Mercurial > libavcodec.hg
view dca.c @ 6517:48759bfbd073 libavcodec
Apply 'cold' attribute to init/uninit functions in libavcodec
| author | zuxy |
|---|---|
| date | Fri, 21 Mar 2008 03:11:20 +0000 |
| parents | 9f397992ddff |
| children | 1b90003d4d60 |
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/* * 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 "dca.h" //#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 av_always_inline int 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 total_channels; ///< number of channels including extensions 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 av_cold void dca_init_vlcs(void) { static int vlcs_initialized = 0; int i, j; if (vlcs_initialized) return; dca_bitalloc_index.offset = 1; dca_bitalloc_index.wrap = 2; 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_initialized = 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 = s->amode; if(s->lfe) s->output |= DCA_LFE; #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->total_channels = get_bits(&s->gb, 3) + 1; s->prim_channels = s->total_channels; if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */ 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 value) { if (level < 5) { /* huffman encoded */ value += get_bitalloc(gb, &dca_scalefactor, level); } 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 if (s->bitalloc_huffman[j] == 7) { av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n"); return -1; } else { s->bitalloc[j][k] = get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[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++) { const 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 = scale_factor_quant7; else scale_table = 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], 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], 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], 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) { if(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); } } else { int am = s->amode & DCA_CHANNEL_MASK; for (j = 0; j < s->prim_channels; j++) { s->downmix_coef[j][0] = dca_default_coeffs[am][j][0]; s->downmix_coef[j][1] = dca_default_coeffs[am][j][1]; } } } /* 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) { int source_channel = s->joint_intensity[j] - 1; 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){ int lfe_samples = 2 * s->lfe * s->subsubframes; 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) { const 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 = fir_32bands_nonperfect; else /* Perfect reconstruction */ prCoeff = 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, rs, coef) \ samples[i] += samples[si1] * coef[rs][0]; \ samples[i+256] += samples[si1] * coef[rs][1]; #define MIX_REAR2(samples, si1, si2, rs, coef) \ samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \ samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1]; #define MIX_FRONT3(samples, coef) \ t = samples[i]; \ samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \ samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1]; #define DOWNMIX_TO_STEREO(op1, op2) \ for(i = 0; i < 256; i++){ \ op1 \ op2 \ } static void dca_downmix(float *samples, int srcfmt, int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]) { int i; float t; float coef[DCA_PRIM_CHANNELS_MAX][2]; for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) { coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]]; coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]]; } 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, coef),); break; case DCA_2F1R: DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),); break; case DCA_3F1R: DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), MIX_REAR1(samples, i + 768, 3, coef)); break; case DCA_2F2R: DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),); break; case DCA_3F2R: DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), MIX_REAR2(samples, i + 768, i + 1024, 3, coef)); 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; const 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 = lossless_quant_d; else quant_step_table = 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, s->downmix_coef); } /* 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], 256.0, 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(const uint8_t * src, int src_size, uint8_t * dst, int max_size) { uint32_t mrk; int i, tmp; const uint16_t *ssrc = (const uint16_t *) src; uint16_t *sdst = (uint16_t *) dst; PutBitContext pb; if((unsigned)src_size > (unsigned)max_size) { av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n"); return -1; } 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, const 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 valid 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 *data_size=0; return buf_size; } //set AVCodec values with parsed data avctx->sample_rate = s->sample_rate; avctx->bit_rate = s->bit_rate; channels = s->prim_channels + !!s->lfe; if(avctx->request_channels == 2 && s->prim_channels > 2) { channels = 2; s->output = DCA_STEREO; } avctx->channels = channels; 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 av_cold void pre_calc_cosmod(DCAContext * s) { int i, j, k; static int cosmod_initialized = 0; if(cosmod_initialized) 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_initialized = 1; } /** * DCA initialization * * @param avctx pointer to the AVCodecContext */ static av_cold 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); /* allow downmixing to stereo */ if (avctx->channels > 0 && avctx->request_channels < avctx->channels && avctx->request_channels == 2) { avctx->channels = avctx->request_channels; } 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, };