diff options
Diffstat (limited to 'libavcodec/wmavoice.c')
| -rw-r--r-- | libavcodec/wmavoice.c | 530 |
1 files changed, 227 insertions, 303 deletions
diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index 8c7ec7b..68bb659 100644 --- a/libavcodec/wmavoice.c +++ b/libavcodec/wmavoice.c @@ -2,20 +2,20 @@ * Windows Media Audio Voice decoder. * Copyright (c) 2009 Ronald S. Bultje * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -30,10 +30,10 @@ #include "libavutil/channel_layout.h" #include "libavutil/float_dsp.h" #include "libavutil/mem.h" - +#include "libavutil/thread.h" #include "avcodec.h" -#include "bitstream.h" #include "internal.h" +#include "get_bits.h" #include "put_bits.h" #include "wmavoice_data.h" #include "celp_filters.h" @@ -43,7 +43,6 @@ #include "dct.h" #include "rdft.h" #include "sinewin.h" -#include "vlc.h" #define MAX_BLOCKS 8 ///< maximum number of blocks per frame #define MAX_LSPS 16 ///< maximum filter order @@ -106,26 +105,24 @@ static const struct frame_type_desc { uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs ///< (rather than just one single pulse) ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES - uint16_t frame_size; ///< the amount of bits that make up the block - ///< data (per frame) } frame_descs[17] = { - { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 }, - { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 }, - { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 }, - { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 }, - { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 }, - { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 }, - { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 }, - { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 }, - { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 }, - { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 }, - { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 }, - { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 }, - { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 }, - { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 }, - { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 }, - { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 }, - { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 } + { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0 }, + { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0 }, + { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0 }, + { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, + { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, + { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0 }, + { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, + { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, + { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, + { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, + { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, + { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, + { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, + { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, + { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, + { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, + { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 } }; /** @@ -136,7 +133,7 @@ typedef struct WMAVoiceContext { * @name Global values specified in the stream header / extradata or used all over. * @{ */ - BitstreamContext bc; ///< packet bitreader. During decoder init, + GetBitContext gb; ///< packet bitreader. During decoder init, ///< it contains the extradata from the ///< demuxer. During decoding, it contains ///< packet data. @@ -162,10 +159,6 @@ typedef struct WMAVoiceContext { int lsp_q_mode; ///< defines quantizer defaults [0, 1] int lsp_def_mode; ///< defines different sets of LSP defaults ///< [0, 1] - int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded - ///< per-frame (independent coding) - int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded - ///< per superframe (residual coding) int min_pitch_val; ///< base value for pitch parsing code int max_pitch_val; ///< max value + 1 for pitch parsing @@ -253,6 +246,7 @@ typedef struct WMAVoiceContext { int frame_cntr; ///< current frame index [0 - 0xFFFE]; is ///< only used for comfort noise in #pRNG() + int nb_superframes; ///< number of superframes in current packet float gain_pred_err[6]; ///< cache for gain prediction float excitation_history[MAX_SIGNAL_HISTORY]; ///< cache of the signal of previous @@ -296,20 +290,20 @@ typedef struct WMAVoiceContext { /** * Set up the variable bit mode (VBM) tree from container extradata. - * @param bc bit I/O context. - * The bit context (s->bc) should be loaded with byte 23-46 of the + * @param gb bit I/O context. + * The