diff options
Diffstat (limited to 'libavcodec/dca.c')
| -rw-r--r-- | libavcodec/dca.c | 1322 |
1 files changed, 1322 insertions, 0 deletions
diff --git a/libavcodec/dca.c b/libavcodec/dca.c new file mode 100644 index 0000000000..a57dcdc442 --- /dev/null +++ b/libavcodec/dca.c @@ -0,0 +1,1322 @@ +/* + * 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 */ |