diff options
Diffstat (limited to 'contrib/ffmpeg/libavcodec/dca.c')
| -rw-r--r-- | contrib/ffmpeg/libavcodec/dca.c | 1322 |
1 files changed, 0 insertions, 1322 deletions
diff --git a/contrib/ffmpeg/libavcodec/dca.c b/contrib/ffmpeg/libavcodec/dca.c deleted file mode 100644 index a57dcdc44..000000000 --- a/contrib/ffmpeg/libavcodec/dca.c +++ /dev/null @@ -1,1322 +0,0 @@ -/* - * DCA compatible decoder - * Copyright (C) 2004 Gildas Bazin - * Copyright (C) 2004 Benjamin Zores - * Copyright (C) 2006 Benjamin Larsson - * Copyright (C) 2007 Konstantin Shishkov - * - * 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 dca.c - */ - -#include <math.h> -#include <stddef.h> -#include <stdio.h> - -#include "avcodec.h" -#include "dsputil.h" -#include "bitstream.h" -#include "dcadata.h" -#include "dcahuff.h" -#include "parser.h" - -/** DCA syncwords, also used for bitstream type detection */ -//@{ -#define DCA_MARKER_RAW_BE 0x7FFE8001 -#define DCA_MARKER_RAW_LE 0xFE7F0180 -#define DCA_MARKER_14B_BE 0x1FFFE800 -#define DCA_MARKER_14B_LE 0xFF1F00E8 -//@} - -//#define TRACE - -#define DCA_PRIM_CHANNELS_MAX (5) -#define DCA_SUBBANDS (32) -#define DCA_ABITS_MAX (32) /* Should be 28 */ -#define DCA_SUBSUBFAMES_MAX (4) -#define DCA_LFE_MAX (3) - -enum DCAMode { - DCA_MONO = 0, - DCA_CHANNEL, - DCA_STEREO, - DCA_STEREO_SUMDIFF, - DCA_STEREO_TOTAL, - DCA_3F, - DCA_2F1R, - DCA_3F1R, - DCA_2F2R, - DCA_3F2R, - DCA_4F2R -}; - -#define DCA_DOLBY 101 /* FIXME */ - -#define DCA_CHANNEL_BITS 6 -#define DCA_CHANNEL_MASK 0x3F - -#define DCA_LFE 0x80 - -#define HEADER_SIZE 14 -#define CONVERT_BIAS 384 - -#define DCA_MAX_FRAME_SIZE 16383 - -/** Bit allocation */ -typedef struct { - int offset; ///< code values offset - int maxbits[8]; ///< max bits in VLC - int wrap; ///< wrap for get_vlc2() - VLC vlc[8]; ///< actual codes -} BitAlloc; - -static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select -static BitAlloc dca_tmode; ///< transition mode VLCs -static BitAlloc dca_scalefactor; ///< scalefactor VLCs -static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs - -/** Pre-calculated cosine modulation coefs for the QMF */ -static float cos_mod[544]; - -static int av_always_inline get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx) -{ - return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset; -} - -typedef struct { - AVCodecContext *avctx; - /* Frame header */ - int frame_type; ///< type of the current frame - int samples_deficit; ///< deficit sample count - int crc_present; ///< crc is present in the bitstream - int sample_blocks; ///< number of PCM sample blocks - int frame_size; ///< primary frame byte size - int amode; ///< audio channels arrangement - int sample_rate; ///< audio sampling rate - int bit_rate; ///< transmission bit rate - - int downmix; ///< embedded downmix enabled - int dynrange; ///< embedded dynamic range flag - int timestamp; ///< embedded time stamp flag - int aux_data; ///< auxiliary data flag - int hdcd; ///< source material is mastered in HDCD - int ext_descr; ///< extension audio descriptor flag - int ext_coding; ///< extended coding flag - int aspf; ///< audio sync word insertion flag - int lfe; ///< low frequency effects flag - int predictor_history; ///< predictor history flag - int header_crc; ///< header crc check bytes - int multirate_inter; ///< multirate interpolator switch - int version; ///< encoder software revision - int copy_history; ///< copy history - int source_pcm_res; ///< source pcm resolution - int front_sum; ///< front sum/difference flag - int surround_sum; ///< surround sum/difference flag - int dialog_norm; ///< dialog normalisation parameter - - /* Primary audio coding header */ - int subframes; ///< number of subframes - int prim_channels; ///< number of primary audio channels - int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count - int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband - int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index - int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book - int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book - int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select - int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select - float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment - - /* Primary audio coding side information */ - int subsubframes; ///< number of subsubframes - int partial_samples; ///< partial subsubframe samples count - int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not) - int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs - int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index - int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients) - int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient) - int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook - int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors - int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients - int dynrange_coef; ///< dynamic range coefficient - - int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands - - float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX * - 2 /*history */ ]; ///< Low frequency effect data - int lfe_scale_factor; - - /* Subband samples history (for ADPCM) */ - float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; - float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]; - float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64]; - - int output; ///< type of output - int bias; ///< output bias - - DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */ - DECLARE_ALIGNED_16(int16_t, tsamples[1536]); - - uint8_t dca_buffer[DCA_MAX_FRAME_SIZE]; - int dca_buffer_size; ///< how much data is in the dca_buffer - - GetBitContext gb; - /* Current position in DCA frame */ - int current_subframe; - int current_subsubframe; - - int debug_flag; ///< used for suppressing repeated error messages output - DSPContext dsp; -} DCAContext; - -static void dca_init_vlcs() -{ - static int vlcs_inited = 0; - int i, j; - - if (vlcs_inited) - return; - - dca_bitalloc_index.offset = 1; - dca_bitalloc_index.wrap = 1; - for (i = 0; i < 5; i++) - init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12, - bitalloc_12_bits[i], 1, 1, - bitalloc_12_codes[i], 2, 2, 1); - dca_scalefactor.offset = -64; - dca_scalefactor.wrap = 2; - for (i = 0; i < 5; i++) - init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129, - scales_bits[i], 1, 1, - scales_codes[i], 2, 2, 1); - dca_tmode.offset = 0; - dca_tmode.wrap = 1; - for (i = 0; i < 4; i++) - init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4, - tmode_bits[i], 1, 1, - tmode_codes[i], 2, 2, 1); - - for(i = 0; i < 10; i++) - for(j = 0; j < 7; j++){ - if(!bitalloc_codes[i][j]) break; - dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i]; - dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4); - init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j], - bitalloc_sizes[i], - bitalloc_bits[i][j], 1, 1, - bitalloc_codes[i][j], 2, 2, 1); - } - vlcs_inited = 1; -} - -static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) -{ - while(len--) - *dst++ = get_bits(gb, bits); -} - -static int dca_parse_frame_header(DCAContext * s) -{ - int i, j; - static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 }; - static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; - static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; - - s->bias = CONVERT_BIAS; - - init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); - - /* Sync code */ - get_bits(&s->gb, 32); - - /* Frame header */ - s->frame_type = get_bits(&s->gb, 1); - s->samples_deficit = get_bits(&s->gb, 5) + 1; - s->crc_present = get_bits(&s->gb, 1); - s->sample_blocks = get_bits(&s->gb, 7) + 1; - s->frame_size = get_bits(&s->gb, 14) + 1; - if (s->frame_size < 95) - return -1; - s->amode = get_bits(&s->gb, 6); - s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)]; - if (!s->sample_rate) - return -1; - s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)]; - if (!s->bit_rate) - return -1; - - s->downmix = get_bits(&s->gb, 1); - s->dynrange = get_bits(&s->gb, 1); - s->timestamp = get_bits(&s->gb, 1); - s->aux_data = get_bits(&s->gb, 1); - s->hdcd = get_bits(&s->gb, 1); - s->ext_descr = get_bits(&s->gb, 3); - s->ext_coding = get_bits(&s->gb, 1); - s->aspf = get_bits(&s->gb, 1); - s->lfe = get_bits(&s->gb, 2); - s->predictor_history = get_bits(&s->gb, 1); - - /* TODO: check CRC */ - if (s->crc_present) - s->header_crc = get_bits(&s->gb, 16); - - s->multirate_inter = get_bits(&s->gb, 1); - s->version = get_bits(&s->gb, 4); - s->copy_history = get_bits(&s->gb, 2); - s->source_pcm_res = get_bits(&s->gb, 3); - s->front_sum = get_bits(&s->gb, 1); - s->surround_sum = get_bits(&s->gb, 1); - s->dialog_norm = get_bits(&s->gb, 4); - - /* FIXME: channels mixing levels */ - s->output = DCA_STEREO; - -#ifdef TRACE - av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type); - av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit); - av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present); - av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n", - s->sample_blocks, s->sample_blocks * 