bit context (s->gb) should be loaded with byte 23-46 of the * container extradata (i.e. the ones containing the VBM tree). * @param vbm_tree pointer to array to which the decoded VBM tree will be * written. * @return 0 on success, <0 on error. */ -static av_cold int decode_vbmtree(BitstreamContext *bc, int8_t vbm_tree[25]) +static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) { int cntr[8] = { 0 }, n, res; memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); for (n = 0; n < 17; n++) { - res = bitstream_read(bc, 3); + res = get_bits(gb, 3); if (cntr[res] > 3) // should be >= 3 + (res == 7)) return -1; vbm_tree[res * 3 + cntr[res]++] = n; @@ -317,7 +311,7 @@ static av_cold int decode_vbmtree(BitstreamContext *bc, int8_t vbm_tree[25]) return 0; } -static av_cold void wmavoice_init_static_data(AVCodec *codec) +static av_cold void wmavoice_init_static_data(void) { static const uint8_t bits[] = { 2, 2, 2, 4, 4, 4, @@ -339,14 +333,45 @@ static av_cold void wmavoice_init_static_data(AVCodec *codec) bits, 1, 1, codes, 2, 2, 132); } +static av_cold void wmavoice_flush(AVCodecContext *ctx) +{ + WMAVoiceContext *s = ctx->priv_data; + int n; + + s->postfilter_agc = 0; + s->sframe_cache_size = 0; + s->skip_bits_next = 0; + for (n = 0; n < s->lsps; n++) + s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); + memset(s->excitation_history, 0, + sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); + memset(s->synth_history, 0, + sizeof(*s->synth_history) * MAX_LSPS); + memset(s->gain_pred_err, 0, + sizeof(s->gain_pred_err)); + + if (s->do_apf) { + memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, + sizeof(*s->synth_filter_out_buf) * s->lsps); + memset(s->dcf_mem, 0, + sizeof(*s->dcf_mem) * 2); + memset(s->zero_exc_pf, 0, + sizeof(*s->zero_exc_pf) * s->history_nsamples); + memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); + } +} + /** * Set up decoder with parameters from demuxer (extradata etc.). */ static av_cold int wmavoice_decode_init(AVCodecContext *ctx) { + static AVOnce init_static_once = AV_ONCE_INIT; int n, flags, pitch_range, lsp16_flag; WMAVoiceContext *s = ctx->priv_data; + ff_thread_once(&init_static_once, wmavoice_init_static_data); + /** * Extradata layout: * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), @@ -361,6 +386,11 @@ static av_cold int wmavoice_decode_init(AVCodecContext *ctx) ctx->extradata_size); return AVERROR_INVALIDDATA; } + if (ctx->block_align <= 0) { + av_log(ctx, AV_LOG_ERROR, "Invalid block alignment %d.\n", ctx->block_align); + return AVERROR_INVALIDDATA; + } + flags = AV_RL32(ctx->extradata + 18); s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); s->do_apf = flags & 0x1; @@ -391,18 +421,14 @@ static av_cold int wmavoice_decode_init(AVCodecContext *ctx) lsp16_flag = flags & 0x1000; if (lsp16_flag) { s->lsps = 16; - s->frame_lsp_bitsize = 34; - s->sframe_lsp_bitsize = 60; } else { s->lsps = 10; - s->frame_lsp_bitsize = 24; - s->sframe_lsp_bitsize = 48; } for (n = 0; n < s->lsps; n++) s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); - bitstream_init8(&s->bc, ctx->extradata + 22, ctx->extradata_size - 22); - if (decode_vbmtree(&s->bc, s->vbm_tree) < 0) { + init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); + if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); return AVERROR_INVALIDDATA; } @@ -485,7 +511,8 @@ static void adaptive_gain_control(float *out, const float *in, speech_energy += fabsf(speech_synth[i]); postfilter_energy += fabsf(in[i]); } - gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy; + gain_scale_factor = postfilter_energy == 0.0 ? 0.0 : + (1.0 - alpha) * speech_energy / postfilter_energy; for (i = 0; i < size; i++) { mem = alpha * mem + gain_scale_factor; @@ -520,7 +547,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch, float optimal_gain = 0, dot; const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], - *best_hist_ptr; + *best_hist_ptr = NULL; /* find best fitting point in history */ do { @@ -780,7 +807,7 @@ static void postfilter(WMAVoiceContext *s, const float *synth, *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], *synth_filter_in = zero_exc_pf; - assert(size <= MAX_FRAMESIZE / 2); + av_assert0(size <= MAX_FRAMESIZE / 2); /* generate excitation from input signal */ ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); @@ -849,7 +876,6 @@ static void dequant_lsps(double *lsps, int num, /** * @name LSP dequantization routines * LSP dequantization routines, for 10/16LSPs and independent/residual coding. - * @note we assume enough bits are available, caller should check. * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. * @{ @@ -857,7 +883,7 @@ static void dequant_lsps(double *lsps, int num, /** * Parse 10 independently-coded LSPs. */ -static void dequant_lsp10i(BitstreamContext *bc, double *lsps) +static void dequant_lsp10i(GetBitContext *gb, double *lsps) { static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; static const double mul_lsf[4] = { @@ -870,10 +896,10 @@ static void dequant_lsp10i(BitstreamContext *bc, double *lsps) }; uint16_t v[4]; - v[0] = bitstream_read(bc, 8); - v[1] = bitstream_read(bc, 6); - v[2] = bitstream_read(bc, 5); - v[3] = bitstream_read(bc, 5); + v[0] = get_bits(gb, 8); + v[1] = get_bits(gb, 6); + v[2] = get_bits(gb, 5); + v[3] = get_bits(gb, 5); dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, mul_lsf, base_lsf); @@ -883,7 +909,7 @@ static void dequant_lsp10i(BitstreamContext *bc, double *lsps) * Parse 10 independently-coded LSPs, and then derive the tables to * generate LSPs for the other frames from them (residual coding). */ -static void dequant_lsp10r(BitstreamContext *bc, +static void dequant_lsp10r(GetBitContext *gb, double *i_lsps, const double *old, double *a1, double *a2, int q_mode) { @@ -899,12 +925,12 @@ static void dequant_lsp10r(BitstreamContext *bc, uint16_t interpol, v[3]; int n; - dequant_lsp10i(bc, i_lsps); + dequant_lsp10i(gb, i_lsps); - interpol = bitstream_read(bc, 5); - v[0] = bitstream_read(bc, 7); - v[1] = bitstream_read(bc, 6); - v[2] = bitstream_read(bc, 6); + interpol = get_bits(gb, 5); + v[0] = get_bits(gb, 7); + v[1] = get_bits(gb, 6); + v[2] = get_bits(gb, 6); for (n = 0; n < 10; n++) { double delta = old[n] - i_lsps[n]; @@ -919,7 +945,7 @@ static void dequant_lsp10r(BitstreamContext *bc, /** * Parse 16 independently-coded LSPs. */ -static void dequant_lsp16i(BitstreamContext *bc, double *lsps) +static void dequant_lsp16i(GetBitContext *gb, double *lsps) { static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; static const double mul_lsf[5] = { @@ -934,11 +960,11 @@ static void dequant_lsp16i(BitstreamContext *bc, double *lsps) }; uint16_t v[5]; - v[0] = bitstream_read(bc, 8); - v[1] = bitstream_read(bc, 6); - v[2] = bitstream_read(bc, 7); - v[3] = bitstream_read(bc, 6); - v[4] = bitstream_read(bc, 7); + v[0] = get_bits(gb, 8); + v[1] = get_bits(gb, 6); + v[2] = get_bits(gb, 7); + v[3] = get_bits(gb, 6); + v[4] = get_bits(gb, 7); dequant_lsps( lsps, 5, v, vec_sizes, 2, wmavoice_dq_lsp16i1, mul_lsf, base_lsf); @@ -952,7 +978,7 @@ static void dequant_lsp16i(BitstreamContext *bc, double *lsps) * Parse 16 independently-coded LSPs, and then derive the tables to * generate LSPs for the other frames from them (residual coding). */ -static void dequant_lsp16r(BitstreamContext *bc, +static void dequant_lsp16r(GetBitContext *gb, double *i_lsps, const double *old, double *a1, double *a2, int q_mode) { @@ -968,12 +994,12 @@ static void dequant_lsp16r(BitstreamContext *bc, uint16_t interpol, v[3]; int n; - dequant_lsp16i(bc, i_lsps); + dequant_lsp16i(gb, i_lsps); - interpol = bitstream_read(bc, 5); - v[0] = bitstream_read(bc, 7); - v[1] = bitstream_read(bc, 7); - v[2] = bitstream_read(bc, 7); + interpol = get_bits(gb, 5); + v[0] = get_bits(gb, 7); + v[1] = get_bits(gb, 7); + v[2] = get_bits(gb, 7); for (n = 0; n < 16; n++) { double delta = old[n] - i_lsps[n]; @@ -999,10 +1025,10 @@ static void dequant_lsp16r(BitstreamContext *bc, * Parse the offset of the first pitch-adaptive window pulses, and * the distribution of pulses between the two blocks in this frame. * @param s WMA Voice decoding context private data - * @param bc bit I/O context + * @param gb bit I/O context * @param pitch pitch for each block in this frame */ -static void aw_parse_coords(WMAVoiceContext *s, BitstreamContext *bc, +static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, const int *pitch) { static const int16_t start_offset[94] = { @@ -1019,9 +1045,9 @@ static void aw_parse_coords(WMAVoiceContext *s, BitstreamContext *bc, /* position of pulse */ s->aw_idx_is_ext = 0; - if ((bits = bitstream_read(bc, 6)) >= 54) { + if ((bits = get_bits(gb, 6)) >= 54) { s->aw_idx_is_ext = 1; - bits += (bits - 54) * 3 + bitstream_read(bc, 2); + bits += (bits - 54) * 3 + get_bits(gb, 2); } /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count @@ -1049,12 +1075,12 @@ static void aw_parse_coords(WMAVoiceContext *s, BitstreamContext *bc, /** * Apply second set of pitch-adaptive window pulses. * @param s WMA Voice decoding context private data - * @param bc bit I/O context + * @param gb bit I/O context * @param block_idx block index in frame [0, 1] * @param fcb structure containing fixed codebook vector info * @return -1 on error, 0 otherwise */ -static int aw_pulse_set2(WMAVoiceContext *s, BitstreamContext *bc, +static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, int block_idx, AMRFixed *fcb) { uint16_t use_mask_mem[9]; // only 5 are used, rest is padding @@ -1108,7 +1134,7 @@ static int aw_pulse_set2(WMAVoiceContext *s, BitstreamContext *bc, } /* find the 'aidx'th offset that is not excluded */ - aidx = bitstream_read(bc, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); + aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); for (n = 0; n <= aidx; pulse_start++) { for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; if (idx >= MAX_FRAMESIZE / 2) { // find from zero @@ -1128,7 +1154,7 @@ static int aw_pulse_set2(WMAVoiceContext *s, BitstreamContext *bc, } fcb->x[fcb->n] = start_off; - fcb->y[fcb->n] = bitstream_read_bit(bc) ? -1.0 : 1.0; + fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; fcb->n++; /* set offset for next block, relative to start of that block */ @@ -1140,14 +1166,14 @@ static int aw_pulse_set2(WMAVoiceContext *s, BitstreamContext *bc, /** * Apply first set of pitch-adaptive window pulses. * @param s WMA Voice decoding context private data - * @param bc bit I/O context + * @param gb bit I/O context * @param block_idx block index in frame [0, 1] * @param fcb storage location for fixed codebook pulse info */ -static void aw_pulse_set1(WMAVoiceContext *s, BitstreamContext *bc, +static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, int block_idx, AMRFixed *fcb) { - int val = bitstream_read(bc, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); + int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); float v; if (s->aw_n_pulses[block_idx] > 0) { @@ -1241,7 +1267,7 @@ static int pRNG(int frame_cntr, int block_num, int block_size) * Parse hardcoded signal for a single block. * @note see #synth_block(). */ -static void synth_block_hardcoded(WMAVoiceContext *s, BitstreamContext *bc, +static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, int block_idx, int size, const struct frame_type_desc *frame_desc, float *excitation) @@ -1249,15 +1275,15 @@ static void synth_block_hardcoded(WMAVoiceContext *s, BitstreamContext *bc, float gain; int n, r_idx; - assert(size <= MAX_FRAMESIZE); + av_assert0(size <= MAX_FRAMESIZE); /* Set the offset from which we start reading wmavoice_std_codebook */ if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { r_idx = pRNG(s->frame_cntr, block_idx, size); gain = s->silence_gain; } else /* FCB_TYPE_HARDCODED */ { - r_idx = bitstream_read(bc, 8); - gain = wmavoice_gain_universal[bitstream_read(bc, 6)]; + r_idx = get_bits(gb, 8); + gain = wmavoice_gain_universal[get_bits(gb, 6)]; } /* Clear gain prediction parameters */ @@ -1272,7 +1298,7 @@ static void synth_block_hardcoded(WMAVoiceContext *s, BitstreamContext *bc, * Parse FCB/ACB signal for a single block. * @note see #synth_block(). */ -static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, +static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, int block_idx, int size, int block_pitch_sh2, const struct frame_type_desc *frame_desc, @@ -1285,7 +1311,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, int n, idx, gain_weight; AMRFixed fcb; - assert(size <= MAX_FRAMESIZE / 2); + av_assert0(size <= MAX_FRAMESIZE / 2); memset(pulses, 0, sizeof(*pulses) * size); fcb.pitch_lag = block_pitch_sh2 >> 2; @@ -1296,8 +1322,8 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, /* For the other frame types, this is where we apply the innovation * (fixed) codebook pulses of the speech signal. */ if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { - aw_pulse_set1(s, bc, block_idx, &fcb); - if (aw_pulse_set2(s, bc, block_idx, &fcb)) { + aw_pulse_set1(s, gb, block_idx, &fcb); + if (aw_pulse_set2(s, gb, block_idx, &fcb)) { /* Conceal the block with silence and return. * Skip the correct amount of bits to read the next * block from the correct offset. */ @@ -1306,7 +1332,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, for (n = 0; n < size; n++) excitation[n] = wmavoice_std_codebook[r_idx + n] * s->silence_gain; - bitstream_skip(bc, 7 + 1); + skip_bits(gb, 7 + 1); return; } } else /* FCB_TYPE_EXC_PULSES */ { @@ -1319,12 +1345,12 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, float sign; int pos1, pos2; - sign = bitstream_read_bit(bc) ? 1.0 : -1.0; - pos1 = bitstream_read(bc, offset_nbits); + sign = get_bits1(gb) ? 