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nest-open-source / nest-android-app / ffmpeg / 8e1cc55b733ae723de9931f20a13731bed8162f5 / . / libavcodec / dca_core.c
blob: 48a8f618f79010c423e76e7e4e6f0d09635981ba [file] [log] [blame]
/*
* Copyright (C) 2016 foo86
*
* 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
*/
#include "dcadec.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dcamath.h"
#include "dca_syncwords.h"
#if ARCH_ARM
#include "arm/dca.h"
#endif
enum HeaderType {
HEADER_CORE,
HEADER_XCH,
HEADER_XXCH
};
enum AudioMode {
AMODE_MONO, // Mode 0: A (mono)
AMODE_MONO_DUAL, // Mode 1: A + B (dual mono)
AMODE_STEREO, // Mode 2: L + R (stereo)
AMODE_STEREO_SUMDIFF, // Mode 3: (L+R) + (L-R) (sum-diff)
AMODE_STEREO_TOTAL, // Mode 4: LT + RT (left and right total)
AMODE_3F, // Mode 5: C + L + R
AMODE_2F1R, // Mode 6: L + R + S
AMODE_3F1R, // Mode 7: C + L + R + S
AMODE_2F2R, // Mode 8: L + R + SL + SR
AMODE_3F2R, // Mode 9: C + L + R + SL + SR
AMODE_COUNT
};
enum ExtAudioType {
EXT_AUDIO_XCH = 0,
EXT_AUDIO_X96 = 2,
EXT_AUDIO_XXCH = 6
};
enum LFEFlag {
LFE_FLAG_NONE,
LFE_FLAG_128,
LFE_FLAG_64,
LFE_FLAG_INVALID
};
static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = {
{ DCA_SPEAKER_C, -1, -1, -1, -1 },
{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
{ DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 },
{ DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , -1, -1 },
{ DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Cs, -1, -1 },
{ DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , DCA_SPEAKER_Cs, -1 },
{ DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs, -1 },
{ DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs }
};
static const uint8_t audio_mode_ch_mask[AMODE_COUNT] = {
DCA_SPEAKER_LAYOUT_MONO,
DCA_SPEAKER_LAYOUT_STEREO,
DCA_SPEAKER_LAYOUT_STEREO,
DCA_SPEAKER_LAYOUT_STEREO,
DCA_SPEAKER_LAYOUT_STEREO,
DCA_SPEAKER_LAYOUT_3_0,
DCA_SPEAKER_LAYOUT_2_1,
DCA_SPEAKER_LAYOUT_3_1,
DCA_SPEAKER_LAYOUT_2_2,
DCA_SPEAKER_LAYOUT_5POINT0
};
static const uint8_t block_code_nbits[7] = {
7, 10, 12, 13, 15, 17, 19
};
static const uint8_t quant_index_sel_nbits[DCA_CODE_BOOKS] = {
1, 2, 2, 2, 2, 3, 3, 3, 3, 3
};
static const uint8_t quant_index_group_size[DCA_CODE_BOOKS] = {
1, 3, 3, 3, 3, 7, 7, 7, 7, 7
};
typedef struct DCAVLC {
int offset; ///< Code values offset
int max_depth; ///< Parameter for get_vlc2()
VLC vlc[7]; ///< Actual codes
} DCAVLC;
static DCAVLC vlc_bit_allocation;
static DCAVLC vlc_transition_mode;
static DCAVLC vlc_scale_factor;
static DCAVLC vlc_quant_index[DCA_CODE_BOOKS];
static av_cold void dca_init_vlcs(void)
{
static VLC_TYPE dca_table[23622][2];
static int vlcs_initialized = 0;
int i, j, k;
if (vlcs_initialized)
return;
#define DCA_INIT_VLC(vlc, a, b, c, d) \
do { \
vlc.table = &dca_table[ff_dca_vlc_offs[k]]; \
vlc.table_allocated = ff_dca_vlc_offs[k + 1] - ff_dca_vlc_offs[k]; \
init_vlc(&vlc, a, b, c, 1, 1, d, 2, 2, INIT_VLC_USE_NEW_STATIC); \
} while (0)
vlc_bit_allocation.offset = 1;
vlc_bit_allocation.max_depth = 2;
for (i = 0, k = 0; i < 5; i++, k++)
DCA_INIT_VLC(vlc_bit_allocation.vlc[i], bitalloc_12_vlc_bits[i], 12,
bitalloc_12_bits[i], bitalloc_12_codes[i]);
vlc_scale_factor.offset = -64;
vlc_scale_factor.max_depth = 2;
for (i = 0; i < 5; i++, k++)
DCA_INIT_VLC(vlc_scale_factor.vlc[i], SCALES_VLC_BITS, 129,
