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author | Michael Niedermayer <michaelni@gmx.at> | 2012-07-04 20:39:50 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-07-04 21:03:28 +0200 |
commit | 039e9fe01ca27606a0ec1a944d51041541e10aab (patch) | |
tree | 9fc96837e878cdb2c13b6016d9f3d69785570488 /libavcodec/flacdec.c | |
parent | 8b421fad24acbba69935caf2a2775bd04f8a707a (diff) | |
parent | 7c29377b702783680b223a12503df784b1808086 (diff) | |
download | ffmpeg-streaming-039e9fe01ca27606a0ec1a944d51041541e10aab.zip ffmpeg-streaming-039e9fe01ca27606a0ec1a944d51041541e10aab.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (29 commits)
lavfi: reclassify showfiltfmts as a TESTPROG
graph2dot: fix printf format specifier
swscale: yuv2planeX 8bit >=sse2 functions need aligned stack on x86-32.
vp8: loopfilter >=sse2 functions need aligned stack on x86-32.
amr: remove shift out of the AMR_BIT() macro.
dsputilenc: group yasm and inline asm function pointer assignment.
mov: use forward declaration of a function instead of a table.
Clarify Doxygen comment for FF_API_* #defines.
configure: simplify get_version()
Create version.h headers for libraries that lack them
gitignore: Use full path instead of relative path to specify patterns
mpegvideo: remove VLAs
Add XTEA encryption support in libavutil
Add Blowfish encryption support in libavutil
eval: Add the isinf() function and tests for it
flacdec: move lpc filter to flacdsp
flacdec: split off channel decorrelation as flacdsp
avplay: Add an option for not limiting the input buffer size
FATE: add a test for WMA cover art.
FATE: add a test for apetag cover art
...
Conflicts:
.gitignore
configure
ffplay.c
libavcodec/Makefile
libavcodec/error_resilience.c
libavcodec/mpegvideo.c
libavcodec/ratecontrol.c
libavdevice/avdevice.h
libavfilter/Makefile
libavfilter/filtfmts.c
libavfilter/version.h
libavformat/mov.c
libavformat/version.h
libavutil/Makefile
libavutil/avutil.h
libavutil/version.h
libswscale/swscale.h
libswscale/x86/swscale_mmx.c
tests/fate/libavutil.mak
tests/lavfi-regression.sh
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/flacdec.c')
-rw-r--r-- | libavcodec/flacdec.c | 150 |
1 files changed, 45 insertions, 105 deletions
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c index 02ed463..13ce95d 100644 --- a/libavcodec/flacdec.c +++ b/libavcodec/flacdec.c @@ -42,6 +42,7 @@ #include "golomb.h" #include "flac.h" #include "flacdata.h" +#include "flacdsp.h" #undef NDEBUG #include <assert.h> @@ -54,13 +55,13 @@ typedef struct FLACContext { GetBitContext gb; ///< GetBitContext initialized to start at the current frame int blocksize; ///< number of samples in the current frame - int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits int sample_shift; ///< shift required to make output samples 16-bit or 32-bit - int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit int ch_mode; ///< channel decorrelation type in the current frame int got_streaminfo; ///< indicates if the STREAMINFO has been read int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples + + FLACDSPContext dsp; } FLACContext; static const int64_t flac_channel_layouts[6] = { @@ -101,6 +102,17 @@ int avpriv_flac_is_extradata_valid(AVCodecContext *avctx, return 1; } +static void flac_set_bps(FLACContext *s) +{ + if (s->bps > 16) { + s->avctx->sample_fmt = AV_SAMPLE_FMT_S32; + s->sample_shift = 32 - s->bps; + } else { + s->avctx->sample_fmt = AV_SAMPLE_FMT_S16; + s->sample_shift = 16 - s->bps; + } +} + static av_cold int flac_decode_init(AVCodecContext *avctx) { enum FLACExtradataFormat format; @@ -118,11 +130,9 @@ static av_cold int flac_decode_init(AVCodecContext *avctx) /* initialize based on the demuxer-supplied streamdata header */ avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo); - if (s->bps > 16) - avctx->sample_fmt = AV_SAMPLE_FMT_S32; - else - avctx->sample_fmt = AV_SAMPLE_FMT_S16; allocate_buffers(s); + flac_set_bps(s); + ff_flacdsp_init(&s->dsp, avctx->sample_fmt); s->got_streaminfo = 1; avcodec_get_frame_defaults(&s->frame); @@ -150,8 +160,7 @@ static void allocate_buffers(FLACContext *s) assert(s->max_blocksize); for (i = 0; i < s->channels; i++) { - s->decoded[i] = av_realloc(s->decoded[i], - sizeof(int32_t)*s->max_blocksize); + s->decoded[i] = av_malloc(sizeof(int32_t)*s->max_blocksize); } } @@ -223,6 +232,8 @@ static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) } avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]); allocate_buffers(s); + flac_set_bps(s); + ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt); s->got_streaminfo = 1; return 0; @@ -295,7 +306,8 @@ static int decode_residuals(FLACContext *s, int channel, int pred_order) return 0; } -static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order) +static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order, + int bps) { const int blocksize = s->blocksize; int32_t *decoded = s->decoded[channel]; @@ -303,7 +315,7 @@ static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order) /* warm up samples */ for (i = 0; i < pred_order; i++) { - decoded[i] = get_sbits_long(&s->gb, s->curr_bps); + decoded[i] = get_sbits_long(&s->gb, bps); } if (decode_residuals(s, channel, pred_order) < 0) @@ -345,16 +357,17 @@ static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order) return 0; } -static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) +static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order, + int bps) { - int i, j; + int i; int coeff_prec, qlevel; int coeffs[32]; int32_t *decoded = s->decoded[channel]; /* warm up samples */ for (i = 0; i < pred_order; i++) { - decoded[i] = get_sbits_long(&s->gb, s->curr_bps); + decoded[i] = get_sbits_long(&s->gb, bps); } coeff_prec = get_bits(&s->gb, 4) + 1; @@ -376,38 +389,7 @@ static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) if (decode_residuals(s, channel, pred_order) < 0) return -1; - if (s->bps > 16) { - int64_t sum; - for (i = pred_order; i < s->blocksize; i++) { - sum = 0; - for (j = 0; j < pred_order; j++) - sum += (int64_t)coeffs[j] * decoded[i-j-1]; - decoded[i] += sum >> qlevel; - } - } else { - for (i = pred_order; i < s->blocksize-1; i += 2) { - int c; - int d = decoded[i-pred_order]; - int s0 = 0, s1 = 0; - for (j = pred_order-1; j > 0; j--) { - c = coeffs[j]; - s0 += c*d; - d = decoded[i-j]; - s1 += c*d; - } - c = coeffs[0]; - s0 += c*d; - d = decoded[i] += s0 >> qlevel; - s1 += c*d; - decoded[i+1] += s1 >> qlevel; - } - if (i < s->blocksize) { - int sum = 0; - for (j = 0; j < pred_order; j++) - sum += coeffs[j] * decoded[i-j-1]; - decoded[i] += sum >> qlevel; - } - } + s->dsp.lpc(decoded, coeffs, pred_order, qlevel, s->blocksize); return 0; } @@ -415,15 +397,15 @@ static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order) static inline int decode_subframe(FLACContext *s, int channel) { int type, wasted = 0; + int bps = s->bps; int i, tmp; - s->curr_bps = s->bps; if (channel == 0) { if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE) - s->curr_bps++; + bps++; } else { if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE) - s->curr_bps++; + bps++; } if (get_bits1(&s->gb)) { @@ -436,35 +418,35 @@ static inline int