1.0 : -1.0; + pos1 = get_bits(gb, offset_nbits); fcb.x[fcb.n] = n + 5 * pos1; fcb.y[fcb.n++] = sign; if (n < frame_desc->dbl_pulses) { - pos2 = bitstream_read(bc, offset_nbits); + pos2 = get_bits(gb, offset_nbits); fcb.x[fcb.n] = n + 5 * pos2; fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; } @@ -1334,7 +1360,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, /* Calculate gain for adaptive & fixed codebook signal. * see ff_amr_set_fixed_gain(). */ - idx = bitstream_read(bc, 7); + idx = get_bits(gb, 7); fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, gain_coeff, 6) - 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); @@ -1393,10 +1419,9 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, /** * Parse data in a single block. - * @note we assume enough bits are available, caller should check. * * @param s WMA Voice decoding context private data - * @param bc bit I/O context + * @param gb bit I/O context * @param block_idx index of the to-be-read block * @param size amount of samples to be read in this block * @param block_pitch_sh2 pitch for this block << 2 @@ -1407,7 +1432,7 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc, * @param synth target memory for the speech synthesis filter output * @return 0 on success, <0 on error. */ -static void synth_block(WMAVoiceContext *s, BitstreamContext *bc, +static void synth_block(WMAVoiceContext *s, GetBitContext *gb, int block_idx, int size, int block_pitch_sh2, const double *lsps, const double *prev_lsps, @@ -1420,9 +1445,9 @@ static void synth_block(WMAVoiceContext *s, BitstreamContext *bc, int n; if (frame_desc->acb_type == ACB_TYPE_NONE) - synth_block_hardcoded(s, bc, block_idx, size, frame_desc, excitation); + synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); else - synth_block_fcb_acb(s, bc, block_idx, size, block_pitch_sh2, + synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, frame_desc, excitation); /* convert interpolated LSPs to LPCs */ @@ -1437,10 +1462,9 @@ static void synth_block(WMAVoiceContext *s, BitstreamContext *bc, /** * Synthesize output samples for a single frame. - * @note we assume enough bits are available, caller should check. * * @param ctx WMA Voice decoder context - * @param bc bit I/O context (s->bc or one for cross-packet superframes) + * @param gb bit I/O context (s->gb or one for cross-packet superframes) * @param frame_idx Frame number within superframe [0-2] * @param samples pointer to output sample buffer, has space for at least 160 * samples @@ -1450,17 +1474,17 @@ static void synth_block(WMAVoiceContext *s, BitstreamContext *bc, * @param synth target buffer for synthesized speech data * @return 0 on success, <0 on error. */ -static int synth_frame(AVCodecContext *ctx, BitstreamContext *bc, - int frame_idx, float *samples, +static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, + float *samples, const double *lsps, const double *prev_lsps, float *excitation, float *synth) { WMAVoiceContext *s = ctx->priv_data; - int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val; - int pitch[MAX_BLOCKS], last_block_pitch; + int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val); + int pitch[MAX_BLOCKS], av_uninit(last_block_pitch); /* Parse frame type ("frame header"), see frame_descs */ - int bd_idx = s->vbm_tree[bitstream_read_vlc(bc, frame_type_vlc.table, 6, 3)], block_nsamples; + int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples; if (bd_idx < 0) { av_log(ctx, AV_LOG_ERROR, @@ -1478,7 +1502,7 @@ static int synth_frame(AVCodecContext *ctx, BitstreamContext *bc, * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; - cur_pitch_val = s->min_pitch_val + bitstream_read(bc, s->pitch_nbits); + cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); if (s->last_acb_type == ACB_TYPE_NONE || 20 * abs(cur_pitch_val - s->last_pitch_val) > @@ -1502,10 +1526,10 @@ static int synth_frame(AVCodecContext *ctx, BitstreamContext *bc, /* Global gain (if silence) and pitch-adaptive window coordinates */ switch (frame_descs[bd_idx].fcb_type) { case FCB_TYPE_SILENCE: - s->silence_gain = wmavoice_gain_silence[bitstream_read(bc, 8)]; + s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; break; case FCB_TYPE_AW_PULSES: - aw_parse_coords(s, bc, pitch); + aw_parse_coords(s, gb, pitch); break; } @@ -1526,10 +1550,10 @@ static int synth_frame(AVCodecContext *ctx, BitstreamContext *bc, t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; if (n == 0) { - block_pitch = bitstream_read(bc, s->block_pitch_nbits); + block_pitch = get_bits(gb, s->block_pitch_nbits); } else block_pitch = last_block_pitch - s->block_delta_pitch_hrange + - bitstream_read(bc, s->block_delta_pitch_nbits); + get_bits(gb, s->block_delta_pitch_nbits); /* Convert last_ so that any next delta is within _range */ last_block_pitch = av_clip(block_pitch, s->block_delta_pitch_hrange, @@ -1567,7 +1591,7 @@ static int synth_frame(AVCodecContext *ctx, BitstreamContext *bc, break; } - synth_block(s, bc, n, block_nsamples, bl_pitch_sh2, + synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, lsps, prev_lsps, &frame_descs[bd_idx], &excitation[n * block_nsamples], &synth[n * block_nsamples]); @@ -1656,85 +1680,9 @@ static void stabilize_lsps(double *lsps, int num) } /** - * Test if there's enough bits to read 1 superframe. - * - * @param orig_bc bit I/O context used for reading. This function - * does not modify the state of the bitreader; it - * only uses it to copy the current stream position - * @param s WMA