scales_bits[i], scales_codes[i]);
vlc_transition_mode.offset = 0;
vlc_transition_mode.max_depth = 1;
for (i = 0; i < 4; i++, k++)
DCA_INIT_VLC(vlc_transition_mode.vlc[i], tmode_vlc_bits[i], 4,
tmode_bits[i], tmode_codes[i]);
for (i = 0; i < DCA_CODE_BOOKS; i++) {
vlc_quant_index[i].offset = bitalloc_offsets[i];
vlc_quant_index[i].max_depth = 1 + (i > 4);
for (j = 0; j < quant_index_group_size[i]; j++, k++)
DCA_INIT_VLC(vlc_quant_index[i].vlc[j], bitalloc_maxbits[i][j],
bitalloc_sizes[i], bitalloc_bits[i][j], bitalloc_codes[i][j]);
}
vlcs_initialized = 1;
}
static int dca_get_vlc(GetBitContext *s, DCAVLC *v, int i)
{
return get_vlc2(s, v->vlc[i].table, v->vlc[i].bits, v->max_depth) + v->offset;
}
static void get_array(GetBitContext *s, int32_t *array, int size, int n)
{
int i;
for (i = 0; i < size; i++)
array[i] = get_sbits(s, n);
}
// 5.3.1 - Bit stream header
static int parse_frame_header(DCACoreDecoder *s)
{
int normal_frame, pcmr_index;
// Frame type
normal_frame = get_bits1(&s->gb);
// Deficit sample count
if (get_bits(&s->gb, 5) != DCA_PCMBLOCK_SAMPLES - 1) {
av_log(s->avctx, AV_LOG_ERROR, "Deficit samples are not supported\n");
return normal_frame ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
}
// CRC present flag
s->crc_present = get_bits1(&s->gb);
// Number of PCM sample blocks
s->npcmblocks = get_bits(&s->gb, 7) + 1;
if (s->npcmblocks & (DCA_SUBBAND_SAMPLES - 1)) {
av_log(s->avctx, AV_LOG_ERROR, "Unsupported number of PCM sample blocks (%d)\n", s->npcmblocks);
return (s->npcmblocks < 6 || normal_frame) ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
}
// Primary frame byte size
s->frame_size = get_bits(&s->gb, 14) + 1;
if (s->frame_size < 96) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid core frame size (%d bytes)\n", s->frame_size);
return AVERROR_INVALIDDATA;
}
// Audio channel arrangement
s->audio_mode = get_bits(&s->gb, 6);
if (s->audio_mode >= AMODE_COUNT) {
av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", s->audio_mode);
return AVERROR_PATCHWELCOME;
}
// Core audio sampling frequency
s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
if (!s->sample_rate) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n");
return AVERROR_INVALIDDATA;
}
// Transmission bit rate
s->bit_rate = ff_dca_bit_rates[get_bits(&s->gb, 5)];
// Reserved field
skip_bits1(&s->gb);
// Embedded dynamic range flag
s->drc_present = get_bits1(&s->gb);
// Embedded time stamp flag
s->ts_present = get_bits1(&s->gb);
// Auxiliary data flag
s->aux_present = get_bits1(&s->gb);
// HDCD mastering flag
skip_bits1(&s->gb);
// Extension audio descriptor flag
s->ext_audio_type = get_bits(&s->gb, 3);
// Extended coding flag
s->ext_audio_present = get_bits1(&s->gb);
// Audio sync word insertion flag
s->sync_ssf = get_bits1(&s->gb);
// Low frequency effects flag
s->lfe_present = get_bits(&s->gb, 2);
if (s->lfe_present == LFE_FLAG_INVALID) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n");
return AVERROR_INVALIDDATA;
}
// Predictor history flag switch
s->predictor_history = get_bits1(&s->gb);
// Header CRC check bytes
if (s->crc_present)
skip_bits(&s->gb, 16);
// Multirate interpolator switch
s->filter_perfect = get_bits1(&s->gb);
// Encoder software revision
skip_bits(&s->gb, 4);
// Copy history
skip_bits(&s->gb, 2);