32); - av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size); - av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n", - s->amode, dca_channels[s->amode]); - av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n", - s->sample_rate, dca_sample_rates[s->sample_rate]); - av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n", - s->bit_rate, dca_bit_rates[s->bit_rate]); - av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix); - av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange); - av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp); - av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data); - av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd); - av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr); - av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding); - av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf); - av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe); - av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n", - s->predictor_history); - av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc); - av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n", - s->multirate_inter); - av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version); - av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history); - av_log(s->avctx, AV_LOG_DEBUG, - "source pcm resolution: %i (%i bits/sample)\n", - s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]); - av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum); - av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum); - av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); -#endif - - /* Primary audio coding header */ - s->subframes = get_bits(&s->gb, 4) + 1; - s->prim_channels = get_bits(&s->gb, 3) + 1; - - - for (i = 0; i < s->prim_channels; i++) { - s->subband_activity[i] = get_bits(&s->gb, 5) + 2; - if (s->subband_activity[i] > DCA_SUBBANDS) - s->subband_activity[i] = DCA_SUBBANDS; - } - for (i = 0; i < s->prim_channels; i++) { - s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; - if (s->vq_start_subband[i] > DCA_SUBBANDS) - s->vq_start_subband[i] = DCA_SUBBANDS; - } - get_array(&s->gb, s->joint_intensity, s->prim_channels, 3); - get_array(&s->gb, s->transient_huffman, s->prim_channels, 2); - get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3); - get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3); - - /* Get codebooks quantization indexes */ - memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); - for (j = 1; j < 11; j++) - for (i = 0; i < s->prim_channels; i++) - s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); - - /* Get scale factor adjustment */ - for (j = 0; j < 11; j++) - for (i = 0; i < s->prim_channels; i++) - s->scalefactor_adj[i][j] = 1; - - for (j = 1; j < 11; j++) - for (i = 0; i < s->prim_channels; i++) - if (s->quant_index_huffman[i][j] < thr[j]) - s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; - - if (s->crc_present) { - /* Audio header CRC check */ - get_bits(&s->gb, 16); - } - - s->current_subframe = 0; - s->current_subsubframe = 0; - -#ifdef TRACE - av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes); - av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels); - for(i = 0; i < s->prim_channels; i++){ - av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]); - av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]); - av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]); - av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]); - av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]); - av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]); - av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:"); - for (j = 0; j < 11; j++) - av_log(s->avctx, AV_LOG_DEBUG, " %i", - s->quant_index_huffman[i][j]); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:"); - for (j = 0; j < 11; j++) - av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } -#endif - - return 0; -} - - -static inline int get_scale(GetBitContext *gb, int level, int index, int value) -{ - if (level < 5) { - /* huffman encoded */ - value += get_bitalloc(gb, &dca_scalefactor, index); - } else if(level < 8) - value = get_bits(gb, level + 1); - return value; -} - -static int dca_subframe_header(DCAContext * s) -{ - /* Primary audio coding side information */ - int j, k; - - s->subsubframes = get_bits(&s->gb, 2) + 1; - s->partial_samples = get_bits(&s->gb, 3); - for (j = 0; j < s->prim_channels; j++) { - for (k = 0; k < s->subband_activity[j]; k++) - s->prediction_mode[j][k] = get_bits(&s->gb, 1); - } - - /* Get prediction codebook */ - for (j = 0; j < s->prim_channels; j++) { - for (k = 0; k < s->subband_activity[j]; k++) { - if (s->prediction_mode[j][k] > 0) { - /* (Prediction coefficient VQ address) */ - s->prediction_vq[j][k] = get_bits(&s->gb, 