Voice decoding context private data - * @return < 0 on error, 1 on not enough bits or 0 if OK. - */ -static int check_bits_for_superframe(BitstreamContext *orig_bc, - WMAVoiceContext *s) -{ - BitstreamContext s_bc, *bc = &s_bc; - int n, need_bits, bd_idx; - const struct frame_type_desc *frame_desc; - - /* initialize a copy */ - *bc = *orig_bc; - - /* superframe header */ - if (bitstream_bits_left(bc) < 14) - return 1; - if (!bitstream_read_bit(bc)) - return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe - if (bitstream_read_bit(bc)) bitstream_skip(bc, 12); // number of samples in superframe - if (s->has_residual_lsps) { // residual LSPs (for all frames) - if (bitstream_bits_left(bc) < s->sframe_lsp_bitsize) - return 1; - bitstream_skip(bc, s->sframe_lsp_bitsize); - } - - /* frames */ - for (n = 0; n < MAX_FRAMES; n++) { - int aw_idx_is_ext = 0; - - if (!s->has_residual_lsps) { // independent LSPs (per-frame) - if (bitstream_bits_left(bc) < s->frame_lsp_bitsize) - return 1; - bitstream_skip(bc, s->frame_lsp_bitsize); - } - bd_idx = s->vbm_tree[bitstream_read_vlc(bc, frame_type_vlc.table, 6, 3)]; - if (bd_idx < 0) - return AVERROR_INVALIDDATA; // invalid frame type VLC code - frame_desc = &frame_descs[bd_idx]; - if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { - if (bitstream_bits_left(bc) < s->pitch_nbits) - return 1; - bitstream_skip(bc, s->pitch_nbits); - } - if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { - bitstream_skip(bc, 8); - } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { - int tmp = bitstream_read(bc, 6); - if (tmp >= 0x36) { - bitstream_skip(bc, 2); - aw_idx_is_ext = 1; - } - } - - /* blocks */ - if (frame_desc->acb_type == ACB_TYPE_HAMMING) { - need_bits = s->block_pitch_nbits + - (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; - } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { - need_bits = 2 * !aw_idx_is_ext; - } else - need_bits = 0; - need_bits += frame_desc->frame_size; - if (bitstream_bits_left(bc) < need_bits) - return 1; - bitstream_skip(bc, need_bits); - } - - return 0; -} - -/** * Synthesize output samples for a single superframe. If we have any data * cached in s->sframe_cache, that will be used instead of whatever is loaded - * in s->bc. + * in s->gb. * * WMA Voice superframes contain 3 frames, each containing 160 audio samples, * to give a total of 480 samples per frame. See #synth_frame() for frame @@ -1752,8 +1700,8 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, int *got_frame_ptr) { WMAVoiceContext *s = ctx->priv_data; - BitstreamContext *bc = &s->bc, s_bc; - int n, res, n_samples = 480; + GetBitContext *gb = &s->gb, s_gb; + int n, res, n_samples = MAX_SFRAMESIZE; double lsps[MAX_FRAMES][MAX_LSPS]; const double *mean_lsf = s->lsps == 16 ? wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; @@ -1767,35 +1715,30 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, s->history_nsamples * sizeof(*excitation)); if (s->sframe_cache_size > 0) { - bc = &s_bc; - bitstream_init(bc, s->sframe_cache, s->sframe_cache_size); + gb = &s_gb; + init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); s->sframe_cache_size = 0; } - if ((res = check_bits_for_superframe(bc, s)) == 1) { - *got_frame_ptr = 0; - return 1; - } else if (res < 0) - return res; - /* First bit is speech/music bit, it differentiates between WMAVoice * speech samples (the actual codec) and WMAVoice music samples, which * are really WMAPro-in-WMAVoice-superframes. I've never seen those in * the wild yet. */ - if (!bitstream_read_bit(bc)) { + if (!get_bits1(gb)) { avpriv_request_sample(ctx, "WMAPro-in-WMAVoice"); return AVERROR_PATCHWELCOME; } /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ - if (bitstream_read_bit(bc)) { - if ((n_samples = bitstream_read(bc, 12)) > 480) { + if (get_bits1(gb)) { + if ((n_samples = get_bits(gb, 12)) > MAX_SFRAMESIZE) { av_log(ctx, AV_LOG_ERROR, - "Superframe encodes >480 samples (%d), not allowed\n", - n_samples); + "Superframe encodes > %d samples (%d), not allowed\n", + MAX_SFRAMESIZE, n_samples); return AVERROR_INVALIDDATA; } } + /* Parse LSPs, if global for the superframe (can also be per-frame). */ if (s->has_residual_lsps) { double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; @@ -1804,9 +1747,9 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; if (s->lsps == 10) { - dequant_lsp10r(bc, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); + dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); } else /* s->lsps == 16 */ - dequant_lsp16r(bc, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); + dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); for (n = 0; n < s->lsps; n++) { lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); @@ -1817,12 +1760,14 