decode_subframe(FLACContext *s, int channel) int left = get_bits_left(&s->gb); wasted = 1; if ( left < 0 || - (left < s->curr_bps && !show_bits_long(&s->gb, left)) || - !show_bits_long(&s->gb, s->curr_bps)) { + (left < bps && !show_bits_long(&s->gb, left)) || + !show_bits_long(&s->gb, bps)) { av_log(s->avctx, AV_LOG_ERROR, "Invalid number of wasted bits > available bits (%d) - left=%d\n", - s->curr_bps, left); + bps, left); return AVERROR_INVALIDDATA; } while (!get_bits1(&s->gb)) wasted++; - s->curr_bps -= wasted; + bps -= wasted; } - if (s->curr_bps > 32) { + if (bps > 32) { av_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0); return -1; } //FIXME use av_log2 for types if (type == 0) { - tmp = get_sbits_long(&s->gb, s->curr_bps); + tmp = get_sbits_long(&s->gb, bps); for (i = 0; i < s->blocksize; i++) s->decoded[channel][i] = tmp; } else if (type == 1) { for (i = 0; i < s->blocksize; i++) - s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps); + s->decoded[channel][i] = get_sbits_long(&s->gb, bps); } else if ((type >= 8) && (type <= 12)) { - if (decode_subframe_fixed(s, channel, type & ~0x8) < 0) + if (decode_subframe_fixed(s, channel, type & ~0x8, bps) < 0) return -1; } else if (type >= 32) { - if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0) + if (decode_subframe_lpc(s, channel, (type & ~0x20)+1, bps) < 0) return -1; } else { av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n"); @@ -512,15 +494,7 @@ static int decode_frame(FLACContext *s) } s->bps = s->avctx->bits_per_raw_sample = fi.bps; - if (s->bps > 16) { - s->avctx->sample_fmt = AV_SAMPLE_FMT_S32; - s->sample_shift = 32 - s->bps; - s->is32 = 1; - } else { - s->avctx->sample_fmt = AV_SAMPLE_FMT_S16; - s->sample_shift = 16 - s->bps; - s->is32 = 0; - } + flac_set_bps(s); if (!s->max_blocksize) s->max_blocksize = FLAC_MAX_BLOCKSIZE; @@ -546,6 +520,7 @@ static int decode_frame(FLACContext *s) if (!s->got_streaminfo) { allocate_buffers(s); + ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt); s->got_streaminfo = 1; dump_headers(s->avctx, (FLACStreaminfo *)s); } @@ -572,9 +547,7 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; FLACContext *s = avctx->priv_data; - int i, j = 0, bytes_read = 0; - int16_t *samples_16; - int32_t *samples_32; + int bytes_read = 0; int ret; *got_frame_ptr = 0; @@ -614,42 +587,9 @@ static int flac_decode_frame(AVCodecContext *avctx, void *data, av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } - samples_16 = (int16_t *)s->frame.data[0]; - samples_32 = (int32_t *)s->frame.data[0]; - -#define DECORRELATE(left, right)\ - assert(s->channels == 2);\ - for (i = 0; i < s->blocksize; i++) {\ - int a= s->decoded[0][i];\ - int b= s->decoded[1][i];\ - if (s->is32) {\ - *samples_32++ = (left) << s->sample_shift;\ - *samples_32++ = (right) << s->sample_shift;\ - } else {\ - *samples_16++ = (left) << s->sample_shift;\ - *samples_16++ = (right) << s->sample_shift;\ - }\ - }\ - break; - - switch (s->ch_mode) { - case FLAC_CHMODE_INDEPENDENT: - for (j = 0; j < s->blocksize; j++) { - for (i = 0; i < s->channels; i++) { - if (s->is32) - *samples_32++ = s->decoded[i][j] << s->sample_shift; - else - *samples_16++ = s->decoded[i][j] << s->sample_shift; - } - } - break; - case FLAC_CHMODE_LEFT_SIDE: - DECORRELATE(a,a-b) - case FLAC_CHMODE_RIGHT_SIDE: - DECORRELATE(a+b,b) - case FLAC_CHMODE_MID_SIDE: - DECORRELATE( (a-=b>>1) + b, a) - } + + s->dsp.decorrelate[s->ch_mode](s->frame.data, s->decoded, s->channels, + s->blocksize, s->sample_shift); if (bytes_read > buf_size) { av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size); |