12); - } - } - } - - /* Bit allocation index */ - for (j = 0; j < s->prim_channels; j++) { - for (k = 0; k < s->vq_start_subband[j]; k++) { - if (s->bitalloc_huffman[j] == 6) - s->bitalloc[j][k] = get_bits(&s->gb, 5); - else if (s->bitalloc_huffman[j] == 5) - s->bitalloc[j][k] = get_bits(&s->gb, 4); - else { - s->bitalloc[j][k] = - get_bitalloc(&s->gb, &dca_bitalloc_index, j); - } - - if (s->bitalloc[j][k] > 26) { -// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n", -// j, k, s->bitalloc[j][k]); - return -1; - } - } - } - - /* Transition mode */ - for (j = 0; j < s->prim_channels; j++) { - for (k = 0; k < s->subband_activity[j]; k++) { - s->transition_mode[j][k] = 0; - if (s->subsubframes > 1 && - k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { - s->transition_mode[j][k] = - get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); - } - } - } - - for (j = 0; j < s->prim_channels; j++) { - uint32_t *scale_table; - int scale_sum; - - memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); - - if (s->scalefactor_huffman[j] == 6) - scale_table = (uint32_t *) scale_factor_quant7; - else - scale_table = (uint32_t *) scale_factor_quant6; - - /* When huffman coded, only the difference is encoded */ - scale_sum = 0; - - for (k = 0; k < s->subband_activity[j]; k++) { - if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { - scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], j, scale_sum); - s->scale_factor[j][k][0] = scale_table[scale_sum]; - } - - if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) { - /* Get second scale factor */ - scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], j, scale_sum); - s->scale_factor[j][k][1] = scale_table[scale_sum]; - } - } - } - - /* Joint subband scale factor codebook select */ - for (j = 0; j < s->prim_channels; j++) { - /* Transmitted only if joint subband coding enabled */ - if (s->joint_intensity[j] > 0) - s->joint_huff[j] = get_bits(&s->gb, 3); - } - - /* Scale factors for joint subband coding */ - for (j = 0; j < s->prim_channels; j++) { - int source_channel; - - /* Transmitted only if joint subband coding enabled */ - if (s->joint_intensity[j] > 0) { - int scale = 0; - source_channel = s->joint_intensity[j] - 1; - - /* When huffman coded, only the difference is encoded - * (is this valid as well for joint scales ???) */ - - for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { - scale = get_scale(&s->gb, s->joint_huff[j], j, 0); - scale += 64; /* bias */ - s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ - } - - if (!s->debug_flag & 0x02) { - av_log(s->avctx, AV_LOG_DEBUG, - "Joint stereo coding not supported\n"); - s->debug_flag |= 0x02; - } - } - } - - /* Stereo downmix coefficients */ - if (s->prim_channels > 2 && s->downmix) { - for (j = 0; j < s->prim_channels; j++) { - s->downmix_coef[j][0] = get_bits(&s->gb, 7); - s->downmix_coef[j][1] = get_bits(&s->gb, 7); - } - } - - /* Dynamic range coefficient */ - if (s->dynrange) - s->dynrange_coef = get_bits(&s->gb, 8); - - /* Side information CRC check word */ - if (s->crc_present) { - get_bits(&s->gb, 16); - } - - /* - * Primary audio data arrays - */ - - /* VQ encoded high frequency subbands */ - for (j = 0; j < s->prim_channels; j++) - for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) - /* 1 vector -> 32 samples */ - s->high_freq_vq[j][k] = get_bits(&s->gb, 10); - - /* Low frequency effect data */ - if (s->lfe) { - /* LFE samples */ - int lfe_samples = 2 * s->lfe * s->subsubframes; - float lfe_scale; - - for (j = lfe_samples; j < lfe_samples * 2; j++) { - /* Signed 8 bits int */ - s->lfe_data[j] = get_sbits(&s->gb, 8); - } - - /* Scale factor index */ - s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)]; - - /* Quantization step size * scale factor */ - lfe_scale = 0.035 * s->lfe_scale_factor; - - for (j = lfe_samples; j < lfe_samples * 2; j++) - s->lfe_data[j] *= lfe_scale; - } - -#ifdef TRACE - av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes); - av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n", - s->partial_samples); - for (j = 0; j < s->prim_channels; j++) { - av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:"); - for (k = 0; k < s->subband_activity[j]; k++) - av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } - for (j = 0; j < s->prim_channels; j++) { - for (k = 0; k < s->subband_activity[j]; k++) - av_log(s->avctx, AV_LOG_DEBUG, - "prediction coefs: %f, %f, %f, %f\n", - (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192, - (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192, - (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192, - (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192); - } - for (j = 0; j < s->prim_channels; j++) { - av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: "); - for (k = 0; k < s->vq_start_subband[j]; k++) - av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } - for (j = 0; j < s->prim_channels; j++) { - av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:"); - for (k = 0; k < s->subband_activity[j]; k++) - av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } - for (j = 0; j < s->prim_channels; j++) { - av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:"); - for (k = 0; k < s->subband_activity[j]; k++) { - if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) - av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]); - if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) - av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]); - } - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } - for (j = 0; j < s->prim_channels; j++) { - if (s->joint_intensity[j] > 0) { - av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n"); - for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) - av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } - } - if (s->prim_channels > 2 && s->downmix) { - av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n"); - for (j = 0; j < s->prim_channels; j++) { - av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]); - av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]); - } - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } - for (j = 0; j < s->prim_channels; j++) - for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) - av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]); - if(s->lfe){ - av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n"); - for (j = lfe_samples; j < lfe_samples * 2; j++) - av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]); - av_log(s->avctx, AV_LOG_DEBUG, "\n"); - } -#endif - - return 0; -} - -static void qmf_32_subbands(DCAContext * s, int chans, - float samples_in[32][8], float *samples_out, - float scale, float bias) -{ - float *prCoeff; - int i, j, k; - float praXin[33], *raXin = &praXin[1]; - - float *subband_fir_hist = s->subband_fir_hist[chans]; - float *subband_fir_hist2 = s->subband_fir_noidea[chans]; - - int chindex = 0, subindex; - - praXin[0] = 0.0; - - /* Select filter */ - if (!s->multirate_inter) /* Non-perfect reconstruction */ - prCoeff = (float *) fir_32bands_nonperfect; - else /* Perfect reconstruction */ - prCoeff = (float *) fir_32bands_perfect; - - /* Reconstructed channel sample index */ - for (subindex = 0; subindex < 8; subindex++) { - float t1, t2, sum[16], diff[16]; - - /* Load in one sample from each subband and clear inactive subbands */ - for (i = 0; i < s->subband_activity[chans]; i++) - raXin[i] = samples_in[i][subindex]; - for (; i < 32; i++) - raXin[i] = 0.0; - - /* Multiply by cosine modulation coefficients and - * create temporary arrays SUM and DIFF */ - for (j = 0, k = 0; k < 16; k++) { - t1 = 0.0; - t2 = 0.0; - for (i = 0; i < 16; i++, j++){ - t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j]; - t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256]; - } - sum[k] = t1 + t2; - diff[k] = t1 - t2; - } - - j = 512; - /* Store history */ - for (k = 0; k < 16; k++) - subband_fir_hist[k] = cos_mod[j++] * sum[k]; - for (k = 0; k < 16; k++) - subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k]; - - /* Multiply by filter coefficients */ - for (k = 31, i = 0; i < 32; i++, k--) - for (j = 0; j < 512; j += 64){ - subband_fir_hist2[i] += prCoeff[i+j] * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]); - subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]); - } - - /* Create 32 PCM output samples */ - for (i = 0; i < 32; i++) - samples_out[chindex++] = subband_fir_hist2[i] * scale + bias; - - /* Update working arrays */ - memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float)); - memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float)); - memset(&subband_fir_hist2[32], 0, 32 * sizeof(float)); - } -} - -static void lfe_interpolation_fir(int decimation_select, - int num_deci_sample, float *samples_in, - float *samples_out, float scale, - float bias) -{ - /* samples_in: An array holding decimated samples. - * Samples in current subframe starts from samples_in[0], - * while samples_in[-1], samples_in[-2], ..., stores samples - * from last subframe as history. - * - * samples_out: An array holding interpolated samples - */ - - int decifactor, k, j; - const float *prCoeff; - - int interp_index = 0; /* Index to the interpolated samples */ - int deciindex; - - /* Select decimation filter */ - if (decimation_select == 1) { - decifactor = 