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, stabilize_lsps(lsps[n], s->lsps); } + /* synth_superframe can run multiple times per packet + * free potential previous frame */ + av_frame_unref(frame); + /* get output buffer */ - frame->nb_samples = 480; - if ((res = ff_get_buffer(ctx, frame, 0)) < 0) { - av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n"); + frame->nb_samples = MAX_SFRAMESIZE; + if ((res = ff_get_buffer(ctx, frame, 0)) < 0) return res; - } frame->nb_samples = n_samples; samples = (float *)frame->data[0]; @@ -1832,16 +1777,16 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, int m; if (s->lsps == 10) { - dequant_lsp10i(bc, lsps[n]); + dequant_lsp10i(gb, lsps[n]); } else /* s->lsps == 16 */ - dequant_lsp16i(bc, lsps[n]); + dequant_lsp16i(gb, lsps[n]); for (m = 0; m < s->lsps; m++) lsps[n][m] += mean_lsf[m]; stabilize_lsps(lsps[n], s->lsps); } - if ((res = synth_frame(ctx, bc, n, + if ((res = synth_frame(ctx, gb, n, &samples[n * MAX_FRAMESIZE], lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], &excitation[s->history_nsamples + n * MAX_FRAMESIZE], @@ -1854,9 +1799,14 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, /* Statistics? FIXME - we don't check for length, a slight overrun * will be caught by internal buffer padding, and anything else * will be skipped, not read. */ - if (bitstream_read_bit(bc)) { - res = bitstream_read(bc, 4); - bitstream_skip(bc, 10 * (res + 1)); + if (get_bits1(gb)) { + res = get_bits(gb, 4); + skip_bits(gb, 10 * (res + 1)); + } + + if (get_bits_left(gb) < 0) { + wmavoice_flush(ctx); + return AVERROR_INVALIDDATA; } *got_frame_ptr = 1; @@ -1880,35 +1830,32 @@ static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, * decoder). * * @param s WMA Voice decoding context private data - * @return 1 if not enough bits were available, or 0 on success. + * @return <0 on error, nb_superframes on success. */ static int parse_packet_header(WMAVoiceContext *s) { - BitstreamContext *bc = &s->bc; - unsigned int res; + GetBitContext *gb = &s->gb; + unsigned int res, n_superframes = 0; - if (bitstream_bits_left(bc) < 11) - return 1; - bitstream_skip(bc, 4); // packet sequence number - s->has_residual_lsps = bitstream_read_bit(bc); + skip_bits(gb, 4); // packet sequence number + s->has_residual_lsps = get_bits1(gb); do { - res = bitstream_read(bc, 6); // number of superframes per packet - // (minus first one if there is spillover) - if (bitstream_bits_left(bc) < 6 * (res == 0x3F) + s->spillover_bitsize) - return 1; + res = get_bits(gb, 6); // number of superframes per packet + // (minus first one if there is spillover) + n_superframes += res; } while (res == 0x3F); - s->spillover_nbits = bitstream_read(bc, s->spillover_bitsize); + s->spillover_nbits = get_bits(gb, s->spillover_bitsize); - return 0; + return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA; } /** - * Copy (unaligned) bits from bc/data/size to pb. + * Copy (unaligned) bits from gb/data/size to pb. * * @param pb target buffer to copy bits into * @param data source buffer to copy bits from * @param size size of the source data, in bytes - * @param bc bit I/O context specifying the current position in the source. + * @param gb bit I/O context specifying the current position in the source. * data. This function might use this to align the bit position to * a whole-byte boundary before calling #avpriv_copy_bits() on aligned * source data @@ -1919,18 +1866,18 @@ static int parse_packet_header(WMAVoiceContext *s) */ static void copy_bits(PutBitContext *pb, const uint8_t *data, int size, - BitstreamContext *bc, int nbits) + GetBitContext *gb, int nbits) { int rmn_bytes, rmn_bits; - rmn_bits = rmn_bytes = bitstream_bits_left(bc); + rmn_bits = rmn_bytes = get_bits_left(gb); if (rmn_bits < nbits) return; if (nbits > pb->size_in_bits - put_bits_count(pb)) return; rmn_bits &= 7; rmn_bytes >>= 3; if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) - put_bits(pb, rmn_bits, bitstream_read(bc, rmn_bits)); + put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); avpriv_copy_bits(pb, data + size - rmn_bytes, FFMIN(nbits - rmn_bits, rmn_bytes << 3)); } @@ -1950,70 +1897,76 @@ static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { WMAVoiceContext *s = ctx->priv_data; - BitstreamContext *bc = &s->bc; + GetBitContext *gb = &s->gb; int size, res, pos; /* Packets are sometimes a multiple of ctx->block_align, with a packet - * header at each ctx->block_align bytes. However, Libav's ASF demuxer + * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer * feeds us ASF packets, which may concatenate multiple "codec" packets * in a single "muxer" packet, so we artificially emulate that by * capping the packet size at ctx->block_align. */ for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); - if (!size) { - *got_frame_ptr = 0; - return 0; - } - bitstream_init8(&s->bc, avpkt->data, size); + init_get_bits8(&s->gb, avpkt->data, size); /* size == ctx->block_align is used to indicate whether we are dealing with * a new packet or a packet of which we already read the packet header * previously. */ - if (size == ctx->block_align) { // new packet header - if ((res = parse_packet_header(s)) < 0) - return res; + if (!