// Source PCM resolution
s->source_pcm_res = ff_dca_bits_per_sample[pcmr_index = get_bits(&s->gb, 3)];
if (!s->source_pcm_res) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid source PCM resolution\n");
return AVERROR_INVALIDDATA;
}
s->es_format = pcmr_index & 1;
// Front sum/difference flag
s->sumdiff_front = get_bits1(&s->gb);
// Surround sum/difference flag
s->sumdiff_surround = get_bits1(&s->gb);
// Dialog normalization / unspecified
skip_bits(&s->gb, 4);
return 0;
}
// 5.3.2 - Primary audio coding header
static int parse_coding_header(DCACoreDecoder *s, enum HeaderType header, int xch_base)
{
int n, ch, nchannels, header_size = 0, header_pos = get_bits_count(&s->gb);
unsigned int mask, index;
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
switch (header) {
case HEADER_CORE:
// Number of subframes
s->nsubframes = get_bits(&s->gb, 4) + 1;
// Number of primary audio channels
s->nchannels = get_bits(&s->gb, 3) + 1;
if (s->nchannels != ff_dca_channels[s->audio_mode]) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid number of primary audio channels (%d) for audio channel arrangement (%d)\n", s->nchannels, s->audio_mode);
return AVERROR_INVALIDDATA;
}
av_assert1(s->nchannels <= DCA_CHANNELS - 2);
s->ch_mask = audio_mode_ch_mask[s->audio_mode];
// Add LFE channel if present
if (s->lfe_present)
s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
break;
case HEADER_XCH:
s->nchannels = ff_dca_channels[s->audio_mode] + 1;
av_assert1(s->nchannels <= DCA_CHANNELS - 1);
s->ch_mask |= DCA_SPEAKER_MASK_Cs;
break;
case HEADER_XXCH:
// Channel set header length
header_size = get_bits(&s->gb, 7) + 1;
// Check CRC
if (s->xxch_crc_present
&& (s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
&& ff_dca_check_crc(&s->gb, header_pos, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH channel set header checksum\n");
return AVERROR_INVALIDDATA;
}
// Number of channels in a channel set
nchannels = get_bits(&s->gb, 3) + 1;
if (nchannels > DCA_XXCH_CHANNELS_MAX) {
avpriv_request_sample(s->avctx, "%d XXCH channels", nchannels);
return AVERROR_PATCHWELCOME;
}
s->nchannels = ff_dca_channels[s->audio_mode] + nchannels;
av_assert1(s->nchannels <= DCA_CHANNELS);
// Loudspeaker layout mask
mask = get_bits_long(&s->gb, s->xxch_mask_nbits - DCA_SPEAKER_Cs);
s->xxch_spkr_mask = mask << DCA_SPEAKER_Cs;
if (av_popcount(s->xxch_spkr_mask) != nchannels) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH speaker layout mask (%#x)\n", s->xxch_spkr_mask);
return AVERROR_INVALIDDATA;
}
if (s->xxch_core_mask & s->xxch_spkr_mask) {
av_log(s->avctx, AV_LOG_ERROR, "XXCH speaker layout mask (%#x) overlaps with core (%#x)\n", s->xxch_spkr_mask, s->xxch_core_mask);
return AVERROR_INVALIDDATA;
}
// Combine core and XXCH masks together
s->ch_mask = s->xxch_core_mask | s->xxch_spkr_mask;
// Downmix coefficients present in stream
if (get_bits1(&s->gb)) {
int *coeff_ptr = s->xxch_dmix_coeff;
// Downmix already performed by encoder
s->xxch_dmix_embedded = get_bits1(&s->gb);
// Downmix scale factor
index = get_bits(&s->gb, 6) * 4 - FF_DCA_DMIXTABLE_OFFSET - 3;
if (index >= FF_DCA_INV_DMIXTABLE_SIZE) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix scale index (%d)\n", index);
return AVERROR_INVALIDDATA;
}
s->xxch_dmix_scale_inv = ff_dca_inv_dmixtable[index];
// Downmix channel mapping mask