128; - prCoeff = lfe_fir_128; - } else { - decifactor = 64; - prCoeff = lfe_fir_64; - } - /* Interpolation */ - for (deciindex = 0; deciindex < num_deci_sample; deciindex++) { - /* One decimated sample generates decifactor interpolated ones */ - for (k = 0; k < decifactor; k++) { - float rTmp = 0.0; - //FIXME the coeffs are symetric, fix that - for (j = 0; j < 512 / decifactor; j++) - rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor]; - samples_out[interp_index++] = rTmp / scale + bias; - } - } -} - -/* downmixing routines */ -#define MIX_REAR1(samples, si1) \ - samples[i] += samples[si1]; \ - samples[i+256] += samples[si1]; - -#define MIX_REAR2(samples, si1, si2) \ - samples[i] += samples[si1]; \ - samples[i+256] += samples[si2]; - -#define MIX_FRONT3(samples) \ - t = samples[i]; \ - samples[i] += samples[i+256]; \ - samples[i+256] = samples[i+512] + t; - -#define DOWNMIX_TO_STEREO(op1, op2) \ - for(i = 0; i < 256; i++){ \ - op1 \ - op2 \ - } - -static void dca_downmix(float *samples, int srcfmt) -{ - int i; - float t; - - switch (srcfmt) { - case DCA_MONO: - case DCA_CHANNEL: - case DCA_STEREO_TOTAL: - case DCA_STEREO_SUMDIFF: - case DCA_4F2R: - av_log(NULL, 0, "Not implemented!\n"); - break; - case DCA_STEREO: - break; - case DCA_3F: - DOWNMIX_TO_STEREO(MIX_FRONT3(samples),); - break; - case DCA_2F1R: - DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512),); - break; - case DCA_3F1R: - DOWNMIX_TO_STEREO(MIX_FRONT3(samples), - MIX_REAR1(samples, i + 768)); - break; - case DCA_2F2R: - DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768),); - break; - case DCA_3F2R: - DOWNMIX_TO_STEREO(MIX_FRONT3(samples), - MIX_REAR2(samples, i + 768, i + 1024)); - break; - } -} - - -/* Very compact version of the block code decoder that does not use table - * look-up but is slightly slower */ -static int decode_blockcode(int code, int levels, int *values) -{ - int i; - int offset = (levels - 1) >> 1; - - for (i = 0; i < 4; i++) { - values[i] = (code % levels) - offset; - code /= levels; - } - - if (code == 0) - return 0; - else { - av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n"); - return -1; - } -} - -static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; -static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; - -static int dca_subsubframe(DCAContext * s) -{ - int k, l; - int subsubframe = s->current_subsubframe; - - float *quant_step_table; - - /* FIXME */ - float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; - - /* - * Audio data - */ - - /* Select quantization step size table */ - if (s->bit_rate == 0x1f) - quant_step_table = (float *) lossless_quant_d; - else - quant_step_table = (float *) lossy_quant_d; - - for (k = 0; k < s->prim_channels; k++) { - for (l = 0; l < s->vq_start_subband[k]; l++) { - int m; - - /* Select the mid-tread linear quantizer */ - int abits = s->bitalloc[k][l]; - - float quant_step_size = quant_step_table[abits]; - float rscale; - - /* - * Determine quantization index code book and its type - */ - - /* Select quantization index code book */ - int sel = s->quant_index_huffman[k][abits]; - - /* - * Extract bits from the bit stream - */ - if(!abits){ - memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0])); - }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){ - if(abits <= 7){ - /* Block code */ - int block_code1, block_code2, size, levels; - int block[8]; - - size = abits_sizes[abits-1]; - levels = abits_levels[abits-1]; - - block_code1 = get_bits(&s->gb, size); - /* FIXME Should test return value */ - decode_blockcode(block_code1, levels, block); - block_code2 = get_bits(&s->gb, size); - decode_blockcode(block_code2, levels, &block[4]); - for (m = 0; m < 8; m++) - subband_samples[k][l][m] = block[m]; - }else{ - /* no coding */ - for (m = 0; m < 8; m++) - subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3); - } - }else{ - /* Huffman coded */ - for (m = 0; m < 8; m++) - subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel); - } - - /* Deal with transients */ - if (s->transition_mode[k][l] && - subsubframe >= s->transition_mode[k][l]) - rscale = quant_step_size * s->scale_factor[k][l][1]; - else - rscale = quant_step_size * s->scale_factor[k][l][0]; - - rscale *= s->scalefactor_adj[k][sel]; - - for (m = 0; m < 8; m++) - subband_samples[k][l][m] *= rscale; - - /* - * Inverse ADPCM if in prediction mode - */ - if (s->prediction_mode[k][l]) { - int n; - for (m = 0; m < 8; m++) { - for (n = 1; n <= 4; n++) - if (m >= n) - subband_samples[k][l][m] += - (adpcm_vb[s->prediction_vq[k][l]][n - 1] * - subband_samples[k][l][m - n] / 8192); - else if (s->predictor_history) - subband_samples[k][l][m] += - (adpcm_vb[s->prediction_vq[k][l]][n - 1] * - s->subband_samples_hist[k][l][m - n + - 4] / 8192); - } - } - } - - /* - * Decode VQ encoded high frequencies - */ - for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) { - /* 1 vector -> 32 samples but we only need the 8 samples - * for this subsubframe. */ - int m; - - if (!s->debug_flag & 0x01) { - av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); - s->debug_flag |= 0x01; - } - - for (m = 0; m < 8; m++) { - subband_samples[k][l][m] = - high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 + - m] - * (float) s->scale_factor[k][l][0] / 16.0; - } - } - } - - /* Check for DSYNC after subsubframe */ - if (s->aspf || subsubframe == s->subsubframes - 1) { - if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */ -#ifdef TRACE - av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n"); -#endif - } else { - av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); - } - } - - /* Backup predictor history for adpcm */ - for (k = 0; k < s->prim_channels; k++) - for (l = 0; l < s->vq_start_subband[k]; l++) - memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4], - 4 * sizeof(subband_samples[0][0][0])); - - /* 32 subbands QMF */ - for (k = 0; k < s->prim_channels; k++) { -/* static float pcm_to_double[8] = - {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/ - qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k], - 2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ , - 0 /*s->bias */ ); - } - - /* Down mixing */ - - if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) { - dca_downmix(s->samples, s->amode); - } - - /* Generate LFE samples for this subsubframe FIXME!!! */ - if (s->output & DCA_LFE) { - int lfe_samples = 2 * s->lfe * s->subsubframes; - int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK]; - - lfe_interpolation_fir(s->lfe, 2 * s->lfe, - s->lfe_data + lfe_samples + - 2 * s->lfe * subsubframe, - &s->samples[256 * i_channels], - 8388608.0, s->bias); - /* Outputs 20bits pcm samples */ - } - - return 0; -} - - -static int dca_subframe_footer(DCAContext * s) -{ - int aux_data_count = 0, i; - int lfe_samples; - - /* - * Unpack optional information - */ - - if (s->timestamp) - get_bits(&s->gb, 32); - - if (s->aux_data) - aux_data_count = get_bits(&s->gb, 6); - - for (i = 0; i < aux_data_count; i++) - get_bits(&s->gb, 8); - - if (s->crc_present && (s->downmix || s->dynrange)) - get_bits(&s->gb, 16); - - lfe_samples = 2 * s->lfe * s->subsubframes; - for (i = 0; i < lfe_samples; i++) { - s->lfe_data[i] = s->lfe_data[i + lfe_samples]; - } - - return 0; -} - -/** - * Decode a dca frame block - * - * @param s pointer to the DCAContext - */ - -static int dca_decode_block(DCAContext * s) -{ - - /* Sanity check */ - if (s->current_subframe >= s->subframes) { - av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", - s->current_subframe, s->subframes); - return -1; - } - - if (!s->current_subsubframe) { -#ifdef TRACE - av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n"); -#endif - /* Read subframe header */ - if (dca_subframe_header(s)) - return -1; - } - - /* Read subsubframe */ -#ifdef TRACE - av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n"); -#endif - if (dca_subsubframe(s)) - return -1; - - /* Update state */ - s->current_subsubframe++; - if (s->current_subsubframe >= s->subsubframes) { - s->current_subsubframe = 0; - s->current_subframe++; - } - if (s->current_subframe >= s->subframes) { -#ifdef TRACE - av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n"); -#endif - /* Read subframe footer */ - if (dca_subframe_footer(s)) - return -1; - } - - return 0; -} - -/** - * Convert bitstream to one representation based on sync marker - */ -static int dca_convert_bitstream(uint8_t * src, int src_size, uint8_t * dst, - int max_size) -{ - uint32_t mrk; - int i, tmp; - uint16_t *ssrc = (uint16_t *) src, *sdst = (uint16_t *) dst; - PutBitContext pb; - - mrk = AV_RB32(src); - switch (mrk) { - case DCA_MARKER_RAW_BE: - memcpy(dst, src, FFMIN(src_size, max_size)); - return FFMIN(src_size, max_size); - case DCA_MARKER_RAW_LE: - for (i = 0; i < (FFMIN(src_size, max_size) + 1) >> 1; i++) - *sdst++ = bswap_16(*ssrc++); - return FFMIN(src_size, max_size); - case DCA_MARKER_14B_BE: - case DCA_MARKER_14B_LE: - init_put_bits(&pb, dst, max_size); - for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) { - tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF; - put_bits(&pb, 14, tmp); - } - flush_put_bits(&pb); - return (put_bits_count(&pb) + 7) >> 3; - default: - return -1; - } -} - -/** - * Main frame decoding function - * FIXME add arguments - */ -static int dca_decode_frame(AVCodecContext * avctx, - void *data, int *data_size, - uint8_t * buf, int buf_size) -{ - - int i, j, k; - int16_t *samples = data; - DCAContext *s = avctx->priv_data; - int channels; - - - s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE); - if (s->dca_buffer_size == -1) { - av_log(avctx, AV_LOG_ERROR, "Not a DCA frame\n"); - return -1; - } - - init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); - if (dca_parse_frame_header(s) < 0) { - //seems like the frame is corrupt, try with the next one - return buf_size; - } - //set AVCodec values with parsed data - avctx->sample_rate = s->sample_rate; - avctx->channels = 2; //FIXME - avctx->bit_rate = s->bit_rate; - - channels = dca_channels[s->output]; - if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels) - return -1; - *data_size = 0; - for (i = 0; i < (s->sample_blocks / 8); i++) { - dca_decode_block(s); - s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels); - /* interleave samples */ - for (j = 0; j < 256; j++) { - for (k = 0; k < channels; k++) - samples[k] = s->tsamples[j + k * 256]; - samples += channels; - } - *data_size += 256 * sizeof(int16_t) * channels; - } - - return buf_size; -} - - - -/** - * Build the cosine modulation tables for the QMF - * - * @param s pointer to the DCAContext - */ - -static void pre_calc_cosmod(DCAContext * s) -{ - int i, j, k; - static int cosmod_inited = 0; - - if(cosmod_inited) return; - for (j = 0, k = 0; k < 16; k++) - for (i = 0; i < 16; i++) - cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64); - - for (k = 0; k < 16; k++) - for (i = 0; i < 16; i++) - cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32); - - for (k = 0; k < 16; k++) - cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128)); - - for (k = 0; k < 16; k++) - cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128)); - - cosmod_inited = 1; -} - - -/** - * DCA initialization - * - * @param avctx pointer to the AVCodecContext - */ - -static int dca_decode_init(AVCodecContext * avctx) -{ - DCAContext *s = avctx->priv_data; - - s->avctx = avctx; - dca_init_vlcs(); - pre_calc_cosmod(s); - - dsputil_init(&s->dsp, avctx); - return 0; -} - - -AVCodec dca_decoder = { - .name = "dca", - .type = CODEC_TYPE_AUDIO, - .id = CODEC_ID_DTS, - .priv_data_size = sizeof(DCAContext), - .init = dca_decode_init, - .decode = dca_decode_frame, -}; - -#ifdef CONFIG_DCA_PARSER - -typedef struct DCAParseContext { - ParseContext pc; - uint32_t lastmarker; -} DCAParseContext; - -#define IS_MARKER(state, i, buf, buf_size) \ - ((state == DCA_MARKER_14B_LE && (i < buf_size-2) && (buf[i+1] & 0xF0) == 0xF0 && buf[i+2] == 0x07) \ - || (state == DCA_MARKER_14B_BE && (i < buf_size-2) && buf[i+1] == 0x07 && (buf[i+2] & 0xF0) == 0xF0) \ - || state == DCA_MARKER_RAW_LE || state == DCA_MARKER_RAW_BE) - -/** - * finds the end of the current frame in the bitstream. - * @return the position of the first byte of the next frame, or -1 - */ -static int dca_find_frame_end(DCAParseContext * pc1, const uint8_t * buf, - int buf_size) -{ - int start_found, i; - uint32_t state; - ParseContext *pc = &pc1->pc; - - start_found = pc->frame_start_found; - state = pc->state; - - i = 0; - if (!start_found) { - for (i = 0; i < buf_size; i++) { - state = (state << 8) | buf[i]; - if (IS_MARKER(state, i, buf, buf_size)) { - if (pc1->lastmarker && state == pc1->lastmarker) { - start_found = 1; - break; - } else if (!pc1->lastmarker) { - start_found = 1; - pc1->lastmarker = state; - break; - } - } - } - } - if (start_found) { - for (; i < buf_size; i++) { - state = (state << 8) | buf[i]; - if (state == pc1->lastmarker && IS_MARKER(state, i, buf, buf_size)) { - pc->frame_start_found = 0; - pc->state = -1; - return i - 3; - } - } - } - pc->frame_start_found = start_found; - pc->state = state; - return END_NOT_FOUND; -} - -static int dca_parse_init(AVCodecParserContext * s) -{ - DCAParseContext *pc1 = s->priv_data; - - pc1->lastmarker = 0; - return 0; -} - -static int dca_parse(AVCodecParserContext * s, - AVCodecContext * avctx, - uint8_t ** poutbuf, int *poutbuf_size, - const uint8_t * buf, int buf_size) -{ - DCAParseContext *pc1 = s->priv_data; - ParseContext *pc = &pc1->pc; - int next; - - if (s->flags & PARSER_FLAG_COMPLETE_FRAMES) { - next = buf_size; - } else { - next = dca_find_frame_end(pc1, buf, buf_size); - - if (ff_combine_frame(pc, next, (uint8_t **) & buf, &buf_size) < 0) { - *poutbuf = NULL; - *poutbuf_size = 0; - return buf_size; - } - } - *poutbuf = (uint8_t *) buf; - *poutbuf_size = buf_size; - return next; -} - -AVCodecParser dca_parser = { - {CODEC_ID_DTS}, - sizeof(DCAParseContext), - dca_parse_init, - dca_parse, - ff_parse_close, -}; -#endif /* CONFIG_DCA_PARSER */ |