \ + LIBAVCORE_VERSION_MINOR, \ + LIBAVCORE_VERSION_MICRO) +#define LIBAVCORE_VERSION AV_VERSION(LIBAVCORE_VERSION_MAJOR, \ + LIBAVCORE_VERSION_MINOR, \ + LIBAVCORE_VERSION_MICRO) +#define LIBAVCORE_BUILD LIBAVCORE_VERSION_INT + +#define LIBAVCORE_IDENT "Lavcore" AV_STRINGIFY(LIBAVCORE_VERSION) + +/** + * Return the LIBAVCORE_VERSION_INT constant. + */ +unsigned avcore_version(void); + +/** + * Return the libavcore build-time configuration. + */ +const char *avcore_configuration(void); + +/** + * Return the libavcore license. + */ +const char *avcore_license(void); + +#endif /* AVCORE_AVCORE_H */ diff --git a/thirdparties/android/include/libavcore/imgutils.h b/thirdparties/android/include/libavcore/imgutils.h new file mode 100644 index 0000000..4a4d6a1 --- /dev/null +++ b/thirdparties/android/include/libavcore/imgutils.h @@ -0,0 +1,53 @@ +/* + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCORE_IMGUTILS_H +#define AVCORE_IMGUTILS_H + +/** + * @file + * misc image utilities + */ + +#include "libavutil/pixfmt.h" +#include "avcore.h" + +/** + * Fill plane linesizes for an image with pixel format pix_fmt and + * width width. + * + * @param linesizes array to be filled with the linesize for each plane + * @return >= 0 in case of success, a negative error code otherwise + */ +int av_fill_image_linesizes(int linesizes[4], enum PixelFormat pix_fmt, int width); + +/** + * Fill plane data pointers for an image with pixel format pix_fmt and + * height height. + * + * @param data pointers array to be filled with the pointer for each image plane + * @param ptr the pointer to a buffer which will contain the image + * @param linesizes[4] the array containing the linesize for each + * plane, should be filled by av_fill_image_linesizes() + * @return the size in bytes required for the image buffer, a negative + * error code in case of failure + */ +int av_fill_image_pointers(uint8_t *data[4], enum PixelFormat pix_fmt, int height, + uint8_t *ptr, const int linesizes[4]); + +#endif /* AVCORE_IMGUTILS_H */ diff --git a/thirdparties/android/include/libavcore/parseutils.h b/thirdparties/android/include/libavcore/parseutils.h new file mode 100644 index 0000000..a81cc18 --- /dev/null +++ b/thirdparties/android/include/libavcore/parseutils.h @@ -0,0 +1,53 @@ +/* + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCORE_PARSEUTILS_H +#define AVCORE_PARSEUTILS_H + +#include <libavutil/rational.h> + +/** + * @file + * misc parsing utilities for libavcore + */ + +/** + * Parse str and put in width_ptr and height_ptr the detected values. + * + * @param[in,out] width_ptr pointer to the variable which will contain the detected + * width value + * @param[in,out] height_ptr pointer to the variable which will contain the detected + * height value + * @param[in] str the string to parse: it has to be a string in the format + * width x height or a valid video size abbreviation. + * @return >= 0 on success, a negative error code otherwise + */ +int av_parse_video_size(int *width_ptr, int *height_ptr, const char *str); + +/** + * Parse str and store the detected values in *rate. + * + * @param[in,out] rate pointer to the AVRational which will contain the detected + * frame rate + * @param[in] str the string to parse: it has to be a string in the format + * rate_num / rate_den, a float number or a valid video rate abbreviation + * @return >= 0 on success, a negative error code otherwise + */ +int av_parse_video_rate(AVRational *rate, const char *str); + +#endif /* AVCORE_PARSEUTILS_H */ diff --git a/thirdparties/android/include/libavutil/adler32.h b/thirdparties/android/include/libavutil/adler32.h new file mode 100644 index 0000000..0b890bc --- /dev/null +++ b/thirdparties/android/include/libavutil/adler32.h @@ -0,0 +1,42 @@ +/* + * copyright (c) 2006 Mans Rullgard + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_ADLER32_H +#define AVUTIL_ADLER32_H + +#include <stdint.h> +#include "attributes.h" + +/** + * Calculate the Adler32 checksum of a buffer. + * + * Passing the return value to a subsequent av_adler32_update() call + * allows the checksum of multiple buffers to be calculated as though + * they were concatenated. + * + * @param adler initial checksum value + * @param buf pointer to input buffer + * @param len size of input buffer + * @return updated checksum + */ +unsigned long av_adler32_update(unsigned long adler, const uint8_t *buf, + unsigned int len) av_pure; + +#endif /* AVUTIL_ADLER32_H */ diff --git a/thirdparties/android/include/libavutil/attributes.h b/thirdparties/android/include/libavutil/attributes.h new file mode 100644 index 0000000..a95bb02 --- /dev/null +++ b/thirdparties/android/include/libavutil/attributes.h @@ -0,0 +1,121 @@ +/* + * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * Macro definitions for various function/variable attributes + */ + +#ifndef AVUTIL_ATTRIBUTES_H +#define AVUTIL_ATTRIBUTES_H + +#ifdef __GNUC__ +# define AV_GCC_VERSION_AT_LEAST(x,y) (__GNUC__ > x || __GNUC__ == x && __GNUC_MINOR__ >= y) +#else +# define AV_GCC_VERSION_AT_LEAST(x,y) 0 +#endif + +#ifndef av_always_inline +#if AV_GCC_VERSION_AT_LEAST(3,1) +# define av_always_inline __attribute__((always_inline)) inline +#else +# define av_always_inline inline +#endif +#endif + +#ifndef av_noinline +#if AV_GCC_VERSION_AT_LEAST(3,1) +# define av_noinline __attribute__((noinline)) +#else +# define av_noinline +#endif +#endif + +#ifndef av_pure +#if AV_GCC_VERSION_AT_LEAST(3,1) +# define av_pure __attribute__((pure)) +#else +# define av_pure +#endif +#endif + +#ifndef av_const +#if AV_GCC_VERSION_AT_LEAST(2,6) +# define av_const __attribute__((const)) +#else +# define av_const +#endif +#endif + +#ifndef av_cold +#if (!defined(__ICC) || __ICC > 1110) && AV_GCC_VERSION_AT_LEAST(4,3) +# define av_cold __attribute__((cold)) +#else +# define av_cold +#endif +#endif + +#ifndef av_flatten +#if (!defined(__ICC) || __ICC > 1110) && AV_GCC_VERSION_AT_LEAST(4,1) +# define av_flatten __attribute__((flatten)) +#else +# define av_flatten +#endif +#endif + +#ifndef attribute_deprecated +#if AV_GCC_VERSION_AT_LEAST(3,1) +# define attribute_deprecated __attribute__((deprecated)) +#else +# define attribute_deprecated +#endif +#endif + +#ifndef av_unused +#if defined(__GNUC__) +# define av_unused __attribute__((unused)) +#else +# define av_unused +#endif +#endif + +#ifndef av_alias +#if (!defined(__ICC) || __ICC > 1110) && AV_GCC_VERSION_AT_LEAST(3,3) +# define av_alias __attribute__((may_alias)) +#else +# define av_alias +#endif +#endif + +#ifndef av_uninit +#if defined(__GNUC__) && !defined(__ICC) +# define av_uninit(x) x=x +#else +# define av_uninit(x) x +#endif +#endif + +#ifdef __GNUC__ +# define av_builtin_constant_p __builtin_constant_p +#else +# define av_builtin_constant_p(x) 0 +#endif + +#endif /* AVUTIL_ATTRIBUTES_H */ diff --git a/thirdparties/android/include/libavutil/avconfig.h b/thirdparties/android/include/libavutil/avconfig.h new file mode 100644 index 0000000..36f72aa --- /dev/null +++ b/thirdparties/android/include/libavutil/avconfig.h @@ -0,0 +1,6 @@ +/* Generated by ffconf */ +#ifndef AVUTIL_AVCONFIG_H +#define AVUTIL_AVCONFIG_H +#define AV_HAVE_BIGENDIAN 0 +#define AV_HAVE_FAST_UNALIGNED 0 +#endif /* AVUTIL_AVCONFIG_H */ diff --git a/thirdparties/android/include/libavutil/avstring.h b/thirdparties/android/include/libavutil/avstring.h new file mode 100644 index 0000000..01c2391 --- /dev/null +++ b/thirdparties/android/include/libavutil/avstring.h @@ -0,0 +1,117 @@ +/* + * Copyright (c) 2007 Mans Rullgard + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_AVSTRING_H +#define AVUTIL_AVSTRING_H + +#include <stddef.h> + +/** + * Return non-zero if pfx is a prefix of str. If it is, *ptr is set to + * the address of the first character in str after the prefix. + * + * @param str input string + * @param pfx prefix to test + * @param ptr updated if the prefix is matched inside str + * @return non-zero if the prefix matches, zero otherwise + */ +int av_strstart(const char *str, const char *pfx, const char **ptr); + +/** + * Return non-zero if pfx is a prefix of str independent of case. If + * it is, *ptr is set to the address of the first character in str + * after the prefix. + * + * @param str input string + * @param pfx prefix to test + * @param ptr updated if the prefix is matched inside str + * @return non-zero if the prefix matches, zero otherwise + */ +int av_stristart(const char *str, const char *pfx, const char **ptr); + +/** + * Locate the first case-independent occurrence in the string haystack + * of the string needle. A zero-length string needle is considered to + * match at the start of haystack. + * + * This function is a case-insensitive version of the standard strstr(). + * + * @param haystack string to search in + * @param needle string to search for + * @return pointer to the located match within haystack + * or a null pointer if no match + */ +char *av_stristr(const char *haystack, const char *needle); + +/** + * Copy the string src to dst, but no more than size - 1 bytes, and + * null-terminate dst. + * + * This function is the same as BSD strlcpy(). + * + * @param dst destination buffer + * @param src source string + * @param size size of destination buffer + * @return the length of src + * + * WARNING: since the return value is the length of src, src absolutely + * _must_ be a properly 0-terminated string, otherwise this will read beyond + * the end of the buffer and possibly crash. + */ +size_t av_strlcpy(char *dst, const char *src, size_t size); + +/** + * Append the string src to the string dst, but to a total length of + * no more than size - 1 bytes, and null-terminate dst. + * + * This function is similar to BSD strlcat(), but differs when + * size <= strlen(dst). + * + * @param dst destination buffer + * @param src source string + * @param size size of destination buffer + * @return the total length of src and dst + * + * WARNING: since the return value use the length of src and dst, these absolutely + * _must_ be a properly 0-terminated strings, otherwise this will read beyond + * the end of the buffer and possibly crash. + */ +size_t av_strlcat(char *dst, const char *src, size_t size); + +/** + * Append output to a string, according to a format. Never write out of + * the destination buffer, and always put a terminating 0 within + * the buffer. + * @param dst destination buffer (string to which the output is + * appended) + * @param size total size of the destination buffer + * @param fmt printf-compatible format string, specifying how the + * following parameters are used + * @return the length of the string that would have been generated + * if enough space had been available + */ +size_t av_strlcatf(char *dst, size_t size, const char *fmt, ...); + +/** + * Convert a number to a av_malloced string. + */ +char *av_d2str(double d); + +#endif /* AVUTIL_AVSTRING_H */ diff --git a/thirdparties/android/include/libavutil/avutil.h b/thirdparties/android/include/libavutil/avutil.h new file mode 100644 index 0000000..f43eedd --- /dev/null +++ b/thirdparties/android/include/libavutil/avutil.h @@ -0,0 +1,89 @@ +/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_AVUTIL_H
+#define AVUTIL_AVUTIL_H
+
+/**
+ * @file
+ * external API header
+ */
+
+
+#define AV_STRINGIFY(s) AV_TOSTRING(s)
+#define AV_TOSTRING(s) #s
+
+#define AV_GLUE(a, b) a ## b
+#define AV_JOIN(a, b) AV_GLUE(a, b)
+
+#define AV_PRAGMA(s) _Pragma(#s)
+
+#define AV_VERSION_INT(a, b, c) (a<<16 | b<<8 | c)
+#define AV_VERSION_DOT(a, b, c) a ##.## b ##.## c
+#define AV_VERSION(a, b, c) AV_VERSION_DOT(a, b, c)
+
+#define LIBAVUTIL_VERSION_MAJOR 50
+#define LIBAVUTIL_VERSION_MINOR 23
+#define LIBAVUTIL_VERSION_MICRO 0
+
+#define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
+ LIBAVUTIL_VERSION_MINOR, \
+ LIBAVUTIL_VERSION_MICRO)
+#define LIBAVUTIL_VERSION AV_VERSION(LIBAVUTIL_VERSION_MAJOR, \
+ LIBAVUTIL_VERSION_MINOR, \
+ LIBAVUTIL_VERSION_MICRO)
+#define LIBAVUTIL_BUILD LIBAVUTIL_VERSION_INT
+
+#define LIBAVUTIL_IDENT "Lavu" AV_STRINGIFY(LIBAVUTIL_VERSION)
+
+/**
+ * Return the LIBAVUTIL_VERSION_INT constant.
+ */
+unsigned avutil_version(void);
+
+/**
+ * Return the libavutil build-time configuration.
+ */
+const char *avutil_configuration(void);
+
+/**
+ * Return the libavutil license.
+ */
+const char *avutil_license(void);
+
+enum AVMediaType {
+ AVMEDIA_TYPE_UNKNOWN = -1,
+ AVMEDIA_TYPE_VIDEO,
+ AVMEDIA_TYPE_AUDIO,
+ AVMEDIA_TYPE_DATA,
+ AVMEDIA_TYPE_SUBTITLE,
+ AVMEDIA_TYPE_ATTACHMENT,
+ AVMEDIA_TYPE_NB
+};
+
+#include "common.h"
+#include "error.h"
+#include "mathematics.h"
+#include "rational.h"
+#include "intfloat_readwrite.h"
+#include "log.h"
+#include "pixfmt.h"
+
+#endif /* AVUTIL_AVUTIL_H */
diff --git a/thirdparties/android/include/libavutil/base64.h b/thirdparties/android/include/libavutil/base64.h new file mode 100644 index 0000000..092980b --- /dev/null +++ b/thirdparties/android/include/libavutil/base64.h @@ -0,0 +1,54 @@ +/* + * Copyright (c) 2006 Ryan Martell. (rdm4@martellventures.com) + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_BASE64_H +#define AVUTIL_BASE64_H + +#include <stdint.h> + +/** + * Decode a base64-encoded string. + * + * @param out buffer for decoded data + * @param in null-terminated input string + * @param out_size size in bytes of the out buffer, must be at + * least 3/4 of the length of in + * @return number of bytes written, or a negative value in case of + * invalid input + */ +int av_base64_decode(uint8_t *out, const char *in, int out_size); + +/** + * Encode data to base64 and null-terminate. + * + * @param out buffer for encoded data + * @param out_size size in bytes of the output buffer, must be at + * least AV_BASE64_SIZE(in_size) + * @param in_size size in bytes of the 'in' buffer + * @return 'out' or NULL in case of error + */ +char *av_base64_encode(char *out, int out_size, const uint8_t *in, int in_size); + +/** + * Calculate the output size needed to base64-encode x bytes. + */ +#define AV_BASE64_SIZE(x) (((x)+2) / 3 * 4 + 1) + +#endif /* AVUTIL_BASE64_H */ diff --git a/thirdparties/android/include/libavutil/bswap.h b/thirdparties/android/include/libavutil/bswap.h new file mode 100644 index 0000000..0bf0eba --- /dev/null +++ b/thirdparties/android/include/libavutil/bswap.h @@ -0,0 +1,124 @@ +/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * byte swapping routines
+ */
+
+#ifndef AVUTIL_BSWAP_H
+#define AVUTIL_BSWAP_H
+
+#include <stdint.h>
+#include "libavutil/avconfig.h"
+#include "attributes.h"
+
+#ifdef HAVE_AV_CONFIG_H
+
+#include "config.h"
+
+#if ARCH_ARM
+# include "arm/bswap.h"
+#elif ARCH_AVR32
+# include "avr32/bswap.h"
+#elif ARCH_BFIN
+# include "bfin/bswap.h"
+#elif ARCH_SH4
+# include "sh4/bswap.h"
+#elif ARCH_X86
+# include "x86/bswap.h"
+#endif
+
+#endif /* HAVE_AV_CONFIG_H */
+
+#define AV_BSWAP16C(x) (((x) << 8 & 0xff00) | ((x) >> 8 & 0x00ff))
+#define AV_BSWAP32C(x) (AV_BSWAP16C(x) << 16 | AV_BSWAP16C((x) >> 16))
+#define AV_BSWAP64C(x) (AV_BSWAP32C(x) << 32 | AV_BSWAP32C((x) >> 32))
+
+#define AV_BSWAPC(s, x) AV_BSWAP##s##C(x)
+
+#ifndef av_bswap16
+static av_always_inline av_const uint16_t av_bswap16(uint16_t x)
+{
+ x= (x>>8) | (x<<8);
+ return x;
+}
+#endif
+
+#ifndef av_bswap32
+static av_always_inline av_const uint32_t av_bswap32(uint32_t x)
+{
+ x= ((x<<8)&0xFF00FF00) | ((x>>8)&0x00FF00FF);
+ x= (x>>16) | (x<<16);
+ return x;
+}
+#endif
+
+#ifndef av_bswap64
+static inline uint64_t av_const av_bswap64(uint64_t x)
+{
+#if 0
+ x= ((x<< 8)&0xFF00FF00FF00FF00ULL) | ((x>> 8)&0x00FF00FF00FF00FFULL);
+ x= ((x<<16)&0xFFFF0000FFFF0000ULL) | ((x>>16)&0x0000FFFF0000FFFFULL);
+ return (x>>32) | (x<<32);
+#else
+ union {
+ uint64_t ll;
+ uint32_t l[2];
+ } w, r;
+ w.ll = x;
+ r.l[0] = av_bswap32 (w.l[1]);
+ r.l[1] = av_bswap32 (w.l[0]);
+ return r.ll;
+#endif
+}
+#endif
+
+// be2ne ... big-endian to native-endian
+// le2ne ... little-endian to native-endian
+
+#if AV_HAVE_BIGENDIAN
+#define av_be2ne16(x) (x)
+#define av_be2ne32(x) (x)
+#define av_be2ne64(x) (x)
+#define av_le2ne16(x) av_bswap16(x)
+#define av_le2ne32(x) av_bswap32(x)
+#define av_le2ne64(x) av_bswap64(x)
+#define AV_BE2NEC(s, x) (x)
+#define AV_LE2NEC(s, x) AV_BSWAPC(s, x)
+#else
+#define av_be2ne16(x) av_bswap16(x)
+#define av_be2ne32(x) av_bswap32(x)
+#define av_be2ne64(x) av_bswap64(x)
+#define av_le2ne16(x) (x)
+#define av_le2ne32(x) (x)
+#define av_le2ne64(x) (x)
+#define AV_BE2NEC(s, x) AV_BSWAPC(s, x)
+#define AV_LE2NEC(s, x) (x)
+#endif
+
+#define AV_BE2NE16C(x) AV_BE2NEC(16, x)
+#define AV_BE2NE32C(x) AV_BE2NEC(32, x)
+#define AV_BE2NE64C(x) AV_BE2NEC(64, x)
+#define AV_LE2NE16C(x) AV_LE2NEC(16, x)
+#define AV_LE2NE32C(x) AV_LE2NEC(32, x)
+#define AV_LE2NE64C(x) AV_LE2NEC(64, x)
+
+#endif /* AVUTIL_BSWAP_H */
diff --git a/thirdparties/android/include/libavutil/common.h b/thirdparties/android/include/libavutil/common.h new file mode 100644 index 0000000..5eff235 --- /dev/null +++ b/thirdparties/android/include/libavutil/common.h @@ -0,0 +1,346 @@ +/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * common internal and external API header
+ */
+
+#ifndef AVUTIL_COMMON_H
+#define AVUTIL_COMMON_H
+
+#include <ctype.h>
+#include <errno.h>
+#include <inttypes.h>
+#include <limits.h>
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include "attributes.h"
+
+//rounded division & shift
+#define RSHIFT(a,b) ((a) > 0 ? ((a) + ((1<<(b))>>1))>>(b) : ((a) + ((1<<(b))>>1)-1)>>(b))
+/* assume b>0 */
+#define ROUNDED_DIV(a,b) (((a)>0 ? (a) + ((b)>>1) : (a) - ((b)>>1))/(b))
+#define FFABS(a) ((a) >= 0 ? (a) : (-(a)))
+#define FFSIGN(a) ((a) > 0 ? 1 : -1)
+
+#define FFMAX(a,b) ((a) > (b) ? (a) : (b))
+#define FFMAX3(a,b,c) FFMAX(FFMAX(a,b),c)
+#define FFMIN(a,b) ((a) > (b) ? (b) : (a))
+#define FFMIN3(a,b,c) FFMIN(FFMIN(a,b),c)
+
+#define FFSWAP(type,a,b) do{type SWAP_tmp= b; b= a; a= SWAP_tmp;}while(0)
+#define FF_ARRAY_ELEMS(a) (sizeof(a) / sizeof((a)[0]))
+#define FFALIGN(x, a) (((x)+(a)-1)&~((a)-1))
+
+/* misc math functions */
+extern const uint8_t ff_log2_tab[256];
+
+extern const uint8_t av_reverse[256];
+
+static inline av_const int av_log2_c(unsigned int v)
+{
+ int n = 0;
+ if (v & 0xffff0000) {
+ v >>= 16;
+ n += 16;
+ }
+ if (v & 0xff00) {
+ v >>= 8;
+ n += 8;
+ }
+ n += ff_log2_tab[v];
+
+ return n;
+}
+
+static inline av_const int av_log2_16bit_c(unsigned int v)
+{
+ int n = 0;
+ if (v & 0xff00) {
+ v >>= 8;
+ n += 8;
+ }
+ n += ff_log2_tab[v];
+
+ return n;
+}
+
+#ifdef HAVE_AV_CONFIG_H
+# include "config.h"
+# include "intmath.h"
+#endif
+
+/* Pull in unguarded fallback defines at the end of this file. */
+#include "common.h"
+
+/**
+ * Clip a signed integer value into the amin-amax range.
+ * @param a value to clip
+ * @param amin minimum value of the clip range
+ * @param amax maximum value of the clip range
+ * @return clipped value
+ */
+static inline av_const int av_clip_c(int a, int amin, int amax)
+{
+ if (a < amin) return amin;
+ else if (a > amax) return amax;
+ else return a;
+}
+
+/**
+ * Clip a signed integer value into the 0-255 range.
+ * @param a value to clip
+ * @return clipped value
+ */
+static inline av_const uint8_t av_clip_uint8_c(int a)
+{
+ if (a&(~0xFF)) return (-a)>>31;
+ else return a;
+}
+
+/**
+ * Clip a signed integer value into the -128,127 range.
+ * @param a value to clip
+ * @return clipped value
+ */
+static inline av_const int8_t av_clip_int8_c(int a)
+{
+ if ((a+0x80) & ~0xFF) return (a>>31) ^ 0x7F;
+ else return a;
+}
+
+/**
+ * Clip a signed integer value into the 0-65535 range.
+ * @param a value to clip
+ * @return clipped value
+ */
+static inline av_const uint16_t av_clip_uint16_c(int a)
+{
+ if (a&(~0xFFFF)) return (-a)>>31;
+ else return a;
+}
+
+/**
+ * Clip a signed integer value into the -32768,32767 range.
+ * @param a value to clip
+ * @return clipped value
+ */
+static inline av_const int16_t av_clip_int16_c(int a)
+{
+ if ((a+0x8000) & ~0xFFFF) return (a>>31) ^ 0x7FFF;
+ else return a;
+}
+
+/**
+ * Clip a signed 64-bit integer value into the -2147483648,2147483647 range.
+ * @param a value to clip
+ * @return clipped value
+ */
+static inline av_const int32_t av_clipl_int32_c(int64_t a)
+{
+ if ((a+0x80000000u) & ~UINT64_C(0xFFFFFFFF)) return (a>>63) ^ 0x7FFFFFFF;
+ else return a;
+}
+
+/**
+ * Clip a float value into the amin-amax range.
+ * @param a value to clip
+ * @param amin minimum value of the clip range
+ * @param amax maximum value of the clip range
+ * @return clipped value
+ */
+static inline av_const float av_clipf_c(float a, float amin, float amax)
+{
+ if (a < amin) return amin;
+ else if (a > amax) return amax;
+ else return a;
+}
+
+/** Compute ceil(log2(x)).
+ * @param x value used to compute ceil(log2(x))
+ * @return computed ceiling of log2(x)
+ */
+static inline av_const int av_ceil_log2_c(int x)
+{
+ return av_log2((x - 1) << 1);
+}
+
+#define MKTAG(a,b,c,d) ((a) | ((b) << 8) | ((c) << 16) | ((d) << 24))
+#define MKBETAG(a,b,c,d) ((d) | ((c) << 8) | ((b) << 16) | ((a) << 24))
+
+/**
+ * Convert a UTF-8 character (up to 4 bytes) to its 32-bit UCS-4 encoded form.
+ *
+ * @param val Output value, must be an lvalue of type uint32_t.
+ * @param GET_BYTE Expression reading one byte from the input.
+ * Evaluated up to 7 times (4 for the currently
+ * assigned Unicode range). With a memory buffer
+ * input, this could be *ptr++.
+ * @param ERROR Expression to be evaluated on invalid input,
+ * typically a goto statement.
+ */
+#define GET_UTF8(val, GET_BYTE, ERROR)\
+ val= GET_BYTE;\
+ {\
+ int ones= 7 - av_log2(val ^ 255);\
+ if(ones==1)\
+ ERROR\
+ val&= 127>>ones;\
+ while(--ones > 0){\
+ int tmp= GET_BYTE - 128;\
+ if(tmp>>6)\
+ ERROR\
+ val= (val<<6) + tmp;\
+ }\
+ }
+
+/**
+ * Convert a UTF-16 character (2 or 4 bytes) to its 32-bit UCS-4 encoded form.
+ *
+ * @param val Output value, must be an lvalue of type uint32_t.
+ * @param GET_16BIT Expression returning two bytes of UTF-16 data converted
+ * to native byte order. Evaluated one or two times.
+ * @param ERROR Expression to be evaluated on invalid input,
+ * typically a goto statement.
+ */
+#define GET_UTF16(val, GET_16BIT, ERROR)\
+ val = GET_16BIT;\
+ {\
+ unsigned int hi = val - 0xD800;\
+ if (hi < 0x800) {\
+ val = GET_16BIT - 0xDC00;\
+ if (val > 0x3FFU || hi > 0x3FFU)\
+ ERROR\
+ val += (hi<<10) + 0x10000;\
+ }\
+ }\
+
+/*!
+ * \def PUT_UTF8(val, tmp, PUT_BYTE)
+ * Convert a 32-bit Unicode character to its UTF-8 encoded form (up to 4 bytes long).
+ * \param val is an input-only argument and should be of type uint32_t. It holds
+ * a UCS-4 encoded Unicode character that is to be converted to UTF-8. If
+ * val is given as a function it is executed only once.
+ * \param tmp is a temporary variable and should be of type uint8_t. It
+ * represents an intermediate value during conversion that is to be
+ * output by PUT_BYTE.
+ * \param PUT_BYTE writes the converted UTF-8 bytes to any proper destination.
+ * It could be a function or a statement, and uses tmp as the input byte.
+ * For example, PUT_BYTE could be "*output++ = tmp;" PUT_BYTE will be
+ * executed up to 4 times for values in the valid UTF-8 range and up to
+ * 7 times in the general case, depending on the length of the converted
+ * Unicode character.
+ */
+#define PUT_UTF8(val, tmp, PUT_BYTE)\
+ {\
+ int bytes, shift;\
+ uint32_t in = val;\
+ if (in < 0x80) {\
+ tmp = in;\
+ PUT_BYTE\
+ } else {\
+ bytes = (av_log2(in) + 4) / 5;\
+ shift = (bytes - 1) * 6;\
+ tmp = (256 - (256 >> bytes)) | (in >> shift);\
+ PUT_BYTE\
+ while (shift >= 6) {\
+ shift -= 6;\
+ tmp = 0x80 | ((in >> shift) & 0x3f);\
+ PUT_BYTE\
+ }\
+ }\
+ }
+
+/*!
+ * \def PUT_UTF16(val, tmp, PUT_16BIT)
+ * Convert a 32-bit Unicode character to its UTF-16 encoded form (2 or 4 bytes).
+ * \param val is an input-only argument and should be of type uint32_t. It holds
+ * a UCS-4 encoded Unicode character that is to be converted to UTF-16. If
+ * val is given as a function it is executed only once.
+ * \param tmp is a temporary variable and should be of type uint16_t. It
+ * represents an intermediate value during conversion that is to be
+ * output by PUT_16BIT.
+ * \param PUT_16BIT writes the converted UTF-16 data to any proper destination
+ * in desired endianness. It could be a function or a statement, and uses tmp
+ * as the input byte. For example, PUT_BYTE could be "*output++ = tmp;"
+ * PUT_BYTE will be executed 1 or 2 times depending on input character.
+ */
+#define PUT_UTF16(val, tmp, PUT_16BIT)\
+ {\
+ uint32_t in = val;\
+ if (in < 0x10000) {\
+ tmp = in;\
+ PUT_16BIT\
+ } else {\
+ tmp = 0xD800 | ((in - 0x10000) >> 10);\
+ PUT_16BIT\
+ tmp = 0xDC00 | ((in - 0x10000) & 0x3FF);\
+ PUT_16BIT\
+ }\
+ }\
+
+
+
+#include "mem.h"
+
+#ifdef HAVE_AV_CONFIG_H
+# include "internal.h"
+#endif /* HAVE_AV_CONFIG_H */
+
+#endif /* AVUTIL_COMMON_H */
+
+/*
+ * The following definitions are outside the multiple inclusion guard
+ * to ensure they are immediately available in intmath.h.
+ */
+
+#ifndef av_log2
+# define av_log2 av_log2_c
+#endif
+#ifndef av_log2_16bit
+# define av_log2_16bit av_log2_16bit_c
+#endif
+#ifndef av_ceil_log2
+# define av_ceil_log2 av_ceil_log2_c
+#endif
+#ifndef av_clip
+# define av_clip av_clip_c
+#endif
+#ifndef av_clip_uint8
+# define av_clip_uint8 av_clip_uint8_c
+#endif
+#ifndef av_clip_int8
+# define av_clip_int8 av_clip_int8_c
+#endif
+#ifndef av_clip_uint16
+# define av_clip_uint16 av_clip_uint16_c
+#endif
+#ifndef av_clip_int16
+# define av_clip_int16 av_clip_int16_c
+#endif
+#ifndef av_clipl_int32
+# define av_clipl_int32 av_clipl_int32_c
+#endif
+#ifndef av_clipf
+# define av_clipf av_clipf_c
+#endif
diff --git a/thirdparties/android/include/libavutil/crc.h b/thirdparties/android/include/libavutil/crc.h new file mode 100644 index 0000000..b83ced1 --- /dev/null +++ b/thirdparties/android/include/libavutil/crc.h @@ -0,0 +1,44 @@ +/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_CRC_H
+#define AVUTIL_CRC_H
+
+#include <stdint.h>
+#include <stddef.h>
+#include "attributes.h"
+
+typedef uint32_t AVCRC;
+
+typedef enum {
+ AV_CRC_8_ATM,
+ AV_CRC_16_ANSI,
+ AV_CRC_16_CCITT,
+ AV_CRC_32_IEEE,
+ AV_CRC_32_IEEE_LE, /*< reversed bitorder version of AV_CRC_32_IEEE */
+ AV_CRC_MAX, /*< Not part of public API! Do not use outside libavutil. */
+}AVCRCId;
+
+int av_crc_init(AVCRC *ctx, int le, int bits, uint32_t poly, int ctx_size);
+const AVCRC *av_crc_get_table(AVCRCId crc_id);
+uint32_t av_crc(const AVCRC *ctx, uint32_t start_crc, const uint8_t *buffer, size_t length) av_pure;
+
+#endif /* AVUTIL_CRC_H */
+
diff --git a/thirdparties/android/include/libavutil/error.h b/thirdparties/android/include/libavutil/error.h new file mode 100644 index 0000000..28fa925 --- /dev/null +++ b/thirdparties/android/include/libavutil/error.h @@ -0,0 +1,74 @@ +/* + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * error code definitions + */ + +#ifndef AVUTIL_ERROR_H +#define AVUTIL_ERROR_H + +#include <errno.h> +#include "avutil.h" + +/* error handling */ +#if EDOM > 0 +#define AVERROR(e) (-(e)) ///< Returns a negative error code from a POSIX error code, to return from library functions. +#define AVUNERROR(e) (-(e)) ///< Returns a POSIX error code from a library function error return value. +#else +/* Some platforms have E* and errno already negated. */ +#define AVERROR(e) (e) +#define AVUNERROR(e) (e) +#endif + +#if LIBAVUTIL_VERSION_MAJOR < 51 +#define AVERROR_INVALIDDATA AVERROR(EINVAL) ///< Invalid data found when processing input +#define AVERROR_IO AVERROR(EIO) ///< I/O error +#define AVERROR_NOENT AVERROR(ENOENT) ///< No such file or directory +#define AVERROR_NOFMT AVERROR(EILSEQ) ///< Unknown format +#define AVERROR_NOMEM AVERROR(ENOMEM) ///< Not enough memory +#define AVERROR_NOTSUPP AVERROR(ENOSYS) ///< Operation not supported +#define AVERROR_NUMEXPECTED AVERROR(EDOM) ///< Number syntax expected in filename +#define AVERROR_UNKNOWN AVERROR(EINVAL) ///< Unknown error +#endif + +#define AVERROR_EOF AVERROR(EPIPE) ///< End of file + +#define AVERROR_PATCHWELCOME (-MKTAG('P','A','W','E')) ///< Not yet implemented in FFmpeg, patches welcome + +#if LIBAVUTIL_VERSION_MAJOR > 50 +#define AVERROR_INVALIDDATA (-MKTAG('I','N','D','A')) ///< Invalid data found when processing input +#define AVERROR_NUMEXPECTED (-MKTAG('N','U','E','X')) ///< Number syntax expected in filename +#endif + +/** + * Put a description of the AVERROR code errnum in errbuf. + * In case of failure the global variable errno is set to indicate the + * error. Even in case of failure av_strerror() will print a generic + * error message indicating the errnum provided to errbuf. + * + * @param errnum error code to describe + * @param errbuf buffer to which description is written + * @param errbuf_size the size in bytes of errbuf + * @return 0 on success, a negative value if a description for errnum + * cannot be found + */ +int av_strerror(int errnum, char *errbuf, size_t errbuf_size); + +#endif /* AVUTIL_ERROR_H */ diff --git a/thirdparties/android/include/libavutil/eval.h b/thirdparties/android/include/libavutil/eval.h new file mode 100644 index 0000000..7a4f5f0 --- /dev/null +++ b/thirdparties/android/include/libavutil/eval.h @@ -0,0 +1,111 @@ +/* + * Copyright (c) 2002 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +/** + * @file + * simple arithmetic expression evaluator + */ + +#ifndef AVUTIL_EVAL_H +#define AVUTIL_EVAL_H + +typedef struct AVExpr AVExpr; + +/** + * Parse and evaluate an expression. + * Note, this is significantly slower than av_eval_expr(). + * + * @param res a pointer to a double where is put the result value of + * the expression, or NAN in case of error + * @param s expression as a zero terminated string, for example "1+2^3+5*5+sin(2/3)" + * @param const_names NULL terminated array of zero terminated strings of constant identifiers, for example {"PI", "E", 0} + * @param const_values a zero terminated array of values for the identifiers from const_names + * @param func1_names NULL terminated array of zero terminated strings of funcs1 identifiers + * @param funcs1 NULL terminated array of function pointers for functions which take 1 argument + * @param func2_names NULL terminated array of zero terminated strings of funcs2 identifiers + * @param funcs2 NULL terminated array of function pointers for functions which take 2 arguments + * @param opaque a pointer which will be passed to all functions from funcs1 and funcs2 + * @param log_ctx parent logging context + * @return 0 in case of success, a negative value corresponding to an + * AVERROR code otherwise + */ +int av_parse_and_eval_expr(double *res, const char *s, + const char * const *const_names, const double *const_values, + const char * const *func1_names, double (* const *funcs1)(void *, double), + const char * const *func2_names, double (* const *funcs2)(void *, double, double), + void *opaque, int log_offset, void *log_ctx); + +/** + * Parse an expression. + * + * @param expr a pointer where is put an AVExpr containing the parsed + * value in case of successfull parsing, or NULL otherwise. + * The pointed to AVExpr must be freed with av_free_expr() by the user + * when it is not needed anymore. + * @param s expression as a zero terminated string, for example "1+2^3+5*5+sin(2/3)" + * @param const_names NULL terminated array of zero terminated strings of constant identifiers, for example {"PI", "E", 0} + * @param func1_names NULL terminated array of zero terminated strings of funcs1 identifiers + * @param funcs1 NULL terminated array of function pointers for functions which take 1 argument + * @param func2_names NULL terminated array of zero terminated strings of funcs2 identifiers + * @param funcs2 NULL terminated array of function pointers for functions which take 2 arguments + * @param log_ctx parent logging context + * @return 0 in case of success, a negative value corresponding to an + * AVERROR code otherwise + */ +int av_parse_expr(AVExpr **expr, const char *s, + const char * const *const_names, + const char * const *func1_names, double (* const *funcs1)(void *, double), + const char * const *func2_names, double (* const *funcs2)(void *, double, double), + int log_offset, void *log_ctx); + +/** + * Evaluate a previously parsed expression. + * + * @param const_values a zero terminated array of values for the identifiers from av_parse_expr() const_names + * @param opaque a pointer which will be passed to all functions from funcs1 and funcs2 + * @return the value of the expression + */ +double av_eval_expr(AVExpr *e, const double *const_values, void *opaque); + +/** + * Free a parsed expression previously created with av_parse_expr(). + */ +void av_free_expr(AVExpr *e); + +/** + * Parse the string in numstr and return its value as a double. If + * the string is empty, contains only whitespaces, or does not contain + * an initial substring that has the expected syntax for a + * floating-point number, no conversion is performed. In this case, + * returns a value of zero and the value returned in tail is the value + * of numstr. + * + * @param numstr a string representing a number, may contain one of + * the International System number postfixes, for example 'K', 'M', + * 'G'. If 'i' is appended after the postfix, powers of 2 are used + * instead of powers of 10. The 'B' postfix multiplies the value for + * 8, and can be appended after another postfix or used alone. This + * allows using for example 'KB', 'MiB', 'G' and 'B' as postfix. + * @param tail if non-NULL puts here the pointer to the char next + * after the last parsed character + */ +double av_strtod(const char *numstr, char **tail); + +#endif /* AVUTIL_EVAL_H */ diff --git a/thirdparties/android/include/libavutil/fifo.h b/thirdparties/android/include/libavutil/fifo.h new file mode 100644 index 0000000..d12e523 --- /dev/null +++ b/thirdparties/android/include/libavutil/fifo.h @@ -0,0 +1,116 @@ +/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * a very simple circular buffer FIFO implementation
+ */
+
+#ifndef AVUTIL_FIFO_H
+#define AVUTIL_FIFO_H
+
+#include <stdint.h>
+
+typedef struct AVFifoBuffer {
+ uint8_t *buffer;
+ uint8_t *rptr, *wptr, *end;
+ uint32_t rndx, wndx;
+} AVFifoBuffer;
+
+/**
+ * Initialize an AVFifoBuffer.