(size % ctx->block_align)) { // new packet header + if (!size) { + s->spillover_nbits = 0; + s->nb_superframes = 0; + } else { + if ((res = parse_packet_header(s)) < 0) + return res; + s->nb_superframes = res; + } /* If the packet header specifies a s->spillover_nbits, then we want * to push out all data of the previous packet (+ spillover) before * continuing to parse new superframes in the current packet. */ - if (s->spillover_nbits > 0) { - if (s->sframe_cache_size > 0) { - int cnt = bitstream_tell(bc); - copy_bits(&s->pb, avpkt->data, size, bc, s->spillover_nbits); - flush_put_bits(&s->pb); - s->sframe_cache_size += s->spillover_nbits; - if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 && - *got_frame_ptr) { - cnt += s->spillover_nbits; - s->skip_bits_next = cnt & 7; - return cnt >> 3; - } else - bitstream_skip (bc, s->spillover_nbits - cnt + - bitstream_tell(bc)); // resync + if (s->sframe_cache_size > 0) { + int cnt = get_bits_count(gb); + if (cnt + s->spillover_nbits > avpkt->size * 8) { + s->spillover_nbits = avpkt->size * 8 - cnt; + } + copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); + flush_put_bits(&s->pb); + s->sframe_cache_size += s->spillover_nbits; + if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 && + *got_frame_ptr) { + cnt += s->spillover_nbits; + s->skip_bits_next = cnt & 7; + res = cnt >> 3; + return res; } else - bitstream_skip(bc, s->spillover_nbits); // resync + skip_bits_long (gb, s->spillover_nbits - cnt + + get_bits_count(gb)); // resync + } else if (s->spillover_nbits) { + skip_bits_long(gb, s->spillover_nbits); // resync } } else if (s->skip_bits_next) - bitstream_skip(bc, s->skip_bits_next); + skip_bits(gb, s->skip_bits_next); /* Try parsing superframes in current packet */ s->sframe_cache_size = 0; s->skip_bits_next = 0; - pos = bitstream_bits_left(bc); - if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { - return res; - } else if (*got_frame_ptr) { - int cnt = bitstream_tell(bc); - s->skip_bits_next = cnt & 7; - return cnt >> 3; + pos = get_bits_left(gb); + if (s->nb_superframes-- == 0) { + *got_frame_ptr = 0; + return size; + } else if (s->nb_superframes > 0) { + if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { + return res; + } else if (*got_frame_ptr) { + int cnt = get_bits_count(gb); + s->skip_bits_next = cnt & 7; + res = cnt >> 3; + return res; + } } else if ((s->sframe_cache_size = pos) > 0) { - /* rewind bit reader to start of last (incomplete) superframe... */ - bitstream_init8(bc, avpkt->data, size); - bitstream_skip(bc, (size << 3) - pos); - assert(bitstream_bits_left(bc) == pos); - - /* ...and cache it for spillover in next packet */ + /* ... cache it for spillover in next packet */ init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); - copy_bits(&s->pb, avpkt->data, size, bc, s->sframe_cache_size); + copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); // FIXME bad - just copy bytes as whole and add use the // skip_bits_next field } @@ -2035,34 +1988,6 @@ static av_cold int wmavoice_decode_end(AVCodecContext *ctx) return 0; } -static av_cold void wmavoice_flush(AVCodecContext *ctx) -{ - WMAVoiceContext *s = ctx->priv_data; - int n; - - s->postfilter_agc = 0; - s->sframe_cache_size = 0; - s->skip_bits_next = 0; - for (n = 0; n < s->lsps; n++) - s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); - memset(s->excitation_history, 0, - sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); - memset(s->synth_history, 0, - sizeof(*s->synth_history) * MAX_LSPS); - memset(s->gain_pred_err, 0, - sizeof(s->gain_pred_err)); - - if (s->do_apf) { - memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, - sizeof(*s->synth_filter_out_buf) * s->lsps); - memset(s->dcf_mem, 0, - sizeof(*s->dcf_mem) * 2); - memset(s->zero_exc_pf, 0, - sizeof(*s->zero_exc_pf) * s->history_nsamples); - memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); - } -} - AVCodec ff_wmavoice_decoder = { .name = "wmavoice", .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), @@ -2070,9 +1995,8 @@ AVCodec ff_wmavoice_decoder = { .id = AV_CODEC_ID_WMAVOICE, .priv_data_size = sizeof(WMAVoiceContext), .init = wmavoice_decode_init, - .init_static_data = wmavoice_init_static_data, .close = wmavoice_decode_end, .decode = wmavoice_decode_packet, - .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, + .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, .flush = wmavoice_flush, }; |