for (ch = 0; ch < nchannels; ch++) {
mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
if ((mask & s->xxch_core_mask) != mask) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix channel mapping mask (%#x)\n", mask);
return AVERROR_INVALIDDATA;
}
s->xxch_dmix_mask[ch] = mask;
}
// Downmix coefficients
for (ch = 0; ch < nchannels; ch++) {
for (n = 0; n < s->xxch_mask_nbits; n++) {
if (s->xxch_dmix_mask[ch] & (1U << n)) {
int code = get_bits(&s->gb, 7);
int sign = (code >> 6) - 1;
if (code &= 63) {
index = code * 4 - 3;
if (index >= FF_DCA_DMIXTABLE_SIZE) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix coefficient index (%d)\n", index);
return AVERROR_INVALIDDATA;
}
*coeff_ptr++ = (ff_dca_dmixtable[index] ^ sign) - sign;
} else {
*coeff_ptr++ = 0;
}
}
}
}
} else {
s->xxch_dmix_embedded = 0;
}
break;
}
// Subband activity count
for (ch = xch_base; ch < s->nchannels; ch++) {
s->nsubbands[ch] = get_bits(&s->gb, 5) + 2;
if (s->nsubbands[ch] > DCA_SUBBANDS) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid subband activity count\n");
return AVERROR_INVALIDDATA;
}
}
// High frequency VQ start subband
for (ch = xch_base; ch < s->nchannels; ch++)
s->subband_vq_start[ch] = get_bits(&s->gb, 5) + 1;
// Joint intensity coding index
for (ch = xch_base; ch < s->nchannels; ch++) {
if ((n = get_bits(&s->gb, 3)) && header == HEADER_XXCH)
n += xch_base - 1;
if (n > s->nchannels) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid joint intensity coding index\n");
return AVERROR_INVALIDDATA;
}
s->joint_intensity_index[ch] = n;
}
// Transient mode code book
for (ch = xch_base; ch < s->nchannels; ch++)
s->transition_mode_sel[ch] = get_bits(&s->gb, 2);
// Scale factor code book
for (ch = xch_base; ch < s->nchannels; ch++) {
s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
if (s->scale_factor_sel[ch] == 7) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor code book\n");
return AVERROR_INVALIDDATA;
}
}
// Bit allocation quantizer select
for (ch = xch_base; ch < s->nchannels; ch++) {
s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
if (s->bit_allocation_sel[ch] == 7) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation quantizer select\n");
return AVERROR_INVALIDDATA;
}
}
// Quantization index codebook select
for (n = 0; n < DCA_CODE_BOOKS; n++)
for (ch = xch_base; ch < s->nchannels; ch++)
s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);
// Scale factor adjustment index
for (n = 0; n < DCA_CODE_BOOKS; n++)
for (ch = xch_base; ch < s->nchannels; ch++)
if (s->quant_index_sel[ch][n] < quant_index_group_size[n])
s->scale_factor_adj[ch][n] = ff_dca_scale_factor_adj[get_bits(&s->gb, 2)];
if (header == HEADER_XXCH) {
// Reserved
// Byte align
// CRC16 of channel set header
if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set header\n");
return AVERROR_INVALIDDATA;
}
} else {
// Audio header CRC check word
if (s->crc_present)
skip_bits(&s->gb, 16);
}
return 0;
}
static inline int parse_scale(DCACoreDecoder *s, int *scale_index, int sel)
{
const uint32_t *scale_table;
unsigned int scale_size;
// Select the root square table
if (sel > 5) {
scale_table = ff_dca_scale_factor_quant7;
scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
} else {
scale_table = ff_dca_scale_factor_quant6;
scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