+ * @param size of FIFO
+ * @return AVFifoBuffer or NULL in case of memory allocation failure
+ */
+AVFifoBuffer *av_fifo_alloc(unsigned int size);
+
+/**
+ * Free an AVFifoBuffer.
+ * @param *f AVFifoBuffer to free
+ */
+void av_fifo_free(AVFifoBuffer *f);
+
+/**
+ * Reset the AVFifoBuffer to the state right after av_fifo_alloc, in particular it is emptied.
+ * @param *f AVFifoBuffer to reset
+ */
+void av_fifo_reset(AVFifoBuffer *f);
+
+/**
+ * Return the amount of data in bytes in the AVFifoBuffer, that is the
+ * amount of data you can read from it.
+ * @param *f AVFifoBuffer to read from
+ * @return size
+ */
+int av_fifo_size(AVFifoBuffer *f);
+
+/**
+ * Return the amount of space in bytes in the AVFifoBuffer, that is the
+ * amount of data you can write into it.
+ * @param *f AVFifoBuffer to write into
+ * @return size
+ */
+int av_fifo_space(AVFifoBuffer *f);
+
+/**
+ * Feed data from an AVFifoBuffer to a user-supplied callback.
+ * @param *f AVFifoBuffer to read from
+ * @param buf_size number of bytes to read
+ * @param *func generic read function
+ * @param *dest data destination
+ */
+int av_fifo_generic_read(AVFifoBuffer *f, void *dest, int buf_size, void (*func)(void*, void*, int));
+
+/**
+ * Feed data from a user-supplied callback to an AVFifoBuffer.
+ * @param *f AVFifoBuffer to write to
+ * @param *src data source; non-const since it may be used as a
+ * modifiable context by the function defined in func
+ * @param size number of bytes to write
+ * @param *func generic write function; the first parameter is src,
+ * the second is dest_buf, the third is dest_buf_size.
+ * func must return the number of bytes written to dest_buf, or <= 0 to
+ * indicate no more data available to write.
+ * If func is NULL, src is interpreted as a simple byte array for source data.
+ * @return the number of bytes written to the FIFO
+ */
+int av_fifo_generic_write(AVFifoBuffer *f, void *src, int size, int (*func)(void*, void*, int));
+
+/**
+ * Resize an AVFifoBuffer.
+ * @param *f AVFifoBuffer to resize
+ * @param size new AVFifoBuffer size in bytes
+ * @return <0 for failure, >=0 otherwise
+ */
+int av_fifo_realloc2(AVFifoBuffer *f, unsigned int size);
+
+/**
+ * Read and discard the specified amount of data from an AVFifoBuffer.
+ * @param *f AVFifoBuffer to read from
+ * @param size amount of data to read in bytes
+ */
+void av_fifo_drain(AVFifoBuffer *f, int size);
+
+static inline uint8_t av_fifo_peek(AVFifoBuffer *f, int offs)
+{
+ uint8_t *ptr = f->rptr + offs;
+ if (ptr >= f->end)
+ ptr -= f->end - f->buffer;
+ return *ptr;
+}
+#endif /* AVUTIL_FIFO_H */
diff --git a/thirdparties/android/include/libavutil/intfloat_readwrite.h b/thirdparties/android/include/libavutil/intfloat_readwrite.h new file mode 100644 index 0000000..fda48a4 --- /dev/null +++ b/thirdparties/android/include/libavutil/intfloat_readwrite.h @@ -0,0 +1,40 @@ +/*
+ * copyright (c) 2005 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_INTFLOAT_READWRITE_H
+#define AVUTIL_INTFLOAT_READWRITE_H
+
+#include <stdint.h>
+#include "attributes.h"
+
+/* IEEE 80 bits extended float */
+typedef struct AVExtFloat {
+ uint8_t exponent[2];
+ uint8_t mantissa[8];
+} AVExtFloat;
+
+double av_int2dbl(int64_t v) av_const;
+float av_int2flt(int32_t v) av_const;
+double av_ext2dbl(const AVExtFloat ext) av_const;
+int64_t av_dbl2int(double d) av_const;
+int32_t av_flt2int(float d) av_const;
+AVExtFloat av_dbl2ext(double d) av_const;
+
+#endif /* AVUTIL_INTFLOAT_READWRITE_H */
diff --git a/thirdparties/android/include/libavutil/intreadwrite.h b/thirdparties/android/include/libavutil/intreadwrite.h new file mode 100644 index 0000000..1849a64 --- /dev/null +++ b/thirdparties/android/include/libavutil/intreadwrite.h @@ -0,0 +1,522 @@ +/* + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_INTREADWRITE_H +#define AVUTIL_INTREADWRITE_H + +#include <stdint.h> +#include "libavutil/avconfig.h" +#include "attributes.h" +#include "bswap.h" + +typedef union { + uint64_t u64; + uint32_t u32[2]; + uint16_t u16[4]; + uint8_t u8 [8]; + double f64; + float f32[2]; +} av_alias av_alias64; + +typedef union { + uint32_t u32; + uint16_t u16[2]; + uint8_t u8 [4]; + float f32; +} av_alias av_alias32; + +typedef union { + uint16_t u16; + uint8_t u8 [2]; +} av_alias av_alias16; + +/* + * Arch-specific headers can provide any combination of + * AV_[RW][BLN](16|24|32|64) and AV_(COPY|SWAP|ZERO)(64|128) macros. + * Preprocessor symbols must be defined, even if these are implemented + * as inline functions. + */ + +#ifdef HAVE_AV_CONFIG_H + +#include "config.h" + +#if ARCH_ARM +# include "arm/intreadwrite.h" +#elif ARCH_AVR32 +# include "avr32/intreadwrite.h" +#elif ARCH_MIPS +# include "mips/intreadwrite.h" +#elif ARCH_PPC +# include "ppc/intreadwrite.h" +#elif ARCH_TOMI +# include "tomi/intreadwrite.h" +#elif ARCH_X86 +# include "x86/intreadwrite.h" +#endif + +#endif /* HAVE_AV_CONFIG_H */ + +/* + * Map AV_RNXX <-> AV_R[BL]XX for all variants provided by per-arch headers. + */ + +#if AV_HAVE_BIGENDIAN + +# if defined(AV_RN16) && !defined(AV_RB16) +# define AV_RB16(p) AV_RN16(p) +# elif !defined(AV_RN16) && defined(AV_RB16) +# define AV_RN16(p) AV_RB16(p) +# endif + +# if defined(AV_WN16) && !defined(AV_WB16) +# define AV_WB16(p, v) AV_WN16(p, v) +# elif !defined(AV_WN16) && defined(AV_WB16) +# define AV_WN16(p, v) AV_WB16(p, v) +# endif + +# if defined(AV_RN24) && !defined(AV_RB24) +# define AV_RB24(p) AV_RN24(p) +# elif !defined(AV_RN24) && defined(AV_RB24) +# define AV_RN24(p) AV_RB24(p) +# endif + +# if defined(AV_WN24) && !defined(AV_WB24) +# define AV_WB24(p, v) AV_WN24(p, v) +# elif !defined(AV_WN24) && defined(AV_WB24) +# define AV_WN24(p, v) AV_WB24(p, v) +# endif + +# if defined(AV_RN32) && !defined(AV_RB32) +# define AV_RB32(p) AV_RN32(p) +# elif !defined(AV_RN32) && defined(AV_RB32) +# define AV_RN32(p) AV_RB32(p) +# endif + +# if defined(AV_WN32) && !defined(AV_WB32) +# define AV_WB32(p, v) AV_WN32(p, v) +# elif !defined(AV_WN32) && defined(AV_WB32) +# define AV_WN32(p, v) AV_WB32(p, v) +# endif + +# if defined(AV_RN64) && !defined(AV_RB64) +# define AV_RB64(p) AV_RN64(p) +# elif !defined(AV_RN64) && defined(AV_RB64) +# define AV_RN64(p) AV_RB64(p) +# endif + +# if defined(AV_WN64) && !defined(AV_WB64) +# define AV_WB64(p, v) AV_WN64(p, v) +# elif !defined(AV_WN64) && defined(AV_WB64) +# define AV_WN64(p, v) AV_WB64(p, v) +# endif + +#else /* AV_HAVE_BIGENDIAN */ + +# if defined(AV_RN16) && !defined(AV_RL16) +# define AV_RL16(p) AV_RN16(p) +# elif !defined(AV_RN16) && defined(AV_RL16) +# define AV_RN16(p) AV_RL16(p) +# endif + +# if defined(AV_WN16) && !defined(AV_WL16) +# define AV_WL16(p, v) AV_WN16(p, v) +# elif !defined(AV_WN16) && defined(AV_WL16) +# define AV_WN16(p, v) AV_WL16(p, v) +# endif + +# if defined(AV_RN24) && !defined(AV_RL24) +# define AV_RL24(p) AV_RN24(p) +# elif !defined(AV_RN24) && defined(AV_RL24) +# define AV_RN24(p) AV_RL24(p) +# endif + +# if defined(AV_WN24) && !defined(AV_WL24) +# define AV_WL24(p, v) AV_WN24(p, v) +# elif !defined(AV_WN24) && defined(AV_WL24) +# define AV_WN24(p, v) AV_WL24(p, v) +# endif + +# if defined(AV_RN32) && !defined(AV_RL32) +# define AV_RL32(p) AV_RN32(p) +# elif !defined(AV_RN32) && defined(AV_RL32) +# define AV_RN32(p) AV_RL32(p) +# endif + +# if defined(AV_WN32) && !defined(AV_WL32) +# define AV_WL32(p, v) AV_WN32(p, v) +# elif !defined(AV_WN32) && defined(AV_WL32) +# define AV_WN32(p, v) AV_WL32(p, v) +# endif + +# if defined(AV_RN64) && !defined(AV_RL64) +# define AV_RL64(p) AV_RN64(p) +# elif !defined(AV_RN64) && defined(AV_RL64) +# define AV_RN64(p) AV_RL64(p) +# endif + +# if defined(AV_WN64) && !defined(AV_WL64) +# define AV_WL64(p, v) AV_WN64(p, v) +# elif !defined(AV_WN64) && defined(AV_WL64) +# define AV_WN64(p, v) AV_WL64(p, v) +# endif + +#endif /* !AV_HAVE_BIGENDIAN */ + +/* + * Define AV_[RW]N helper macros to simplify definitions not provided + * by per-arch headers. + */ + +#if defined(__GNUC__) && !defined(__TI_COMPILER_VERSION__) + +union unaligned_64 { uint64_t l; } __attribute__((packed)) av_alias; +union unaligned_32 { uint32_t l; } __attribute__((packed)) av_alias; +union unaligned_16 { uint16_t l; } __attribute__((packed)) av_alias; + +# define AV_RN(s, p) (((const union unaligned_##s *) (p))->l) +# define AV_WN(s, p, v) ((((union unaligned_##s *) (p))->l) = (v)) + +#elif defined(__DECC) + +# define AV_RN(s, p) (*((const __unaligned uint##s##_t*)(p))) +# define AV_WN(s, p, v) (*((__unaligned uint##s##_t*)(p)) = (v)) + +#elif AV_HAVE_FAST_UNALIGNED + +# define AV_RN(s, p) (((const av_alias##s*)(p))->u##s) +# define AV_WN(s, p, v) (((av_alias##s*)(p))->u##s = (v)) + +#else + +#ifndef AV_RB16 +# define AV_RB16(x) \ + ((((const uint8_t*)(x))[0] << 8) | \ + ((const uint8_t*)(x))[1]) +#endif +#ifndef AV_WB16 +# define AV_WB16(p, d) do { \ + ((uint8_t*)(p))[1] = (d); \ + ((uint8_t*)(p))[0] = (d)>>8; \ + } while(0) +#endif + +#ifndef AV_RL16 +# define AV_RL16(x) \ + ((((const uint8_t*)(x))[1] << 8) | \ + ((const uint8_t*)(x))[0]) +#endif +#ifndef AV_WL16 +# define AV_WL16(p, d) do { \ + ((uint8_t*)(p))[0] = (d); \ + ((uint8_t*)(p))[1] = (d)>>8; \ + } while(0) +#endif + +#ifndef AV_RB32 +# define AV_RB32(x) \ + ((((const uint8_t*)(x))[0] << 24) | \ + (((const uint8_t*)(x))[1] << 16) | \ + (((const uint8_t*)(x))[2] << 8) | \ + ((const uint8_t*)(x))[3]) +#endif +#ifndef AV_WB32 +# define AV_WB32(p, d) do { \ + ((uint8_t*)(p))[3] = (d); \ + ((uint8_t*)(p))[2] = (d)>>8; \ + ((uint8_t*)(p))[1] = (d)>>16; \ + ((uint8_t*)(p))[0] = (d)>>24; \ + } while(0) +#endif + +#ifndef AV_RL32 +# define AV_RL32(x) \ + ((((const uint8_t*)(x))[3] << 24) | \ + (((const uint8_t*)(x))[2] << 16) | \ + (((const uint8_t*)(x))[1] << 8) | \ + ((const uint8_t*)(x))[0]) +#endif +#ifndef AV_WL32 +# define AV_WL32(p, d) do { \ + ((uint8_t*)(p))[0] = (d); \ + ((uint8_t*)(p))[1] = (d)>>8; \ + ((uint8_t*)(p))[2] = (d)>>16; \ + ((uint8_t*)(p))[3] = (d)>>24; \ + } while(0) +#endif + +#ifndef AV_RB64 +# define AV_RB64(x) \ + (((uint64_t)((const uint8_t*)(x))[0] << 56) | \ + ((uint64_t)((const uint8_t*)(x))[1] << 48) | \ + ((uint64_t)((const uint8_t*)(x))[2] << 40) | \ + ((uint64_t)((const uint8_t*)(x))[3] << 32) | \ + ((uint64_t)((const uint8_t*)(x))[4] << 24) | \ + ((uint64_t)((const uint8_t*)(x))[5] << 16) | \ + ((uint64_t)((const uint8_t*)(x))[6] << 8) | \ + (uint64_t)((const uint8_t*)(x))[7]) +#endif +#ifndef AV_WB64 +# define AV_WB64(p, d) do { \ + ((uint8_t*)(p))[7] = (d); \ + ((uint8_t*)(p))[6] = (d)>>8; \ + ((uint8_t*)(p))[5] = (d)>>16; \ + ((uint8_t*)(p))[4] = (d)>>24; \ + ((uint8_t*)(p))[3] = (d)>>32; \ + ((uint8_t*)(p))[2] = (d)>>40; \ + ((uint8_t*)(p))[1] = (d)>>48; \ + ((uint8_t*)(p))[0] = (d)>>56; \ + } while(0) +#endif + +#ifndef AV_RL64 +# define AV_RL64(x) \ + (((uint64_t)((const uint8_t*)(x))[7] << 56) | \ + ((uint64_t)((const uint8_t*)(x))[6] << 48) | \ + ((uint64_t)((const uint8_t*)(x))[5] << 40) | \ + ((uint64_t)((const uint8_t*)(x))[4] << 32) | \ + ((uint64_t)((const uint8_t*)(x))[3] << 24) | \ + ((uint64_t)((const uint8_t*)(x))[2] << 16) | \ + ((uint64_t)((const uint8_t*)(x))[1] << 8) | \ + (uint64_t)((const uint8_t*)(x))[0]) +#endif +#ifndef AV_WL64 +# define AV_WL64(p, d) do { \ + ((uint8_t*)(p))[0] = (d); \ + ((uint8_t*)(p))[1] = (d)>>8; \ + ((uint8_t*)(p))[2] = (d)>>16; \ + ((uint8_t*)(p))[3] = (d)>>24; \ + ((uint8_t*)(p))[4] = (d)>>32; \ + ((uint8_t*)(p))[5] = (d)>>40; \ + ((uint8_t*)(p))[6] = (d)>>48; \ + ((uint8_t*)(p))[7] = (d)>>56; \ + } while(0) +#endif + +#if AV_HAVE_BIGENDIAN +# define AV_RN(s, p) AV_RB##s(p) +# define AV_WN(s, p, v) AV_WB##s(p, v) +#else +# define AV_RN(s, p) AV_RL##s(p) +# define AV_WN(s, p, v) AV_WL##s(p, v) +#endif + +#endif /* HAVE_FAST_UNALIGNED */ + +#ifndef AV_RN16 +# define AV_RN16(p) AV_RN(16, p) +#endif + +#ifndef AV_RN32 +# define AV_RN32(p) AV_RN(32, p) +#endif + +#ifndef AV_RN64 +# define AV_RN64(p) AV_RN(64, p) +#endif + +#ifndef AV_WN16 +# define AV_WN16(p, v) AV_WN(16, p, v) +#endif + +#ifndef AV_WN32 +# define AV_WN32(p, v) AV_WN(32, p, v) +#endif + +#ifndef AV_WN64 +# define AV_WN64(p, v) AV_WN(64, p, v) +#endif + +#if AV_HAVE_BIGENDIAN +# define AV_RB(s, p) AV_RN##s(p) +# define AV_WB(s, p, v) AV_WN##s(p, v) +# define AV_RL(s, p) av_bswap##s(AV_RN##s(p)) +# define AV_WL(s, p, v) AV_WN##s(p, av_bswap##s(v)) +#else +# define AV_RB(s, p) av_bswap##s(AV_RN##s(p)) +# define AV_WB(s, p, v) AV_WN##s(p, av_bswap##s(v)) +# define AV_RL(s, p) AV_RN##s(p) +# define AV_WL(s, p, v) AV_WN##s(p, v) +#endif + +#define AV_RB8(x) (((const uint8_t*)(x))[0]) +#define AV_WB8(p, d) do { ((uint8_t*)(p))[0] = (d); } while(0) + +#define AV_RL8(x) AV_RB8(x) +#define AV_WL8(p, d) AV_WB8(p, d) + +#ifndef AV_RB16 +# define AV_RB16(p) AV_RB(16, p) +#endif +#ifndef AV_WB16 +# define AV_WB16(p, v) AV_WB(16, p, v) +#endif + +#ifndef AV_RL16 +# define AV_RL16(p) AV_RL(16, p) +#endif +#ifndef AV_WL16 +# define AV_WL16(p, v) AV_WL(16, p, v) +#endif + +#ifndef AV_RB32 +# define AV_RB32(p) AV_RB(32, p) +#endif +#ifndef AV_WB32 +# define AV_WB32(p, v) AV_WB(32, p, v) +#endif + +#ifndef AV_RL32 +# define AV_RL32(p) AV_RL(32, p) +#endif +#ifndef AV_WL32 +# define AV_WL32(p, v) AV_WL(32, p, v) +#endif + +#ifndef AV_RB64 +# define AV_RB64(p) AV_RB(64, p) +#endif +#ifndef AV_WB64 +# define AV_WB64(p, v) AV_WB(64, p, v) +#endif + +#ifndef AV_RL64 +# define AV_RL64(p) AV_RL(64, p) +#endif +#ifndef AV_WL64 +# define AV_WL64(p, v) AV_WL(64, p, v) +#endif + +#ifndef AV_RB24 +# define AV_RB24(x) \ + ((((const uint8_t*)(x))[0] << 16) | \ + (((const uint8_t*)(x))[1] << 8) | \ + ((const uint8_t*)(x))[2]) +#endif +#ifndef AV_WB24 +# define AV_WB24(p, d) do { \ + ((uint8_t*)(p))[2] = (d); \ + ((uint8_t*)(p))[1] = (d)>>8; \ + ((uint8_t*)(p))[0] = (d)>>16; \ + } while(0) +#endif + +#ifndef AV_RL24 +# define AV_RL24(x) \ + ((((const uint8_t*)(x))[2] << 16) | \ + (((const uint8_t*)(x))[1] << 8) | \ + ((const uint8_t*)(x))[0]) +#endif +#ifndef AV_WL24 +# define AV_WL24(p, d) do { \ + ((uint8_t*)(p))[0] = (d); \ + ((uint8_t*)(p))[1] = (d)>>8; \ + ((uint8_t*)(p))[2] = (d)>>16; \ + } while(0) +#endif + +/* + * The AV_[RW]NA macros access naturally aligned data + * in a type-safe way. + */ + +#define AV_RNA(s, p) (((const av_alias##s*)(p))->u##s) +#define AV_WNA(s, p, v) (((av_alias##s*)(p))->u##s = (v)) + +#ifndef AV_RN16A +# define AV_RN16A(p) AV_RNA(16, p) +#endif + +#ifndef AV_RN32A +# define AV_RN32A(p) AV_RNA(32, p) +#endif + +#ifndef AV_RN64A +# define AV_RN64A(p) AV_RNA(64, p) +#endif + +#ifndef AV_WN16A +# define AV_WN16A(p, v) AV_WNA(16, p, v) +#endif + +#ifndef AV_WN32A +# define AV_WN32A(p, v) AV_WNA(32, p, v) +#endif + +#ifndef AV_WN64A +# define AV_WN64A(p, v) AV_WNA(64, p, v) +#endif + +/* Parameters for AV_COPY*, AV_SWAP*, AV_ZERO* must be + * naturally aligned. They may be implemented using MMX, + * so emms_c() must be called before using any float code + * afterwards. + */ + +#define AV_COPY(n, d, s) \ + (((av_alias##n*)(d))->u##n = ((const av_alias##n*)(s))->u##n) + +#ifndef AV_COPY16 +# define AV_COPY16(d, s) AV_COPY(16, d, s) +#endif + +#ifndef AV_COPY32 +# define AV_COPY32(d, s) AV_COPY(32, d, s) +#endif + +#ifndef AV_COPY64 +# define AV_COPY64(d, s) AV_COPY(64, d, s) +#endif + +#ifndef AV_COPY128 +# define AV_COPY128(d, s) \ + do { \ + AV_COPY64(d, s); \ + AV_COPY64((char*)(d)+8, (char*)(s)+8); \ + } while(0) +#endif + +#define AV_SWAP(n, a, b) FFSWAP(av_alias##n, *(av_alias##n*)(a), *(av_alias##n*)(b)) + +#ifndef AV_SWAP64 +# define AV_SWAP64(a, b) AV_SWAP(64, a, b) +#endif + +#define AV_ZERO(n, d) (((av_alias##n*)(d))->u##n = 0) + +#ifndef AV_ZERO16 +# define AV_ZERO16(d) AV_ZERO(16, d) +#endif + +#ifndef AV_ZERO32 +# define AV_ZERO32(d) AV_ZERO(32, d) +#endif + +#ifndef AV_ZERO64 +# define AV_ZERO64(d) AV_ZERO(64, d) +#endif + +#ifndef AV_ZERO128 +# define AV_ZERO128(d) \ + do { \ + AV_ZERO64(d); \ + AV_ZERO64((char*)(d)+8); \ + } while(0) +#endif + +#endif /* AVUTIL_INTREADWRITE_H */ diff --git a/thirdparties/android/include/libavutil/lfg.h b/thirdparties/android/include/libavutil/lfg.h new file mode 100644 index 0000000..0e89ea3 --- /dev/null +++ b/thirdparties/android/include/libavutil/lfg.h @@ -0,0 +1,62 @@ +/* + * Lagged Fibonacci PRNG + * Copyright (c) 2008 Michael Niedermayer + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_LFG_H +#define AVUTIL_LFG_H + +typedef struct { + unsigned int state[64]; + int index; +} AVLFG; + +void av_lfg_init(AVLFG *c, unsigned int seed); + +/** + * Get the next random unsigned 32-bit number using an ALFG. + * + * Please also consider a simple LCG like state= state*1664525+1013904223, + * it may be good enough and faster for your specific use case. + */ +static inline unsigned int av_lfg_get(AVLFG *c){ + c->state[c->index & 63] = c->state[(c->index-24) & 63] + c->state[(c->index-55) & 63]; + return c->state[c->index++ & 63]; +} + +/** + * Get the next random unsigned 32-bit number using a MLFG. + * + * Please also consider av_lfg_get() above, it is faster. + */ +static inline unsigned int av_mlfg_get(AVLFG *c){ + unsigned int a= c->state[(c->index-55) & 63]; + unsigned int b= c->state[(c->index-24) & 63]; + return c->state[c->index++ & 63] = 2*a*b+a+b; +} + +/** + * Get the next two numbers generated by a Box-Muller Gaussian + * generator using the random numbers issued by lfg. + * + * @param out[2] array where the two generated numbers are placed + */ +void av_bmg_get(AVLFG *lfg, double out[2]); + +#endif /* AVUTIL_LFG_H */ diff --git a/thirdparties/android/include/libavutil/log.h b/thirdparties/android/include/libavutil/log.h new file mode 100644 index 0000000..831c26e --- /dev/null +++ b/thirdparties/android/include/libavutil/log.h @@ -0,0 +1,138 @@ +/* + * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_LOG_H +#define AVUTIL_LOG_H + +#include <stdarg.h> +#include "avutil.h" + +/** + * Describe the class of an AVClass context structure. That is an + * arbitrary struct of which the first field is a pointer to an + * AVClass struct (e.g. AVCodecContext, AVFormatContext etc.). + */ +typedef struct { + /** + * The name of the class; usually it is the same name as the + * context structure type to which the AVClass is associated. + */ + const char* class_name; + + /** + * A pointer to a function which returns the name of a context + * instance ctx associated with the class. + */ + const char* (*item_name)(void* ctx); + + /** + * a pointer to the first option specified in the class if any or NULL + * + * @see av_set_default_options() + */ + const struct AVOption *option; + + /** + * LIBAVUTIL_VERSION with which this structure was created. + * This is used to allow fields to be added without requiring major + * version bumps everywhere. + */ + + int version; + + /** + * Offset in the structure where log_level_offset is stored. + * 0 means there is no such variable + */ + int log_level_offset_offset; + + /** + * Offset in the structure where a pointer to the parent context for loging is stored. + * for example a decoder that uses eval.c could pass its AVCodecContext to eval as such + * parent context. And a av_log() implementation could then display the parent context + * can be NULL of course + */ + int parent_log_context_offset; +} AVClass; + +/* av_log API */ + +#define AV_LOG_QUIET -8 + +/** + * Something went really wrong and we will crash now. + */ +#define AV_LOG_PANIC 0 + +/** + * Something went wrong and recovery is not possible. + * For example, no header was found for a format which depends + * on headers or an illegal combination of parameters is used. + */ +#define AV_LOG_FATAL 8 + +/** + * Something went wrong and cannot losslessly be recovered. + * However, not all future data is affected. + */ +#define AV_LOG_ERROR 16 + +/** + * Something somehow does not look correct. This may or may not + * lead to problems. An example would be the use of '-vstrict -2'. + */ +#define AV_LOG_WARNING 24 + +#define AV_LOG_INFO 32 +#define AV_LOG_VERBOSE 40 + +/** + * Stuff which is only useful for libav* developers. + */ +#define AV_LOG_DEBUG 48 + +/** + * Send the specified message to the log if the level is less than or equal + * to the current av_log_level. By default, all logging messages are sent to + * stderr. This behavior can be altered by setting a different av_vlog callback + * function. + * + * @param avcl A pointer to an arbitrary struct of which the first field is a + * pointer to an AVClass struct. + * @param level The importance level of the message, lower values signifying + * higher importance. + * @param fmt The format string (printf-compatible) that specifies how + * subsequent arguments are converted to output. + * @see av_vlog + */ +#ifdef __GNUC__ +void av_log(void *avcl, int level, const char *fmt, ...) __attribute__ ((__format__ (__printf__, 3, 4))); +#else +void av_log(void *avcl, int level, const char *fmt, ...); +#endif + +void av_vlog(void *avcl, int level, const char *fmt, va_list); +int av_log_get_level(void); +void av_log_set_level(int); +void av_log_set_callback(void (*)(void*, int, const char*, va_list)); +void av_log_default_callback(void* ptr, int level, const char* fmt, va_list vl); +const char* av_default_item_name(void* ctx); + +#endif /* AVUTIL_LOG_H */ diff --git a/thirdparties/android/include/libavutil/lzo.h b/thirdparties/android/include/libavutil/lzo.h new file mode 100644 index 0000000..deef81b --- /dev/null +++ b/thirdparties/android/include/libavutil/lzo.h @@ -0,0 +1,66 @@ +/*
+ * LZO 1x decompression
+ * copyright (c) 2006 Reimar Doeffinger
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_LZO_H
+#define AVUTIL_LZO_H
+
+#include <stdint.h>
+
+/** \defgroup errflags Error flags returned by av_lzo1x_decode
+ * \{ */
+//! end of the input buffer reached before decoding finished
+#define AV_LZO_INPUT_DEPLETED 1
+//! decoded data did not fit into output buffer
+#define AV_LZO_OUTPUT_FULL 2
+//! a reference to previously decoded data was wrong
+#define AV_LZO_INVALID_BACKPTR 4
+//! a non-specific error in the compressed bitstream
+#define AV_LZO_ERROR 8
+/** \} */
+
+#define AV_LZO_INPUT_PADDING 8
+#define AV_LZO_OUTPUT_PADDING 12
+
+/**
+ * \brief Decodes LZO 1x compressed data.
+ * \param out output buffer
+ * \param outlen size of output buffer, number of bytes left are returned here
+ * \param in input buffer
+ * \param inlen size of input buffer, number of bytes left are returned here
+ * \return 0 on success, otherwise a combination of the error flags above
+ *
+ * Make sure all buffers are appropriately padded, in must provide
+ * AV_LZO_INPUT_PADDING, out must provide AV_LZO_OUTPUT_PADDING additional bytes.
+ */
+int av_lzo1x_decode(void *out, int *outlen, const void *in, int *inlen);
+
+/**
+ * \brief deliberately overlapping memcpy implementation
+ * \param dst destination buffer; must be padded with 12 additional bytes
+ * \param back how many bytes back we start (the initial size of the overlapping window)
+ * \param cnt number of bytes to copy, must be >= 0
+ *
+ * cnt > back is valid, this will copy the bytes we just copied,
+ * thus creating a repeating pattern with a period length of back.
+ */
+void av_memcpy_backptr(uint8_t *dst, int back, int cnt);
+
+#endif /* AVUTIL_LZO_H */
diff --git a/thirdparties/android/include/libavutil/mathematics.h b/thirdparties/android/include/libavutil/mathematics.h new file mode 100644 index 0000000..df2821f --- /dev/null +++ b/thirdparties/android/include/libavutil/mathematics.h @@ -0,0 +1,112 @@ +/*
+ * copyright (c) 2005 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_MATHEMATICS_H
+#define AVUTIL_MATHEMATICS_H
+
+#include <stdint.h>
+#include <math.h>
+#include "attributes.h"
+#include "rational.h"
+
+#ifndef M_E
+#define M_E 2.7182818284590452354 /* e */
+#endif
+#ifndef M_LN2
+#define M_LN2 0.69314718055994530942 /* log_e 2 */
+#endif
+#ifndef M_LN10
+#define M_LN10 2.30258509299404568402 /* log_e 10 */
+#endif
+#ifndef M_LOG2_10
+#define M_LOG2_10 3.32192809488736234787 /* log_2 10 */
+#endif
+#ifndef M_PHI
+#define M_PHI 1.61803398874989484820 /* phi / golden ratio */
+#endif
+#ifndef M_PI
+#define M_PI 3.14159265358979323846 /* pi */
+#endif
+#ifndef M_SQRT1_2
+#define M_SQRT1_2 0.70710678118654752440 /* 1/sqrt(2) */
+#endif
+#ifndef M_SQRT2
+#define M_SQRT2 1.41421356237309504880 /* sqrt(2) */
+#endif
+#ifndef NAN
+#define NAN (0.0/0.0)
+#endif
+#ifndef INFINITY
+#define INFINITY (1.0/0.0)
+#endif
+
+enum AVRounding {
+ AV_ROUND_ZERO = 0, ///< Round toward zero.
+ AV_ROUND_INF = 1, ///< Round away from zero.
+ AV_ROUND_DOWN = 2, ///< Round toward -infinity.
+ AV_ROUND_UP = 3, ///< Round toward +infinity.
+ AV_ROUND_NEAR_INF = 5, ///< Round to nearest and halfway cases away from zero.
+};
+
+/**
+ * Return the greatest common divisor of a and b.
+ * If both a and b are 0 or either or both are <0 then behavior is
+ * undefined.
+ */
+int64_t av_const av_gcd(int64_t a, int64_t b);
+
+/**
+ * Rescale a 64-bit integer with rounding to nearest.
+ * A simple a*b/c isn't possible as it can overflow.
+ */
+int64_t av_rescale(int64_t a, int64_t b, int64_t c) av_const;
+
+/**
+ * Rescale a 64-bit integer with specified rounding.
+ * A simple a*b/c isn't possible as it can overflow.
+ */
+int64_t av_rescale_rnd(int64_t a, int64_t b, int64_t c, enum AVRounding) av_const;
+
+/**
+ * Rescale a 64-bit integer by 2 rational numbers.
+ */
+int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq) av_const;
+
+/**
+ * Compare 2 timestamps each in its own timebases.
+ * The result of the function is undefined if one of the timestamps
+ * is outside the int64_t range when represented in the others timebase.
+ * @return -1 if ts_a is before ts_b, 1 if ts_a is after ts_b or 0 if they represent the same position
+ */
+int av_compare_ts(int64_t ts_a, AVRational tb_a, int64_t ts_b, AVRational tb_b);
+
+/**
+ * Compare 2 integers modulo mod.
+ * That is we compare integers a and b for which only the least
+ * significant log2(mod) bits are known.
+ *
+ * @param mod must be a power of 2
+ * @return a negative value if a is smaller than b
+ * a positive value if a is greater than b
+ * 0 if a equals b
+ */
+int64_t av_compare_mod(uint64_t a, uint64_t b, uint64_t mod);
+
+#endif /* AVUTIL_MATHEMATICS_H */
diff --git a/thirdparties/android/include/libavutil/md5.h b/thirdparties/android/include/libavutil/md5.h new file mode 100644 index 0000000..969202a --- /dev/null +++ b/thirdparties/android/include/libavutil/md5.h @@ -0,0 +1,36 @@ +/* + * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_MD5_H +#define AVUTIL_MD5_H + +#include <stdint.h> + +extern const int av_md5_size; + +struct AVMD5; + +void av_md5_init(struct AVMD5 *ctx); +void av_md5_update(struct AVMD5 *ctx, const uint8_t *src, const int len); +void av_md5_final(struct AVMD5 *ctx, uint8_t *dst); +void av_md5_sum(uint8_t *dst, const uint8_t *src, const int len); + +#endif /* AVUTIL_MD5_H */ + diff --git a/thirdparties/android/include/libavutil/mem.h b/thirdparties/android/include/libavutil/mem.h new file mode 100644 index 0000000..cc21305 --- /dev/null +++ b/thirdparties/android/include/libavutil/mem.h @@ -0,0 +1,126 @@ +/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * memory handling functions
+ */
+
+#ifndef AVUTIL_MEM_H
+#define AVUTIL_MEM_H
+
+#include "attributes.h"
+#include "avutil.h"
+
+#if defined(__ICC) && _ICC < 1200 || defined(__SUNPRO_C)
+ #define DECLARE_ALIGNED(n,t,v) t __attribute__ ((aligned (n))) v
+ #define DECLARE_ASM_CONST(n,t,v) const t __attribute__ ((aligned (n))) v
+#elif defined(__TI_COMPILER_VERSION__)
+ #define DECLARE_ALIGNED(n,t,v) \
+ AV_PRAGMA(DATA_ALIGN(v,n)) \
+ t __attribute__((aligned(n))) v
+ #define DECLARE_ASM_CONST(n,t,v) \
+ AV_PRAGMA(DATA_ALIGN(v,n)) \
+ static const t __attribute__((aligned(n))) v
+#elif defined(__GNUC__)
+ #define DECLARE_ALIGNED(n,t,v) t __attribute__ ((aligned (n))) v
+ #define DECLARE_ASM_CONST(n,t,v) static const t attribute_used __attribute__ ((aligned (n))) v
+#elif defined(_MSC_VER)
+ #define DECLARE_ALIGNED(n,t,v) __declspec(align(n)) t v
+ #define DECLARE_ASM_CONST(n,t,v) __declspec(align(n)) static const t v
+#else
+ #define DECLARE_ALIGNED(n,t,v) t v
+ #define DECLARE_ASM_CONST(n,t,v) static const t v
+#endif
+
+#if AV_GCC_VERSION_AT_LEAST(3,1)
+ #define av_malloc_attrib __attribute__((__malloc__))
+#else
+ #define av_malloc_attrib
+#endif
+
+#if (!defined(__ICC) || __ICC > 1110) && AV_GCC_VERSION_AT_LEAST(4,3)
+ #define av_alloc_size(n) __attribute__((alloc_size(n)))
+#else
+ #define av_alloc_size(n)
+#endif
+
+/**
+ * Allocate a block of size bytes with alignment suitable for all
+ * memory accesses (including vectors if available on the CPU).
+ * @param size Size in bytes for the memory block to be allocated.
+ * @return Pointer to the allocated block, NULL if the block cannot
+ * be allocated.
+ * @see av_mallocz()
+ */
+void *av_malloc(unsigned int size) av_malloc_attrib av_alloc_size(1);
+
+/**
+ * Allocate or reallocate a block of memory.
+ * If ptr is NULL and size > 0, allocate a new block. If
+ * size is zero, free the memory block pointed to by ptr.
+ * @param size Size in bytes for the memory block to be allocated or
+ * reallocated.
+ * @param ptr Pointer to a memory block already allocated with
+ * av_malloc(z)() or av_realloc() or NULL.
+ * @return Pointer to a newly reallocated block or NULL if the block
+ * cannot be reallocated or the function is used to free the memory block.
+ * @see av_fast_realloc()
+ */
+void *av_realloc(void *ptr, unsigned int size) av_alloc_size(2);
+
+/**
+ * Free a memory block which has been allocated with av_malloc(z)() or
+ * av_realloc().
+ * @param ptr Pointer to the memory block which should be freed.
+ * @note ptr = NULL is explicitly allowed.
+ * @note It is recommended that you use av_freep() instead.
+ * @see av_freep()
+ */
+void av_free(void *ptr);
+
+/**
+ * Allocate a block of size bytes with alignment suitable for all
+ * memory accesses (including vectors if available on the CPU) and
+ * zero all the bytes of the block.
+ * @param size Size in bytes for the memory block to be allocated.
+ * @return Pointer to the allocated block, NULL if it cannot be allocated.
+ * @see av_malloc()
+ */
+void *av_mallocz(unsigned int size) av_malloc_attrib av_alloc_size(1);
+
+/**
+ * Duplicate the string s.
+ * @param s string to be duplicated
+ * @return Pointer to a newly allocated string containing a
+ * copy of s or NULL if the string cannot be allocated.
+ */
+char *av_strdup(const char *s) av_malloc_attrib;
+
+/**
+ * Free a memory block which has been allocated with av_malloc(z)() or
+ * av_realloc() and set the pointer pointing to it to NULL.
+ * @param ptr Pointer to the pointer to the memory block which should
+ * be freed.
+ * @see av_free()
+ */
+void av_freep(void *ptr);
+
+#endif /* AVUTIL_MEM_H */
diff --git a/thirdparties/android/include/libavutil/pixdesc.h b/thirdparties/android/include/libavutil/pixdesc.h new file mode 100644 index 0000000..8d131be --- /dev/null +++ b/thirdparties/android/include/libavutil/pixdesc.h @@ -0,0 +1,154 @@ +/* + * pixel format descriptor + * Copyright (c) 2009 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_PIXDESC_H +#define AVUTIL_PIXDESC_H + +#include <inttypes.h> + +typedef struct AVComponentDescriptor{ + uint16_t plane :2; ///< which of the 4 planes contains the component + + /** + * Number of elements between 2 horizontally consecutive pixels minus 1. + * Elements are bits for bitstream formats, bytes otherwise. + */ + uint16_t step_minus1 :3; + + /** + * Number of elements before the component of the first pixel plus 1. + * Elements are bits for bitstream formats, bytes otherwise. + */ + uint16_t offset_plus1 :3; + uint16_t shift :3; ///< number of least significant bits that must be shifted away to get the value + uint16_t depth_minus1 :4; ///< number of bits in the component minus 1 +}AVComponentDescriptor; + +/** + * Descriptor that unambiguously describes how the bits of a pixel are + * stored in the up to 4 data planes of an image. It also stores the + * subsampling factors and number of components. + * + * @note This is separate of the colorspace (RGB, YCbCr, YPbPr, JPEG-style YUV + * and all the YUV variants) AVPixFmtDescriptor just stores how values + * are stored not what these values represent. + */ +typedef struct AVPixFmtDescriptor{ + const char *name; + uint8_t nb_components; ///< The number of components each pixel has, (1-4) + + /** + * Amount to shift the luma width right to find the chroma width. + * For YV12 this is 1 for example. + * chroma_width = -((-luma_width) >> log2_chroma_w) + * The note above is needed to ensure rounding up. + * This value only refers to the chroma components. + */ + uint8_t log2_chroma_w; ///< chroma_width = -((-luma_width )>>log2_chroma_w) + + /** + * Amount to shift the luma height right to find the chroma height. + * For YV12 this is 1 for example. + * chroma_height= -((-luma_height) >> log2_chroma_h) + * The note above is needed to ensure rounding up. + * This value only refers to the chroma components. + */ + uint8_t log2_chroma_h; + uint8_t flags; + + /** + * Parameters that describe how pixels are packed. If the format + * has chroma components, they must be stored in comp[1] and + * comp[2]. + */ + AVComponentDescriptor comp[4]; +}AVPixFmtDescriptor; + +#define PIX_FMT_BE 1 ///< Pixel format is big-endian. +#define PIX_FMT_PAL 2 ///< Pixel format has a palette in data[1], values are indexes in this palette. +#define PIX_FMT_BITSTREAM 4 ///< All values of a component are bit-wise packed end to end. +#define PIX_FMT_HWACCEL 8 ///< Pixel format is an HW accelerated format. + +/** + * The array of all the pixel format descriptors. + */ +extern const AVPixFmtDescriptor av_pix_fmt_descriptors[]; + +/** + * Read a line from an image, and write the values of the + * pixel format component c to dst. + * + * @param data the array containing the pointers to the planes of the image + * @param linesize the array containing the linesizes of the image + * @param desc the pixel format descriptor for the image + * @param x the horizontal coordinate of the first pixel to read + * @param y the vertical coordinate of the first pixel to read + * @param w the width of the line to read, that is the number of + * values to write to dst + * @param read_pal_component if not zero and the format is a paletted + * format writes the values corresponding to the palette + * component c in data[1] to dst, rather than the palette indexes in + * data[0]. The behavior is undefined if the format is not paletted. + */ +void av_read_image_line(uint16_t *dst, const uint8_t *data[4], const int linesize[4], + const AVPixFmtDescriptor *desc, int x, int y, int c, int w, int read_pal_component); + +/** + * Write the values from src to the pixel format component c of an + * image line. + * + * @param src array containing the values to write + * @param data the array containing the pointers to the planes of the + * image to write into. It is supposed to be zeroed. + * @param linesize the array containing the linesizes of the image + * @param desc the pixel format descriptor for the image + * @param x the horizontal coordinate of the first pixel to write + * @param y the vertical coordinate of the first pixel to write + * @param w the width of the line to write, that is the number of + * values to write to the image line + */ +void av_write_image_line(const uint16_t *src, uint8_t *data[4], const int linesize[4], + const AVPixFmtDescriptor *desc, int x, int y, int c, int w); + +/** + * Return the pixel format corresponding to name. + * + * If there is no pixel format with name name, then looks for a + * pixel format with the name corresponding to the native endian + * format of name. + * For example in a little-endian system, first looks for "gray16", + * then for "gray16le". + * + * Finally if no pixel format has been found, returns PIX_FMT_NONE. + */ +enum PixelFormat av_get_pix_fmt(const char *name); + +/** + * Return the number of bits per pixel used by the pixel format + * described by pixdesc. + * + * The returned number of bits refers to the number of bits actually + * used for storing the pixel information, that is padding bits are + * not counted. + */ +int av_get_bits_per_pixel(const AVPixFmtDescriptor *pixdesc); + +#endif /* AVUTIL_PIXDESC_H */ diff --git a/thirdparties/android/include/libavutil/pixfmt.h b/thirdparties/android/include/libavutil/pixfmt.h new file mode 100644 index 0000000..e7a150d --- /dev/null +++ b/thirdparties/android/include/libavutil/pixfmt.h @@ -0,0 +1,163 @@ +/*
+ * copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVUTIL_PIXFMT_H
+#define AVUTIL_PIXFMT_H
+
+/**
+ * @file
+ * pixel format definitions
+ *
+ * @warning This file has to be considered an internal but installed
+ * header, so it should not be directly included in your projects.
+ */
+
+#include "libavutil/avconfig.h"
+
+/**
+ * Pixel format. Notes:
+ *
+ * PIX_FMT_RGB32 is handled in an endian-specific manner. An RGBA
+ * color is put together as:
+ * (A << 24) | (R << 16) | (G << 8) | B
+ * This is stored as BGRA on little-endian CPU architectures and ARGB on
+ * big-endian CPUs.
+ *
+ * When the pixel format is palettized RGB (PIX_FMT_PAL8), the palettized
+ * image data is stored in AVFrame.data[0]. The palette is transported in
+ * AVFrame.data[1], is 1024 bytes long (256 4-byte entries) and is
+ * formatted the same as in PIX_FMT_RGB32 described above (i.e., it is
+ * also endian-specific). Note also that the individual RGB palette
+ * components stored in AVFrame.data[1] should be in the range 0..255.
+ * This is important as many custom PAL8 video codecs that were designed
+ * to run on the IBM VGA graphics adapter use 6-bit palette components.
+ *
+ * For all the 8bit per pixel formats, an RGB32 palette is in data[1] like
+ * for pal8. This palette is filled in automatically by the function
+ * allocating the picture.
+ *
+ * Note, make sure that all newly added big endian formats have pix_fmt&1==1
+ * and that all newly added little endian formats have pix_fmt&1==0
+ * this allows simpler detection of big vs little endian.
+ */
+enum PixelFormat {
+ PIX_FMT_NONE= -1,
+ PIX_FMT_YUV420P, ///< planar YUV 4:2:0, 12bpp, (1 Cr & Cb sample per 2x2 Y samples)
+ PIX_FMT_YUYV422, ///< packed YUV 4:2:2, 16bpp, Y0 Cb Y1 Cr
+ PIX_FMT_RGB24, ///< packed RGB 8:8:8, 24bpp, RGBRGB...
+ PIX_FMT_BGR24, ///< packed RGB 8:8:8, 24bpp, BGRBGR...
+ PIX_FMT_YUV422P, ///< planar YUV 4:2:2, 16bpp, (1 Cr & Cb sample per 2x1 Y samples)
+ PIX_FMT_YUV444P, ///< planar YUV 4:4:4, 24bpp, (1 Cr & Cb sample per 1x1 Y samples)
+ PIX_FMT_YUV410P, ///< planar YUV 4:1:0, 9bpp, (1 Cr & Cb sample per 4x4 Y samples)
+ PIX_FMT_YUV411P, ///< planar YUV 4:1:1, 12bpp, (1 Cr & Cb sample per 4x1 Y samples)
+ PIX_FMT_GRAY8, ///< Y , 8bpp
+ PIX_FMT_MONOWHITE, ///< Y , 1bpp, 0 is white, 1 is black, in each byte pixels are ordered from the msb to the lsb
+ PIX_FMT_MONOBLACK, ///< Y , 1bpp, 0 is black, 1 is white, in each byte pixels are ordered from the msb to the lsb
+ PIX_FMT_PAL8, ///< 8 bit with PIX_FMT_RGB32 palette
+ PIX_FMT_YUVJ420P, ///< planar YUV 4:2:0, 12bpp, full scale (JPEG), deprecated in favor of PIX_FMT_YUV420P and setting color_range
+ PIX_FMT_YUVJ422P, ///< planar YUV 4:2:2, 16bpp, full scale (JPEG), deprecated in favor of PIX_FMT_YUV422P and setting color_range
+ PIX_FMT_YUVJ444P, ///< planar YUV 4:4:4, 24bpp, full scale (JPEG), deprecated in favor of PIX_FMT_YUV444P and setting color_range
+ PIX_FMT_XVMC_MPEG2_MC,///< XVideo Motion Acceleration via common packet passing
+ PIX_FMT_XVMC_MPEG2_IDCT,
+ PIX_FMT_UYVY422, ///< packed YUV 4:2:2, 16bpp, Cb Y0 Cr Y1
+ PIX_FMT_UYYVYY411, ///< packed YUV 4:1:1, 12bpp, Cb Y0 Y1 Cr Y2 Y3
+ PIX_FMT_BGR8, ///< packed RGB 3:3:2, 8bpp, (msb)2B 3G 3R(lsb)
+ PIX_FMT_BGR4, ///< packed RGB 1:2:1 bitstream, 4bpp, (msb)1B 2G 1R(lsb), a byte contains two pixels, the first pixel in the byte is the one composed by the 4 msb bits
+ PIX_FMT_BGR4_BYTE, ///< packed RGB 1:2:1, 8bpp, (msb)1B 2G 1R(lsb)
+ PIX_FMT_RGB8, ///< packed RGB 3:3:2, 8bpp, (msb)2R 3G 3B(lsb)
+ PIX_FMT_RGB4, ///< packed RGB 1:2:1 bitstream, 4bpp, (msb)1R 2G 1B(lsb), a byte contains two pixels, the first pixel in the byte is the one composed by the 4 msb bits
+ PIX_FMT_RGB4_BYTE, ///< packed RGB 1:2:1, 8bpp, (msb)1R 2G 1B(lsb)
+ PIX_FMT_NV12, ///< planar YUV 4:2:0, 12bpp, 1 plane for Y and 1 plane for the UV components, which are interleaved (first byte U and the following byte V)
+ PIX_FMT_NV21, ///< as above, but U and V bytes are swapped
+
+ PIX_FMT_ARGB, ///< packed ARGB 8:8:8:8, 32bpp, ARGBARGB...
+ PIX_FMT_RGBA, ///< packed RGBA 8:8:8:8, 32bpp, RGBARGBA...
+ PIX_FMT_ABGR, ///< packed ABGR 8:8:8:8, 32bpp, ABGRABGR...
+ PIX_FMT_BGRA, ///< packed BGRA 8:8:8:8, 32bpp, BGRABGRA...
+
+ PIX_FMT_GRAY16BE, ///< Y , 16bpp, big-endian
+ PIX_FMT_GRAY16LE, ///< Y , 16bpp, little-endian
+ PIX_FMT_YUV440P, ///< planar YUV 4:4:0 (1 Cr & Cb sample per 1x2 Y samples)
+ PIX_FMT_YUVJ440P, ///< planar YUV 4:4:0 full scale (JPEG), deprecated in favor of PIX_FMT_YUV440P and setting color_range
+ PIX_FMT_YUVA420P, ///< planar YUV 4:2:0, 20bpp, (1 Cr & Cb sample per 2x2 Y & A samples)
+ PIX_FMT_VDPAU_H264,///< H.264 HW decoding with VDPAU, data[0] contains a vdpau_render_state struct which contains the bitstream of the slices as well as various fields extracted from headers
+ PIX_FMT_VDPAU_MPEG1,///< MPEG-1 HW decoding with VDPAU, data[0] contains a vdpau_render_state struct which contains the bitstream of the slices as well as various fields extracted from headers
+ PIX_FMT_VDPAU_MPEG2,///< MPEG-2 HW decoding with VDPAU, data[0] contains a vdpau_render_state struct which contains the bitstream of the slices as well as various fields extracted from headers
+ PIX_FMT_VDPAU_WMV3,///< WMV3 HW decoding with VDPAU, data[0] contains a vdpau_render_state struct which contains the bitstream of the slices as well as various fields extracted from headers
+ PIX_FMT_VDPAU_VC1, ///< VC-1 HW decoding with VDPAU, data[0] contains a vdpau_render_state struct which contains the bitstream of the slices as well as various fields extracted from headers
+ PIX_FMT_RGB48BE, ///< packed RGB 16:16:16, 48bpp, 16R, 16G, 16B, the 2-byte value for each R/G/B component is stored as big-endian
+ PIX_FMT_RGB48LE, ///< packed RGB 16:16:16, 48bpp, 16R, 16G, 16B, the 2-byte value for each R/G/B component is stored as little-endian
+
+ PIX_FMT_RGB565BE, ///< packed RGB 5:6:5, 16bpp, (msb) 5R 6G 5B(lsb), big-endian
+ PIX_FMT_RGB565LE, ///< packed RGB 5:6:5, 16bpp, (msb) 5R 6G 5B(lsb), little-endian
+ PIX_FMT_RGB555BE, ///< packed RGB 5:5:5, 16bpp, (msb)1A 5R 5G 5B(lsb), big-endian, most significant bit to 0
+ PIX_FMT_RGB555LE, ///< packed RGB 5:5:5, 16bpp, (msb)1A 5R 5G 5B(lsb), little-endian, most significant bit to 0
+
+ PIX_FMT_BGR565BE, ///< packed BGR 5:6:5, 16bpp, (msb) 5B 6G 5R(lsb), big-endian
+ PIX_FMT_BGR565LE, ///< packed BGR 5:6:5, 16bpp, (msb) 5B 6G 5R(lsb), little-endian
+ PIX_FMT_BGR555BE, ///< packed BGR 5:5:5, 16bpp, (msb)1A 5B 5G 5R(lsb), big-endian, most significant bit to 1
+ PIX_FMT_BGR555LE, ///< packed BGR 5:5:5, 16bpp, (msb)1A 5B 5G 5R(lsb), little-endian, most significant bit to 1
+
+ PIX_FMT_VAAPI_MOCO, ///< HW acceleration through VA API at motion compensation entry-point, Picture.data[3] contains a vaapi_render_state struct which contains macroblocks as well as various fields extracted from headers
+ PIX_FMT_VAAPI_IDCT, ///< HW acceleration through VA API at IDCT entry-point, Picture.data[3] contains a vaapi_render_state struct which contains fields extracted from headers
+ PIX_FMT_VAAPI_VLD, ///< HW decoding through VA API, Picture.data[3] contains a vaapi_render_state struct which contains the bitstream of the slices as well as various fields extracted from headers
+
+ PIX_FMT_YUV420P16LE, ///< planar YUV 4:2:0, 24bpp, (1 Cr & Cb sample per 2x2 Y samples), little-endian
+ PIX_FMT_YUV420P16BE, ///< planar YUV 4:2:0, 24bpp, (1 Cr & Cb sample per 2x2 Y samples), big-endian
+ PIX_FMT_YUV422P16LE, ///< planar YUV 4:2:2, 32bpp, (1 Cr & Cb sample per 2x1 Y samples), little-endian
+ PIX_FMT_YUV422P16BE, ///< planar YUV 4:2:2, 32bpp, (1 Cr & Cb sample per 2x1 Y samples), big-endian
+ PIX_FMT_YUV444P16LE, ///< planar YUV 4:4:4, 48bpp, (1 Cr & Cb sample per 1x1 Y samples), little-endian
+ PIX_FMT_YUV444P16BE, ///< planar YUV 4:4:4, 48bpp, (1 Cr & Cb sample per 1x1 Y samples), big-endian
+ PIX_FMT_VDPAU_MPEG4, ///< MPEG4 HW decoding with VDPAU, data[0] contains a vdpau_render_state struct which contains the bitstream of the slices as well as various fields extracted from headers
+ PIX_FMT_DXVA2_VLD, ///< HW decoding through DXVA2, Picture.data[3] contains a LPDIRECT3DSURFACE9 pointer
+
+ PIX_FMT_RGB444BE, ///< packed RGB 4:4:4, 16bpp, (msb)4A 4R 4G 4B(lsb), big-endian, most significant bits to 0
+ PIX_FMT_RGB444LE, ///< packed RGB 4:4:4, 16bpp, (msb)4A 4R 4G 4B(lsb), little-endian, most significant bits to 0
+ PIX_FMT_BGR444BE, ///< packed BGR 4:4:4, 16bpp, (msb)4A 4B 4G 4R(lsb), big-endian, most significant bits to 1
+ PIX_FMT_BGR444LE, ///< packed BGR 4:4:4, 16bpp, (msb)4A 4B 4G 4R(lsb), little-endian, most significant bits to 1
+ PIX_FMT_Y400A, ///< 8bit gray, 8bit alpha
+ PIX_FMT_NB, ///< number of pixel formats, DO NOT USE THIS if you want to link with shared libav* because the number of formats might differ between versions
+};
+
+#if AV_HAVE_BIGENDIAN
+# define PIX_FMT_NE(be, le) PIX_FMT_##be
+#else
+# define PIX_FMT_NE(be, le) PIX_FMT_##le
+#endif
+
+#define PIX_FMT_RGB32 PIX_FMT_NE(ARGB, BGRA)
+#define PIX_FMT_RGB32_1 PIX_FMT_NE(RGBA, ABGR)
+#define PIX_FMT_BGR32 PIX_FMT_NE(ABGR, RGBA)
+#define PIX_FMT_BGR32_1 PIX_FMT_NE(BGRA, ARGB)
+
+#define PIX_FMT_GRAY16 PIX_FMT_NE(GRAY16BE, GRAY16LE)
+#define PIX_FMT_RGB48 PIX_FMT_NE(RGB48BE, RGB48LE)
+#define PIX_FMT_RGB565 PIX_FMT_NE(RGB565BE, RGB565LE)
+#define PIX_FMT_RGB555 PIX_FMT_NE(RGB555BE, RGB555LE)
+#define PIX_FMT_RGB444 PIX_FMT_NE(RGB444BE, RGB444LE)
+#define PIX_FMT_BGR565 PIX_FMT_NE(BGR565BE, BGR565LE)
+#define PIX_FMT_BGR555 PIX_FMT_NE(BGR555BE, BGR555LE)
+#define PIX_FMT_BGR444 PIX_FMT_NE(BGR444BE, BGR444LE)
+
+#define PIX_FMT_YUV420P16 PIX_FMT_NE(YUV420P16BE, YUV420P16LE)
+#define PIX_FMT_YUV422P16 PIX_FMT_NE(YUV422P16BE, YUV422P16LE)
+#define PIX_FMT_YUV444P16 PIX_FMT_NE(YUV444P16BE, YUV444P16LE)
+
+#endif /* AVUTIL_PIXFMT_H */
diff --git a/thirdparties/android/include/libavutil/random_seed.h b/thirdparties/android/include/libavutil/random_seed.h new file mode 100644 index 0000000..7f75063 --- /dev/null +++ b/thirdparties/android/include/libavutil/random_seed.h @@ -0,0 +1,31 @@ +/* + * Copyright (c) 2009 Baptiste Coudurier <baptiste.coudurier@gmail.com> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_RANDOM_SEED_H +#define AVUTIL_RANDOM_SEED_H + +#include <stdint.h> + +/** + * Get a seed to use in conjunction with random functions. + */ +uint32_t av_get_random_seed(void); + +#endif /* AVUTIL_RANDOM_SEED_H */ diff --git a/thirdparties/android/include/libavutil/rational.h b/thirdparties/android/include/libavutil/rational.h new file mode 100644 index 0000000..edbc674 --- /dev/null +++ b/thirdparties/android/include/libavutil/rational.h @@ -0,0 +1,129 @@ +/*
+ * rational numbers
+ * Copyright (c) 2003 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * rational numbers
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#ifndef AVUTIL_RATIONAL_H
+#define AVUTIL_RATIONAL_H
+
+#include <stdint.h>
+#include "attributes.h"
+
+/**
+ * rational number numerator/denominator
+ */
+typedef struct AVRational{
+ int num; ///< numerator
+ int den; ///< denominator
+} AVRational;
+
+/**
+ * Compare two rationals.
+ * @param a first rational
+ * @param b second rational
+ * @return 0 if a==b, 1 if a>b and -1 if a<b
+ */
+static inline int av_cmp_q(AVRational a, AVRational b){
+ const int64_t tmp= a.num * (int64_t)b.den - b.num * (int64_t)a.den;
+
+ if(tmp) return (tmp>>63)|1;
+ else return 0;
+}
+
+/**
+ * Convert rational to double.
+ * @param a rational to convert
+ * @return (double) a
+ */
+static inline double av_q2d(AVRational a){
+ return a.num / (double) a.den;
+}
+
+/**
+ * Reduce a fraction.
+ * This is useful for framerate calculations.
+ * @param dst_num destination numerator
+ * @param dst_den destination denominator
+ * @param num source numerator
+ * @param den source denominator
+ * @param max the maximum allowed for dst_num & dst_den
+ * @return 1 if exact, 0 otherwise
+ */
+int av_reduce(int *dst_num, int *dst_den, int64_t num, int64_t den, int64_t max);
+
+/**
+ * Multiply two rationals.
+ * @param b first rational
+ * @param c second rational
+ * @return b*c
+ */
+AVRational av_mul_q(AVRational b, AVRational c) av_const;
+
+/**
+ * Divide one rational by another.
+ * @param b first rational
+ * @param c second rational
+ * @return b/c
+ */
+AVRational av_div_q(AVRational b, AVRational c) av_const;
+
+/**
+ * Add two rationals.
+ * @param b first rational
+ * @param c second rational
+ * @return b+c
+ */
+AVRational av_add_q(AVRational b, AVRational c) av_const;
+
+/**
+ * Subtract one rational from another.
+ * @param b first rational
+ * @param c second rational
+ * @return b-c
+ */
+AVRational av_sub_q(AVRational b, AVRational c) av_const;
+
+/**
+ * Convert a double precision floating point number to a rational.
+ * @param d double to convert
+ * @param max the maximum allowed numerator and denominator
+ * @return (AVRational) d
+ */
+AVRational av_d2q(double d, int max) av_const;
+
+/**
+ * @return 1 if q1 is nearer to q than q2, -1 if q2 is nearer
+ * than q1, 0 if they have the same distance.
+ */
+int av_nearer_q(AVRational q, AVRational q1, AVRational q2);
+
+/**
+ * Find the nearest value in q_list to q.
+ * @param q_list an array of rationals terminated by {0, 0}
+ * @return the index of the nearest value found in the array
+ */
+int av_find_nearest_q_idx(AVRational q, const AVRational* q_list);
+
+#endif /* AVUTIL_RATIONAL_H */
diff --git a/thirdparties/android/include/libavutil/sha1.h b/thirdparties/android/include/libavutil/sha1.h new file mode 100644 index 0000000..3ff5804 --- /dev/null +++ b/thirdparties/android/include/libavutil/sha1.h @@ -0,0 +1,57 @@ +/* + * Copyright (C) 2007 Michael Niedermayer <michaelni@gmx.at> + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVUTIL_SHA1_H +#define AVUTIL_SHA1_H + +#include <stdint.h> + +extern const int av_sha1_size; + +struct AVSHA1; + +/** + * Initialize SHA-1 hashing. + * + * @param context pointer to the function context (of size av_sha_size) + * @deprecated use av_sha_init() instead + */ +void av_sha1_init(struct AVSHA1* context); + +/** + * Update hash value. + * + * @param context hash function context + * @param data input data to update hash with + * @param len input data length + * @deprecated use av_sha_update() instead + */ +void av_sha1_update(struct AVSHA1* context, const uint8_t* data, unsigned int len); + +/** + * Finish hashing and output digest value. + * + * @param context hash function context + * @param digest buffer where output digest value is stored + * @deprecated use av_sha_final() instead + */ +void av_sha1_final(struct AVSHA1* context, uint8_t digest[20]); + +#endif /* AVUTIL_SHA1_H */ diff --git a/thirdparties/android/include/libswscale/swscale.h b/thirdparties/android/include/libswscale/swscale.h new file mode 100644 index 0000000..1f6c5c9 --- /dev/null +++ b/thirdparties/android/include/libswscale/swscale.h @@ -0,0 +1,331 @@ +/*
+ * Copyright (C) 2001-2003 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef SWSCALE_SWSCALE_H
+#define SWSCALE_SWSCALE_H
+
+/**
+ * @file
+ * @brief
+ * external api for the swscale stuff
+ */
+
+#include "libavutil/avutil.h"
+
+#define LIBSWSCALE_VERSION_MAJOR 0
+#define LIBSWSCALE_VERSION_MINOR 11
+#define LIBSWSCALE_VERSION_MICRO 0
+
+#define LIBSWSCALE_VERSION_INT AV_VERSION_INT(LIBSWSCALE_VERSION_MAJOR, \
+ LIBSWSCALE_VERSION_MINOR, \
+ LIBSWSCALE_VERSION_MICRO)
+#define LIBSWSCALE_VERSION AV_VERSION(LIBSWSCALE_VERSION_MAJOR, \
+ LIBSWSCALE_VERSION_MINOR, \
+ LIBSWSCALE_VERSION_MICRO)
+#define LIBSWSCALE_BUILD LIBSWSCALE_VERSION_INT
+
+#define LIBSWSCALE_IDENT "SwS" AV_STRINGIFY(LIBSWSCALE_VERSION)
+
+/**
+ * Returns the LIBSWSCALE_VERSION_INT constant.
+ */
+unsigned swscale_version(void);
+
+/**
+ * Returns the libswscale build-time configuration.
+ */
+const char *swscale_configuration(void);
+
+/**
+ * Returns the libswscale license.
+ */
+const char *swscale_license(void);
+
+/* values for the flags, the stuff on the command line is different */
+#define SWS_FAST_BILINEAR 1
+#define SWS_BILINEAR 2
+#define SWS_BICUBIC 4
+#define SWS_X 8
+#define SWS_POINT 0x10
+#define SWS_AREA 0x20
+#define SWS_BICUBLIN 0x40
+#define SWS_GAUSS 0x80
+#define SWS_SINC 0x100
+#define SWS_LANCZOS 0x200
+#define SWS_SPLINE 0x400
+
+#define SWS_SRC_V_CHR_DROP_MASK 0x30000
+#define SWS_SRC_V_CHR_DROP_SHIFT 16
+
+#define SWS_PARAM_DEFAULT 123456
+
+#define SWS_PRINT_INFO 0x1000
+
+//the following 3 flags are not completely implemented
+//internal chrominace subsampling info
+#define SWS_FULL_CHR_H_INT 0x2000
+//input subsampling info
+#define SWS_FULL_CHR_H_INP 0x4000
+#define SWS_DIRECT_BGR 0x8000
+#define SWS_ACCURATE_RND 0x40000
+#define SWS_BITEXACT 0x80000
+
+#define SWS_CPU_CAPS_MMX 0x80000000
+#define SWS_CPU_CAPS_MMX2 0x20000000
+#define SWS_CPU_CAPS_3DNOW 0x40000000
+#define SWS_CPU_CAPS_ALTIVEC 0x10000000
+#define SWS_CPU_CAPS_BFIN 0x01000000
+
+#define SWS_MAX_REDUCE_CUTOFF 0.002
+
+#define SWS_CS_ITU709 1
+#define SWS_CS_FCC 4
+#define SWS_CS_ITU601 5
+#define SWS_CS_ITU624 5
+#define SWS_CS_SMPTE170M 5
+#define SWS_CS_SMPTE240M 7
+#define SWS_CS_DEFAULT 5
+
+/**
+ * Returns a pointer to yuv<->rgb coefficients for the given colorspace
+ * suitable for sws_setColorspaceDetails().
+ *
+ * @param colorspace One of the SWS_CS_* macros. If invalid,
+ * SWS_CS_DEFAULT is used.
+ */
+const int *sws_getCoefficients(int colorspace);
+
+
+// when used for filters they must have an odd number of elements
+// coeffs cannot be shared between vectors
+typedef struct {
+ double *coeff; ///< pointer to the list of coefficients
+ int length; ///< number of coefficients in the vector
+} SwsVector;
+
+// vectors can be shared
+typedef struct {
+ SwsVector *lumH;
+ SwsVector *lumV;
+ SwsVector *chrH;
+ SwsVector *chrV;
+} SwsFilter;
+
+struct SwsContext;
+
+/**
+ * Returns a positive value if pix_fmt is a supported input format, 0
+ * otherwise.
+ */
+int sws_isSupportedInput(enum PixelFormat pix_fmt);
+
+/**
+ * Returns a positive value if pix_fmt is a supported output format, 0
+ * otherwise.
+ */
+int sws_isSupportedOutput(enum PixelFormat pix_fmt);
+
+/**
+ * Frees the swscaler context swsContext.
+ * If swsContext is NULL, then does nothing.
+ */
+void sws_freeContext(struct SwsContext *swsContext);
+
+/**
+ * Allocates and returns a SwsContext. You need it to perform
+ * scaling/conversion operations using sws_scale().
+ *
+ * @param srcW the width of the source image
+ * @param srcH the height of the source image
+ * @param srcFormat the source image format
+ * @param dstW the width of the destination image
+ * @param dstH the height of the destination image
+ * @param dstFormat the destination image format
+ * @param flags specify which algorithm and options to use for rescaling
+ * @return a pointer to an allocated context, or NULL in case of error
+ */
+struct SwsContext *sws_getContext(int srcW, int srcH, enum PixelFormat srcFormat,
+ int dstW, int dstH, enum PixelFormat dstFormat,
+ int flags, SwsFilter *srcFilter,
+ SwsFilter *dstFilter, const double *param);
+
+/**
+ * Scales the image slice in srcSlice and puts the resulting scaled
+ * slice in the image in dst. A slice is a sequence of consecutive
+ * rows in an image.
+ *
+ * Slices have to be provided in sequential order, either in
+ * top-bottom or bottom-top order. If slices are provided in
+ * non-sequential order the behavior of the function is undefined.
+ *
+ * @param context the scaling context previously created with
+ * sws_getContext()
+ * @param srcSlice the array containing the pointers to the planes of
+ * the source slice
+ * @param srcStride the array containing the strides for each plane of
+ * the source image
+ * @param srcSliceY the position in the source image of the slice to
+ * process, that is the number (counted starting from
+ * zero) in the image of the first row of the slice
+ * @param srcSliceH the height of the source slice, that is the number
+ * of rows in the slice
+ * @param dst the array containing the pointers to the planes of
+ * the destination image
+ * @param dstStride the array containing the strides for each plane of
+ * the destination image
+ * @return the height of the output slice
+ */
+int sws_scale(struct SwsContext *context, const uint8_t* const srcSlice[], const int srcStride[],
+ int srcSliceY, int srcSliceH, uint8_t* const dst[], const int dstStride[]);
+
+#if LIBSWSCALE_VERSION_MAJOR < 1
+/**
+ * @deprecated Use sws_scale() instead.
+ */
+int sws_scale_ordered(struct SwsContext *context, const uint8_t* const src[],
+ int srcStride[], int srcSliceY, int srcSliceH,
+ uint8_t* dst[], int dstStride[]) attribute_deprecated;
+#endif
+
+/**
+ * @param inv_table the yuv2rgb coefficients, normally ff_yuv2rgb_coeffs[x]
+ * @param fullRange if 1 then the luma range is 0..255 if 0 it is 16..235
+ * @return -1 if not supported
+ */
+int sws_setColorspaceDetails(struct SwsContext *c, const int inv_table[4],
+ int srcRange, const int table[4], int dstRange,
+ int brightness, int contrast, int saturation);
+
+/**
+ * @return -1 if not supported
+ */
+int sws_getColorspaceDetails(struct SwsContext *c, int **inv_table,
+ int *srcRange, int **table, int *dstRange,
+ int *brightness, int *contrast, int *saturation);
+
+/**
+ * Allocates and returns an uninitialized vector with length coefficients.
+ */
+SwsVector *sws_allocVec(int length);
+
+/**
+ * Returns a normalized Gaussian curve used to filter stuff
+ * quality=3 is high quality, lower is lower quality.
+ */
+SwsVector *sws_getGaussianVec(double variance, double quality);
+
+/**
+ * Allocates and returns a vector with length coefficients, all
+ * with the same value c.
+ */
+SwsVector *sws_getConstVec(double c, int length);
+
+/**
+ * Allocates and returns a vector with just one coefficient, with
+ * value 1.0.
+ */
+SwsVector *sws_getIdentityVec(void);
+
+/**
+ * Scales all the coefficients of a by the scalar value.
+ */
+void sws_scaleVec(SwsVector *a, double scalar);
+
+/**
+ * Scales all the coefficients of a so that their sum equals height.
+ */
+void sws_normalizeVec(SwsVector *a, double height);
+void sws_convVec(SwsVector *a, SwsVector *b);
+void sws_addVec(SwsVector *a, SwsVector *b);
+void sws_subVec(SwsVector *a, SwsVector *b);
+void sws_shiftVec(SwsVector *a, int shift);
+
+/**
+ * Allocates and returns a clone of the vector a, that is a vector
+ * with the same coefficients as a.
+ */
+SwsVector *sws_cloneVec(SwsVector *a);
+
+#if LIBSWSCALE_VERSION_MAJOR < 1
+/**
+ * @deprecated Use sws_printVec2() instead.
+ */
+attribute_deprecated void sws_printVec(SwsVector *a);
+#endif
+
+/**
+ * Prints with av_log() a textual representation of the vector a
+ * if log_level <= av_log_level.