}
// If Huffman code was used, the difference of scales was encoded
if (sel < 5)
*scale_index += dca_get_vlc(&s->gb, &vlc_scale_factor, sel);
else
*scale_index = get_bits(&s->gb, sel + 1);
// Look up scale factor from the root square table
if ((unsigned int)*scale_index >= scale_size) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor index\n");
return AVERROR_INVALIDDATA;
}
return scale_table[*scale_index];
}
static inline int parse_joint_scale(DCACoreDecoder *s, int sel)
{
int scale_index;
// Absolute value was encoded even when Huffman code was used
if (sel < 5)
scale_index = dca_get_vlc(&s->gb, &vlc_scale_factor, sel);
else
scale_index = get_bits(&s->gb, sel + 1);
// Bias by 64
scale_index += 64;
// Look up joint scale factor
if ((unsigned int)scale_index >= FF_ARRAY_ELEMS(ff_dca_joint_scale_factors)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor index\n");
return AVERROR_INVALIDDATA;
}
return ff_dca_joint_scale_factors[scale_index];
}
// 5.4.1 - Primary audio coding side information
static int parse_subframe_header(DCACoreDecoder *s, int sf,
enum HeaderType header, int xch_base)
{
int ch, band, ret;
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
if (header == HEADER_CORE) {
// Subsubframe count
s->nsubsubframes[sf] = get_bits(&s->gb, 2) + 1;
// Partial subsubframe sample count
skip_bits(&s->gb, 3);
}
// Prediction mode
for (ch = xch_base; ch < s->nchannels; ch++)
for (band = 0; band < s->nsubbands[ch]; band++)
s->prediction_mode[ch][band] = get_bits1(&s->gb);
// Prediction coefficients VQ address
for (ch = xch_base; ch < s->nchannels; ch++)
for (band = 0; band < s->nsubbands[ch]; band++)
if (s->prediction_mode[ch][band])
s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);
// Bit allocation index
for (ch = xch_base; ch < s->nchannels; ch++) {
int sel = s->bit_allocation_sel[ch];
for (band = 0; band < s->subband_vq_start[ch]; band++) {
int abits;
if (sel < 5)
abits = dca_get_vlc(&s->gb, &vlc_bit_allocation, sel);
else
abits = get_bits(&s->gb, sel - 1);
if (abits > DCA_ABITS_MAX) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n");
return AVERROR_INVALIDDATA;
}
s->bit_allocation[ch][band] = abits;
}
}
// Transition mode
for (ch = xch_base; ch < s->nchannels; ch++) {
// Clear transition mode for all subbands
memset(s->transition_mode[sf][ch], 0, sizeof(s->transition_mode[0][0]));
// Transient possible only if more than one subsubframe
if (s->nsubsubframes[sf] > 1) {
int sel = s->transition_mode_sel[ch];
for (band = 0; band < s->subband_vq_start[ch]; band++)
if (s->bit_allocation[ch][band])
s->transition_mode[sf][ch][band] = dca_get_vlc(&s->gb, &vlc_transition_mode, sel);
}
}
// Scale factors
for (ch = xch_base; ch < s->nchannels; ch++) {
int sel = s->scale_factor_sel[ch];
int scale_index = 0;
// Extract scales for subbands up to VQ
for (band = 0; band < s->subband_vq_start[ch]; band++) {
if (s->bit_allocation[ch][band]) {
if ((ret = parse_scale(s, &scale_index, sel)) < 0)
return ret;
s->scale_factors[ch][band][0] = ret;
if (s->transition_mode[sf][ch][band]) {
if ((ret = parse_scale(s, &scale_index, sel)) < 0)
return ret;
s->scale_factors[ch][band][1] = ret;
}
} else {
s->scale_factors[ch][band][0] = 0;
}
}
// High frequency VQ subbands
for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++) {