+ */
+void sws_printVec2(SwsVector *a, AVClass *log_ctx, int log_level);
+
+void sws_freeVec(SwsVector *a);
+
+SwsFilter *sws_getDefaultFilter(float lumaGBlur, float chromaGBlur,
+ float lumaSharpen, float chromaSharpen,
+ float chromaHShift, float chromaVShift,
+ int verbose);
+void sws_freeFilter(SwsFilter *filter);
+
+/**
+ * Checks if context can be reused, otherwise reallocates a new
+ * one.
+ *
+ * If context is NULL, just calls sws_getContext() to get a new
+ * context. Otherwise, checks if the parameters are the ones already
+ * saved in context. If that is the case, returns the current
+ * context. Otherwise, frees context and gets a new context with
+ * the new parameters.
+ *
+ * Be warned that srcFilter and dstFilter are not checked, they
+ * are assumed to remain the same.
+ */
+struct SwsContext *sws_getCachedContext(struct SwsContext *context,
+ int srcW, int srcH, enum PixelFormat srcFormat,
+ int dstW, int dstH, enum PixelFormat dstFormat,
+ int flags, SwsFilter *srcFilter,
+ SwsFilter *dstFilter, const double *param);
+
+/**
+ * Converts an 8bit paletted frame into a frame with a color depth of 32-bits.
+ *
+ * The output frame will have the same packed format as the palette.
+ *
+ * @param src source frame buffer
+ * @param dst destination frame buffer
+ * @param num_pixels number of pixels to convert
+ * @param palette array with [256] entries, which must match color arrangement (RGB or BGR) of src
+ */
+void sws_convertPalette8ToPacked32(const uint8_t *src, uint8_t *dst, long num_pixels, const uint8_t *palette);
+
+/**
+ * Converts an 8bit paletted frame into a frame with a color depth of 24 bits.
+ *
+ * With the palette format "ABCD", the destination frame ends up with the format "ABC".
+ *
+ * @param src source frame buffer
+ * @param dst destination frame buffer
+ * @param num_pixels number of pixels to convert
+ * @param palette array with [256] entries, which must match color arrangement (RGB or BGR) of src
+ */
+void sws_convertPalette8ToPacked24(const uint8_t *src, uint8_t *dst, long num_pixels, const uint8_t *palette);
+
+
+#endif /* SWSCALE_SWSCALE_H */
diff --git a/thirdparties/android/include/ogg/config_types.h b/thirdparties/android/include/ogg/config_types.h new file mode 100644 index 0000000..a320fc8 --- /dev/null +++ b/thirdparties/android/include/ogg/config_types.h @@ -0,0 +1,11 @@ +#ifndef __CONFIG_TYPES_H__ +#define __CONFIG_TYPES_H__ + +/* these are filled in by configure */ +typedef short ogg_int16_t; +typedef unsigned short ogg_uint16_t; +typedef int ogg_int32_t; +typedef unsigned int ogg_uint32_t; +typedef long long ogg_int64_t; + +#endif diff --git a/thirdparties/android/include/ogg/ogg.h b/thirdparties/android/include/ogg/ogg.h new file mode 100644 index 0000000..ae0cfd5 --- /dev/null +++ b/thirdparties/android/include/ogg/ogg.h @@ -0,0 +1,208 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: toplevel libogg include + last mod: $Id: ogg.h 16051 2009-05-27 05:00:06Z xiphmont $ + + ********************************************************************/ +#ifndef _OGG_H +#define _OGG_H + +#ifdef __cplusplus +extern "C" { +#endif + +#include <stddef.h> +#include <ogg/os_types.h> + +typedef struct { + void *iov_base; + size_t iov_len; +} ogg_iovec_t; + +typedef struct { + long endbyte; + int endbit; + + unsigned char *buffer; + unsigned char *ptr; + long storage; +} oggpack_buffer; + +/* ogg_page is used to encapsulate the data in one Ogg bitstream page *****/ + +typedef struct { + unsigned char *header; + long header_len; + unsigned char *body; + long body_len; +} ogg_page; + +/* ogg_stream_state contains the current encode/decode state of a logical + Ogg bitstream **********************************************************/ + +typedef struct { + unsigned char *body_data; /* bytes from packet bodies */ + long body_storage; /* storage elements allocated */ + long body_fill; /* elements stored; fill mark */ + long body_returned; /* elements of fill returned */ + + + int *lacing_vals; /* The values that will go to the segment table */ + ogg_int64_t *granule_vals; /* granulepos values for headers. Not compact + this way, but it is simple coupled to the + lacing fifo */ + long lacing_storage; + long lacing_fill; + long lacing_packet; + long lacing_returned; + + unsigned char header[282]; /* working space for header encode */ + int header_fill; + + int e_o_s; /* set when we have buffered the last packet in the + logical bitstream */ + int b_o_s; /* set after we've written the initial page + of a logical bitstream */ + long serialno; + long pageno; + ogg_int64_t packetno; /* sequence number for decode; the framing + knows where there's a hole in the data, + but we need coupling so that the codec + (which is in a seperate abstraction + layer) also knows about the gap */ + ogg_int64_t granulepos; + +} ogg_stream_state; + +/* ogg_packet is used to encapsulate the data and metadata belonging + to a single raw Ogg/Vorbis packet *************************************/ + +typedef struct { + unsigned char *packet; + long bytes; + long b_o_s; + long e_o_s; + + ogg_int64_t granulepos; + + ogg_int64_t packetno; /* sequence number for decode; the framing + knows where there's a hole in the data, + but we need coupling so that the codec + (which is in a seperate abstraction + layer) also knows about the gap */ +} ogg_packet; + +typedef struct { + unsigned char *data; + int storage; + int fill; + int returned; + + int unsynced; + int headerbytes; + int bodybytes; +} ogg_sync_state; + +/* Ogg BITSTREAM PRIMITIVES: bitstream ************************/ + +extern void oggpack_writeinit(oggpack_buffer *b); +extern int oggpack_writecheck(oggpack_buffer *b); +extern void oggpack_writetrunc(oggpack_buffer *b,long bits); +extern void oggpack_writealign(oggpack_buffer *b); +extern void oggpack_writecopy(oggpack_buffer *b,void *source,long bits); +extern void oggpack_reset(oggpack_buffer *b); +extern void oggpack_writeclear(oggpack_buffer *b); +extern void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes); +extern void oggpack_write(oggpack_buffer *b,unsigned long value,int bits); +extern long oggpack_look(oggpack_buffer *b,int bits); +extern long oggpack_look1(oggpack_buffer *b); +extern void oggpack_adv(oggpack_buffer *b,int bits); +extern void oggpack_adv1(oggpack_buffer *b); +extern long oggpack_read(oggpack_buffer *b,int bits); +extern long oggpack_read1(oggpack_buffer *b); +extern long oggpack_bytes(oggpack_buffer *b); +extern long oggpack_bits(oggpack_buffer *b); +extern unsigned char *oggpack_get_buffer(oggpack_buffer *b); + +extern void oggpackB_writeinit(oggpack_buffer *b); +extern int oggpackB_writecheck(oggpack_buffer *b); +extern void oggpackB_writetrunc(oggpack_buffer *b,long bits); +extern void oggpackB_writealign(oggpack_buffer *b); +extern void oggpackB_writecopy(oggpack_buffer *b,void *source,long bits); +extern void oggpackB_reset(oggpack_buffer *b); +extern void oggpackB_writeclear(oggpack_buffer *b); +extern void oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes); +extern void oggpackB_write(oggpack_buffer *b,unsigned long value,int bits); +extern long oggpackB_look(oggpack_buffer *b,int bits); +extern long oggpackB_look1(oggpack_buffer *b); +extern void oggpackB_adv(oggpack_buffer *b,int bits); +extern void oggpackB_adv1(oggpack_buffer *b); +extern long oggpackB_read(oggpack_buffer *b,int bits); +extern long oggpackB_read1(oggpack_buffer *b); +extern long oggpackB_bytes(oggpack_buffer *b); +extern long oggpackB_bits(oggpack_buffer *b); +extern unsigned char *oggpackB_get_buffer(oggpack_buffer *b); + +/* Ogg BITSTREAM PRIMITIVES: encoding **************************/ + +extern int ogg_stream_packetin(ogg_stream_state *os, ogg_packet *op); +extern int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov, + int count, long e_o_s, ogg_int64_t granulepos); +extern int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og); +extern int ogg_stream_flush(ogg_stream_state *os, ogg_page *og); + +/* Ogg BITSTREAM PRIMITIVES: decoding **************************/ + +extern int ogg_sync_init(ogg_sync_state *oy); +extern int ogg_sync_clear(ogg_sync_state *oy); +extern int ogg_sync_reset(ogg_sync_state *oy); +extern int ogg_sync_destroy(ogg_sync_state *oy); +extern int ogg_sync_check(ogg_sync_state *oy); + +extern char *ogg_sync_buffer(ogg_sync_state *oy, long size); +extern int ogg_sync_wrote(ogg_sync_state *oy, long bytes); +extern long ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og); +extern int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og); +extern int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og); +extern int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op); +extern int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op); + +/* Ogg BITSTREAM PRIMITIVES: general ***************************/ + +extern int ogg_stream_init(ogg_stream_state *os,int serialno); +extern int ogg_stream_clear(ogg_stream_state *os); +extern int ogg_stream_reset(ogg_stream_state *os); +extern int ogg_stream_reset_serialno(ogg_stream_state *os,int serialno); +extern int ogg_stream_destroy(ogg_stream_state *os); +extern int ogg_stream_check(ogg_stream_state *os); +extern int ogg_stream_eos(ogg_stream_state *os); + +extern void ogg_page_checksum_set(ogg_page *og); + +extern int ogg_page_version(const ogg_page *og); +extern int ogg_page_continued(const ogg_page *og); +extern int ogg_page_bos(const ogg_page *og); +extern int ogg_page_eos(const ogg_page *og); +extern ogg_int64_t ogg_page_granulepos(const ogg_page *og); +extern int ogg_page_serialno(const ogg_page *og); +extern long ogg_page_pageno(const ogg_page *og); +extern int ogg_page_packets(const ogg_page *og); + +extern void ogg_packet_clear(ogg_packet *op); + + +#ifdef __cplusplus +} +#endif + +#endif /* _OGG_H */ diff --git a/thirdparties/android/include/ogg/os_types.h b/thirdparties/android/include/ogg/os_types.h new file mode 100644 index 0000000..a42f800 --- /dev/null +++ b/thirdparties/android/include/ogg/os_types.h @@ -0,0 +1,139 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: #ifdef jail to whip a few platforms into the UNIX ideal. + last mod: $Id: os_types.h 14997 2008-06-04 03:27:18Z ivo $ + + ********************************************************************/ +#ifndef _OS_TYPES_H +#define _OS_TYPES_H + +/* make it easy on the folks that want to compile the libs with a + different malloc than stdlib */ +#define _ogg_malloc malloc +#define _ogg_calloc calloc +#define _ogg_realloc realloc +#define _ogg_free free + +#if defined(_WIN32) + +# if defined(__CYGWIN__) +# include <stdint.h> + typedef int16_t ogg_int16_t; + typedef uint16_t ogg_uint16_t; + typedef int32_t ogg_int32_t; + typedef uint32_t ogg_uint32_t; + typedef int64_t ogg_int64_t; + typedef uint64_t ogg_uint64_t; +# elif defined(__MINGW32__) +# include <sys/types.h> + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + typedef unsigned long long ogg_uint64_t; +# elif defined(__MWERKS__) + typedef long long ogg_int64_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; +# else + /* MSVC/Borland */ + typedef __int64 ogg_int64_t; + typedef __int32 ogg_int32_t; + typedef unsigned __int32 ogg_uint32_t; + typedef __int16 ogg_int16_t; + typedef unsigned __int16 ogg_uint16_t; +# endif + +#elif defined(__MACOS__) + +# include <sys/types.h> + typedef SInt16 ogg_int16_t; + typedef UInt16 ogg_uint16_t; + typedef SInt32 ogg_int32_t; + typedef UInt32 ogg_uint32_t; + typedef SInt64 ogg_int64_t; + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include <sys/types.h> + typedef int16_t ogg_int16_t; + typedef u_int16_t ogg_uint16_t; + typedef int32_t ogg_int32_t; + typedef u_int32_t ogg_uint32_t; + typedef int64_t ogg_int64_t; + +#elif defined(__HAIKU__) + + /* Haiku */ +# include <sys/types.h> + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + +#elif defined(__BEOS__) + + /* Be */ +# include <inttypes.h> + typedef int16_t ogg_int16_t; + typedef u_int16_t ogg_uint16_t; + typedef int32_t ogg_int32_t; + typedef u_int32_t ogg_uint32_t; + typedef int64_t ogg_int64_t; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short ogg_int16_t; + typedef int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long ogg_int64_t; + +#elif defined(R5900) + + /* PS2 EE */ + typedef long ogg_int64_t; + typedef int ogg_int32_t; + typedef unsigned ogg_uint32_t; + typedef short ogg_int16_t; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short ogg_int16_t; + typedef unsigned short ogg_uint16_t; + typedef signed int ogg_int32_t; + typedef unsigned int ogg_uint32_t; + typedef long long int ogg_int64_t; + +#else + +# include <sys/types.h> +# include <ogg/config_types.h> + +#endif + +#endif /* _OS_TYPES_H */ diff --git a/thirdparties/android/include/opencore-amrnb/interf_dec.h b/thirdparties/android/include/opencore-amrnb/interf_dec.h new file mode 100644 index 0000000..98051f4 --- /dev/null +++ b/thirdparties/android/include/opencore-amrnb/interf_dec.h @@ -0,0 +1,34 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2009 Martin Storsjo + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#ifndef OPENCORE_AMRNB_INTERF_DEC_H +#define OPENCORE_AMRNB_INTERF_DEC_H + +#ifdef __cplusplus +extern "C" { +#endif + +void* Decoder_Interface_init(void); +void Decoder_Interface_exit(void* state); +void Decoder_Interface_Decode(void* state, const unsigned char* in, short* out, int bfi); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/thirdparties/android/include/opencore-amrnb/interf_enc.h b/thirdparties/android/include/opencore-amrnb/interf_enc.h new file mode 100644 index 0000000..b89b0c7 --- /dev/null +++ b/thirdparties/android/include/opencore-amrnb/interf_enc.h @@ -0,0 +1,50 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2009 Martin Storsjo + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#ifndef OPENCORE_AMRNB_INTERF_ENC_H +#define OPENCORE_AMRNB_INTERF_ENC_H + +#ifdef __cplusplus +extern "C" { +#endif + +#ifndef AMRNB_WRAPPER_INTERNAL +/* Copied from enc/src/gsmamr_enc.h */ +enum Mode { + MR475 = 0,/* 4.75 kbps */ + MR515, /* 5.15 kbps */ + MR59, /* 5.90 kbps */ + MR67, /* 6.70 kbps */ + MR74, /* 7.40 kbps */ + MR795, /* 7.95 kbps */ + MR102, /* 10.2 kbps */ + MR122, /* 12.2 kbps */ + MRDTX, /* DTX */ + N_MODES /* Not Used */ +}; +#endif + +void* Encoder_Interface_init(int dtx); +void Encoder_Interface_exit(void* state); +int Encoder_Interface_Encode(void* state, enum Mode mode, const short* speech, unsigned char* out, int forceSpeech); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/thirdparties/android/include/opencore-amrwb/dec_if.h b/thirdparties/android/include/opencore-amrwb/dec_if.h new file mode 100644 index 0000000..56acdaa --- /dev/null +++ b/thirdparties/android/include/opencore-amrwb/dec_if.h @@ -0,0 +1,36 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2009 Martin Storsjo + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#ifndef OPENCORE_AMRWB_DEC_IF_H +#define OPENCORE_AMRWB_DEC_IF_H + +#ifdef __cplusplus +extern "C" { +#endif + +#define _good_frame 0 + +void* D_IF_init(void); +void D_IF_decode(void* state, const unsigned char* bits, short* synth, int bfi); +void D_IF_exit(void* state); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/thirdparties/android/include/opencore-amrwb/if_rom.h b/thirdparties/android/include/opencore-amrwb/if_rom.h new file mode 100644 index 0000000..8977e03 --- /dev/null +++ b/thirdparties/android/include/opencore-amrwb/if_rom.h @@ -0,0 +1,33 @@ +/* ------------------------------------------------------------------ + * Copyright (C) 2009 Martin Storsjo + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either + * express or implied. + * See the License for the specific language governing permissions + * and limitations under the License. + * ------------------------------------------------------------------- + */ + +#ifndef OPENCORE_AMRWB_IF_ROM_H +#define OPENCORE_AMRWB_IF_ROM_H + +#ifdef __cplusplus +extern "C" { +#endif + +#include <stdint.h> +typedef int16_t Word16; + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/thirdparties/android/include/speex/speex.h b/thirdparties/android/include/speex/speex.h new file mode 100644 index 0000000..82ba016 --- /dev/null +++ b/thirdparties/android/include/speex/speex.h @@ -0,0 +1,424 @@ +/* Copyright (C) 2002-2006 Jean-Marc Valin*/ +/** + @file speex.h + @brief Describes the different modes of the codec +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +*/ + +#ifndef SPEEX_H +#define SPEEX_H +/** @defgroup Codec Speex encoder and decoder + * This is the Speex codec itself. + * @{ + */ + +#include "speex/speex_bits.h" +#include "speex/speex_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/* Values allowed for *ctl() requests */ + +/** Set enhancement on/off (decoder only) */ +#define SPEEX_SET_ENH 0 +/** Get enhancement state (decoder only) */ +#define SPEEX_GET_ENH 1 + +/*Would be SPEEX_SET_FRAME_SIZE, but it's (currently) invalid*/ +/** Obtain frame size used by encoder/decoder */ +#define SPEEX_GET_FRAME_SIZE 3 + +/** Set quality value */ +#define SPEEX_SET_QUALITY 4 +/** Get current quality setting */ +/* #define SPEEX_GET_QUALITY 5 -- Doesn't make much sense, does it? */ + +/** Set sub-mode to use */ +#define SPEEX_SET_MODE 6 +/** Get current sub-mode in use */ +#define SPEEX_GET_MODE 7 + +/** Set low-band sub-mode to use (wideband only)*/ +#define SPEEX_SET_LOW_MODE 8 +/** Get current low-band mode in use (wideband only)*/ +#define SPEEX_GET_LOW_MODE 9 + +/** Set high-band sub-mode to use (wideband only)*/ +#define SPEEX_SET_HIGH_MODE 10 +/** Get current high-band mode in use (wideband only)*/ +#define SPEEX_GET_HIGH_MODE 11 + +/** Set VBR on (1) or off (0) */ +#define SPEEX_SET_VBR 12 +/** Get VBR status (1 for on, 0 for off) */ +#define SPEEX_GET_VBR 13 + +/** Set quality value for VBR encoding (0-10) */ +#define SPEEX_SET_VBR_QUALITY 14 +/** Get current quality value for VBR encoding (0-10) */ +#define SPEEX_GET_VBR_QUALITY 15 + +/** Set complexity of the encoder (0-10) */ +#define SPEEX_SET_COMPLEXITY 16 +/** Get current complexity of the encoder (0-10) */ +#define SPEEX_GET_COMPLEXITY 17 + +/** Set bit-rate used by the encoder (or lower) */ +#define SPEEX_SET_BITRATE 18 +/** Get current bit-rate used by the encoder or decoder */ +#define SPEEX_GET_BITRATE 19 + +/** Define a handler function for in-band Speex request*/ +#define SPEEX_SET_HANDLER 20 + +/** Define a handler function for in-band user-defined request*/ +#define SPEEX_SET_USER_HANDLER 22 + +/** Set sampling rate used in bit-rate computation */ +#define SPEEX_SET_SAMPLING_RATE 24 +/** Get sampling rate used in bit-rate computation */ +#define SPEEX_GET_SAMPLING_RATE 25 + +/** Reset the encoder/decoder memories to zero*/ +#define SPEEX_RESET_STATE 26 + +/** Get VBR info (mostly used internally) */ +#define SPEEX_GET_RELATIVE_QUALITY 29 + +/** Set VAD status (1 for on, 0 for off) */ +#define SPEEX_SET_VAD 30 + +/** Get VAD status (1 for on, 0 for off) */ +#define SPEEX_GET_VAD 31 + +/** Set Average Bit-Rate (ABR) to n bits per seconds */ +#define SPEEX_SET_ABR 32 +/** Get Average Bit-Rate (ABR) setting (in bps) */ +#define SPEEX_GET_ABR 33 + +/** Set DTX status (1 for on, 0 for off) */ +#define SPEEX_SET_DTX 34 +/** Get DTX status (1 for on, 0 for off) */ +#define SPEEX_GET_DTX 35 + +/** Set submode encoding in each frame (1 for yes, 0 for no, setting to no breaks the standard) */ +#define SPEEX_SET_SUBMODE_ENCODING 36 +/** Get submode encoding in each frame */ +#define SPEEX_GET_SUBMODE_ENCODING 37 + +/*#define SPEEX_SET_LOOKAHEAD 38*/ +/** Returns the lookahead used by Speex */ +#define SPEEX_GET_LOOKAHEAD 39 + +/** Sets tuning for packet-loss concealment (expected loss rate) */ +#define SPEEX_SET_PLC_TUNING 40 +/** Gets tuning for PLC */ +#define SPEEX_GET_PLC_TUNING 41 + +/** Sets the max bit-rate allowed in VBR mode */ +#define SPEEX_SET_VBR_MAX_BITRATE 42 +/** Gets the max bit-rate allowed in VBR mode */ +#define SPEEX_GET_VBR_MAX_BITRATE 43 + +/** Turn on/off input/output high-pass filtering */ +#define SPEEX_SET_HIGHPASS 44 +/** Get status of input/output high-pass filtering */ +#define SPEEX_GET_HIGHPASS 45 + +/** Get "activity level" of the last decoded frame, i.e. + how much damage we cause if we remove the frame */ +#define SPEEX_GET_ACTIVITY 47 + + +/* Preserving compatibility:*/ +/** Equivalent to SPEEX_SET_ENH */ +#define SPEEX_SET_PF 0 +/** Equivalent to SPEEX_GET_ENH */ +#define SPEEX_GET_PF 1 + + + + +/* Values allowed for mode queries */ +/** Query the frame size of a mode */ +#define SPEEX_MODE_FRAME_SIZE 0 + +/** Query the size of an encoded frame for a particular sub-mode */ +#define SPEEX_SUBMODE_BITS_PER_FRAME 1 + + + +/** Get major Speex version */ +#define SPEEX_LIB_GET_MAJOR_VERSION 1 +/** Get minor Speex version */ +#define SPEEX_LIB_GET_MINOR_VERSION 3 +/** Get micro Speex version */ +#define SPEEX_LIB_GET_MICRO_VERSION 5 +/** Get extra Speex version */ +#define SPEEX_LIB_GET_EXTRA_VERSION 7 +/** Get Speex version string */ +#define SPEEX_LIB_GET_VERSION_STRING 9 + +/*#define SPEEX_LIB_SET_ALLOC_FUNC 10 +#define SPEEX_LIB_GET_ALLOC_FUNC 11 +#define SPEEX_LIB_SET_FREE_FUNC 12 +#define SPEEX_LIB_GET_FREE_FUNC 13 + +#define SPEEX_LIB_SET_WARNING_FUNC 14 +#define SPEEX_LIB_GET_WARNING_FUNC 15 +#define SPEEX_LIB_SET_ERROR_FUNC 16 +#define SPEEX_LIB_GET_ERROR_FUNC 17 +*/ + +/** Number of defined modes in Speex */ +#define SPEEX_NB_MODES 3 + +/** modeID for the defined narrowband mode */ +#define SPEEX_MODEID_NB 0 + +/** modeID for the defined wideband mode */ +#define SPEEX_MODEID_WB 1 + +/** modeID for the defined ultra-wideband mode */ +#define SPEEX_MODEID_UWB 2 + +struct SpeexMode; + + +/* Prototypes for mode function pointers */ + +/** Encoder state initialization function */ +typedef void *(*encoder_init_func)(const struct SpeexMode *mode); + +/** Encoder state destruction function */ +typedef void (*encoder_destroy_func)(void *st); + +/** Main encoding function */ +typedef int (*encode_func)(void *state, void *in, SpeexBits *bits); + +/** Function for controlling the encoder options */ +typedef int (*encoder_ctl_func)(void *state, int request, void *ptr); + +/** Decoder state initialization function */ +typedef void *(*decoder_init_func)(const struct SpeexMode *mode); + +/** Decoder state destruction function */ +typedef void (*decoder_destroy_func)(void *st); + +/** Main decoding function */ +typedef int (*decode_func)(void *state, SpeexBits *bits, void *out); + +/** Function for controlling the decoder options */ +typedef int (*decoder_ctl_func)(void *state, int request, void *ptr); + + +/** Query function for a mode */ +typedef int (*mode_query_func)(const void *mode, int request, void *ptr); + +/** Struct defining a Speex mode */ +typedef struct SpeexMode { + /** Pointer to the low-level mode data */ + const void *mode; + + /** Pointer to the mode query function */ + mode_query_func query; + + /** The name of the mode (you should not rely on this to identify the mode)*/ + const char *modeName; + + /**ID of the mode*/ + int modeID; + + /**Version number of the bitstream (incremented every time we break + bitstream compatibility*/ + int bitstream_version; + + /** Pointer to encoder initialization function */ + encoder_init_func enc_init; + + /** Pointer to encoder destruction function */ + encoder_destroy_func enc_destroy; + + /** Pointer to frame encoding function */ + encode_func enc; + + /** Pointer to decoder initialization function */ + decoder_init_func dec_init; + + /** Pointer to decoder destruction function */ + decoder_destroy_func dec_destroy; + + /** Pointer to frame decoding function */ + decode_func dec; + + /** ioctl-like requests for encoder */ + encoder_ctl_func enc_ctl; + + /** ioctl-like requests for decoder */ + decoder_ctl_func dec_ctl; + +} SpeexMode; + +/** + * Returns a handle to a newly created Speex encoder state structure. For now, + * the "mode" argument can be &nb_mode or &wb_mode . In the future, more modes + * may be added. Note that for now if you have more than one channels to + * encode, you need one state per channel. + * + * @param mode The mode to use (either speex_nb_mode or speex_wb.mode) + * @return A newly created encoder state or NULL if state allocation fails + */ +void *speex_encoder_init(const SpeexMode *mode); + +/** Frees all resources associated to an existing Speex encoder state. + * @param state Encoder state to be destroyed */ +void speex_encoder_destroy(void *state); + +/** Uses an existing encoder state to encode one frame of speech pointed to by + "in". The encoded bit-stream is saved in "bits". + @param state Encoder state + @param in Frame that will be encoded with a +-2^15 range. This data MAY be + overwritten by the encoder and should be considered uninitialised + after the call. + @param bits Bit-stream where the data will be written + @return 0 if frame needs not be transmitted (DTX only), 1 otherwise + */ +int speex_encode(void *state, float *in, SpeexBits *bits); + +/** Uses an existing encoder state to encode one frame of speech pointed to by + "in". The encoded bit-stream is saved in "bits". + @param state Encoder state + @param in Frame that will be encoded with a +-2^15 range + @param bits Bit-stream where the data will be written + @return 0 if frame needs not be transmitted (DTX only), 1 otherwise + */ +int speex_encode_int(void *state, spx_int16_t *in, SpeexBits *bits); + +/** Used like the ioctl function to control the encoder parameters + * + * @param state Encoder state + * @param request ioctl-type request (one of the SPEEX_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter + */ +int speex_encoder_ctl(void *state, int request, void *ptr); + + +/** Returns a handle to a newly created decoder state structure. For now, + * the mode argument can be &nb_mode or &wb_mode . In the future, more modes + * may be added. Note that for now if you have more than one channels to + * decode, you need one state per channel. + * + * @param mode Speex mode (one of speex_nb_mode or speex_wb_mode) + * @return A newly created decoder state or NULL if state allocation fails + */ +void *speex_decoder_init(const SpeexMode *mode); + +/** Frees all resources associated to an existing decoder state. + * + * @param state State to be destroyed + */ +void speex_decoder_destroy(void *state); + +/** Uses an existing decoder state to decode one frame of speech from + * bit-stream bits. The output speech is saved written to out. + * + * @param state Decoder state + * @param bits Bit-stream from which to decode the frame (NULL if the packet was lost) + * @param out Where to write the decoded frame + * @return return status (0 for no error, -1 for end of stream, -2 corrupt stream) + */ +int speex_decode(void *state, SpeexBits *bits, float *out); + +/** Uses an existing decoder state to decode one frame of speech from + * bit-stream bits. The output speech is saved written to out. + * + * @param state Decoder state + * @param bits Bit-stream from which to decode the frame (NULL if the packet was lost) + * @param out Where to write the decoded frame + * @return return status (0 for no error, -1 for end of stream, -2 corrupt stream) + */ +int speex_decode_int(void *state, SpeexBits *bits, spx_int16_t *out); + +/** Used like the ioctl function to control the encoder parameters + * + * @param state Decoder state + * @param request ioctl-type request (one of the SPEEX_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter + */ +int speex_decoder_ctl(void *state, int request, void *ptr); + + +/** Query function for mode information + * + * @param mode Speex mode + * @param request ioctl-type request (one of the SPEEX_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter + */ +int speex_mode_query(const SpeexMode *mode, int request, void *ptr); + +/** Functions for controlling the behavior of libspeex + * @param request ioctl-type request (one of the SPEEX_LIB_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown, -2 for invalid parameter + */ +int speex_lib_ctl(int request, void *ptr); + +/** Default narrowband mode */ +extern const SpeexMode speex_nb_mode; + +/** Default wideband mode */ +extern const SpeexMode speex_wb_mode; + +/** Default "ultra-wideband" mode */ +extern const SpeexMode speex_uwb_mode; + +/** List of all modes available */ +extern const SpeexMode * const speex_mode_list[SPEEX_NB_MODES]; + +/** Obtain one of the modes available */ +const SpeexMode * speex_lib_get_mode (int mode); + +#ifndef WIN32 +/* We actually override the function in the narrowband case so that we can avoid linking in the wideband stuff */ +#define speex_lib_get_mode(mode) ((mode)==SPEEX_MODEID_NB ? &speex_nb_mode : speex_lib_get_mode (mode)) +#endif + +#ifdef __cplusplus +} +#endif + +/** @}*/ +#endif diff --git a/thirdparties/android/include/speex/speex_bits.h b/thirdparties/android/include/speex/speex_bits.h new file mode 100644 index 0000000..a26fb4c --- /dev/null +++ b/thirdparties/android/include/speex/speex_bits.h @@ -0,0 +1,174 @@ +/* Copyright (C) 2002 Jean-Marc Valin */ +/** + @file speex_bits.h + @brief Handles bit packing/unpacking +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +*/ + +#ifndef BITS_H +#define BITS_H +/** @defgroup SpeexBits SpeexBits: Bit-stream manipulations + * This is the structure that holds the bit-stream when encoding or decoding + * with Speex. It allows some manipulations as well. + * @{ + */ + +#ifdef __cplusplus +extern "C" { +#endif + +/** Bit-packing data structure representing (part of) a bit-stream. */ +typedef struct SpeexBits { + char *chars; /**< "raw" data */ + int nbBits; /**< Total number of bits stored in the stream*/ + int charPtr; /**< Position of the byte "cursor" */ + int bitPtr; /**< Position of the bit "cursor" within the current char */ + int owner; /**< Does the struct "own" the "raw" buffer (member "chars") */ + int overflow;/**< Set to one if we try to read past the valid data */ + int buf_size;/**< Allocated size for buffer */ + int reserved1; /**< Reserved for future use */ + void *reserved2; /**< Reserved for future use */ +} SpeexBits; + +/** Initializes and allocates resources for a SpeexBits struct */ +void speex_bits_init(SpeexBits *bits); + +/** Initializes SpeexBits struct using a pre-allocated buffer*/ +void speex_bits_init_buffer(SpeexBits *bits, void *buff, int buf_size); + +/** Sets the bits in a SpeexBits struct to use data from an existing buffer (for decoding without copying data) */ +void speex_bits_set_bit_buffer(SpeexBits *bits, void *buff, int buf_size); + +/** Frees all resources associated to a SpeexBits struct. Right now this does nothing since no resources are allocated, but this could change in the future.*/ +void speex_bits_destroy(SpeexBits *bits); + +/** Resets bits to initial value (just after initialization, erasing content)*/ +void speex_bits_reset(SpeexBits *bits); + +/** Rewind the bit-stream to the beginning (ready for read) without erasing the content */ +void speex_bits_rewind(SpeexBits *bits); + +/** Initializes the bit-stream from the data in an area of memory */ +void speex_bits_read_from(SpeexBits *bits, char *bytes, int len); + +/** Append bytes to the bit-stream + * + * @param bits Bit-stream to operate on + * @param bytes pointer to the bytes what will be appended + * @param len Number of bytes of append + */ +void speex_bits_read_whole_bytes(SpeexBits *bits, char *bytes, int len); + +/** Write the content of a bit-stream to an area of memory + * + * @param bits Bit-stream to operate on + * @param bytes Memory location where to write the bits + * @param max_len Maximum number of bytes to write (i.e. size of the "bytes" buffer) + * @return Number of bytes written to the "bytes" buffer +*/ +int speex_bits_write(SpeexBits *bits, char *bytes, int max_len); + +/** Like speex_bits_write, but writes only the complete bytes in the stream. Also removes the written bytes from the stream */ +int speex_bits_write_whole_bytes(SpeexBits *bits, char *bytes, int max_len); + +/** Append bits to the bit-stream + * @param bits Bit-stream to operate on + * @param data Value to append as integer + * @param nbBits number of bits to consider in "data" + */ +void speex_bits_pack(SpeexBits *bits, int data, int nbBits); + +/** Interpret the next bits in the bit-stream as a signed integer + * + * @param bits Bit-stream to operate on + * @param nbBits Number of bits to interpret + * @return A signed integer represented by the bits read + */ +int speex_bits_unpack_signed(SpeexBits *bits, int nbBits); + +/** Interpret the next bits in the bit-stream as an unsigned integer + * + * @param bits Bit-stream to operate on + * @param nbBits Number of bits to interpret + * @return An unsigned integer represented by the bits read + */ +unsigned int speex_bits_unpack_unsigned(SpeexBits *bits, int nbBits); + +/** Returns the number of bytes in the bit-stream, including the last one even if it is not "full" + * + * @param bits Bit-stream to operate on + * @return Number of bytes in the stream + */ +int speex_bits_nbytes(SpeexBits *bits); + +/** Same as speex_bits_unpack_unsigned, but without modifying the cursor position + * + * @param bits Bit-stream to operate on + * @param nbBits Number of bits to look for + * @return Value of the bits peeked, interpreted as unsigned + */ +unsigned int speex_bits_peek_unsigned(SpeexBits *bits, int nbBits); + +/** Get the value of the next bit in the stream, without modifying the + * "cursor" position + * + * @param bits Bit-stream to operate on + * @return Value of the bit peeked (one bit only) + */ +int speex_bits_peek(SpeexBits *bits); + +/** Advances the position of the "bit cursor" in the stream + * + * @param bits Bit-stream to operate on + * @param n Number of bits to advance + */ +void speex_bits_advance(SpeexBits *bits, int n); + +/** Returns the number of bits remaining to be read in a stream + * + * @param bits Bit-stream to operate on + * @return Number of bits that can still be read from the stream + */ +int speex_bits_remaining(SpeexBits *bits); + +/** Insert a terminator so that the data can be sent as a packet while auto-detecting + * the number of frames in each packet + * + * @param bits Bit-stream to operate on + */ +void speex_bits_insert_terminator(SpeexBits *bits); + +#ifdef __cplusplus +} +#endif + +/* @} */ +#endif diff --git a/thirdparties/android/include/speex/speex_buffer.h b/thirdparties/android/include/speex/speex_buffer.h new file mode 100644 index 0000000..df56f5f --- /dev/null +++ b/thirdparties/android/include/speex/speex_buffer.h @@ -0,0 +1,68 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: speex_buffer.h + This is a very simple ring buffer implementation. It is not thread-safe + so you need to do your own locking. + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SPEEX_BUFFER_H +#define SPEEX_BUFFER_H + +#include "speex/speex_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +struct SpeexBuffer_; +typedef struct SpeexBuffer_ SpeexBuffer; + +SpeexBuffer *speex_buffer_init(int size); + +void speex_buffer_destroy(SpeexBuffer *st); + +int speex_buffer_write(SpeexBuffer *st, void *data, int len); + +int speex_buffer_writezeros(SpeexBuffer *st, int len); + +int speex_buffer_read(SpeexBuffer *st, void *data, int len); + +int speex_buffer_get_available(SpeexBuffer *st); + +int speex_buffer_resize(SpeexBuffer *st, int len); + +#ifdef __cplusplus +} +#endif + +#endif + + + + diff --git a/thirdparties/android/include/speex/speex_callbacks.h b/thirdparties/android/include/speex/speex_callbacks.h new file mode 100644 index 0000000..6f450b3 --- /dev/null +++ b/thirdparties/android/include/speex/speex_callbacks.h @@ -0,0 +1,134 @@ +/* Copyright (C) 2002 Jean-Marc Valin*/ +/** + @file speex_callbacks.h + @brief Describes callback handling and in-band signalling +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +*/ + +#ifndef SPEEX_CALLBACKS_H +#define SPEEX_CALLBACKS_H +/** @defgroup SpeexCallbacks Various definitions for Speex callbacks supported by the decoder. + * @{ + */ + +#include "speex.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** Total number of callbacks */ +#define SPEEX_MAX_CALLBACKS 16 + +/* Describes all the in-band requests */ + +/*These are 1-bit requests*/ +/** Request for perceptual enhancement (1 for on, 0 for off) */ +#define SPEEX_INBAND_ENH_REQUEST 0 +/** Reserved */ +#define SPEEX_INBAND_RESERVED1 1 + +/*These are 4-bit requests*/ +/** Request for a mode change */ +#define SPEEX_INBAND_MODE_REQUEST 2 +/** Request for a low mode change */ +#define SPEEX_INBAND_LOW_MODE_REQUEST 3 +/** Request for a high mode change */ +#define SPEEX_INBAND_HIGH_MODE_REQUEST 4 +/** Request for VBR (1 on, 0 off) */ +#define SPEEX_INBAND_VBR_QUALITY_REQUEST 5 +/** Request to be sent acknowledge */ +#define SPEEX_INBAND_ACKNOWLEDGE_REQUEST 6 +/** Request for VBR (1 for on, 0 for off) */ +#define SPEEX_INBAND_VBR_REQUEST 7 + +/*These are 8-bit requests*/ +/** Send a character in-band */ +#define SPEEX_INBAND_CHAR 8 +/** Intensity stereo information */ +#define SPEEX_INBAND_STEREO 9 + +/*These are 16-bit requests*/ +/** Transmit max bit-rate allowed */ +#define SPEEX_INBAND_MAX_BITRATE 10 + +/*These are 32-bit requests*/ +/** Acknowledge packet reception */ +#define SPEEX_INBAND_ACKNOWLEDGE 12 + +/** Callback function type */ +typedef int (*speex_callback_func)(SpeexBits *bits, void *state, void *data); + +/** Callback information */ +typedef struct SpeexCallback { + int callback_id; /**< ID associated to the callback */ + speex_callback_func func; /**< Callback handler function */ + void *data; /**< Data that will be sent to the handler */ + void *reserved1; /**< Reserved for future use */ + int reserved2; /**< Reserved for future use */ +} SpeexCallback; + +/** Handle in-band request */ +int speex_inband_handler(SpeexBits *bits, SpeexCallback *callback_list, void *state); + +/** Standard handler for mode request (change mode, no questions asked) */ +int speex_std_mode_request_handler(SpeexBits *bits, void *state, void *data); + +/** Standard handler for high mode request (change high mode, no questions asked) */ +int speex_std_high_mode_request_handler(SpeexBits *bits, void *state, void *data); + +/** Standard handler for in-band characters (write to stderr) */ +int speex_std_char_handler(SpeexBits *bits, void *state, void *data); + +/** Default handler for user-defined requests: in this case, just ignore */ +int speex_default_user_handler(SpeexBits *bits, void *state, void *data); + + + +/** Standard handler for low mode request (change low mode, no questions asked) */ +int speex_std_low_mode_request_handler(SpeexBits *bits, void *state, void *data); + +/** Standard handler for VBR request (Set VBR, no questions asked) */ +int speex_std_vbr_request_handler(SpeexBits *bits, void *state, void *data); + +/** Standard handler for enhancer request (Turn enhancer on/off, no questions asked) */ +int speex_std_enh_request_handler(SpeexBits *bits, void *state, void *data); + +/** Standard handler for VBR quality request (Set VBR quality, no questions asked) */ +int speex_std_vbr_quality_request_handler(SpeexBits *bits, void *state, void *data); + + +#ifdef __cplusplus +} +#endif + +/** @} */ +#endif diff --git a/thirdparties/android/include/speex/speex_echo.h b/thirdparties/android/include/speex/speex_echo.h new file mode 100644 index 0000000..53bcd28 --- /dev/null +++ b/thirdparties/android/include/speex/speex_echo.h @@ -0,0 +1,170 @@ +/* Copyright (C) Jean-Marc Valin */ +/** + @file speex_echo.h + @brief Echo cancellation +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SPEEX_ECHO_H +#define SPEEX_ECHO_H +/** @defgroup SpeexEchoState SpeexEchoState: Acoustic echo canceller + * This is the acoustic echo canceller module. + * @{ + */ +#include "speex/speex_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** Obtain frame size used by the AEC */ +#define SPEEX_ECHO_GET_FRAME_SIZE 3 + +/** Set sampling rate */ +#define SPEEX_ECHO_SET_SAMPLING_RATE 24 +/** Get sampling rate */ +#define SPEEX_ECHO_GET_SAMPLING_RATE 25 + +/* Can't set window sizes */ +/** Get size of impulse response (int32) */ +#define SPEEX_ECHO_GET_IMPULSE_RESPONSE_SIZE 27 + +/* Can't set window content */ +/** Get impulse response (int32[]) */ +#define SPEEX_ECHO_GET_IMPULSE_RESPONSE 29 + +/** Internal echo canceller state. Should never be accessed directly. */ +struct SpeexEchoState_; + +/** @class SpeexEchoState + * This holds the state of the echo canceller. You need one per channel. +*/ + +/** Internal echo canceller state. Should never be accessed directly. */ +typedef struct SpeexEchoState_ SpeexEchoState; + +/** Creates a new echo canceller state + * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms) + * @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms) + * @return Newly-created echo canceller state + */ +SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length); + +/** Creates a new multi-channel echo canceller state + * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms) + * @param filter_length Number of samples of echo to cancel (should generally correspond to 100-500 ms) + * @param nb_mic Number of microphone channels + * @param nb_speakers Number of speaker channels + * @return Newly-created echo canceller state + */ +SpeexEchoState *speex_echo_state_init_mc(int frame_size, int filter_length, int nb_mic, int nb_speakers); + +/** Destroys an echo canceller state + * @param st Echo canceller state +*/ +void speex_echo_state_destroy(SpeexEchoState *st); + +/** Performs echo cancellation a frame, based on the audio sent to the speaker (no delay is added + * to playback in this form) + * + * @param st Echo canceller state + * @param rec Signal from the microphone (near end + far end echo) + * @param play Signal played to the speaker (received from far end) + * @param out Returns near-end signal with echo removed + */ +void speex_echo_cancellation(SpeexEchoState *st, const spx_int16_t *rec, const spx_int16_t *play, spx_int16_t *out); + +/** Performs echo cancellation a frame (deprecated) */ +void speex_echo_cancel(SpeexEchoState *st, const spx_int16_t *rec, const spx_int16_t *play, spx_int16_t *out, spx_int32_t *Yout); + +/** Perform echo cancellation using internal playback buffer, which is delayed by two frames + * to account for the delay introduced by most soundcards (but it could be off!) + * @param st Echo canceller state + * @param rec Signal from the microphone (near end + far end echo) + * @param out Returns near-end signal with echo removed +*/ +void speex_echo_capture(SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out); + +/** Let the echo canceller know that a frame was just queued to the soundcard + * @param st Echo canceller state + * @param play Signal played to the speaker (received from far end) +*/ +void speex_echo_playback(SpeexEchoState *st, const spx_int16_t *play); + +/** Reset the echo canceller to its original state + * @param st Echo canceller state + */ +void speex_echo_state_reset(SpeexEchoState *st); + +/** Used like the ioctl function to control the echo canceller parameters + * + * @param st Echo canceller state + * @param request ioctl-type request (one of the SPEEX_ECHO_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown + */ +int speex_echo_ctl(SpeexEchoState *st, int request, void *ptr); + + + +struct SpeexDecorrState_; + +typedef struct SpeexDecorrState_ SpeexDecorrState; + + +/** Create a state for the channel decorrelation algorithm + This is useful for multi-channel echo cancellation only + * @param rate Sampling rate + * @param channels Number of channels (it's a bit pointless if you don't have at least 2) + * @param frame_size Size of the frame to process at ones (counting samples *per* channel) +*/ +SpeexDecorrState *speex_decorrelate_new(int rate, int channels, int frame_size); + +/** Remove correlation between the channels by modifying the phase and possibly + adding noise in a way that is not (or little) perceptible. + * @param st Decorrelator state + * @param in Input audio in interleaved format + * @param out Result of the decorrelation (out *may* alias in) + * @param strength How much alteration of the audio to apply from 0 to 100. +*/ +void speex_decorrelate(SpeexDecorrState *st, const spx_int16_t *in, spx_int16_t *out, int strength); + +/** Destroy a Decorrelation state + * @param st State to destroy +*/ +void speex_decorrelate_destroy(SpeexDecorrState *st); + + +#ifdef __cplusplus +} +#endif + + +/** @}*/ +#endif diff --git a/thirdparties/android/include/speex/speex_header.h b/thirdparties/android/include/speex/speex_header.h new file mode 100644 index 0000000..f85b249 --- /dev/null +++ b/thirdparties/android/include/speex/speex_header.h @@ -0,0 +1,94 @@ +/* Copyright (C) 2002 Jean-Marc Valin */ +/** + @file speex_header.h + @brief Describes the Speex header +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +*/ + + +#ifndef SPEEX_HEADER_H +#define SPEEX_HEADER_H +/** @defgroup SpeexHeader SpeexHeader: Makes it easy to write/parse an Ogg/Speex header + * This is the Speex header for the Ogg encapsulation. You don't need that if you just use RTP. + * @{ + */ + +#include "speex/speex_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +struct SpeexMode; + +/** Length of the Speex header identifier */ +#define SPEEX_HEADER_STRING_LENGTH 8 + +/** Maximum number of characters for encoding the Speex version number in the header */ +#define SPEEX_HEADER_VERSION_LENGTH 20 + +/** Speex header info for file-based formats */ +typedef struct SpeexHeader { + char speex_string[SPEEX_HEADER_STRING_LENGTH]; /**< Identifies a Speex bit-stream, always set to "Speex " */ + char speex_version[SPEEX_HEADER_VERSION_LENGTH]; /**< Speex version */ + spx_int32_t speex_version_id; /**< Version for Speex (for checking compatibility) */ + spx_int32_t header_size; /**< Total size of the header ( sizeof(SpeexHeader) ) */ + spx_int32_t rate; /**< Sampling rate used */ + spx_int32_t mode; /**< Mode used (0 for narrowband, 1 for wideband) */ + spx_int32_t mode_bitstream_version; /**< Version ID of the bit-stream */ + spx_int32_t nb_channels; /**< Number of channels encoded */ + spx_int32_t bitrate; /**< Bit-rate used */ + spx_int32_t frame_size; /**< Size of frames */ + spx_int32_t vbr; /**< 1 for a VBR encoding, 0 otherwise */ + spx_int32_t frames_per_packet; /**< Number of frames stored per Ogg packet */ + spx_int32_t extra_headers; /**< Number of additional headers after the comments */ + spx_int32_t reserved1; /**< Reserved for future use, must be zero */ + spx_int32_t reserved2; /**< Reserved for future use, must be zero */ +} SpeexHeader; + +/** Initializes a SpeexHeader using basic information */ +void speex_init_header(SpeexHeader *header, int rate, int nb_channels, const struct SpeexMode *m); + +/** Creates the header packet from the header itself (mostly involves endianness conversion) */ +char *speex_header_to_packet(SpeexHeader *header, int *size); + +/** Creates a SpeexHeader from a packet */ +SpeexHeader *speex_packet_to_header(char *packet, int size); + +/** Frees the memory allocated by either speex_header_to_packet() or speex_packet_to_header() */ +void speex_header_free(void *ptr); + +#ifdef __cplusplus +} +#endif + +/** @} */ +#endif diff --git a/thirdparties/android/include/speex/speex_jitter.h b/thirdparties/android/include/speex/speex_jitter.h new file mode 100644 index 0000000..d68674b --- /dev/null +++ b/thirdparties/android/include/speex/speex_jitter.h @@ -0,0 +1,197 @@ +/* Copyright (C) 2002 Jean-Marc Valin */ +/** + @file speex_jitter.h + @brief Adaptive jitter buffer for Speex +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +*/ + +#ifndef SPEEX_JITTER_H +#define SPEEX_JITTER_H +/** @defgroup JitterBuffer JitterBuffer: Adaptive jitter buffer + * This is the jitter buffer that reorders UDP/RTP packets and adjusts the buffer size + * to maintain good quality and low latency. + * @{ + */ + +#include "speex/speex_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** Generic adaptive jitter buffer state */ +struct JitterBuffer_; + +/** Generic adaptive jitter buffer state */ +typedef struct JitterBuffer_ JitterBuffer; + +/** Definition of an incoming packet */ +typedef struct _JitterBufferPacket JitterBufferPacket; + +/** Definition of an incoming packet */ +struct _JitterBufferPacket { + char *data; /**< Data bytes contained in the packet */ + spx_uint32_t len; /**< Length of the packet in bytes */ + spx_uint32_t timestamp; /**< Timestamp for the packet */ + spx_uint32_t span; /**< Time covered by the packet (same units as timestamp) */ + spx_uint16_t sequence; /**< RTP Sequence number if available (0 otherwise) */ + spx_uint32_t user_data; /**< Put whatever data you like here (it's ignored by the jitter buffer) */ +}; + +/** Packet has been retrieved */ +#define JITTER_BUFFER_OK 0 +/** Packet is lost or is late */ +#define JITTER_BUFFER_MISSING 1 +/** A "fake" packet is meant to be inserted here to increase buffering */ +#define JITTER_BUFFER_INSERTION 2 +/** There was an error in the jitter buffer */ +#define JITTER_BUFFER_INTERNAL_ERROR -1 +/** Invalid argument */ +#define JITTER_BUFFER_BAD_ARGUMENT -2 + + +/** Set minimum amount of extra buffering required (margin) */ +#define JITTER_BUFFER_SET_MARGIN 0 +/** Get minimum amount of extra buffering required (margin) */ +#define JITTER_BUFFER_GET_MARGIN 1 +/* JITTER_BUFFER_SET_AVAILABLE_COUNT wouldn't make sense */ + +/** Get the amount of available packets currently buffered */ +#define JITTER_BUFFER_GET_AVAILABLE_COUNT 3 +/** Included because of an early misspelling (will remove in next release) */ +#define JITTER_BUFFER_GET_AVALIABLE_COUNT 3 + +/** Assign a function to destroy unused packet. When setting that, the jitter + buffer no longer copies packet data. */ +#define JITTER_BUFFER_SET_DESTROY_CALLBACK 4 +/** */ +#define JITTER_BUFFER_GET_DESTROY_CALLBACK 5 + +/** Tell the jitter buffer to only adjust the delay in multiples of the step parameter provided */ +#define JITTER_BUFFER_SET_DELAY_STEP 6 +/** */ +#define JITTER_BUFFER_GET_DELAY_STEP 7 + +/** Tell the jitter buffer to only do concealment in multiples of the size parameter provided */ +#define JITTER_BUFFER_SET_CONCEALMENT_SIZE 8 +#define JITTER_BUFFER_GET_CONCEALMENT_SIZE 9 + +/** Absolute max amount of loss that can be tolerated regardless of the delay. Typical loss + should be half of that or less. */ +#define JITTER_BUFFER_SET_MAX_LATE_RATE 10 +#define JITTER_BUFFER_GET_MAX_LATE_RATE 11 + +/** Equivalent cost of one percent late packet in timestamp units */ +#define JITTER_BUFFER_SET_LATE_COST 12 +#define JITTER_BUFFER_GET_LATE_COST 13 + + +/** Initialises jitter buffer + * + * @param step_size Starting value for the size of concleanment packets and delay + adjustment steps. Can be changed at any time using JITTER_BUFFER_SET_DELAY_STEP + and JITTER_BUFFER_GET_CONCEALMENT_SIZE. + * @return Newly created jitter buffer state + */ +JitterBuffer *jitter_buffer_init(int step_size); + +/** Restores jitter buffer to its original state + * + * @param jitter Jitter buffer state + */ +void jitter_buffer_reset(JitterBuffer *jitter); + +/** Destroys jitter buffer + * + * @param jitter Jitter buffer state + */ +void jitter_buffer_destroy(JitterBuffer *jitter); + +/** Put one packet into the jitter buffer + * + * @param jitter Jitter buffer state + * @param packet Incoming packet +*/ +void jitter_buffer_put(JitterBuffer *jitter, const JitterBufferPacket *packet); + +/** Get one packet from the jitter buffer + * + * @param jitter Jitter buffer state + * @param packet Returned packet + * @param desired_span Number of samples (or units) we wish to get from the buffer (no guarantee) + * @param current_timestamp Timestamp for the returned packet +*/ +int jitter_buffer_get(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t desired_span, spx_int32_t *start_offset); + +/** Used right after jitter_buffer_get() to obtain another packet that would have the same timestamp. + * This is mainly useful for media where a single "frame" can be split into several packets. + * + * @param jitter Jitter buffer state + * @param packet Returned packet + */ +int jitter_buffer_get_another(JitterBuffer *jitter, JitterBufferPacket *packet); + +/** Get pointer timestamp of jitter buffer + * + * @param jitter Jitter buffer state +*/ +int jitter_buffer_get_pointer_timestamp(JitterBuffer *jitter); + +/** Advance by one tick + * + * @param jitter Jitter buffer state +*/ +void jitter_buffer_tick(JitterBuffer *jitter); + +/** Telling the jitter buffer about the remaining data in the application buffer + * @param jitter Jitter buffer state + * @param rem Amount of data buffered by the application (timestamp units) + */ +void jitter_buffer_remaining_span(JitterBuffer *jitter, spx_uint32_t rem); + +/** Used like the ioctl function to control the jitter buffer parameters + * + * @param jitter Jitter buffer state + * @param request ioctl-type request (one of the JITTER_BUFFER_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown +*/ +int jitter_buffer_ctl(JitterBuffer *jitter, int request, void *ptr); + +int jitter_buffer_update_delay(JitterBuffer *jitter, JitterBufferPacket *packet, spx_int32_t *start_offset); + +/* @} */ + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/thirdparties/android/include/speex/speex_preprocess.h b/thirdparties/android/include/speex/speex_preprocess.h new file mode 100644 index 0000000..f8eef2c --- /dev/null +++ b/thirdparties/android/include/speex/speex_preprocess.h @@ -0,0 +1,219 @@ +/* Copyright (C) 2003 Epic Games + Written by Jean-Marc Valin */ +/** + * @file speex_preprocess.h + * @brief Speex preprocessor. The preprocess can do noise suppression, + * residual echo suppression (after using the echo canceller), automatic + * gain control (AGC) and voice activity detection (VAD). +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef SPEEX_PREPROCESS_H +#define SPEEX_PREPROCESS_H +/** @defgroup SpeexPreprocessState SpeexPreprocessState: The Speex preprocessor + * This is the Speex preprocessor. The preprocess can do noise suppression, + * residual echo suppression (after using the echo canceller), automatic + * gain control (AGC) and voice activity detection (VAD). + * @{ + */ + +#include "speex/speex_types.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** State of the preprocessor (one per channel). Should never be accessed directly. */ +struct SpeexPreprocessState_; + +/** State of the preprocessor (one per channel). Should never be accessed directly. */ +typedef struct SpeexPreprocessState_ SpeexPreprocessState; + + +/** Creates a new preprocessing state. You MUST create one state per channel processed. + * @param frame_size Number of samples to process at one time (should correspond to 10-20 ms). Must be + * the same value as that used for the echo canceller for residual echo cancellation to work. + * @param sampling_rate Sampling rate used for the input. + * @return Newly created preprocessor state +*/ +SpeexPreprocessState *speex_preprocess_state_init(int frame_size, int sampling_rate); + +/** Destroys a preprocessor state + * @param st Preprocessor state to destroy +*/ +void speex_preprocess_state_destroy(SpeexPreprocessState *st); + +/** Preprocess a frame + * @param st Preprocessor state + * @param x Audio sample vector (in and out). Must be same size as specified in speex_preprocess_state_init(). + * @return Bool value for voice activity (1 for speech, 0 for noise/silence), ONLY if VAD turned on. +*/ +int speex_preprocess_run(SpeexPreprocessState *st, spx_int16_t *x); + +/** Preprocess a frame (deprecated, use speex_preprocess_run() instead)*/ +int speex_preprocess(SpeexPreprocessState *st, spx_int16_t *x, spx_int32_t *echo); + +/** Update preprocessor state, but do not compute the output + * @param st Preprocessor state + * @param x Audio sample vector (in only). Must be same size as specified in speex_preprocess_state_init(). +*/ +void speex_preprocess_estimate_update(SpeexPreprocessState *st, spx_int16_t *x); + +/** Used like the ioctl function to control the preprocessor parameters + * @param st Preprocessor state + * @param request ioctl-type request (one of the SPEEX_PREPROCESS_* macros) + * @param ptr Data exchanged to-from function + * @return 0 if no error, -1 if request in unknown +*/ +int speex_preprocess_ctl(SpeexPreprocessState *st, int request, void *ptr); + + + +/** Set preprocessor denoiser state */ +#define SPEEX_PREPROCESS_SET_DENOISE 0 +/** Get preprocessor denoiser state */ +#define SPEEX_PREPROCESS_GET_DENOISE 1 + +/** Set preprocessor Automatic Gain Control state */ +#define SPEEX_PREPROCESS_SET_AGC 2 +/** Get preprocessor Automatic Gain Control state */ +#define SPEEX_PREPROCESS_GET_AGC 3 + +/** Set preprocessor Voice Activity Detection state */ +#define SPEEX_PREPROCESS_SET_VAD 4 +/** Get preprocessor Voice Activity Detection state */ +#define SPEEX_PREPROCESS_GET_VAD 5 + +/** Set preprocessor Automatic Gain Control level (float) */ +#define SPEEX_PREPROCESS_SET_AGC_LEVEL 6 +/** Get preprocessor Automatic Gain Control level (float) */ +#define SPEEX_PREPROCESS_GET_AGC_LEVEL 7 + +/** Set preprocessor dereverb state */ +#define SPEEX_PREPROCESS_SET_DEREVERB 8 +/** Get preprocessor dereverb state */ +#define SPEEX_PREPROCESS_GET_DEREVERB 9 + +/** Set preprocessor dereverb level */ +#define SPEEX_PREPROCESS_SET_DEREVERB_LEVEL 10 +/** Get preprocessor dereverb level */ +#define SPEEX_PREPROCESS_GET_DEREVERB_LEVEL 11 + +/** Set preprocessor dereverb decay */ +#define SPEEX_PREPROCESS_SET_DEREVERB_DECAY 12 +/** Get preprocessor dereverb decay */ +#define SPEEX_PREPROCESS_GET_DEREVERB_DECAY 13 + +/** Set probability required for the VAD to go from silence to voice */ +#define SPEEX_PREPROCESS_SET_PROB_START 14 +/** Get probability required for the VAD to go from silence to voice */ +#define SPEEX_PREPROCESS_GET_PROB_START 15 + +/** Set probability required for the VAD to stay in the voice state (integer percent) */ +#define SPEEX_PREPROCESS_SET_PROB_CONTINUE 16 +/** Get probability required for the VAD to stay in the voice state (integer percent) */ +#define SPEEX_PREPROCESS_GET_PROB_CONTINUE 17 + +/** Set maximum attenuation of the noise in dB (negative number) */ +#define SPEEX_PREPROCESS_SET_NOISE_SUPPRESS 18 +/** Get maximum attenuation of the noise in dB (negative number) */ +#define SPEEX_PREPROCESS_GET_NOISE_SUPPRESS 19 + +/** Set maximum attenuation of the residual echo in dB (negative number) */ +#define SPEEX_PREPROCESS_SET_ECHO_SUPPRESS 20 +/** Get maximum attenuation of the residual echo in dB (negative number) */ +#define SPEEX_PREPROCESS_GET_ECHO_SUPPRESS 21 + +/** Set maximum attenuation of the residual echo in dB when near end is active (negative number) */ +#define SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE 22 +/** Get maximum attenuation of the residual echo in dB when near end is active (negative number) */ +#define SPEEX_PREPROCESS_GET_ECHO_SUPPRESS_ACTIVE 23 + +/** Set the corresponding echo canceller state so that residual echo suppression can be performed (NULL for no residual echo suppression) */ +#define SPEEX_PREPROCESS_SET_ECHO_STATE 24 +/** Get the corresponding echo canceller state */ +#define SPEEX_PREPROCESS_GET_ECHO_STATE 25 + +/** Set maximal gain increase in dB/second (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_INCREMENT 26 + +/** Get maximal gain increase in dB/second (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_INCREMENT 27 + +/** Set maximal gain decrease in dB/second (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_DECREMENT 28 + +/** Get maximal gain decrease in dB/second (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_DECREMENT 29 + +/** Set maximal gain in dB (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_MAX_GAIN 30 + +/** Get maximal gain in dB (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_MAX_GAIN 31 + +/* Can't set loudness */ +/** Get loudness */ +#define SPEEX_PREPROCESS_GET_AGC_LOUDNESS 33 + +/* Can't set gain */ +/** Get current gain (int32 percent) */ +#define SPEEX_PREPROCESS_GET_AGC_GAIN 35 + +/* Can't set spectrum size */ +/** Get spectrum size for power spectrum (int32) */ +#define SPEEX_PREPROCESS_GET_PSD_SIZE 37 + +/* Can't set power spectrum */ +/** Get power spectrum (int32[] of squared values) */ +#define SPEEX_PREPROCESS_GET_PSD 39 + +/* Can't set noise size */ +/** Get spectrum size for noise estimate (int32) */ +#define SPEEX_PREPROCESS_GET_NOISE_PSD_SIZE 41 + +/* Can't set noise estimate */ +/** Get noise estimate (int32[] of squared values) */ +#define SPEEX_PREPROCESS_GET_NOISE_PSD 43 + +/* Can't set speech probability */ +/** Get speech probability in last frame (int32). */ +#define SPEEX_PREPROCESS_GET_PROB 45 + +/** Set preprocessor Automatic Gain Control level (int32) */ +#define SPEEX_PREPROCESS_SET_AGC_TARGET 46 +/** Get preprocessor Automatic Gain Control level (int32) */ +#define SPEEX_PREPROCESS_GET_AGC_TARGET 47 + +#ifdef __cplusplus +} +#endif + +/** @}*/ +#endif diff --git a/thirdparties/android/include/speex/speex_resampler.h b/thirdparties/android/include/speex/speex_resampler.h new file mode 100644 index 0000000..54eef8d --- /dev/null +++ b/thirdparties/android/include/speex/speex_resampler.h @@ -0,0 +1,340 @@ +/* Copyright (C) 2007 Jean-Marc Valin + + File: speex_resampler.h + Resampling code + + The design goals of this code are: + - Very fast algorithm + - Low memory requirement + - Good *perceptual* quality (and not best SNR) + + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions are + met: + + 1. Redistributions of source code must retain the above copyright notice, + this list of conditions and the following disclaimer. + + 2. Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + 3. The name of the author may not be used to endorse or promote products + derived from this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR + IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES + OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE + DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, + INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES + (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR + SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, + STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + POSSIBILITY OF SUCH DAMAGE. +*/ + + +#ifndef SPEEX_RESAMPLER_H +#define SPEEX_RESAMPLER_H + +#ifdef OUTSIDE_SPEEX + +/********* WARNING: MENTAL SANITY ENDS HERE *************/ + +/* If the resampler is defined outside of Speex, we change the symbol names so that + there won't be any clash if linking with Speex later on. */ + +/* #define RANDOM_PREFIX your software name here */ +#ifndef RANDOM_PREFIX +#error "Please define RANDOM_PREFIX (above) to something specific to your project to prevent symbol name clashes" +#endif + +#define CAT_PREFIX2(a,b) a ## b +#define CAT_PREFIX(a,b) CAT_PREFIX2(a, b) + +#define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init) +#define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac) +#define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy) +#define speex_resampler_process_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_float) +#define speex_resampler_process_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_int) +#define speex_resampler_process_interleaved_float CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_float) +#define speex_resampler_process_interleaved_int CAT_PREFIX(RANDOM_PREFIX,_resampler_process_interleaved_int) +#define speex_resampler_set_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate) +#define speex_resampler_get_rate CAT_PREFIX(RANDOM_PREFIX,_resampler_get_rate) +#define speex_resampler_set_rate_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_set_rate_frac) +#define speex_resampler_get_ratio CAT_PREFIX(RANDOM_PREFIX,_resampler_get_ratio) +#define speex_resampler_set_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_set_quality) +#define speex_resampler_get_quality CAT_PREFIX(RANDOM_PREFIX,_resampler_get_quality) +#define speex_resampler_set_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_input_stride) +#define speex_resampler_get_input_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_stride) +#define speex_resampler_set_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_set_output_stride) +#define speex_resampler_get_output_stride CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_stride) +#define speex_resampler_get_input_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_input_latency) +#define speex_resampler_get_output_latency CAT_PREFIX(RANDOM_PREFIX,_resampler_get_output_latency) +#define speex_resampler_skip_zeros CAT_PREFIX(RANDOM_PREFIX,_resampler_skip_zeros) +#define speex_resampler_reset_mem CAT_PREFIX(RANDOM_PREFIX,_resampler_reset_mem) +#define speex_resampler_strerror CAT_PREFIX(RANDOM_PREFIX,_resampler_strerror) + +#define spx_int16_t short +#define spx_int32_t int +#define spx_uint16_t unsigned short +#define spx_uint32_t unsigned int + +#else /* OUTSIDE_SPEEX */ + +#include "speex/speex_types.h" + +#endif /* OUTSIDE_SPEEX */ + +#ifdef __cplusplus +extern "C" { +#endif + +#define SPEEX_RESAMPLER_QUALITY_MAX 10 +#define SPEEX_RESAMPLER_QUALITY_MIN 0 +#define SPEEX_RESAMPLER_QUALITY_DEFAULT 4 +#define SPEEX_RESAMPLER_QUALITY_VOIP 3 +#define SPEEX_RESAMPLER_QUALITY_DESKTOP 5 + +enum { + RESAMPLER_ERR_SUCCESS = 0, + RESAMPLER_ERR_ALLOC_FAILED = 1, + RESAMPLER_ERR_BAD_STATE = 2, + RESAMPLER_ERR_INVALID_ARG = 3, + RESAMPLER_ERR_PTR_OVERLAP = 4, + + RESAMPLER_ERR_MAX_ERROR +}; + +struct SpeexResamplerState_; +typedef struct SpeexResamplerState_ SpeexResamplerState; + +/** Create a new resampler with integer input and output rates. + * @param nb_channels Number of channels to be processed + * @param in_rate Input sampling rate (integer number of Hz). + * @param out_rate Output sampling rate (integer number of Hz). + * @param quality Resampling quality between 0 and 10, where 0 has poor quality + * and 10 has very high quality. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, + spx_uint32_t in_rate, + spx_uint32_t out_rate, + int quality, + int *err); + +/** Create a new resampler with fractional input/output rates. The sampling + * rate ratio is an arbitrary rational number with both the numerator and + * denominator being 32-bit integers. + * @param nb_channels Number of channels to be processed + * @param ratio_num Numerator of the sampling rate ratio + * @param ratio_den Denominator of the sampling rate ratio + * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). + * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). + * @param quality Resampling quality between 0 and 10, where 0 has poor quality + * and 10 has very high quality. + * @return Newly created resampler state + * @retval NULL Error: not enough memory + */ +SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, + spx_uint32_t out_rate, + int quality, + int *err); + +/** Destroy a resampler state. + * @param st Resampler state + */ +void speex_resampler_destroy(SpeexResamplerState *st); + +/** Resample a float array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param channel_index Index of the channel to process for the multi-channel + * base (0 otherwise) + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the + * number of samples processed + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written + */ +int speex_resampler_process_float(SpeexResamplerState *st, + spx_uint32_t channel_index, + const float *in, + spx_uint32_t *in_len, + float *out, + spx_uint32_t *out_len); + +/** Resample an int array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param channel_index Index of the channel to process for the multi-channel + * base (0 otherwise) + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written + */ +int speex_resampler_process_int(SpeexResamplerState *st, + spx_uint32_t channel_index, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, + spx_uint32_t *out_len); + +/** Resample an interleaved float array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed. This is all per-channel. + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written. + * This is all per-channel. + */ +int speex_resampler_process_interleaved_float(SpeexResamplerState *st, + const float *in, + spx_uint32_t *in_len, + float *out, + spx_uint32_t *out_len); + +/** Resample an interleaved int array. The input and output buffers must *not* overlap. + * @param st Resampler state + * @param in Input buffer + * @param in_len Number of input samples in the input buffer. Returns the number + * of samples processed. This is all per-channel. + * @param out Output buffer + * @param out_len Size of the output buffer. Returns the number of samples written. + * This is all per-channel. + */ +int speex_resampler_process_interleaved_int(SpeexResamplerState *st, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, + spx_uint32_t *out_len); + +/** Set (change) the input/output sampling rates (integer value). + * @param st Resampler state + * @param in_rate Input sampling rate (integer number of Hz). + * @param out_rate Output sampling rate (integer number of Hz). + */ +int speex_resampler_set_rate(SpeexResamplerState *st, + spx_uint32_t in_rate, + spx_uint32_t out_rate); + +/** Get the current input/output sampling rates (integer value). + * @param st Resampler state + * @param in_rate Input sampling rate (integer number of Hz) copied. + * @param out_rate Output sampling rate (integer number of Hz) copied. + */ +void speex_resampler_get_rate(SpeexResamplerState *st, + spx_uint32_t *in_rate, + spx_uint32_t *out_rate); + +/** Set (change) the input/output sampling rates and resampling ratio + * (fractional values in Hz supported). + * @param st Resampler state + * @param ratio_num Numerator of the sampling rate ratio + * @param ratio_den Denominator of the sampling rate ratio + * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). + * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). + */ +int speex_resampler_set_rate_frac(SpeexResamplerState *st, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, + spx_uint32_t out_rate); + +/** Get the current resampling ratio. This will be reduced to the least + * common denominator. + * @param st Resampler state + * @param ratio_num Numerator of the sampling rate ratio copied + * @param ratio_den Denominator of the sampling rate ratio copied + */ +void speex_resampler_get_ratio(SpeexResamplerState *st, + spx_uint32_t *ratio_num, + spx_uint32_t *ratio_den); + +/** Set (change) the conversion quality. + * @param st Resampler state + * @param quality Resampling quality between 0 and 10, where 0 has poor + * quality and 10 has very high quality. + */ +int speex_resampler_set_quality(SpeexResamplerState *st, + int quality); + +/** Get the conversion quality. + * @param st Resampler state + * @param quality Resampling quality between 0 and 10, where 0 has poor + * quality and 10 has very high quality. + */ +void speex_resampler_get_quality(SpeexResamplerState *st, + int *quality); + +/** Set (change) the input stride. + * @param st Resampler state + * @param stride Input stride + */ +void speex_resampler_set_input_stride(SpeexResamplerState *st, + spx_uint32_t stride); + +/** Get the input stride. + * @param st Resampler state + * @param stride Input stride copied + */ +void speex_resampler_get_input_stride(SpeexResamplerState *st, + spx_uint32_t *stride); + +/** Set (change) the output stride. + * @param st Resampler state + * @param stride Output stride + */ +void speex_resampler_set_output_stride(SpeexResamplerState *st, + spx_uint32_t stride); + +/** Get the output stride. + * @param st Resampler state copied + * @param stride Output stride + */ +void speex_resampler_get_output_stride(SpeexResamplerState *st, + spx_uint32_t *stride); + +/** Get the latency in input samples introduced by the resampler. + * @param st Resampler state + */ +int speex_resampler_get_input_latency(SpeexResamplerState *st); + +/** Get the latency in output samples introduced by the resampler. + * @param st Resampler state + */ +int speex_resampler_get_output_latency(SpeexResamplerState *st); + +/** Make sure that the first samples to go out of the resamplers don't have + * leading zeros. This is only useful before starting to use a newly created + * resampler. It is recommended to use that when resampling an audio file, as + * it will generate a file with the same length. For real-time processing, + * it is probably easier not to use this call (so that the output duration + * is the same for the first frame). + * @param st Resampler state + */ +int speex_resampler_skip_zeros(SpeexResamplerState *st); + +/** Reset a resampler so a new (unrelated) stream can be processed. + * @param st Resampler state + */ +int speex_resampler_reset_mem(SpeexResamplerState *st); + +/** Returns the English meaning for an error code + * @param err Error code + * @return English string + */ +const char *speex_resampler_strerror(int err); + +#ifdef __cplusplus +} +#endif + +#endif diff --git a/thirdparties/android/include/speex/speex_stereo.h b/thirdparties/android/include/speex/speex_stereo.h new file mode 100644 index 0000000..a259713 --- /dev/null +++ b/thirdparties/android/include/speex/speex_stereo.h @@ -0,0 +1,91 @@ +/* Copyright (C) 2002 Jean-Marc Valin*/ +/** + @file speex_stereo.h + @brief Describes the handling for intensity stereo +*/ +/* + Redistribution and use in source and binary forms, with or without + modification, are permitted provided that the following conditions + are met: + + - Redistributions of source code must retain the above copyright + notice, this list of conditions and the following disclaimer. + + - Redistributions in binary form must reproduce the above copyright + notice, this list of conditions and the following disclaimer in the + documentation and/or other materials provided with the distribution. + + - Neither the name of the Xiph.org Foundation nor the names of its + contributors may be used to endorse or promote products derived from + this software without specific prior written permission. + + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS + ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT + LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR + A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR + CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, + EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR + PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF + LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING + NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS + SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. +*/ + +#ifndef STEREO_H +#define STEREO_H +/** @defgroup SpeexStereoState SpeexStereoState: Handling Speex stereo files + * This describes the Speex intensity stereo encoding/decoding + * @{ + */ + +#include "speex/speex_types.h" +#include "speex/speex_bits.h" + +#ifdef __cplusplus +extern "C" { +#endif + +/** If you access any of these fields directly, I'll personally come and bite you */ +typedef struct SpeexStereoState { + float balance; /**< Left/right balance info */ + float e_ratio; /**< Ratio of energies: E(left+right)/[E(left)+E(right)] */ + float smooth_left; /**< Smoothed left channel gain */ + float smooth_right; /**< Smoothed right channel gain */ + float reserved1; /**< Reserved for future use */ + float reserved2; /**< Reserved for future use */ +} SpeexStereoState; + +/** Deprecated. Use speex_stereo_state_init() instead. */ +#define SPEEX_STEREO_STATE_INIT {1,.5,1,1,0,0} + +/** Initialise/create a stereo stereo state */ +SpeexStereoState *speex_stereo_state_init(); + +/** Reset/re-initialise an already allocated stereo state */ +void speex_stereo_state_reset(SpeexStereoState *stereo); + +/** Destroy a stereo stereo state */ +void speex_stereo_state_destroy(SpeexStereoState *stereo); + +/** Transforms a stereo frame into a mono frame and stores intensity stereo info in 'bits' */ +void speex_encode_stereo(float *data, int frame_size, SpeexBits *bits); + +/** Transforms a stereo frame into a mono frame and stores intensity stereo info in 'bits' */ +void speex_encode_stereo_int(spx_int16_t *data, int frame_size, SpeexBits *bits); + +/** Transforms a mono frame into a stereo frame using intensity stereo info */ +void speex_decode_stereo(float *data, int frame_size, SpeexStereoState *stereo); + +/** Transforms a mono frame into a stereo frame using intensity stereo info */ +void speex_decode_stereo_int(spx_int16_t *data, int frame_size, SpeexStereoState *stereo); + +/** Callback handler for intensity stereo info */ +int speex_std_stereo_request_handler(SpeexBits *bits, void *state, void *data); + +#ifdef __cplusplus +} +#endif + +/** @} */ +#endif diff --git a/thirdparties/android/include/speex/speex_types.h b/thirdparties/android/include/speex/speex_types.h new file mode 100644 index 0000000..737c5fa --- /dev/null +++ b/thirdparties/android/include/speex/speex_types.h @@ -0,0 +1,133 @@ +/* speex_types.h taken from libogg */ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: #ifdef jail to whip a few platforms into the UNIX ideal. + last mod: $Id: os_types.h 7524 2004-08-11 04:20:36Z conrad $ + + ********************************************************************/ +/** + @file speex_types.h + @brief Speex types +*/ +#ifndef _SPEEX_TYPES_H +#define _SPEEX_TYPES_H + +#if defined(_WIN32) + +# if defined(__CYGWIN__) +# include <_G_config.h> + typedef _G_int32_t spx_int32_t; + typedef _G_uint32_t spx_uint32_t; + typedef _G_int16_t spx_int16_t; + typedef _G_uint16_t spx_uint16_t; +# elif defined(__MINGW32__) + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; +# elif defined(__MWERKS__) + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; +# else + /* MSVC/Borland */ + typedef __int32 spx_int32_t; + typedef unsigned __int32 spx_uint32_t; + typedef __int16 spx_int16_t; + typedef unsigned __int16 spx_uint16_t; +# endif + +#elif defined(__MACOS__) + +# include <sys/types.h> + typedef SInt16 spx_int16_t; + typedef UInt16 spx_uint16_t; + typedef SInt32 spx_int32_t; + typedef UInt32 spx_uint32_t; + +#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */ + +# include <sys/types.h> + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined(__BEOS__) + + /* Be */ +# include <inttypes.h> + typedef int16_t spx_int16_t; + typedef u_int16_t spx_uint16_t; + typedef int32_t spx_int32_t; + typedef u_int32_t spx_uint32_t; + +#elif defined (__EMX__) + + /* OS/2 GCC */ + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined (DJGPP) + + /* DJGPP */ + typedef short spx_int16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(R5900) + + /* PS2 EE */ + typedef int spx_int32_t; + typedef unsigned spx_uint32_t; + typedef short spx_int16_t; + +#elif defined(__SYMBIAN32__) + + /* Symbian GCC */ + typedef signed short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef signed int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef long spx_int32_t; + typedef unsigned long spx_uint32_t; + +#elif defined(CONFIG_TI_C6X) + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#elif ANDROID + + typedef short spx_int16_t; + typedef unsigned short spx_uint16_t; + typedef int spx_int32_t; + typedef unsigned int spx_uint32_t; + +#else + +# include <speex/speex_config_types.h> + +#endif + +#endif /* _SPEEX_TYPES_H */ diff --git a/thirdparties/android/include/theora/codec.h b/thirdparties/android/include/theora/codec.h new file mode 100644 index 0000000..5c26696 --- /dev/null +++ b/thirdparties/android/include/theora/codec.h @@ -0,0 +1,591 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggTheora SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE Theora SOURCE CODE IS COPYRIGHT (C) 2002-2009 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: + last mod: $Id: theora.h,v 1.8 2004/03/15 22:17:32 derf Exp $ + + ********************************************************************/ + +/**\mainpage + * + * \section intro Introduction + * + * This is the documentation for <tt>libtheora</tt> C API. + * The current reference + * implementation for <a href="http://www.theora.org/">Theora</a>, a free, + * patent-unencumbered video codec. + * Theora is derived from On2's VP3 codec with additional features and + * integration with Ogg multimedia formats by + * <a href="http://www.xiph.org/">the Xiph.Org Foundation</a>. + * Complete documentation of the format itself is available in + * <a href="http://www.theora.org/doc/Theora.pdf">the Theora + * specification</a>. + * + * \subsection Organization + * + * The functions documented here are actually subdivided into three + * separate libraries: + * - <tt>libtheoraenc</tt> contains the encoder interface, + * described in \ref encfuncs. + * - <tt>libtheoradec</tt> contains the decoder interface and + * routines shared with the encoder. + * You must also link to this if you link to <tt>libtheoraenc</tt>. + * The routines in this library are described in \ref decfuncs and + * \ref basefuncs. + * - <tt>libtheora</tt> contains the \ref oldfuncs. + * + * New code should link to <tt>libtheoradec</tt> and, if using encoder + * features, <tt>libtheoraenc</tt>. Together these two export both + * the standard and the legacy API, so this is all that is needed by + * any code. The older <tt>libtheora</tt> library is provided just for + * compatibility with older build configurations. + * + * In general the recommended 1.x API symbols can be distinguished + * by their <tt>th_</tt> or <tt>TH_</tt> namespace prefix. + * The older, legacy API uses <tt>theora_</tt> or <tt>OC_</tt> + * prefixes instead. + */ + +/**\file + * The shared <tt>libtheoradec</tt> and <tt>libtheoraenc</tt> C API. + * You don't need to include this directly.