if ((ret = parse_scale(s, &scale_index, sel)) < 0)
return ret;
s->scale_factors[ch][band][0] = ret;
}
}
// Joint subband codebook select
for (ch = xch_base; ch < s->nchannels; ch++) {
if (s->joint_intensity_index[ch]) {
s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
if (s->joint_scale_sel[ch] == 7) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor code book\n");
return AVERROR_INVALIDDATA;
}
}
}
// Scale factors for joint subband coding
for (ch = xch_base; ch < s->nchannels; ch++) {
int src_ch = s->joint_intensity_index[ch] - 1;
if (src_ch >= 0) {
int sel = s->joint_scale_sel[ch];
for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
if ((ret = parse_joint_scale(s, sel)) < 0)
return ret;
s->joint_scale_factors[ch][band] = ret;
}
}
}
// Dynamic range coefficient
if (s->drc_present && header == HEADER_CORE)
skip_bits(&s->gb, 8);
// Side information CRC check word
if (s->crc_present)
skip_bits(&s->gb, 16);
return 0;
}
#ifndef decode_blockcodes
static inline int decode_blockcodes(int code1, int code2, int levels, int32_t *audio)
{
int offset = (levels - 1) / 2;
int n, div;
for (n = 0; n < DCA_SUBBAND_SAMPLES / 2; n++) {
div = FASTDIV(code1, levels);
audio[n] = code1 - div * levels - offset;
code1 = div;
}
for (; n < DCA_SUBBAND_SAMPLES; n++) {
div = FASTDIV(code2, levels);
audio[n] = code2 - div * levels - offset;
code2 = div;
}
return code1 | code2;
}
#endif
static inline int parse_block_codes(DCACoreDecoder *s, int32_t *audio, int abits)
{
// Extract block code indices from the bit stream
int code1 = get_bits(&s->gb, block_code_nbits[abits - 1]);
int code2 = get_bits(&s->gb, block_code_nbits[abits - 1]);
int levels = ff_dca_quant_levels[abits];
// Look up samples from the block code book
if (decode_blockcodes(code1, code2, levels, audio)) {
av_log(s->avctx, AV_LOG_ERROR, "Failed to decode block code(s)\n");
return AVERROR_INVALIDDATA;
}
return 0;
}
static inline int parse_huffman_codes(DCACoreDecoder *s, int32_t *audio, int abits, int sel)
{
int i;
// Extract Huffman codes from the bit stream
for (i = 0; i < DCA_SUBBAND_SAMPLES; i++)
audio[i] = dca_get_vlc(&s->gb, &vlc_quant_index[abits - 1], sel);
return 1;
}
static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, int ch)
{
av_assert1(abits >= 0 && abits <= DCA_ABITS_MAX);
if (abits == 0) {
// No bits allocated
memset(audio, 0, DCA_SUBBAND_SAMPLES * sizeof(*audio));
return 0;
}
if (abits <= DCA_CODE_BOOKS) {
int sel = s->quant_index_sel[ch][abits - 1];
if (sel < quant_index_group_size[abits - 1]) {
// Huffman codes
return parse_huffman_codes(s, audio, abits, sel);
}
if (abits <= 7) {
// Block codes
return parse_block_codes(s, audio, abits);
}
}
// No further encoding
get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
return 0;
}
static inline void dequantize(int32_t *output, const int32_t *input,
int32_t step_size, int32_t scale, int residual)
{
// Account for quantizer step size
int64_t step_scale = (int64_t)step_size * scale;
int n, shift = 0;
// Limit scale factor resolution to 22 bits
if (step_scale > (1 << 23)) {
shift = av_log2(step_scale >> 23) + 1;
step_scale >>= shift;
}
// Scale the samples
if (residual) {
for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
} else {
for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
}
}
static inline void inverse_adpcm(int32_t **subband_samples,
const int16_t *vq_index,
const int8_t *prediction_mode,
int sb_start, int sb_end,
int ofs, int len)
{
int i, j, k;
for (i = sb_start; i < sb_end; i++) {
if (prediction_mode[i]) {
const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]];
int32_t *ptr = subband_samples[i] + ofs;
for (j = 0; j < len; j++) {
int64_t err = 0;
for (k = 0; k < DCA_ADPCM_COEFFS; k++)
err += (int64_t)ptr[j - k - 1] * coeff[k];
ptr[j] = clip23(ptr[j] + clip23(norm13(err)));
}
}
}
}
// 5.5 - Primary audio data arrays
static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType header,
int xch_base, int *sub_pos, int *lfe_pos)
{
int32_t audio[16], scale;
int n, ssf, ofs, ch, band;
// Check number of subband samples in this subframe
int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
if (*sub_pos + nsamples > s->npcmblocks) {
av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
return AVERROR_INVALIDDATA;
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
// VQ encoded subbands
for (ch = xch_base; ch < s->nchannels; ch++) {
int32_t vq_index[DCA_SUBBANDS];
for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++)
// Extract the VQ address from the bit stream
vq_index[band] = get_bits(&s->gb, 10);
if (s->subband_vq_start[ch] < s->nsubbands[ch]) {
s->dcadsp->decode_hf(s->subband_samples[ch], vq_index,
ff_dca_high_freq_vq, s->scale_factors[ch],
s->subband_vq_start[ch], s->nsubbands[ch],
*sub_pos, nsamples);
}
}
// Low frequency effect data
if (s->lfe_present && header == HEADER_CORE) {
unsigned int index;
// Determine number of LFE samples in this subframe
int nlfesamples = 2 * s->lfe_present * s->nsubsubframes[sf];
av_assert1((unsigned int)nlfesamples <= FF_ARRAY_ELEMS(audio));
// Extract LFE samples from the bit stream
get_array(&s->gb, audio, nlfesamples, 8);
// Extract scale factor index from the bit stream
index = get_bits(&s->gb, 8);
if (index >= FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE scale factor index\n");
return AVERROR_INVALIDDATA;
}
// Look up the 7-bit root square quantization table
scale = ff_dca_scale_factor_quant7[index];
// Account for quantizer step size which is 0.035
scale = mul23(4697620 /* 0.035 * (1 << 27) */, scale);
// Scale and take the LFE samples
for (n = 0, ofs = *lfe_pos; n < nlfesamples; n++, ofs++)
s->lfe_samples[ofs] = clip23(audio[n] * scale >> 4);
// Advance LFE sample pointer for the next subframe
*lfe_pos = ofs;
}
// Audio data
for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
for (ch = xch_base; ch < s->nchannels; ch++) {
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
// Not high frequency VQ subbands
for (band = 0; band < s->subband_vq_start[ch]; band++) {
int ret, trans_ssf, abits = s->bit_allocation[ch][band];
int32_t step_size;
// Extract bits from the bit stream
if ((ret = extract_audio(s, audio, abits, ch)) < 0)
return ret;
// Select quantization step size table and look up
// quantization step size
if (s->bit_rate == 3)
step_size = ff_dca_lossless_quant[abits];
else
step_size = ff_dca_lossy_quant[abits];
// Identify transient location
trans_ssf = s->transition_mode[sf][ch][band];
// Determine proper scale factor
if (trans_ssf == 0 || ssf < trans_ssf)
scale = s->scale_factors[ch][band][0];
else
scale = s->scale_factors[ch][band][1];
// Adjust scale factor when SEL indicates Huffman code