*/ + +#if !defined(_O_THEORA_CODEC_H_) +# define _O_THEORA_CODEC_H_ (1) +# include <ogg/ogg.h> + +#if defined(__cplusplus) +extern "C" { +#endif + + + +/**\name Return codes*/ +/*@{*/ +/**An invalid pointer was provided.*/ +#define TH_EFAULT (-1) +/**An invalid argument was provided.*/ +#define TH_EINVAL (-10) +/**The contents of the header were incomplete, invalid, or unexpected.*/ +#define TH_EBADHEADER (-20) +/**The header does not belong to a Theora stream.*/ +#define TH_ENOTFORMAT (-21) +/**The bitstream version is too high.*/ +#define TH_EVERSION (-22) +/**The specified function is not implemented.*/ +#define TH_EIMPL (-23) +/**There were errors in the video data packet.*/ +#define TH_EBADPACKET (-24) +/**The decoded packet represented a dropped frame. + The player can continue to display the current frame, as the contents of the + decoded frame buffer have not changed.*/ +#define TH_DUPFRAME (1) +/*@}*/ + +/**The currently defined color space tags. + * See <a href="http://www.theora.org/doc/Theora.pdf">the Theora + * specification</a>, Chapter 4, for exact details on the meaning + * of each of these color spaces.*/ +typedef enum{ + /**The color space was not specified at the encoder. + It may be conveyed by an external means.*/ + TH_CS_UNSPECIFIED, + /**A color space designed for NTSC content.*/ + TH_CS_ITU_REC_470M, + /**A color space designed for PAL/SECAM content.*/ + TH_CS_ITU_REC_470BG, + /**The total number of currently defined color spaces.*/ + TH_CS_NSPACES +}th_colorspace; + +/**The currently defined pixel format tags. + * See <a href="http://www.theora.org/doc/Theora.pdf">the Theora + * specification</a>, Section 4.4, for details on the precise sample + * locations.*/ +typedef enum{ + /**Chroma decimation by 2 in both the X and Y directions (4:2:0). + The Cb and Cr chroma planes are half the width and half the + height of the luma plane.*/ + TH_PF_420, + /**Currently reserved.*/ + TH_PF_RSVD, + /**Chroma decimation by 2 in the X direction (4:2:2). + The Cb and Cr chroma planes are half the width of the luma plane, but full + height.*/ + TH_PF_422, + /**No chroma decimation (4:4:4). + The Cb and Cr chroma planes are full width and full height.*/ + TH_PF_444, + /**The total number of currently defined pixel formats.*/ + TH_PF_NFORMATS +}th_pixel_fmt; + + + +/**A buffer for a single color plane in an uncompressed image. + * This contains the image data in a left-to-right, top-down format. + * Each row of pixels is stored contiguously in memory, but successive + * rows need not be. + * Use \a stride to compute the offset of the next row. + * The encoder accepts both positive \a stride values (top-down in memory) + * and negative (bottom-up in memory). + * The decoder currently always generates images with positive strides.*/ +typedef struct{ + /**The width of this plane.*/ + int width; + /**The height of this plane.*/ + int height; + /**The offset in bytes between successive rows.*/ + int stride; + /**A pointer to the beginning of the first row.*/ + unsigned char *data; +}th_img_plane; + +/**A complete image buffer for an uncompressed frame. + * The chroma planes may be decimated by a factor of two in either + * direction, as indicated by th_info#pixel_fmt. + * The width and height of the Y' plane must be multiples of 16. + * They may need to be cropped for display, using the rectangle + * specified by th_info#pic_x, th_info#pic_y, th_info#pic_width, + * and th_info#pic_height. + * All samples are 8 bits. + * \note The term YUV often used to describe a colorspace is ambiguous. + * The exact parameters of the RGB to YUV conversion process aside, in + * many contexts the U and V channels actually have opposite meanings. + * To avoid this confusion, we are explicit: the name of the color + * channels are Y'CbCr, and they appear in that order, always. + * The prime symbol denotes that the Y channel is non-linear. + * Cb and Cr stand for "Chroma blue" and "Chroma red", respectively.*/ +typedef th_img_plane th_ycbcr_buffer[3]; + +/**Theora bitstream information. + * This contains the basic playback parameters for a stream, and corresponds to + * the initial 'info' header packet. + * To initialize an encoder, the application fills in this structure and + * passes it to th_encode_alloc(). + * A default encoding mode is chosen based on the values of the #quality and + * #target_bitrate fields. + * On decode, it is filled in by th_decode_headerin(), and then passed to + * th_decode_alloc(). + * + * Encoded Theora frames must be a multiple of 16 in size; + * this is what the #frame_width and #frame_height members represent. + * To handle arbitrary picture sizes, a crop rectangle is specified in the + * #pic_x, #pic_y, #pic_width and #pic_height members. + * + * All frame buffers contain pointers to the full, padded frame. + * However, the current encoder <em>will not</em> reference pixels outside of + * the cropped picture region, and the application does not need to fill them + * in. + * The decoder <em>will</em> allocate storage for a full frame, but the + * application <em>should not</em> rely on the padding containing sensible + * data. + * + * It is also generally recommended that the offsets and sizes should still be + * multiples of 2 to avoid chroma sampling shifts when chroma is sub-sampled. + * See <a href="http://www.theora.org/doc/Theora.pdf">the Theora + * specification</a>, Section 4.4, for more details. + * + * Frame rate, in frames per second, is stored as a rational fraction, as is + * the pixel aspect ratio. + * Note that this refers to the aspect ratio of the individual pixels, not of + * the overall frame itself. + * The frame aspect ratio can be computed from pixel aspect ratio using the + * image dimensions.*/ +typedef struct{ + /**\name Theora version + * Bitstream version information.*/ + /*@{*/ + unsigned char version_major; + unsigned char version_minor; + unsigned char version_subminor; + /*@}*/ + /**The encoded frame width. + * This must be a multiple of 16, and less than 1048576.*/ + ogg_uint32_t frame_width; + /**The encoded frame height. + * This must be a multiple of 16, and less than 1048576.*/ + ogg_uint32_t frame_height; + /**The displayed picture width. + * This must be no larger than width.*/ + ogg_uint32_t pic_width; + /**The displayed picture height. + * This must be no larger than height.*/ + ogg_uint32_t pic_height; + /**The X offset of the displayed picture. + * This must be no larger than #frame_width-#pic_width or 255, whichever is + * smaller.*/ + ogg_uint32_t pic_x; + /**The Y offset of the displayed picture. + * This must be no larger than #frame_height-#pic_height, and + * #frame_height-#pic_height-#pic_y must be no larger than 255. + * This slightly funny restriction is due to the fact that the offset is + * specified from the top of the image for consistency with the standard + * graphics left-handed coordinate system used throughout this API, while + * it is stored in the encoded stream as an offset from the bottom.*/ + ogg_uint32_t pic_y; + /**\name Frame rate + * The frame rate, as a fraction. + * If either is 0, the frame rate is undefined.*/ + /*@{*/ + ogg_uint32_t fps_numerator; + ogg_uint32_t fps_denominator; + /*@}*/ + /**\name Aspect ratio + * The aspect ratio of the pixels. + * If either value is zero, the aspect ratio is undefined. + * If not specified by any external means, 1:1 should be assumed. + * The aspect ratio of the full picture can be computed as + * \code + * aspect_numerator*pic_width/(aspect_denominator*pic_height). + * \endcode */ + /*@{*/ + ogg_uint32_t aspect_numerator; + ogg_uint32_t aspect_denominator; + /*@}*/ + /**The color space.*/ + th_colorspace colorspace; + /**The pixel format.*/ + th_pixel_fmt pixel_fmt; + /**The target bit-rate in bits per second. + If initializing an encoder with this struct, set this field to a non-zero + value to activate CBR encoding by default.*/ + int target_bitrate; + /**The target quality level. + Valid values range from 0 to 63, inclusive, with higher values giving + higher quality. + If initializing an encoder with this struct, and #target_bitrate is set + to zero, VBR encoding at this quality will be activated by default.*/ + /*Currently this is set so that a qi of 0 corresponds to distortions of 24 + times the JND, and each increase by 16 halves that value. + This gives us fine discrimination at low qualities, yet effective rate + control at high qualities. + The qi value 63 is special, however. + For this, the highest quality, we use one half of a JND for our threshold. + Due to the lower bounds placed on allowable quantizers in Theora, we will + not actually be able to achieve quality this good, but this should + provide as close to visually lossless quality as Theora is capable of. + We could lift the quantizer restrictions without breaking VP3.1 + compatibility, but this would result in quantized coefficients that are + too large for the current bitstream to be able to store. + We'd have to redesign the token syntax to store these large coefficients, + which would make transcoding complex.*/ + int quality; + /**The amount to shift to extract the last keyframe number from the granule + * position. + * This can be at most 31. + * th_info_init() will set this to a default value (currently <tt>6</tt>, + * which is good for streaming applications), but you can set it to 0 to + * make every frame a keyframe. + * The maximum distance between key frames is + * <tt>1<<#keyframe_granule_shift</tt>. + * The keyframe frequency can be more finely controlled with + * #TH_ENCCTL_SET_KEYFRAME_FREQUENCY_FORCE, which can also be adjusted + * during encoding (for example, to force the next frame to be a keyframe), + * but it cannot be set larger than the amount permitted by this field after + * the headers have been output.*/ + int keyframe_granule_shift; +}th_info; + +/**The comment information. + * + * This structure holds the in-stream metadata corresponding to + * the 'comment' header packet. + * The comment header is meant to be used much like someone jotting a quick + * note on the label of a video. + * It should be a short, to the point text note that can be more than a couple + * words, but not more than a short paragraph. + * + * The metadata is stored as a series of (tag, value) pairs, in + * length-encoded string vectors. + * The first occurrence of the '=' character delimits the tag and value. + * A particular tag may occur more than once, and order is significant. + * The character set encoding for the strings is always UTF-8, but the tag + * names are limited to ASCII, and treated as case-insensitive. + * See <a href="http://www.theora.org/doc/Theora.pdf">the Theora + * specification</a>, Section 6.3.3 for details. + * + * In filling in this structure, th_decode_headerin() will null-terminate + * the user_comment strings for safety. + * However, the bitstream format itself treats them as 8-bit clean vectors, + * possibly containing null characters, and so the length array should be + * treated as their authoritative length. + */ +typedef struct th_comment{ + /**The array of comment string vectors.*/ + char **user_comments; + /**An array of the corresponding length of each vector, in bytes.*/ + int *comment_lengths; + /**The total number of comment strings.*/ + int comments; + /**The null-terminated vendor string. + This identifies the software used to encode the stream.*/ + char *vendor; +}th_comment; + + + +/**A single base matrix.*/ +typedef unsigned char th_quant_base[64]; + +/**A set of \a qi ranges.*/ +typedef struct{ + /**The number of ranges in the set.*/ + int nranges; + /**The size of each of the #nranges ranges. + These must sum to 63.*/ + const int *sizes; + /**#nranges <tt>+1</tt> base matrices. + Matrices \a i and <tt>i+1</tt> form the endpoints of range \a i.*/ + const th_quant_base *base_matrices; +}th_quant_ranges; + +/**A complete set of quantization parameters. + The quantizer for each coefficient is calculated as: + \code + Q=MAX(MIN(qmin[qti][ci!=0],scale[ci!=0][qi]*base[qti][pli][qi][ci]/100), + 1024). + \endcode + + \a qti is the quantization type index: 0 for intra, 1 for inter. + <tt>ci!=0</tt> is 0 for the DC coefficient and 1 for AC coefficients. + \a qi is the quality index, ranging between 0 (low quality) and 63 (high + quality). + \a pli is the color plane index: 0 for Y', 1 for Cb, 2 for Cr. + \a ci is the DCT coefficient index. + Coefficient indices correspond to the normal 2D DCT block + ordering--row-major with low frequencies first--\em not zig-zag order. + + Minimum quantizers are constant, and are given by: + \code + qmin[2][2]={{4,2},{8,4}}. + \endcode + + Parameters that can be stored in the bitstream are as follows: + - The two scale matrices ac_scale and dc_scale. + \code + scale[2][64]={dc_scale,ac_scale}. + \endcode + - The base matrices for each \a qi, \a qti and \a pli (up to 384 in all). + In order to avoid storing a full 384 base matrices, only a sparse set of + matrices are stored, and the rest are linearly interpolated. + This is done as follows. + For each \a qti and \a pli, a series of \a n \a qi ranges is defined. + The size of each \a qi range can vary arbitrarily, but they must sum to + 63. + Then, <tt>n+1</tt> matrices are specified, one for each endpoint of the + ranges. + For interpolation purposes, each range's endpoints are the first \a qi + value it contains and one past the last \a qi value it contains. + Fractional values are rounded to the nearest integer, with ties rounded + away from zero. + + Base matrices are stored by reference, so if the same matrices are used + multiple times, they will only appear once in the bitstream. + The bitstream is also capable of omitting an entire set of ranges and + its associated matrices if they are the same as either the previous + set (indexed in row-major order) or if the inter set is the same as the + intra set. + + - Loop filter limit values. + The same limits are used for the loop filter in all color planes, despite + potentially differing levels of quantization in each. + + For the current encoder, <tt>scale[ci!=0][qi]</tt> must be no greater + than <tt>scale[ci!=0][qi-1]</tt> and <tt>base[qti][pli][qi][ci]</tt> must + be no greater than <tt>base[qti][pli][qi-1][ci]</tt>. + These two conditions ensure that the actual quantizer for a given \a qti, + \a pli, and \a ci does not increase as \a qi increases. + This is not required by the decoder.*/ +typedef struct{ + /**The DC scaling factors.*/ + ogg_uint16_t dc_scale[64]; + /**The AC scaling factors.*/ + ogg_uint16_t ac_scale[64]; + /**The loop filter limit values.*/ + unsigned char loop_filter_limits[64]; + /**The \a qi ranges for each \a ci and \a pli.*/ + th_quant_ranges qi_ranges[2][3]; +}th_quant_info; + + + +/**The number of Huffman tables used by Theora.*/ +#define TH_NHUFFMAN_TABLES (80) +/**The number of DCT token values in each table.*/ +#define TH_NDCT_TOKENS (32) + +/**A Huffman code for a Theora DCT token. + * Each set of Huffman codes in a given table must form a complete, prefix-free + * code. + * There is no requirement that all the tokens in a table have a valid code, + * but the current encoder is not optimized to take advantage of this. + * If each of the five grouops of 16 tables does not contain at least one table + * with a code for every token, then the encoder may fail to encode certain + * frames. + * The complete table in the first group of 16 does not have to be in the same + * place as the complete table in the other groups, but the complete tables in + * the remaining four groups must all be in the same place.*/ +typedef struct{ + /**The bit pattern for the code, with the LSbit of the pattern aligned in + * the LSbit of the word.*/ + ogg_uint32_t pattern; + /**The number of bits in the code. + * This must be between 0 and 32, inclusive.*/ + int nbits; +}th_huff_code; + + + +/**\defgroup basefuncs Functions Shared by Encode and Decode*/ +/*@{*/ +/**\name Basic shared functions*/ +/*@{*/ +/**Retrieves a human-readable string to identify the library vendor and + * version. + * \return the version string.*/ +extern const char *th_version_string(void); +/**Retrieves the library version number. + * This is the highest bitstream version that the encoder library will produce, + * or that the decoder library can decode. + * This number is composed of a 16-bit major version, 8-bit minor version + * and 8 bit sub-version, composed as follows: + * \code + * (VERSION_MAJOR<<16)+(VERSION_MINOR<<8)+(VERSION_SUBMINOR) + * \endcode + * \return the version number.*/ +extern ogg_uint32_t th_version_number(void); +/**Converts a granule position to an absolute frame index, starting at + * <tt>0</tt>. + * The granule position is interpreted in the context of a given + * #th_enc_ctx or #th_dec_ctx handle (either will suffice). + * \param _encdec A previously allocated #th_enc_ctx or #th_dec_ctx + * handle. + * \param _granpos The granule position to convert. + * \returns The absolute frame index corresponding to \a _granpos. + * \retval -1 The given granule position was invalid (i.e. negative).*/ +extern ogg_int64_t th_granule_frame(void *_encdec,ogg_int64_t _granpos); +/**Converts a granule position to an absolute time in seconds. + * The granule position is interpreted in the context of a given + * #th_enc_ctx or #th_dec_ctx handle (either will suffice). + * \param _encdec A previously allocated #th_enc_ctx or #th_dec_ctx + * handle. + * \param _granpos The granule position to convert. + * \return The absolute time in seconds corresponding to \a _granpos. + * This is the "end time" for the frame, or the latest time it should + * be displayed. + * It is not the presentation time. + * \retval -1 The given granule position was invalid (i.e. negative).*/ +extern double th_granule_time(void *_encdec,ogg_int64_t _granpos); +/**Determines whether a Theora packet is a header or not. + * This function does no verification beyond checking the packet type bit, so + * it should not be used for bitstream identification; use + * th_decode_headerin() for that. + * As per the Theora specification, an empty (0-byte) packet is treated as a + * data packet (a delta frame with no coded blocks). + * \param _op An <tt>ogg_packet</tt> containing encoded Theora data. + * \retval 1 The packet is a header packet + * \retval 0 The packet is a video data packet.*/ +extern int th_packet_isheader(ogg_packet *_op); +/**Determines whether a theora packet is a key frame or not. + * This function does no verification beyond checking the packet type and + * key frame bits, so it should not be used for bitstream identification; use + * th_decode_headerin() for that. + * As per the Theora specification, an empty (0-byte) packet is treated as a + * delta frame (with no coded blocks). + * \param _op An <tt>ogg_packet</tt> containing encoded Theora data. + * \retval 1 The packet contains a key frame. + * \retval 0 The packet contains a delta frame. + * \retval -1 The packet is not a video data packet.*/ +extern int th_packet_iskeyframe(ogg_packet *_op); +/*@}*/ + + +/**\name Functions for manipulating header data*/ +/*@{*/ +/**Initializes a th_info structure. + * This should be called on a freshly allocated #th_info structure before + * attempting to use it. + * \param _info The #th_info struct to initialize.*/ +extern void th_info_init(th_info *_info); +/**Clears a #th_info structure. + * This should be called on a #th_info structure after it is no longer + * needed. + * \param _info The #th_info struct to clear.*/ +extern void th_info_clear(th_info *_info); + +/**Initialize a #th_comment structure. + * This should be called on a freshly allocated #th_comment structure + * before attempting to use it. + * \param _tc The #th_comment struct to initialize.*/ +extern void th_comment_init(th_comment *_tc); +/**Add a comment to an initialized #th_comment structure. + * \note Neither th_comment_add() nor th_comment_add_tag() support + * comments containing null values, although the bitstream format does + * support them. + * To add such comments you will need to manipulate the #th_comment + * structure directly. + * \param _tc The #th_comment struct to add the comment to. + * \param _comment Must be a null-terminated UTF-8 string containing the + * comment in "TAG=the value" form.*/ +extern void th_comment_add(th_comment *_tc, char *_comment); +/**Add a comment to an initialized #th_comment structure. + * \note Neither th_comment_add() nor th_comment_add_tag() support + * comments containing null values, although the bitstream format does + * support them. + * To add such comments you will need to manipulate the #th_comment + * structure directly. + * \param _tc The #th_comment struct to add the comment to. + * \param _tag A null-terminated string containing the tag associated with + * the comment. + * \param _val The corresponding value as a null-terminated string.*/ +extern void th_comment_add_tag(th_comment *_tc,char *_tag,char *_val); +/**Look up a comment value by its tag. + * \param _tc An initialized #th_comment structure. + * \param _tag The tag to look up. + * \param _count The instance of the tag. + * The same tag can appear multiple times, each with a distinct + * value, so an index is required to retrieve them all. + * The order in which these values appear is significant and + * should be preserved. + * Use th_comment_query_count() to get the legal range for + * the \a _count parameter. + * \return A pointer to the queried tag's value. + * This points directly to data in the #th_comment structure. + * It should not be modified or freed by the application, and + * modifications to the structure may invalidate the pointer. + * \retval NULL If no matching tag is found.*/ +extern char *th_comment_query(th_comment *_tc,char *_tag,int _count); +/**Look up the number of instances of a tag. + * Call this first when querying for a specific tag and then iterate over the + * number of instances with separate calls to th_comment_query() to + * retrieve all the values for that tag in order. + * \param _tc An initialized #th_comment structure. + * \param _tag The tag to look up. + * \return The number on instances of this particular tag.*/ +extern int th_comment_query_count(th_comment *_tc,char *_tag); +/**Clears a #th_comment structure. + * This should be called on a #th_comment structure after it is no longer + * needed. + * It will free all memory used by the structure members. + * \param _tc The #th_comment struct to clear.*/ +extern void th_comment_clear(th_comment *_tc); +/*@}*/ +/*@}*/ + + + +#if defined(__cplusplus) +} +#endif + +#endif diff --git a/thirdparties/android/include/theora/theora.h b/thirdparties/android/include/theora/theora.h new file mode 100644 index 0000000..575a889 --- /dev/null +++ b/thirdparties/android/include/theora/theora.h @@ -0,0 +1,784 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggTheora SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE Theora SOURCE CODE IS COPYRIGHT (C) 2002-2009 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: + last mod: $Id: theora.h,v 1.17 2003/12/06 18:06:19 arc Exp $ + + ********************************************************************/ + +#ifndef _O_THEORA_H_ +#define _O_THEORA_H_ + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +#include <stddef.h> /* for size_t */ + +#include <ogg/ogg.h> + +/** \file + * The libtheora pre-1.0 legacy C API. + * + * \ingroup oldfuncs + * + * \section intro Introduction + * + * This is the documentation for the libtheora legacy C API, declared in + * the theora.h header, which describes the old interface used before + * the 1.0 release. This API was widely deployed for several years and + * remains supported, but for new code we recommend the cleaner API + * declared in theoradec.h and theoraenc.h. + * + * libtheora is the reference implementation for + * <a href="http://www.theora.org/">Theora</a>, a free video codec. + * Theora is derived from On2's VP3 codec with improved integration with + * Ogg multimedia formats by <a href="http://www.xiph.org/">Xiph.Org</a>. + * + * \section overview Overview + * + * This library will both decode and encode theora packets to/from raw YUV + * frames. In either case, the packets will most likely either come from or + * need to be embedded in an Ogg stream. Use + * <a href="http://xiph.org/ogg/">libogg</a> or + * <a href="http://www.annodex.net/software/liboggz/index.html">liboggz</a> + * to extract/package these packets. + * + * \section decoding Decoding Process + * + * Decoding can be separated into the following steps: + * -# initialise theora_info and theora_comment structures using + * theora_info_init() and theora_comment_init(): + \verbatim + theora_info info; + theora_comment comment; + + theora_info_init(&info); + theora_comment_init(&comment); + \endverbatim + * -# retrieve header packets from Ogg stream (there should be 3) and decode + * into theora_info and theora_comment structures using + * theora_decode_header(). See \ref identification for more information on + * identifying which packets are theora packets. + \verbatim + int i; + for (i = 0; i < 3; i++) + { + (get a theora packet "op" from the Ogg stream) + theora_decode_header(&info, &comment, op); + } + \endverbatim + * -# initialise the decoder based on the information retrieved into the + * theora_info struct by theora_decode_header(). You will need a + * theora_state struct. + \verbatim + theora_state state; + + theora_decode_init(&state, &info); + \endverbatim + * -# pass in packets and retrieve decoded frames! See the yuv_buffer + * documentation for information on how to retrieve raw YUV data. + \verbatim + yuf_buffer buffer; + while (last packet was not e_o_s) { + (get a theora packet "op" from the Ogg stream) + theora_decode_packetin(&state, op); + theora_decode_YUVout(&state, &buffer); + } + \endverbatim + * + * + * \subsection identification Identifying Theora Packets + * + * All streams inside an Ogg file have a unique serial_no attached to the + * stream. Typically, you will want to + * - retrieve the serial_no for each b_o_s (beginning of stream) page + * encountered within the Ogg file; + * - test the first (only) packet on that page to determine if it is a theora + * packet; + * - once you have found a theora b_o_s page then use the retrieved serial_no + * to identify future packets belonging to the same theora stream. + * + * Note that you \e cannot use theora_packet_isheader() to determine if a + * packet is a theora packet or not, as this function does not perform any + * checking beyond whether a header bit is present. Instead, use the + * theora_decode_header() function and check the return value; or examine the + * header bytes at the beginning of the Ogg page. + */ + + +/** \defgroup oldfuncs Legacy pre-1.0 C API */ +/* @{ */ + +/** + * A YUV buffer for passing uncompressed frames to and from the codec. + * This holds a Y'CbCr frame in planar format. The CbCr planes can be + * subsampled and have their own separate dimensions and row stride + * offsets. Note that the strides may be negative in some + * configurations. For theora the width and height of the largest plane + * must be a multiple of 16. The actual meaningful picture size and + * offset are stored in the theora_info structure; frames returned by + * the decoder may need to be cropped for display. + * + * All samples are 8 bits. Within each plane samples are ordered by + * row from the top of the frame to the bottom. Within each row samples + * are ordered from left to right. + * + * During decode, the yuv_buffer struct is allocated by the user, but all + * fields (including luma and chroma pointers) are filled by the library. + * These pointers address library-internal memory and their contents should + * not be modified. + * + * Conversely, during encode the user allocates the struct and fills out all + * fields. The user also manages the data addressed by the luma and chroma + * pointers. See the encoder_example.c and dump_video.c example files in + * theora/examples/ for more information. + */ +typedef struct { + int y_width; /**< Width of the Y' luminance plane */ + int y_height; /**< Height of the luminance plane */ + int y_stride; /**< Offset in bytes between successive rows */ + + int uv_width; /**< Width of the Cb and Cr chroma planes */ + int uv_height; /**< Height of the chroma planes */ + int uv_stride; /**< Offset between successive chroma rows */ + unsigned char *y; /**< Pointer to start of luminance data */ + unsigned char *u; /**< Pointer to start of Cb data */ + unsigned char *v; /**< Pointer to start of Cr data */ + +} yuv_buffer; + +/** + * A Colorspace. + */ +typedef enum { + OC_CS_UNSPECIFIED, /**< The colorspace is unknown or unspecified */ + OC_CS_ITU_REC_470M, /**< This is the best option for 'NTSC' content */ + OC_CS_ITU_REC_470BG, /**< This is the best option for 'PAL' content */ + OC_CS_NSPACES /**< This marks the end of the defined colorspaces */ +} theora_colorspace; + +/** + * A Chroma subsampling + * + * These enumerate the available chroma subsampling options supported + * by the theora format. See Section 4.4 of the specification for + * exact definitions. + */ +typedef enum { + OC_PF_420, /**< Chroma subsampling by 2 in each direction (4:2:0) */ + OC_PF_RSVD, /**< Reserved value */ + OC_PF_422, /**< Horizonatal chroma subsampling by 2 (4:2:2) */ + OC_PF_444 /**< No chroma subsampling at all (4:4:4) */ +} theora_pixelformat; + +/** + * Theora bitstream info. + * Contains the basic playback parameters for a stream, + * corresponding to the initial 'info' header packet. + * + * Encoded theora frames must be a multiple of 16 in width and height. + * To handle other frame sizes, a crop rectangle is specified in + * frame_height and frame_width, offset_x and * offset_y. The offset + * and size should still be a multiple of 2 to avoid chroma sampling + * shifts. Offset values in this structure are measured from the + * upper left of the image. + * + * Frame rate, in frames per second, is stored as a rational + * fraction. Aspect ratio is also stored as a rational fraction, and + * refers to the aspect ratio of the frame pixels, not of the + * overall frame itself. + * + * See <a href="http://svn.xiph.org/trunk/theora/examples/encoder_example.c"> + * examples/encoder_example.c</a> for usage examples of the + * other paramters and good default settings for the encoder parameters. + */ +typedef struct { + ogg_uint32_t width; /**< encoded frame width */ + ogg_uint32_t height; /**< encoded frame height */ + ogg_uint32_t frame_width; /**< display frame width */ + ogg_uint32_t frame_height; /**< display frame height */ + ogg_uint32_t offset_x; /**< horizontal offset of the displayed frame */ + ogg_uint32_t offset_y; /**< vertical offset of the displayed frame */ + ogg_uint32_t fps_numerator; /**< frame rate numerator **/ + ogg_uint32_t fps_denominator; /**< frame rate denominator **/ + ogg_uint32_t aspect_numerator; /**< pixel aspect ratio numerator */ + ogg_uint32_t aspect_denominator; /**< pixel aspect ratio denominator */ + theora_colorspace colorspace; /**< colorspace */ + int target_bitrate; /**< nominal bitrate in bits per second */ + int quality; /**< Nominal quality setting, 0-63 */ + int quick_p; /**< Quick encode/decode */ + + /* decode only */ + unsigned char version_major; + unsigned char version_minor; + unsigned char version_subminor; + + void *codec_setup; + + /* encode only */ + int dropframes_p; + int keyframe_auto_p; + ogg_uint32_t keyframe_frequency; + ogg_uint32_t keyframe_frequency_force; /* also used for decode init to + get granpos shift correct */ + ogg_uint32_t keyframe_data_target_bitrate; + ogg_int32_t keyframe_auto_threshold; + ogg_uint32_t keyframe_mindistance; + ogg_int32_t noise_sensitivity; + ogg_int32_t sharpness; + + theora_pixelformat pixelformat; /**< chroma subsampling mode to expect */ + +} theora_info; + +/** Codec internal state and context. + */ +typedef struct{ + theora_info *i; + ogg_int64_t granulepos; + + void *internal_encode; + void *internal_decode; + +} theora_state; + +/** + * Comment header metadata. + * + * This structure holds the in-stream metadata corresponding to + * the 'comment' header packet. + * + * Meta data is stored as a series of (tag, value) pairs, in + * length-encoded string vectors. The first occurence of the + * '=' character delimits the tag and value. A particular tag + * may occur more than once. The character set encoding for + * the strings is always UTF-8, but the tag names are limited + * to case-insensitive ASCII. See the spec for details. + * + * In filling in this structure, theora_decode_header() will + * null-terminate the user_comment strings for safety. However, + * the bitstream format itself treats them as 8-bit clean, + * and so the length array should be treated as authoritative + * for their length. + */ +typedef struct theora_comment{ + char **user_comments; /**< An array of comment string vectors */ + int *comment_lengths; /**< An array of corresponding string vector lengths in bytes */ + int comments; /**< The total number of comment string vectors */ + char *vendor; /**< The vendor string identifying the encoder, null terminated */ + +} theora_comment; + + +/**\name theora_control() codes */ +/* \anchor decctlcodes_old + * These are the available request codes for theora_control() + * when called with a decoder instance. + * By convention decoder control codes are odd, to distinguish + * them from \ref encctlcodes_old "encoder control codes" which + * are even. + * + * Note that since the 1.0 release, both the legacy and the final + * implementation accept all the same control codes, but only the + * final API declares the newer codes. + * + * Keep any experimental or vendor-specific values above \c 0x8000.*/ + +/*@{*/ + +/**Get the maximum post-processing level. + * The decoder supports a post-processing filter that can improve + * the appearance of the decoded images. This returns the highest + * level setting for this post-processor, corresponding to maximum + * improvement and computational expense. + */ +#define TH_DECCTL_GET_PPLEVEL_MAX (1) + +/**Set the post-processing level. + * Sets the level of post-processing to use when decoding the + * compressed stream. This must be a value between zero (off) + * and the maximum returned by TH_DECCTL_GET_PPLEVEL_MAX. + */ +#define TH_DECCTL_SET_PPLEVEL (3) + +/**Sets the maximum distance between key frames. + * This can be changed during an encode, but will be bounded by + * <tt>1<<th_info#keyframe_granule_shift</tt>. + * If it is set before encoding begins, th_info#keyframe_granule_shift will + * be enlarged appropriately. + * + * \param[in] buf <tt>ogg_uint32_t</tt>: The maximum distance between key + * frames. + * \param[out] buf <tt>ogg_uint32_t</tt>: The actual maximum distance set. + * \retval OC_FAULT \a theora_state or \a buf is <tt>NULL</tt>. + * \retval OC_EINVAL \a buf_sz is not <tt>sizeof(ogg_uint32_t)</tt>. + * \retval OC_IMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_KEYFRAME_FREQUENCY_FORCE (4) + +/**Set the granule position. + * Call this after a seek, to update the internal granulepos + * in the decoder, to insure that subsequent frames are marked + * properly. If you track timestamps yourself and do not use + * the granule postion returned by the decoder, then you do + * not need to use this control. + */ +#define TH_DECCTL_SET_GRANPOS (5) + +/**\anchor encctlcodes_old */ + +/**Sets the quantization parameters to use. + * The parameters are copied, not stored by reference, so they can be freed + * after this call. + * <tt>NULL</tt> may be specified to revert to the default parameters. + * + * \param[in] buf #th_quant_info + * \retval OC_FAULT \a theora_state is <tt>NULL</tt>. + * \retval OC_EINVAL Encoding has already begun, the quantization parameters + * are not acceptable to this version of the encoder, + * \a buf is <tt>NULL</tt> and \a buf_sz is not zero, + * or \a buf is non-<tt>NULL</tt> and \a buf_sz is + * not <tt>sizeof(#th_quant_info)</tt>. + * \retval OC_IMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_QUANT_PARAMS (2) + +/**Disables any encoder features that would prevent lossless transcoding back + * to VP3. + * This primarily means disabling block-level QI values and not using 4MV mode + * when any of the luma blocks in a macro block are not coded. + * It also includes using the VP3 quantization tables and Huffman codes; if you + * set them explicitly after calling this function, the resulting stream will + * not be VP3-compatible. + * If you enable VP3-compatibility when encoding 4:2:2 or 4:4:4 source + * material, or when using a picture region smaller than the full frame (e.g. + * a non-multiple-of-16 width or height), then non-VP3 bitstream features will + * still be disabled, but the stream will still not be VP3-compatible, as VP3 + * was not capable of encoding such formats. + * If you call this after encoding has already begun, then the quantization + * tables and codebooks cannot be changed, but the frame-level features will + * be enabled or disabled as requested. + * + * \param[in] buf <tt>int</tt>: a non-zero value to enable VP3 compatibility, + * or 0 to disable it (the default). + * \param[out] buf <tt>int</tt>: 1 if all bitstream features required for + * VP3-compatibility could be set, and 0 otherwise. + * The latter will be returned if the pixel format is not + * 4:2:0, the picture region is smaller than the full frame, + * or if encoding has begun, preventing the quantization + * tables and codebooks from being set. + * \retval OC_FAULT \a theora_state or \a buf is <tt>NULL</tt>. + * \retval OC_EINVAL \a buf_sz is not <tt>sizeof(int)</tt>. + * \retval OC_IMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_VP3_COMPATIBLE (10) + +/**Gets the maximum speed level. + * Higher speed levels favor quicker encoding over better quality per bit. + * Depending on the encoding mode, and the internal algorithms used, quality + * may actually improve, but in this case bitrate will also likely increase. + * In any case, overall rate/distortion performance will probably decrease. + * The maximum value, and the meaning of each value, may change depending on + * the current encoding mode (VBR vs. CQI, etc.). + * + * \param[out] buf int: The maximum encoding speed level. + * \retval OC_FAULT \a theora_state or \a buf is <tt>NULL</tt>. + * \retval OC_EINVAL \a buf_sz is not <tt>sizeof(int)</tt>. + * \retval OC_IMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_GET_SPLEVEL_MAX (12) + +/**Sets the speed level. + * By default a speed value of 1 is used. + * + * \param[in] buf int: The new encoding speed level. + * 0 is slowest, larger values use less CPU. + * \retval OC_FAULT \a theora_state or \a buf is <tt>NULL</tt>. + * \retval OC_EINVAL \a buf_sz is not <tt>sizeof(int)</tt>, or the + * encoding speed level is out of bounds. + * The maximum encoding speed level may be + * implementation- and encoding mode-specific, and can be + * obtained via #TH_ENCCTL_GET_SPLEVEL_MAX. + * \retval OC_IMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_SET_SPLEVEL (14) + +/*@}*/ + +#define OC_FAULT -1 /**< General failure */ +#define OC_EINVAL -10 /**< Library encountered invalid internal data */ +#define OC_DISABLED -11 /**< Requested action is disabled */ +#define OC_BADHEADER -20 /**< Header packet was corrupt/invalid */ +#define OC_NOTFORMAT -21 /**< Packet is not a theora packet */ +#define OC_VERSION -22 /**< Bitstream version is not handled */ +#define OC_IMPL -23 /**< Feature or action not implemented */ +#define OC_BADPACKET -24 /**< Packet is corrupt */ +#define OC_NEWPACKET -25 /**< Packet is an (ignorable) unhandled extension */ +#define OC_DUPFRAME 1 /**< Packet is a dropped frame */ + +/** + * Retrieve a human-readable string to identify the encoder vendor and version. + * \returns A version string. + */ +extern const char *theora_version_string(void); + +/** + * Retrieve a 32-bit version number. + * This number is composed of a 16-bit major version, 8-bit minor version + * and 8 bit sub-version, composed as follows: +<pre> + (VERSION_MAJOR<<16) + (VERSION_MINOR<<8) + (VERSION_SUB) +</pre> +* \returns The version number. +*/ +extern ogg_uint32_t theora_version_number(void); + +/** + * Initialize the theora encoder. + * \param th The theora_state handle to initialize for encoding. + * \param ti A theora_info struct filled with the desired encoding parameters. + * \retval 0 Success + */ +extern int theora_encode_init(theora_state *th, theora_info *ti); + +/** + * Submit a YUV buffer to the theora encoder. + * \param t A theora_state handle previously initialized for encoding. + * \param yuv A buffer of YUV data to encode. Note that both the yuv_buffer + * struct and the luma/chroma buffers within should be allocated by + * the user. + * \retval OC_EINVAL Encoder is not ready, or is finished. + * \retval -1 The size of the given frame differs from those previously input + * \retval 0 Success + */ +extern int theora_encode_YUVin(theora_state *t, yuv_buffer *yuv); + +/** + * Request the next packet of encoded video. + * The encoded data is placed in a user-provided ogg_packet structure. + * \param t A theora_state handle previously initialized for encoding. + * \param last_p whether this is the last packet the encoder should produce. + * \param op An ogg_packet structure to fill. libtheora will set all + * elements of this structure, including a pointer to encoded + * data. The memory for the encoded data is owned by libtheora. + * \retval 0 No internal storage exists OR no packet is ready + * \retval -1 The encoding process has completed + * \retval 1 Success + */ +extern int theora_encode_packetout( theora_state *t, int last_p, + ogg_packet *op); + +/** + * Request a packet containing the initial header. + * A pointer to the header data is placed in a user-provided ogg_packet + * structure. + * \param t A theora_state handle previously initialized for encoding. + * \param op An ogg_packet structure to fill. libtheora will set all + * elements of this structure, including a pointer to the header + * data. The memory for the header data is owned by libtheora. + * \retval 0 Success + */ +extern int theora_encode_header(theora_state *t, ogg_packet *op); + +/** + * Request a comment header packet from provided metadata. + * A pointer to the comment data is placed in a user-provided ogg_packet + * structure. + * \param tc A theora_comment structure filled with the desired metadata + * \param op An ogg_packet structure to fill. libtheora will set all + * elements of this structure, including a pointer to the encoded + * comment data. The memory for the comment data is owned by + * libtheora. + * \retval 0 Success + */ +extern int theora_encode_comment(theora_comment *tc, ogg_packet *op); + +/** + * Request a packet containing the codebook tables for the stream. + * A pointer to the codebook data is placed in a user-provided ogg_packet + * structure. + * \param t A theora_state handle previously initialized for encoding. + * \param op An ogg_packet structure to fill. libtheora will set all + * elements of this structure, including a pointer to the codebook + * data. The memory for the header data is owned by libtheora. + * \retval 0 Success + */ +extern int theora_encode_tables(theora_state *t, ogg_packet *op); + +/** + * Decode an Ogg packet, with the expectation that the packet contains + * an initial header, comment data or codebook tables. + * + * \param ci A theora_info structure to fill. This must have been previously + * initialized with theora_info_init(). If \a op contains an initial + * header, theora_decode_header() will fill \a ci with the + * parsed header values. If \a op contains codebook tables, + * theora_decode_header() will parse these and attach an internal + * representation to \a ci->codec_setup. + * \param cc A theora_comment structure to fill. If \a op contains comment + * data, theora_decode_header() will fill \a cc with the parsed + * comments. + * \param op An ogg_packet structure which you expect contains an initial + * header, comment data or codebook tables. + * + * \retval OC_BADHEADER \a op is NULL; OR the first byte of \a op->packet + * has the signature of an initial packet, but op is + * not a b_o_s packet; OR this packet has the signature + * of an initial header packet, but an initial header + * packet has already been seen; OR this packet has the + * signature of a comment packet, but the initial header + * has not yet been seen; OR this packet has the signature + * of a comment packet, but contains invalid data; OR + * this packet has the signature of codebook tables, + * but the initial header or comments have not yet + * been seen; OR this packet has the signature of codebook + * tables, but contains invalid data; + * OR the stream being decoded has a compatible version + * but this packet does not have the signature of a + * theora initial header, comments, or codebook packet + * \retval OC_VERSION The packet data of \a op is an initial header with + * a version which is incompatible with this version of + * libtheora. + * \retval OC_NEWPACKET the stream being decoded has an incompatible (future) + * version and contains an unknown signature. + * \retval 0 Success + * + * \note The normal usage is that theora_decode_header() be called on the + * first three packets of a theora logical bitstream in succession. + */ +extern int theora_decode_header(theora_info *ci, theora_comment *cc, + ogg_packet *op); + +/** + * Initialize a theora_state handle for decoding. + * \param th The theora_state handle to initialize. + * \param c A theora_info struct filled with the desired decoding parameters. + * This is of course usually obtained from a previous call to + * theora_decode_header(). + * \retval 0 Success + */ +extern int theora_decode_init(theora_state *th, theora_info *c); + +/** + * Input a packet containing encoded data into the theora decoder. + * \param th A theora_state handle previously initialized for decoding. + * \param op An ogg_packet containing encoded theora data. + * \retval 0 Success + * \retval OC_BADPACKET \a op does not contain encoded video data + */ +extern int theora_decode_packetin(theora_state *th,ogg_packet *op); + +/** + * Output the next available frame of decoded YUV data. + * \param th A theora_state handle previously initialized for decoding. + * \param yuv A yuv_buffer in which libtheora should place the decoded data. + * Note that the buffer struct itself is allocated by the user, but + * that the luma and chroma pointers will be filled in by the + * library. Also note that these luma and chroma regions should be + * considered read-only by the user. + * \retval 0 Success + */ +extern int theora_decode_YUVout(theora_state *th,yuv_buffer *yuv); + +/** + * Report whether a theora packet is a header or not + * This function does no verification beyond checking the header + * flag bit so it should not be used for bitstream identification; + * use theora_decode_header() for that. + * + * \param op An ogg_packet containing encoded theora data. + * \retval 1 The packet is a header packet + * \retval 0 The packet is not a header packet (and so contains frame data) + * + * Thus function was added in the 1.0alpha4 release. + */ +extern int theora_packet_isheader(ogg_packet *op); + +/** + * Report whether a theora packet is a keyframe or not + * + * \param op An ogg_packet containing encoded theora data. + * \retval 1 The packet contains a keyframe image + * \retval 0 The packet is contains an interframe delta + * \retval -1 The packet is not an image data packet at all + * + * Thus function was added in the 1.0alpha4 release. + */ +extern int theora_packet_iskeyframe(ogg_packet *op); + +/** + * Report the granulepos shift radix + * + * When embedded in Ogg, Theora uses a two-part granulepos, + * splitting the 64-bit field into two pieces. The more-significant + * section represents the frame count at the last keyframe, + * and the less-significant section represents the count of + * frames since the last keyframe. In this way the overall + * field is still non-decreasing with time, but usefully encodes + * a pointer to the last keyframe, which is necessary for + * correctly restarting decode after a seek. + * + * This function reports the number of bits used to represent + * the distance to the last keyframe, and thus how the granulepos + * field must be shifted or masked to obtain the two parts. + * + * Since libtheora returns compressed data in an ogg_packet + * structure, this may be generally useful even if the Theora + * packets are not being used in an Ogg container. + * + * \param ti A previously initialized theora_info struct + * \returns The bit shift dividing the two granulepos fields + * + * This function was added in the 1.0alpha5 release. + */ +int theora_granule_shift(theora_info *ti); + +/** + * Convert a granulepos to an absolute frame index, starting at 0. + * The granulepos is interpreted in the context of a given theora_state handle. + * + * Note that while the granulepos encodes the frame count (i.e. starting + * from 1) this call returns the frame index, starting from zero. Thus + * One can calculate the presentation time by multiplying the index by + * the rate. + * + * \param th A previously initialized theora_state handle (encode or decode) + * \param granulepos The granulepos to convert. + * \returns The frame index corresponding to \a granulepos. + * \retval -1 The given granulepos is undefined (i.e. negative) + * + * Thus function was added in the 1.0alpha4 release. + */ +extern ogg_int64_t theora_granule_frame(theora_state *th,ogg_int64_t granulepos); + +/** + * Convert a granulepos to absolute time in seconds. The granulepos is + * interpreted in the context of a given theora_state handle, and gives + * the end time of a frame's presentation as used in Ogg mux ordering. + * + * \param th A previously initialized theora_state handle (encode or decode) + * \param granulepos The granulepos to convert. + * \returns The absolute time in seconds corresponding to \a granulepos. + * This is the "end time" for the frame, or the latest time it should + * be displayed. + * It is not the presentation time. + * \retval -1. The given granulepos is undefined (i.e. negative), or + * \retval -1. The function has been disabled because floating + * point support is not available. + */ +extern double theora_granule_time(theora_state *th,ogg_int64_t granulepos); + +/** + * Initialize a theora_info structure. All values within the given theora_info + * structure are initialized, and space is allocated within libtheora for + * internal codec setup data. + * \param c A theora_info struct to initialize. + */ +extern void theora_info_init(theora_info *c); + +/** + * Clear a theora_info structure. All values within the given theora_info + * structure are cleared, and associated internal codec setup data is freed. + * \param c A theora_info struct to initialize. + */ +extern void theora_info_clear(theora_info *c); + +/** + * Free all internal data associated with a theora_state handle. + * \param t A theora_state handle. + */ +extern void theora_clear(theora_state *t); + +/** + * Initialize an allocated theora_comment structure + * \param tc An allocated theora_comment structure + **/ +extern void theora_comment_init(theora_comment *tc); + +/** + * Add a comment to an initialized theora_comment structure + * \param tc A previously initialized theora comment structure + * \param comment A null-terminated string encoding the comment in the form + * "TAG=the value" + * + * Neither theora_comment_add() nor theora_comment_add_tag() support + * comments containing null values, although the bitstream format + * supports this. To add such comments you will need to manipulate + * the theora_comment structure directly. + **/ + +extern void theora_comment_add(theora_comment *tc, char *comment); + +/** + * Add a comment to an initialized theora_comment structure. + * \param tc A previously initialized theora comment structure + * \param tag A null-terminated string containing the tag + * associated with the comment. + * \param value The corresponding value as a null-terminated string + * + * Neither theora_comment_add() nor theora_comment_add_tag() support + * comments containing null values, although the bitstream format + * supports this. To add such comments you will need to manipulate + * the theora_comment structure directly. + **/ +extern void theora_comment_add_tag(theora_comment *tc, + char *tag, char *value); + +/** + * Look up a comment value by tag. + * \param tc Tn initialized theora_comment structure + * \param tag The tag to look up + * \param count The instance of the tag. The same tag can appear multiple + * times, each with a distinct and ordered value, so an index + * is required to retrieve them all. + * \returns A pointer to the queried tag's value + * \retval NULL No matching tag is found + * + * \note Use theora_comment_query_count() to get the legal range for the + * count parameter. + **/ + +extern char *theora_comment_query(theora_comment *tc, char *tag, int count); + +/** Look up the number of instances of a tag. + * \param tc An initialized theora_comment structure + * \param tag The tag to look up + * \returns The number on instances of a particular tag. + * + * Call this first when querying for a specific tag and then interate + * over the number of instances with separate calls to + * theora_comment_query() to retrieve all instances in order. + **/ +extern int theora_comment_query_count(theora_comment *tc, char *tag); + +/** + * Clear an allocated theora_comment struct so that it can be freed. + * \param tc An allocated theora_comment structure. + **/ +extern void theora_comment_clear(theora_comment *tc); + +/**Encoder control function. + * This is used to provide advanced control the encoding process. + * \param th A #theora_state handle. + * \param req The control code to process. + * See \ref encctlcodes_old "the list of available + * control codes" for details. + * \param buf The parameters for this control code. + * \param buf_sz The size of the parameter buffer.*/ +extern int theora_control(theora_state *th,int req,void *buf,size_t buf_sz); + +/* @} */ /* end oldfuncs doxygen group */ + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif /* _O_THEORA_H_ */ diff --git a/thirdparties/android/include/theora/theoradec.h b/thirdparties/android/include/theora/theoradec.h new file mode 100644 index 0000000..b20f0e3 --- /dev/null +++ b/thirdparties/android/include/theora/theoradec.h @@ -0,0 +1,325 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggTheora SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE Theora SOURCE CODE IS COPYRIGHT (C) 2002-2009 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: + last mod: $Id: theora.h,v 1.8 2004/03/15 22:17:32 derf Exp $ + + ********************************************************************/ + +/**\file + * The <tt>libtheoradec</tt> C decoding API.*/ + +#if !defined(_O_THEORA_THEORADEC_H_) +# define _O_THEORA_THEORADEC_H_ (1) +# include <stddef.h> +# include <ogg/ogg.h> +# include "codec.h" + +#if defined(__cplusplus) +extern "C" { +#endif + + + +/**\name th_decode_ctl() codes + * \anchor decctlcodes + * These are the available request codes for th_decode_ctl(). + * By convention, these are odd, to distinguish them from the + * \ref encctlcodes "encoder control codes". + * Keep any experimental or vendor-specific values above \c 0x8000.*/ +/*@{*/ +/**Gets the maximum post-processing level. + * The decoder supports a post-processing filter that can improve + * the appearance of the decoded images. This returns the highest + * level setting for this post-processor, corresponding to maximum + * improvement and computational expense. + * + * \param[out] _buf int: The maximum post-processing level. + * \retval TH_EFAULT \a _dec_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_DECCTL_GET_PPLEVEL_MAX (1) +/**Sets the post-processing level. + * By default, post-processing is disabled. + * + * Sets the level of post-processing to use when decoding the + * compressed stream. This must be a value between zero (off) + * and the maximum returned by TH_DECCTL_GET_PPLEVEL_MAX. + * + * \param[in] _buf int: The new post-processing level. + * 0 to disable; larger values use more CPU. + * \retval TH_EFAULT \a _dec_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>, or the + * post-processing level is out of bounds. + * The maximum post-processing level may be + * implementation-specific, and can be obtained via + * #TH_DECCTL_GET_PPLEVEL_MAX. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_DECCTL_SET_PPLEVEL (3) +/**Sets the granule position. + * Call this after a seek, before decoding the first frame, to ensure that the + * proper granule position is returned for all subsequent frames. + * If you track timestamps yourself and do not use the granule position + * returned by the decoder, then you need not call this function. + * + * \param[in] _buf <tt>ogg_int64_t</tt>: The granule position of the next + * frame. + * \retval TH_EFAULT \a _dec_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(ogg_int64_t)</tt>, or the + * granule position is negative.*/ +#define TH_DECCTL_SET_GRANPOS (5) +/**Sets the striped decode callback function. + * If set, this function will be called as each piece of a frame is fully + * decoded in th_decode_packetin(). + * You can pass in a #th_stripe_callback with + * th_stripe_callback#stripe_decoded set to <tt>NULL</tt> to disable the + * callbacks at any point. + * Enabling striped decode does not prevent you from calling + * th_decode_ycbcr_out() after the frame is fully decoded. + * + * \param[in] _buf #th_stripe_callback: The callback parameters. + * \retval TH_EFAULT \a _dec_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not + * <tt>sizeof(th_stripe_callback)</tt>.*/ +#define TH_DECCTL_SET_STRIPE_CB (7) + +/**Enables telemetry and sets the macroblock display mode */ +#define TH_DECCTL_SET_TELEMETRY_MBMODE (9) +/**Enables telemetry and sets the motion vector display mode */ +#define TH_DECCTL_SET_TELEMETRY_MV (11) +/**Enables telemetry and sets the adaptive quantization display mode */ +#define TH_DECCTL_SET_TELEMETRY_QI (13) +/**Enables telemetry and sets the bitstream breakdown visualization mode */ +#define TH_DECCTL_SET_TELEMETRY_BITS (15) +/*@}*/ + + + +/**A callback function for striped decode. + * This is a function pointer to an application-provided function that will be + * called each time a section of the image is fully decoded in + * th_decode_packetin(). + * This allows the application to process the section immediately, while it is + * still in cache. + * Note that the frame is decoded bottom to top, so \a _yfrag0 will steadily + * decrease with each call until it reaches 0, at which point the full frame + * is decoded. + * The number of fragment rows made available in each call depends on the pixel + * format and the number of post-processing filters enabled, and may not even + * be constant for the entire frame. + * If a non-<tt>NULL</tt> \a _granpos pointer is passed to + * th_decode_packetin(), the granule position for the frame will be stored + * in it before the first callback is made. + * If an entire frame is dropped (a 0-byte packet), then no callbacks will be + * made at all for that frame. + * \param _ctx An application-provided context pointer. + * \param _buf The image buffer for the decoded frame. + * \param _yfrag0 The Y coordinate of the first row of 8x8 fragments + * decoded. + * Multiply this by 8 to obtain the pixel row number in the + * luma plane. + * If the chroma planes are subsampled in the Y direction, + * this will always be divisible by two. + * \param _yfrag_end The Y coordinate of the first row of 8x8 fragments past + * the newly decoded section. + * If the chroma planes are subsampled in the Y direction, + * this will always be divisible by two. + * I.e., this section contains fragment rows + * <tt>\a _yfrag0 ...\a _yfrag_end -1</tt>.*/ +typedef void (*th_stripe_decoded_func)(void *_ctx,th_ycbcr_buffer _buf, + int _yfrag0,int _yfrag_end); + +/**The striped decode callback data to pass to #TH_DECCTL_SET_STRIPE_CB.*/ +typedef struct{ + /**An application-provided context pointer. + * This will be passed back verbatim to the application.*/ + void *ctx; + /**The callback function pointer.*/ + th_stripe_decoded_func stripe_decoded; +}th_stripe_callback; + + + +/**\name Decoder state + The following data structures are opaque, and their contents are not + publicly defined by this API. + Referring to their internals directly is unsupported, and may break without + warning.*/ +/*@{*/ +/**The decoder context.*/ +typedef struct th_dec_ctx th_dec_ctx; +/**Setup information. + This contains auxiliary information (Huffman tables and quantization + parameters) decoded from the setup header by th_decode_headerin() to be + passed to th_decode_alloc(). + It can be re-used to initialize any number of decoders, and can be freed + via th_setup_free() at any time.*/ +typedef struct th_setup_info th_setup_info; +/*@}*/ + + + +/**\defgroup decfuncs Functions for Decoding*/ +/*@{*/ +/**\name Functions for decoding + * You must link to <tt>libtheoradec</tt> if you use any of the + * functions in this section. + * + * The functions are listed in the order they are used in a typical decode. + * The basic steps are: + * - Parse the header packets by repeatedly calling th_decode_headerin(). + * - Allocate a #th_dec_ctx handle with th_decode_alloc(). + * - Call th_setup_free() to free any memory used for codec setup + * information. + * - Perform any additional decoder configuration with th_decode_ctl(). + * - For each video data packet: + * - Submit the packet to the decoder via th_decode_packetin(). + * - Retrieve the uncompressed video data via th_decode_ycbcr_out(). + * - Call th_decode_free() to release all decoder memory.*/ +/*@{*/ +/**Decodes the header packets of a Theora stream. + * This should be called on the initial packets of the stream, in succession, + * until it returns <tt>0</tt>, indicating that all headers have been + * processed, or an error is encountered. + * At least three header packets are required, and additional optional header + * packets may follow. + * This can be used on the first packet of any logical stream to determine if + * that stream is a Theora stream. + * \param _info A #th_info structure to fill in. + * This must have been previously initialized with + * th_info_init(). + * The application may immediately begin using the contents of + * this structure after the first header is decoded, though it + * must continue to be passed in on all subsequent calls. + * \param _tc A #th_comment structure to fill in. + * The application may immediately begin using the contents of + * this structure after the second header is decoded, though it + * must continue to be passed in on all subsequent calls. + * \param _setup Returns a pointer to additional, private setup information + * needed by the decoder. + * The contents of this pointer must be initialized to + * <tt>NULL</tt> on the first call, and the returned value must + * continue to be passed in on all subsequent calls. + * \param _op An <tt>ogg_packet</tt> structure which contains one of the + * initial packets of an Ogg logical stream. + * \return A positive value indicates that a Theora header was successfully + * processed. + * \retval 0 The first video data packet was encountered after all + * required header packets were parsed. + * The packet just passed in on this call should be saved + * and fed to th_decode_packetin() to begin decoding + * video data. + * \retval TH_EFAULT One of \a _info, \a _tc, or \a _setup was + * <tt>NULL</tt>. + * \retval TH_EBADHEADER \a _op was <tt>NULL</tt>, the packet was not the next + * header packet in the expected sequence, or the format + * of the header data was invalid. + * \retval TH_EVERSION The packet data was a Theora info header, but for a + * bitstream version not decodable with this version of + * <tt>libtheoradec</tt>. + * \retval TH_ENOTFORMAT The packet was not a Theora header. + */ +extern int th_decode_headerin(th_info *_info,th_comment *_tc, + th_setup_info **_setup,ogg_packet *_op); +/**Allocates a decoder instance. + * + * <b>Security Warning:</b> The Theora format supports very large frame sizes, + * potentially even larger than the address space of a 32-bit machine, and + * creating a decoder context allocates the space for several frames of data. + * If the allocation fails here, your program will crash, possibly at some + * future point because the OS kernel returned a valid memory range and will + * only fail when it tries to map the pages in it the first time they are + * used. + * Even if it succeeds, you may experience a denial of service if the frame + * size is large enough to cause excessive paging. + * If you are integrating libtheora in a larger application where such things + * are undesirable, it is highly recommended that you check the frame size in + * \a _info before calling this function and refuse to decode streams where it + * is larger than some reasonable maximum. + * libtheora will not check this for you, because there may be machines that + * can handle such streams and applications that wish to. + * \param _info A #th_info struct filled via th_decode_headerin(). + * \param _setup A #th_setup_info handle returned via + * th_decode_headerin(). + * \return The initialized #th_dec_ctx handle. + * \retval NULL If the decoding parameters were invalid.*/ +extern th_dec_ctx *th_decode_alloc(const th_info *_info, + const th_setup_info *_setup); +/**Releases all storage used for the decoder setup information. + * This should be called after you no longer want to create any decoders for + * a stream whose headers you have parsed with th_decode_headerin(). + * \param _setup The setup information to free. + * This can safely be <tt>NULL</tt>.*/ +extern void th_setup_free(th_setup_info *_setup); +/**Decoder control function. + * This is used to provide advanced control of the decoding process. + * \param _dec A #th_dec_ctx handle. + * \param _req The control code to process. + * See \ref decctlcodes "the list of available control codes" + * for details. + * \param _buf The parameters for this control code. + * \param _buf_sz The size of the parameter buffer.*/ +extern int th_decode_ctl(th_dec_ctx *_dec,int _req,void *_buf, + size_t _buf_sz); +/**Submits a packet containing encoded video data to the decoder. + * \param _dec A #th_dec_ctx handle. + * \param _op An <tt>ogg_packet</tt> containing encoded video data. + * \param _granpos Returns the granule position of the decoded packet. + * If non-<tt>NULL</tt>, the granule position for this specific + * packet is stored in this location. + * This is computed incrementally from previously decoded + * packets. + * After a seek, the correct granule position must be set via + * #TH_DECCTL_SET_GRANPOS for this to work properly. + * \retval 0 Success. + * A new decoded frame can be retrieved by calling + * th_decode_ycbcr_out(). + * \retval TH_DUPFRAME The packet represented a dropped (0-byte) frame. + * The player can skip the call to th_decode_ycbcr_out(), + * as the contents of the decoded frame buffer have not + * changed. + * \retval TH_EFAULT \a _dec or \a _op was <tt>NULL</tt>. + * \retval TH_EBADPACKET \a _op does not contain encoded video data. + * \retval TH_EIMPL The video data uses bitstream features which this + * library does not support.*/ +extern int th_decode_packetin(th_dec_ctx *_dec,const ogg_packet *_op, + ogg_int64_t *_granpos); +/**Outputs the next available frame of decoded Y'CbCr data. + * If a striped decode callback has been set with #TH_DECCTL_SET_STRIPE_CB, + * then the application does not need to call this function. + * \param _dec A #th_dec_ctx handle. + * \param _ycbcr A video buffer structure to fill in. + * <tt>libtheoradec</tt> will fill in all the members of this + * structure, including the pointers to the uncompressed video + * data. + * The memory for this video data is owned by + * <tt>libtheoradec</tt>. + * It may be freed or overwritten without notification when + * subsequent frames are decoded. + * \retval 0 Success + * \retval TH_EFAULT \a _dec or \a _ycbcr was <tt>NULL</tt>. + */ +extern int th_decode_ycbcr_out(th_dec_ctx *_dec, + th_ycbcr_buffer _ycbcr); +/**Frees an allocated decoder instance. + * \param _dec A #th_dec_ctx handle.*/ +extern void th_decode_free(th_dec_ctx *_dec); +/*@}*/ +/*@}*/ + + + +#if defined(__cplusplus) +} +#endif + +#endif diff --git a/thirdparties/android/include/theora/theoraenc.h b/thirdparties/android/include/theora/theoraenc.h new file mode 100644 index 0000000..f54b234 --- /dev/null +++ b/thirdparties/android/include/theora/theoraenc.h @@ -0,0 +1,500 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggTheora SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE Theora SOURCE CODE IS COPYRIGHT (C) 2002-2009 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: + last mod: $Id: theora.h,v 1.8 2004/03/15 22:17:32 derf Exp $ + + ********************************************************************/ + +/**\file + * The <tt>libtheoraenc</tt> C encoding API.*/ + +#if !defined(_O_THEORA_THEORAENC_H_) +# define _O_THEORA_THEORAENC_H_ (1) +# include <stddef.h> +# include <ogg/ogg.h> +# include "codec.h" + +#if defined(__cplusplus) +extern "C" { +#endif + + + +/**\name th_encode_ctl() codes + * \anchor encctlcodes + * These are the available request codes for th_encode_ctl(). + * By convention, these are even, to distinguish them from the + * \ref decctlcodes "decoder control codes". + * Keep any experimental or vendor-specific values above \c 0x8000.*/ +/*@{*/ +/**Sets the Huffman tables to use. + * The tables are copied, not stored by reference, so they can be freed after + * this call. + * <tt>NULL</tt> may be specified to revert to the default tables. + * + * \param[in] _buf <tt>#th_huff_code[#TH_NHUFFMAN_TABLES][#TH_NDCT_TOKENS]</tt> + * \retval TH_EFAULT \a _enc_ctx is <tt>NULL</tt>. + * \retval TH_EINVAL Encoding has already begun or one or more of the given + * tables is not full or prefix-free, \a _buf is + * <tt>NULL</tt> and \a _buf_sz is not zero, or \a _buf is + * non-<tt>NULL</tt> and \a _buf_sz is not + * <tt>sizeof(#th_huff_code)*#TH_NHUFFMAN_TABLES*#TH_NDCT_TOKENS</tt>. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_HUFFMAN_CODES (0) +/**Sets the quantization parameters to use. + * The parameters are copied, not stored by reference, so they can be freed + * after this call. + * <tt>NULL</tt> may be specified to revert to the default parameters. + * + * \param[in] _buf #th_quant_info + * \retval TH_EFAULT \a _enc_ctx is <tt>NULL</tt>. + * \retval TH_EINVAL Encoding has already begun, \a _buf is + * <tt>NULL</tt> and \a _buf_sz is not zero, + * or \a _buf is non-<tt>NULL</tt> and + * \a _buf_sz is not <tt>sizeof(#th_quant_info)</tt>. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_QUANT_PARAMS (2) +/**Sets the maximum distance between key frames. + * This can be changed during an encode, but will be bounded by + * <tt>1<<th_info#keyframe_granule_shift</tt>. + * If it is set before encoding begins, th_info#keyframe_granule_shift will + * be enlarged appropriately. + * + * \param[in] _buf <tt>ogg_uint32_t</tt>: The maximum distance between key + * frames. + * \param[out] _buf <tt>ogg_uint32_t</tt>: The actual maximum distance set. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(ogg_uint32_t)</tt>. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_KEYFRAME_FREQUENCY_FORCE (4) +/**Disables any encoder features that would prevent lossless transcoding back + * to VP3. + * This primarily means disabling block-adaptive quantization and always coding + * all four luma blocks in a macro block when 4MV is used. + * It also includes using the VP3 quantization tables and Huffman codes; if you + * set them explicitly after calling this function, the resulting stream will + * not be VP3-compatible. + * If you enable VP3-compatibility when encoding 4:2:2 or 4:4:4 source + * material, or when using a picture region smaller than the full frame (e.g. + * a non-multiple-of-16 width or height), then non-VP3 bitstream features will + * still be disabled, but the stream will still not be VP3-compatible, as VP3 + * was not capable of encoding such formats. + * If you call this after encoding has already begun, then the quantization + * tables and codebooks cannot be changed, but the frame-level features will + * be enabled or disabled as requested. + * + * \param[in] _buf <tt>int</tt>: a non-zero value to enable VP3 compatibility, + * or 0 to disable it (the default). + * \param[out] _buf <tt>int</tt>: 1 if all bitstream features required for + * VP3-compatibility could be set, and 0 otherwise. + * The latter will be returned if the pixel format is not + * 4:2:0, the picture region is smaller than the full frame, + * or if encoding has begun, preventing the quantization + * tables and codebooks from being set. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_VP3_COMPATIBLE (10) +/**Gets the maximum speed level. + * Higher speed levels favor quicker encoding over better quality per bit. + * Depending on the encoding mode, and the internal algorithms used, quality + * may actually improve, but in this case bitrate will also likely increase. + * In any case, overall rate/distortion performance will probably decrease. + * The maximum value, and the meaning of each value, may change depending on + * the current encoding mode (VBR vs. constant quality, etc.). + * + * \param[out] _buf <tt>int</tt>: The maximum encoding speed level. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>. + * \retval TH_EIMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_GET_SPLEVEL_MAX (12) +/**Sets the speed level. + * The current speed level may be retrieved using #TH_ENCCTL_GET_SPLEVEL. + * + * \param[in] _buf <tt>int</tt>: The new encoding speed level. + * 0 is slowest, larger values use less CPU. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>, or the + * encoding speed level is out of bounds. + * The maximum encoding speed level may be + * implementation- and encoding mode-specific, and can be + * obtained via #TH_ENCCTL_GET_SPLEVEL_MAX. + * \retval TH_EIMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_SET_SPLEVEL (14) +/**Gets the current speed level. + * The default speed level may vary according to encoder implementation, but if + * this control code is not supported (it returns #TH_EIMPL), the default may + * be assumed to be the slowest available speed (0). + * The maximum encoding speed level may be implementation- and encoding + * mode-specific, and can be obtained via #TH_ENCCTL_GET_SPLEVEL_MAX. + * + * \param[out] _buf <tt>int</tt>: The current encoding speed level. + * 0 is slowest, larger values use less CPU. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>. + * \retval TH_EIMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_GET_SPLEVEL (16) +/**Sets the number of duplicates of the next frame to produce. + * Although libtheora can encode duplicate frames very cheaply, it costs some + * amount of CPU to detect them, and a run of duplicates cannot span a + * keyframe boundary. + * This control code tells the encoder to produce the specified number of extra + * duplicates of the next frame. + * This allows the encoder to make smarter keyframe placement decisions and + * rate control decisions, and reduces CPU usage as well, when compared to + * just submitting the same frame for encoding multiple times. + * This setting only applies to the next frame submitted for encoding. + * You MUST call th_encode_packetout() repeatedly until it returns 0, or the + * extra duplicate frames will be lost. + * + * \param[in] _buf <tt>int</tt>: The number of duplicates to produce. + * If this is negative or zero, no duplicates will be produced. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>, or the + * number of duplicates is greater than or equal to the + * maximum keyframe interval. + * In the latter case, NO duplicate frames will be produced. + * You must ensure that the maximum keyframe interval is set + * larger than the maximum number of duplicates you will + * ever wish to insert prior to encoding. + * \retval TH_EIMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_SET_DUP_COUNT (18) +/**Modifies the default bitrate management behavior. + * Use to allow or disallow frame dropping, and to enable or disable capping + * bit reservoir overflows and underflows. + * See \ref encctlcodes "the list of available flags". + * The flags are set by default to + * <tt>#TH_RATECTL_DROP_FRAMES|#TH_RATECTL_CAP_OVERFLOW</tt>. + * + * \param[in] _buf <tt>int</tt>: Any combination of + * \ref ratectlflags "the available flags": + * - #TH_RATECTL_DROP_FRAMES: Enable frame dropping. + * - #TH_RATECTL_CAP_OVERFLOW: Don't bank excess bits for later + * use. + * - #TH_RATECTL_CAP_UNDERFLOW: Don't try to make up shortfalls + * later. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt> or rate control + * is not enabled. + * \retval TH_EIMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_SET_RATE_FLAGS (20) +/**Sets the size of the bitrate management bit reservoir as a function + * of number of frames. + * The reservoir size affects how quickly bitrate management reacts to + * instantaneous changes in the video complexity. + * Larger reservoirs react more slowly, and provide better overall quality, but + * require more buffering by a client, adding more latency to live streams. + * By default, libtheora sets the reservoir to the maximum distance between + * keyframes, subject to a minimum and maximum limit. + * This call may be used to increase or decrease the reservoir, increasing or + * decreasing the allowed temporary variance in bitrate. + * An implementation may impose some limits on the size of a reservoir it can + * handle, in which case the actual reservoir size may not be exactly what was + * requested. + * The actual value set will be returned. + * + * \param[in] _buf <tt>int</tt>: Requested size of the reservoir measured in + * frames. + * \param[out] _buf <tt>int</tt>: The actual size of the reservoir set. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(int)</tt>, or rate control + * is not enabled. The buffer has an implementation + * defined minimum and maximum size and the value in _buf + * will be adjusted to match the actual value set. + * \retval TH_EIMPL Not supported by this implementation in the current + * encoding mode.*/ +#define TH_ENCCTL_SET_RATE_BUFFER (22) +/**Enable pass 1 of two-pass encoding mode and retrieve the first pass metrics. + * Pass 1 mode must be enabled before the first frame is encoded, and a target + * bitrate must have already been specified to the encoder. + * Although this does not have to be the exact rate that will be used in the + * second pass, closer values may produce better results. + * The first call returns the size of the two-pass header data, along with some + * placeholder content, and sets the encoder into pass 1 mode implicitly. + * This call sets the encoder to pass 1 mode implicitly. + * Then, a subsequent call must be made after each call to + * th_encode_ycbcr_in() to retrieve the metrics for that frame. + * An additional, final call must be made to retrieve the summary data, + * containing such information as the total number of frames, etc. + * This must be stored in place of the placeholder data that was returned + * in the first call, before the frame metrics data. + * All of this data must be presented back to the encoder during pass 2 using + * #TH_ENCCTL_2PASS_IN. + * + * \param[out] <tt>char *</tt>_buf: Returns a pointer to internal storage + * containing the two pass metrics data. + * This storage is only valid until the next call, or until the + * encoder context is freed, and must be copied by the + * application. + * \retval >=0 The number of bytes of metric data available in the + * returned buffer. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL \a _buf_sz is not <tt>sizeof(char *)</tt>, no target + * bitrate has been set, or the first call was made after + * the first frame was submitted for encoding. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_2PASS_OUT (24) +/**Submits two-pass encoding metric data collected the first encoding pass to + * the second pass. + * The first call must be made before the first frame is encoded, and a target + * bitrate must have already been specified to the encoder. + * It sets the encoder to pass 2 mode implicitly; this cannot be disabled. + * The encoder may require reading data from some or all of the frames in + * advance, depending on, e.g., the reservoir size used in the second pass. + * You must call this function repeatedly before each frame to provide data + * until either a) it fails to consume all of the data presented or b) all of + * the pass 1 data has been consumed. + * In the first case, you must save the remaining data to be presented after + * the next frame. + * You can call this function with a NULL argument to get an upper bound on + * the number of bytes that will be required before the next frame. + * + * When pass 2 is first enabled, the default bit reservoir is set to the entire + * file; this gives maximum flexibility but can lead to very high peak rates. + * You can subsequently set it to another value with #TH_ENCCTL_SET_RATE_BUFFER + * (e.g., to set it to the keyframe interval for non-live streaming), however, + * you may then need to provide more data before the next frame. + * + * \param[in] _buf <tt>char[]</tt>: A buffer containing the data returned by + * #TH_ENCCTL_2PASS_OUT in pass 1. + * You may pass <tt>NULL</tt> for \a _buf to return an upper + * bound on the number of additional bytes needed before the + * next frame. + * The summary data returned at the end of pass 1 must be at + * the head of the buffer on the first call with a + * non-<tt>NULL</tt> \a _buf, and the placeholder data + * returned at the start of pass 1 should be omitted. + * After each call you should advance this buffer by the number + * of bytes consumed. + * \retval >0 The number of bytes of metric data required/consumed. + * \retval 0 No more data is required before the next frame. + * \retval TH_EFAULT \a _enc_ctx is <tt>NULL</tt>. + * \retval TH_EINVAL No target bitrate has been set, or the first call was + * made after the first frame was submitted for + * encoding. + * \retval TH_ENOTFORMAT The data did not appear to be pass 1 from a compatible + * implementation of this library. + * \retval TH_EBADHEADER The data was invalid; this may be returned when + * attempting to read an aborted pass 1 file that still + * has the placeholder data in place of the summary + * data. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_2PASS_IN (26) +/**Sets the current encoding quality. + * This is only valid so long as no bitrate has been specified, either through + * the #th_info struct used to initialize the encoder or through + * #TH_ENCCTL_SET_BITRATE (this restriction may be relaxed in a future + * version). + * If it is set before the headers are emitted, the target quality encoded in + * them will be updated. + * + * \param[in] _buf <tt>int</tt>: The new target quality, in the range 0...63, + * inclusive. + * \retval 0 Success. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL A target bitrate has already been specified, or the + * quality index was not in the range 0...63. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_QUALITY (28) +/**Sets the current encoding bitrate. + * Once a bitrate is set, the encoder must use a rate-controlled mode for all + * future frames (this restriction may be relaxed in a future version). + * If it is set before the headers are emitted, the target bitrate encoded in + * them will be updated. + * Due to the buffer delay, the exact bitrate of each section of the encode is + * not guaranteed. + * The encoder may have already used more bits than allowed for the frames it + * has encoded, expecting to make them up in future frames, or it may have + * used fewer, holding the excess in reserve. + * The exact transition between the two bitrates is not well-defined by this + * API, but may be affected by flags set with #TH_ENCCTL_SET_RATE_FLAGS. + * After a number of frames equal to the buffer delay, one may expect further + * output to average at the target bitrate. + * + * \param[in] _buf <tt>long</tt>: The new target bitrate, in bits per second. + * \retval 0 Success. + * \retval TH_EFAULT \a _enc_ctx or \a _buf is <tt>NULL</tt>. + * \retval TH_EINVAL The target bitrate was not positive. + * \retval TH_EIMPL Not supported by this implementation.*/ +#define TH_ENCCTL_SET_BITRATE (30) + +/*@}*/ + + +/**\name TH_ENCCTL_SET_RATE_FLAGS flags + * \anchor ratectlflags + * These are the flags available for use with #TH_ENCCTL_SET_RATE_FLAGS.*/ +/*@{*/ +/**Drop frames to keep within bitrate buffer constraints. + * This can have a severe impact on quality, but is the only way to ensure that + * bitrate targets are met at low rates during sudden bursts of activity.*/ +#define TH_RATECTL_DROP_FRAMES (0x1) +/**Ignore bitrate buffer overflows. + * If the encoder uses so few bits that the reservoir of available bits + * overflows, ignore the excess. + * The encoder will not try to use these extra bits in future frames. + * At high rates this may cause the result to be undersized, but allows a + * client to play the stream using a finite buffer; it should normally be + * enabled.*/ +#define TH_RATECTL_CAP_OVERFLOW (0x2) +/**Ignore bitrate buffer underflows. + * If the encoder uses so many bits that the reservoir of available bits + * underflows, ignore the deficit. + * The encoder will not try to make up these extra bits in future frames. + * At low rates this may cause the result to be oversized; it should normally + * be disabled.*/ +#define TH_RATECTL_CAP_UNDERFLOW (0x4) +/*@}*/ + + + +/**The quantization parameters used by VP3.*/ +extern const th_quant_info TH_VP31_QUANT_INFO; + +/**The Huffman tables used by VP3.*/ +extern const th_huff_code + TH_VP31_HUFF_CODES[TH_NHUFFMAN_TABLES][TH_NDCT_TOKENS]; + + + +/**\name Encoder state + The following data structure is opaque, and its contents are not publicly + defined by this API. + Referring to its internals directly is unsupported, and may break without + warning.*/ +/*@{*/ +/**The encoder context.*/ +typedef struct th_enc_ctx th_enc_ctx; +/*@}*/ + + + +/**\defgroup encfuncs Functions for Encoding*/ +/*@{*/ +/**\name Functions for encoding + * You must link to <tt>libtheoraenc</tt> and <tt>libtheoradec</tt> + * if you use any of the functions in this section. + * + * The functions are listed in the order they are used in a typical encode. + * The basic steps are: + * - Fill in a #th_info structure with details on the format of the video you + * wish to encode. + * - Allocate a #th_enc_ctx handle with th_encode_alloc(). + * - Perform any additional encoder configuration required with + * th_encode_ctl(). + * - Repeatedly call th_encode_flushheader() to retrieve all the header + * packets. + * - For each uncompressed frame: + * - Submit the uncompressed frame via th_encode_ycbcr_in() + * - Repeatedly call th_encode_packetout() to retrieve any video data packets + * that are ready. + * - Call th_encode_free() to release all encoder memory.*/ +/*@{*/ +/**Allocates an encoder instance. + * \param _info A #th_info struct filled with the desired encoding parameters. + * \return The initialized #th_enc_ctx handle. + * \retval NULL If the encoding parameters were invalid.*/ +extern th_enc_ctx *th_encode_alloc(const th_info *_info); +/**Encoder control function. + * This is used to provide advanced control the encoding process. + * \param _enc A #th_enc_ctx handle. + * \param _req The control code to process. + * See \ref encctlcodes "the list of available control codes" + * for details. + * \param _buf The parameters for this control code. + * \param _buf_sz The size of the parameter buffer.*/ +extern int th_encode_ctl(th_enc_ctx *_enc,int _req,void *_buf,size_t _buf_sz); +/**Outputs the next header packet. + * This should be called repeatedly after encoder initialization until it + * returns 0 in order to get all of the header packets, in order, before + * encoding actual video data. + * \param _enc A #th_enc_ctx handle. + * \param _comments The metadata to place in the comment header, when it is + * encoded. + * \param _op An <tt>ogg_packet</tt> structure to fill. + * All of the elements of this structure will be set, + * including a pointer to the header data. + * The memory for the header data is owned by + * <tt>libtheoraenc</tt>, and may be invalidated when the + * next encoder function is called. + * \return A positive value indicates that a header packet was successfully + * produced. + * \retval 0 No packet was produced, and no more header packets remain. + * \retval TH_EFAULT \a _enc, \a _comments, or \a _op was <tt>NULL</tt>.*/ +extern int th_encode_flushheader(th_enc_ctx *_enc, + th_comment *_comments,ogg_packet *_op); +/**Submits an uncompressed frame to the encoder. + * \param _enc A #th_enc_ctx handle. + * \param _ycbcr A buffer of Y'CbCr data to encode. + * If the width and height of the buffer matches the frame size + * the encoder was initialized with, the encoder will only + * reference the portion inside the picture region. + * Any data outside this region will be ignored, and need not map + * to a valid address. + * Alternatively, you can pass a buffer equal to the size of the + * picture region, if this is less than the full frame size. + * When using subsampled chroma planes, odd picture sizes or odd + * picture offsets may require an unexpected chroma plane size, + * and their use is generally discouraged, as they will not be + * well-supported by players and other media frameworks. + * See Section 4.4 of + * <a href="http://www.theora.org/doc/Theora.pdf">the Theora + * specification</a> for details if you wish to use them anyway. + * \retval 0 Success. + * \retval TH_EFAULT \a _enc or \a _ycbcr is <tt>NULL</tt>. + * \retval TH_EINVAL The buffer size matches neither the frame size nor the + * picture size the encoder was initialized with, or + * encoding has already completed.*/ +extern int th_encode_ycbcr_in(th_enc_ctx *_enc,th_ycbcr_buffer _ycbcr); +/**Retrieves encoded video data packets. + * This should be called repeatedly after each frame is submitted to flush any + * encoded packets, until it returns 0. + * The encoder will not buffer these packets as subsequent frames are + * compressed, so a failure to do so will result in lost video data. + * \note Currently the encoder operates in a one-frame-in, one-packet-out + * manner. + * However, this may be changed in the future. + * \param _enc A #th_enc_ctx handle. + * \param _last Set this flag to a non-zero value if no more uncompressed + * frames will be submitted. + * This ensures that a proper EOS flag is set on the last packet. + * \param _op An <tt>ogg_packet</tt> structure to fill. + * All of the elements of this structure will be set, including a + * pointer to the video data. + * The memory for the video data is owned by + * <tt>libtheoraenc</tt>, and may be invalidated when the next + * encoder function is called. + * \return A positive value indicates that a video data packet was successfully + * produced. + * \retval 0 No packet was produced, and no more encoded video data + * remains. + * \retval TH_EFAULT \a _enc or \a _op was <tt>NULL</tt>.*/ +extern int th_encode_packetout(th_enc_ctx *_enc,int _last,ogg_packet *_op); +/**Frees an allocated encoder instance. + * \param _enc A #th_enc_ctx handle.*/ +extern void th_encode_free(th_enc_ctx *_enc); +/*@}*/ +/*@}*/ + + + +#if defined(__cplusplus) +} +#endif + +#endif diff --git a/thirdparties/android/include/vorbis/codec.h b/thirdparties/android/include/vorbis/codec.h new file mode 100644 index 0000000..259798c --- /dev/null +++ b/thirdparties/android/include/vorbis/codec.h @@ -0,0 +1,243 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + + ******************************************************************** + + function: libvorbis codec headers + last mod: $Id: codec.h 16037 2009-05-26 21:10:58Z xiphmont $ + + ********************************************************************/ + +#ifndef _vorbis_codec_h_ +#define _vorbis_codec_h_ + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +#include <ogg/ogg.h> + +typedef struct vorbis_info{ + int version; + int channels; + long rate; + + /* The below bitrate declarations are *hints*. + Combinations of the three values carry the following implications: + + all three set to the same value: + implies a fixed rate bitstream + only nominal set: + implies a VBR stream that averages the nominal bitrate. No hard + upper/lower limit + upper and or lower set: + implies a VBR bitstream that obeys the bitrate limits. nominal + may also be set to give a nominal rate. + none set: + the coder does not care to speculate. + */ + + long bitrate_upper; + long bitrate_nominal; + long bitrate_lower; + long bitrate_window; + + void *codec_setup; +} vorbis_info; + +/* vorbis_dsp_state buffers the current vorbis audio + analysis/synthesis state. The DSP state belongs to a specific + logical bitstream ****************************************************/ +typedef struct vorbis_dsp_state{ + int analysisp; + vorbis_info *vi; + + float **pcm; + float **pcmret; + int pcm_storage; + int pcm_current; + int pcm_returned; + + int preextrapolate; + int eofflag; + + long lW; + long W; + long nW; + long centerW; + + ogg_int64_t granulepos; + ogg_int64_t sequence; + + ogg_int64_t glue_bits; + ogg_int64_t time_bits; + ogg_int64_t floor_bits; + ogg_int64_t res_bits; + + void *backend_state; +} vorbis_dsp_state; + +typedef struct vorbis_block{ + /* necessary stream state for linking to the framing abstraction */ + float **pcm; /* this is a pointer into local storage */ + oggpack_buffer opb; + + long lW; + long W; + long nW; + int pcmend; + int mode; + + int eofflag; + ogg_int64_t granulepos; + ogg_int64_t sequence; + vorbis_dsp_state *vd; /* For read-only access of configuration */ + + /* local storage to avoid remallocing; it's up to the mapping to + structure it */ + void *localstore; + long localtop; + long localalloc; + long totaluse; + struct alloc_chain *reap; + + /* bitmetrics for the frame */ + long glue_bits; + long time_bits; + long floor_bits; + long res_bits; + + void *internal; + +} vorbis_block; + +/* vorbis_block is a single block of data to be processed as part of +the analysis/synthesis stream; it belongs to a specific logical +bitstream, but is independant from other vorbis_blocks belonging to +that logical bitstream. *************************************************/ + +struct alloc_chain{ + void *ptr; + struct alloc_chain *next; +}; + +/* vorbis_info contains all the setup information specific to the + specific compression/decompression mode in progress (eg, + psychoacoustic settings, channel setup, options, codebook + etc). vorbis_info and substructures are in backends.h. +*********************************************************************/ + +/* the comments are not part of vorbis_info so that vorbis_info can be + static storage */ +typedef struct vorbis_comment{ + /* unlimited user comment fields. libvorbis writes 'libvorbis' + whatever vendor is set to in encode */ + char **user_comments; + int *comment_lengths; + int comments; + char *vendor; + +} vorbis_comment; + + +/* libvorbis encodes in two abstraction layers; first we perform DSP + and produce a packet (see docs/analysis.txt). The packet is then + coded into a framed OggSquish bitstream by the second layer (see + docs/framing.txt). Decode is the reverse process; we sync/frame + the bitstream and extract individual packets, then decode the + packet back into PCM audio. + + The extra framing/packetizing is used in streaming formats, such as + files. Over the net (such as with UDP), the framing and + packetization aren't necessary as they're provided by the transport + and the streaming layer is not used */ + +/* Vorbis PRIMITIVES: general ***************************************/ + +extern void vorbis_info_init(vorbis_info *vi); +extern void vorbis_info_clear(vorbis_info *vi); +extern int vorbis_info_blocksize(vorbis_info *vi,int zo); +extern void vorbis_comment_init(vorbis_comment *vc); +extern void vorbis_comment_add(vorbis_comment *vc, const char *comment); +extern void vorbis_comment_add_tag(vorbis_comment *vc, + const char *tag, const char *contents); +extern char *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count); +extern int vorbis_comment_query_count(vorbis_comment *vc, const char *tag); +extern void vorbis_comment_clear(vorbis_comment *vc); + +extern int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb); +extern int vorbis_block_clear(vorbis_block *vb); +extern void vorbis_dsp_clear(vorbis_dsp_state *v); +extern double vorbis_granule_time(vorbis_dsp_state *v, + ogg_int64_t granulepos); + +extern const char *vorbis_version_string(void); + +/* Vorbis PRIMITIVES: analysis/DSP layer ****************************/ + +extern int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi); +extern int vorbis_commentheader_out(vorbis_comment *vc, ogg_packet *op); +extern int vorbis_analysis_headerout(vorbis_dsp_state *v, + vorbis_comment *vc, + ogg_packet *op, + ogg_packet *op_comm, + ogg_packet *op_code); +extern float **vorbis_analysis_buffer(vorbis_dsp_state *v,int vals); +extern int vorbis_analysis_wrote(vorbis_dsp_state *v,int vals); +extern int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb); +extern int vorbis_analysis(vorbis_block *vb,ogg_packet *op); + +extern int vorbis_bitrate_addblock(vorbis_block *vb); +extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd, + ogg_packet *op); + +/* Vorbis PRIMITIVES: synthesis layer *******************************/ +extern int vorbis_synthesis_idheader(ogg_packet *op); +extern int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc, + ogg_packet *op); + +extern int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi); +extern int vorbis_synthesis_restart(vorbis_dsp_state *v); +extern int vorbis_synthesis(vorbis_block *vb,ogg_packet *op); +extern int vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op); +extern int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb); +extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm); +extern int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm); +extern int vorbis_synthesis_read(vorbis_dsp_state *v,int samples); +extern long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op); + +extern int vorbis_synthesis_halfrate(vorbis_info *v,int flag); +extern int vorbis_synthesis_halfrate_p(vorbis_info *v); + +/* Vorbis ERRORS and return codes ***********************************/ + +#define OV_FALSE -1 +#define OV_EOF -2 +#define OV_HOLE -3 + +#define OV_EREAD -128 +#define OV_EFAULT -129 +#define OV_EIMPL -130 +#define OV_EINVAL -131 +#define OV_ENOTVORBIS -132 +#define OV_EBADHEADER -133 +#define OV_EVERSION -134 +#define OV_ENOTAUDIO -135 +#define OV_EBADPACKET -136 +#define OV_EBADLINK -137 +#define OV_ENOSEEK -138 + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif + diff --git a/thirdparties/android/include/vorbis/vorbisenc.h b/thirdparties/android/include/vorbis/vorbisenc.h new file mode 100644 index 0000000..bddd83d --- /dev/null +++ b/thirdparties/android/include/vorbis/vorbisenc.h @@ -0,0 +1,112 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: vorbis encode-engine setup + last mod: $Id: vorbisenc.h 16037 2009-05-26 21:10:58Z xiphmont $ + + ********************************************************************/ + +#ifndef _OV_ENC_H_ +#define _OV_ENC_H_ + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +#include "codec.h" + +extern int vorbis_encode_init(vorbis_info *vi, + long channels, + long rate, + + long max_bitrate, + long nominal_bitrate, + long min_bitrate); + +extern int vorbis_encode_setup_managed(vorbis_info *vi, + long channels, + long rate, + + long max_bitrate, + long nominal_bitrate, + long min_bitrate); + +extern int vorbis_encode_setup_vbr(vorbis_info *vi, + long channels, + long rate, + + float quality /* quality level from 0. (lo) to 1. (hi) */ + ); + +extern int vorbis_encode_init_vbr(vorbis_info *vi, + long channels, + long rate, + + float base_quality /* quality level from 0. (lo) to 1. (hi) */ + ); + +extern int vorbis_encode_setup_init(vorbis_info *vi); + +extern int vorbis_encode_ctl(vorbis_info *vi,int number,void *arg); + + /* deprecated rate management supported only for compatability */ +#define OV_ECTL_RATEMANAGE_GET 0x10 +#define OV_ECTL_RATEMANAGE_SET 0x11 +#define OV_ECTL_RATEMANAGE_AVG 0x12 +#define OV_ECTL_RATEMANAGE_HARD 0x13 + +struct ovectl_ratemanage_arg { + int management_active; + + long bitrate_hard_min; + long bitrate_hard_max; + double bitrate_hard_window; + + long bitrate_av_lo; + long bitrate_av_hi; + double bitrate_av_window; + double bitrate_av_window_center; +}; + + + /* new rate setup */ +#define OV_ECTL_RATEMANAGE2_GET 0x14 +#define OV_ECTL_RATEMANAGE2_SET 0x15 + +struct ovectl_ratemanage2_arg { + int management_active; + + long bitrate_limit_min_kbps; + long bitrate_limit_max_kbps; + long bitrate_limit_reservoir_bits; + double bitrate_limit_reservoir_bias; + + long bitrate_average_kbps; + double bitrate_average_damping; +}; + + + +#define OV_ECTL_LOWPASS_GET 0x20 +#define OV_ECTL_LOWPASS_SET 0x21 + +#define OV_ECTL_IBLOCK_GET 0x30 +#define OV_ECTL_IBLOCK_SET 0x31 + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif + + diff --git a/thirdparties/android/include/vorbis/vorbisfile.h b/thirdparties/android/include/vorbis/vorbisfile.h new file mode 100644 index 0000000..ef2a36d --- /dev/null +++ b/thirdparties/android/include/vorbis/vorbisfile.h @@ -0,0 +1,206 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. * + * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS * + * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE * + * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. * + * * + * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.h 16243 2009-07-10 02:49:31Z xiphmont $ + + ********************************************************************/ + +#ifndef _OV_FILE_H_ +#define _OV_FILE_H_ + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +#include <stdio.h> +#include "codec.h" + +/* The function prototypes for the callbacks are basically the same as for + * the stdio functions fread, fseek, fclose, ftell. + * The one difference is that the FILE * arguments have been replaced with + * a void * - this is to be used as a pointer to whatever internal data these + * functions might need. In the stdio case, it's just a FILE * cast to a void * + * + * If you use other functions, check the docs for these functions and return + * the right values. For seek_func(), you *MUST* return -1 if the stream is + * unseekable + */ +typedef struct { + size_t (*read_func) (void *ptr, size_t size, size_t nmemb, void *datasource); + int (*seek_func) (void *datasource, ogg_int64_t offset, int whence); + int (*close_func) (void *datasource); + long (*tell_func) (void *datasource); +} ov_callbacks; + +#ifndef OV_EXCLUDE_STATIC_CALLBACKS + +/* a few sets of convenient callbacks, especially for use under + * Windows where ov_open_callbacks() should always be used instead of + * ov_open() to avoid problems with incompatable crt.o version linking + * issues. */ + +static int _ov_header_fseek_wrap(FILE *f,ogg_int64_t off,int whence){ + if(f==NULL)return(-1); + +#ifdef __MINGW32__ + return fseeko64(f,off,whence); +#elif defined (_WIN32) + return _fseeki64(f,off,whence); +#else + return fseek(f,off,whence); +#endif +} + +/* These structs below (OV_CALLBACKS_DEFAULT etc) are defined here as + * static data. That means that every file which includes this header + * will get its own copy of these structs whether it uses them or + * not unless it #defines OV_EXCLUDE_STATIC_CALLBACKS. + * These static symbols are essential on platforms such as Windows on + * which several different versions of stdio support may be linked to + * by different DLLs, and we need to be certain we know which one + * we're using (the same one as the main application). + */ + +static ov_callbacks OV_CALLBACKS_DEFAULT = { + (size_t (*)(void *, size_t, size_t, void *)) fread, + (int (*)(void *, ogg_int64_t, int)) _ov_header_fseek_wrap, + (int (*)(void *)) fclose, + (long (*)(void *)) ftell +}; + +static ov_callbacks OV_CALLBACKS_NOCLOSE = { + (size_t (*)(void *, size_t, size_t, void *)) fread, + (int (*)(void *, ogg_int64_t, int)) _ov_header_fseek_wrap, + (int (*)(void *)) NULL, + (long (*)(void *)) ftell +}; + +static ov_callbacks OV_CALLBACKS_STREAMONLY = { + (size_t (*)(void *, size_t, size_t, void *)) fread, + (int (*)(void *, ogg_int64_t, int)) NULL, + (int (*)(void *)) fclose, + (long (*)(void *)) NULL +}; + +static ov_callbacks OV_CALLBACKS_STREAMONLY_NOCLOSE = { + (size_t (*)(void *, size_t, size_t, void *)) fread, + (int (*)(void *, ogg_int64_t, int)) NULL, + (int (*)(void *)) NULL, + (long (*)(void *)) NULL +}; + +#endif + +#define NOTOPEN 0 +#define PARTOPEN 1 +#define OPENED 2 +#define STREAMSET 3 +#define INITSET 4 + +typedef struct OggVorbis_File { + void *datasource; /* Pointer to a FILE *, etc. */ + int seekable; + ogg_int64_t offset; + ogg_int64_t end; + ogg_sync_state oy; + + /* If the FILE handle isn't seekable (eg, a pipe), only the current + stream appears */ + int links; + ogg_int64_t *offsets; + ogg_int64_t *dataoffsets; + long *serialnos; + ogg_int64_t *pcmlengths; /* overloaded to maintain binary + compatability; x2 size, stores both + beginning and end values */ + vorbis_info *vi; + vorbis_comment *vc; + + /* Decoding working state local storage */ + ogg_int64_t pcm_offset; + int ready_state; + long current_serialno; + int current_link; + + double bittrack; + double samptrack; + + ogg_stream_state os; /* take physical pages, weld into a logical + stream of packets */ + vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ + vorbis_block vb; /* local working space for packet->PCM decode */ + + ov_callbacks callbacks; + +} OggVorbis_File; + + +extern int ov_clear(OggVorbis_File *vf); +extern int ov_fopen(char *path,OggVorbis_File *vf); +extern int ov_open(FILE *f,OggVorbis_File *vf,char *initial,long ibytes); +extern int ov_open_callbacks(void *datasource, OggVorbis_File *vf, + char *initial, long ibytes, ov_callbacks callbacks); + +extern int ov_test(FILE *f,OggVorbis_File *vf,char *initial,long ibytes); +extern int ov_test_callbacks(void *datasource, OggVorbis_File *vf, + char *initial, long ibytes, ov_callbacks callbacks); +extern int ov_test_open(OggVorbis_File *vf); + +extern long ov_bitrate(OggVorbis_File *vf,int i); +extern long ov_bitrate_instant(OggVorbis_File *vf); +extern long ov_streams(OggVorbis_File *vf); +extern long ov_seekable(OggVorbis_File *vf); +extern long ov_serialnumber(OggVorbis_File *vf,int i); + +extern ogg_int64_t ov_raw_total(OggVorbis_File *vf,int i); +extern ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i); +extern double ov_time_total(OggVorbis_File *vf,int i); + +extern int ov_raw_seek(OggVorbis_File *vf,ogg_int64_t pos); +extern int ov_pcm_seek(OggVorbis_File *vf,ogg_int64_t pos); +extern int ov_pcm_seek_page(OggVorbis_File *vf,ogg_int64_t pos); +extern int ov_time_seek(OggVorbis_File *vf,double pos); +extern int ov_time_seek_page(OggVorbis_File *vf,double pos); + +extern int ov_raw_seek_lap(OggVorbis_File *vf,ogg_int64_t pos); +extern int ov_pcm_seek_lap(OggVorbis_File *vf,ogg_int64_t pos); +extern int ov_pcm_seek_page_lap(OggVorbis_File *vf,ogg_int64_t pos); +extern int ov_time_seek_lap(OggVorbis_File *vf,double pos); +extern int ov_time_seek_page_lap(OggVorbis_File *vf,double pos); + +extern ogg_int64_t ov_raw_tell(OggVorbis_File *vf); +extern ogg_int64_t ov_pcm_tell(OggVorbis_File *vf); +extern double ov_time_tell(OggVorbis_File *vf); + +extern vorbis_info *ov_info(OggVorbis_File *vf,int link); +extern vorbis_comment *ov_comment(OggVorbis_File *vf,int link); + +extern long ov_read_float(OggVorbis_File *vf,float ***pcm_channels,int samples, + int *bitstream); +extern long ov_read_filter(OggVorbis_File *vf,char *buffer,int length, + int bigendianp,int word,int sgned,int *bitstream, + void (*filter)(float **pcm,long channels,long samples,void *filter_param),void *filter_param); +extern long ov_read(OggVorbis_File *vf,char *buffer,int length, + int bigendianp,int word,int sgned,int *bitstream); +extern int ov_crosslap(OggVorbis_File *vf1,OggVorbis_File *vf2); + +extern int ov_halfrate(OggVorbis_File *vf,int flag); +extern int ov_halfrate_p(OggVorbis_File *vf); + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif + diff --git a/thirdparties/android/lib/pkgconfig/libavcodec.pc b/thirdparties/android/lib/pkgconfig/libavcodec.pc new file mode 100644 index 0000000..53071b9 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/libavcodec.pc @@ -0,0 +1,14 @@ +prefix=./android +exec_prefix=${prefix} +libdir=${prefix}/lib +includedir=${prefix}/include + +Name: libavcodec +Description: FFmpeg codec library +Version: 52.78.0 +Requires: libavutil = 50.19.0 +Requires.private: +Conflicts: +Libs: -L${libdir} -lavcodec -lm +Libs.private: +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/libavutil.pc b/thirdparties/android/lib/pkgconfig/libavutil.pc new file mode 100644 index 0000000..b072544 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/libavutil.pc @@ -0,0 +1,14 @@ +prefix=./android +exec_prefix=${prefix} +libdir=${prefix}/lib +includedir=${prefix}/include + +Name: libavutil +Description: FFmpeg utility library +Version: 50.19.0 +Requires: +Requires.private: +Conflicts: +Libs: -L${libdir} -lavutil +Libs.private: +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/libswscale.pc b/thirdparties/android/lib/pkgconfig/libswscale.pc new file mode 100644 index 0000000..7f81447 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/libswscale.pc @@ -0,0 +1,14 @@ +prefix=./android +exec_prefix=${prefix} +libdir=${prefix}/lib +includedir=${prefix}/include + +Name: libswscale +Description: FFmpeg image rescaling library +Version: 0.11.0 +Requires: libavutil = 50.19.0 +Requires.private: +Conflicts: +Libs: -L${libdir} -lswscale +Libs.private: +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/ogg.pc b/thirdparties/android/lib/pkgconfig/ogg.pc new file mode 100644 index 0000000..6b1c3d1 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/ogg.pc @@ -0,0 +1,14 @@ +# ogg pkg-config file + +prefix=/home/inexbee/Bureau/theorarm/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: ogg +Description: ogg is a library for manipulating ogg bitstreams +Version: 1.1.4 +Requires: +Conflicts: +Libs: -L${libdir} -logg +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/opencore-amrnb.pc b/thirdparties/android/lib/pkgconfig/opencore-amrnb.pc new file mode 100644 index 0000000..67cbbd6 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/opencore-amrnb.pc @@ -0,0 +1,10 @@ +prefix=/home/inexbee/Bureau/opencore-amr-0.1.2/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include/opencore-amrnb + +Name: OpenCORE AMR-NB +Description: Adaptive Multi-Rate speech codec library +Version: 0.1.2 +Libs: -L${libdir} -lopencore-amrnb +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/opencore-amrwb.pc b/thirdparties/android/lib/pkgconfig/opencore-amrwb.pc new file mode 100644 index 0000000..430cc32 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/opencore-amrwb.pc @@ -0,0 +1,10 @@ +prefix=/home/inexbee/Bureau/opencore-amr-0.1.2/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include/opencore-amrwb + +Name: OpenCORE AMR-WB +Description: Adaptive Multi-Rate Wideband speech codec library +Version: 0.1.2 +Libs: -L${libdir} -lopencore-amrwb +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/theora.pc b/thirdparties/android/lib/pkgconfig/theora.pc new file mode 100644 index 0000000..787ecca --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/theora.pc @@ -0,0 +1,14 @@ +# theora installed pkg-config file + +prefix=/home/inexbee/Bureau/theorarm/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: theora +Description: Theora video codec +Version: 1.1.1+svn +Requires: ogg >= 1.1 +Conflicts: +Libs: -L${libdir} -ltheora +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/theoradec.pc b/thirdparties/android/lib/pkgconfig/theoradec.pc new file mode 100644 index 0000000..c58d854 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/theoradec.pc @@ -0,0 +1,14 @@ +# theoradec installed pkg-config file + +prefix=/home/inexbee/Bureau/theorarm/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: theora +Description: Theora video codec (decoder) +Version: 1.1.1+svn +Requires: ogg >= 1.1 +Conflicts: +Libs: -L${libdir} -ltheoradec +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/theoraenc.pc b/thirdparties/android/lib/pkgconfig/theoraenc.pc new file mode 100644 index 0000000..27d4b41 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/theoraenc.pc @@ -0,0 +1,14 @@ +# theoraenc installed pkg-config file + +prefix=/home/inexbee/Bureau/theorarm/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: theora +Description: Theora video codec (encoder) +Version: 1.1.1+svn +Requires: theoradec, ogg >= 1.1 +Conflicts: +Libs: -L${libdir} -ltheoraenc +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/vorbis.pc b/thirdparties/android/lib/pkgconfig/vorbis.pc new file mode 100644 index 0000000..59c385e --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/vorbis.pc @@ -0,0 +1,14 @@ +# libvorbis pkg-config source file + +prefix=/home/inexbee/Bureau/theorarm/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: vorbis +Description: vorbis is the primary Ogg Vorbis library +Version: 1.2.3 +Requires: ogg +Conflicts: +Libs: -L${libdir} -lvorbis -lm +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/vorbisenc.pc b/thirdparties/android/lib/pkgconfig/vorbisenc.pc new file mode 100644 index 0000000..87834f9 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/vorbisenc.pc @@ -0,0 +1,14 @@ +# libvorbisenc pkg-config source file + +prefix=/home/inexbee/Bureau/theorarm/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: vorbisenc +Description: vorbisenc is a library that provides a convenient API for setting up an encoding environment using libvorbis +Version: 1.2.3 +Requires: vorbis +Conflicts: +Libs: -L${libdir} -lvorbisenc +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/vorbisfile.pc b/thirdparties/android/lib/pkgconfig/vorbisfile.pc new file mode 100644 index 0000000..e8f2f3a --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/vorbisfile.pc @@ -0,0 +1,14 @@ +# libvorbisfile pkg-config source file + +prefix=/home/inexbee/Bureau/theorarm/android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: vorbisfile +Description: vorbisfile is a library that provides a convenient high-level API for decoding and basic manipulation of all Vorbis I audio streams +Version: 1.2.3 +Requires: vorbis +Conflicts: +Libs: -L${libdir} -lvorbisfile +Cflags: -I${includedir} diff --git a/thirdparties/android/lib/pkgconfig/x264.pc b/thirdparties/android/lib/pkgconfig/x264.pc new file mode 100644 index 0000000..7f02eb2 --- /dev/null +++ b/thirdparties/android/lib/pkgconfig/x264.pc @@ -0,0 +1,10 @@ +prefix=./android +exec_prefix=${prefix} +libdir=${exec_prefix}/lib +includedir=${prefix}/include + +Name: x264 +Description: H.264 (MPEG4 AVC) encoder library +Version: 0.100.x +Libs: -L${exec_prefix}/lib -lx264 +Cflags: -I${prefix}/include |