if (ret > 0) {
int64_t adj = s->scale_factor_adj[ch][abits - 1];
scale = clip23(adj * scale >> 22);
}
dequantize(s->subband_samples[ch][band] + ofs,
audio, step_size, scale, 0);
}
}
// DSYNC
if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
av_log(s->avctx, AV_LOG_ERROR, "DSYNC check failed\n");
return AVERROR_INVALIDDATA;
}
ofs += DCA_SUBBAND_SAMPLES;
}
// Inverse ADPCM
for (ch = xch_base; ch < s->nchannels; ch++) {
inverse_adpcm(s->subband_samples[ch], s->prediction_vq_index[ch],
s->prediction_mode[ch], 0, s->nsubbands[ch],
*sub_pos, nsamples);
}
// Joint subband coding
for (ch = xch_base; ch < s->nchannels; ch++) {
int src_ch = s->joint_intensity_index[ch] - 1;
if (src_ch >= 0) {
s->dcadsp->decode_joint(s->subband_samples[ch], s->subband_samples[src_ch],
s->joint_scale_factors[ch], s->nsubbands[ch],
s->nsubbands[src_ch], *sub_pos, nsamples);
}
}
// Advance subband sample pointer for the next subframe
*sub_pos = ofs;
return 0;
}
static void erase_adpcm_history(DCACoreDecoder *s)
{
int ch, band;
// Erase ADPCM history from previous frame if
// predictor history switch was disabled
for (ch = 0; ch < DCA_CHANNELS; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
AV_ZERO128(s->subband_samples[ch][band] - DCA_ADPCM_COEFFS);
emms_c();
}
static int alloc_sample_buffer(DCACoreDecoder *s)
{
int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS;
int nlfesamples = DCA_LFE_HISTORY + s->npcmblocks / 2;
unsigned int size = s->subband_size;
int ch, band;
// Reallocate subband sample buffer
av_fast_mallocz(&s->subband_buffer, &s->subband_size,
(nframesamples + nlfesamples) * sizeof(int32_t));
if (!s->subband_buffer)
return AVERROR(ENOMEM);
if (size != s->subband_size) {
for (ch = 0; ch < DCA_CHANNELS; ch++)
for (band = 0; band < DCA_SUBBANDS; band++)
s->subband_samples[ch][band] = s->subband_buffer +
(ch * DCA_SUBBANDS + band) * nchsamples + DCA_ADPCM_COEFFS;
s->lfe_samples = s->subband_buffer + nframesamples;
}
if (!s->predictor_history)
erase_adpcm_history(s);
return 0;
}
static int parse_frame_data(DCACoreDecoder *s, enum HeaderType header, int xch_base)
{
int sf, ch, ret, band, sub_pos, lfe_pos;
if ((ret = parse_coding_header(s, header, xch_base)) < 0)
return ret;
for (sf = 0, sub_pos = 0, lfe_pos = DCA_LFE_HISTORY; sf < s->nsubframes; sf++) {
if ((ret = parse_subframe_header(s, sf, header, xch_base)) < 0)
return ret;
if ((ret = parse_subframe_audio(s, sf, header, xch_base, &sub_pos, &lfe_pos)) < 0)
return ret;
}
for (ch = xch_base; ch < s->nchannels; ch++) {
// Determine number of active subbands for this channel
int nsubbands = s->nsubbands[ch];
if (s->joint_intensity_index[ch])
nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);
// Update history for ADPCM
for (band = 0; band < nsubbands; band++) {
int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
AV_COPY128(samples, samples + s->npcmblocks);
}
// Clear inactive subbands
for (; band < DCA_SUBBANDS; band++) {
int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
}
}
emms_c();
return 0;
}
static int parse_xch_frame(DCACoreDecoder *s)
{
int ret;
if (s->ch_mask & DCA_SPEAKER_MASK_Cs) {
av_log(s->avctx, AV_LOG_ERROR, "XCH with Cs speaker already present\n");
return AVERROR_INVALIDDATA;
}
if ((ret = parse_frame_data(s, HEADER_XCH, s->nchannels)) < 0)
return ret;
// Seek to the end of core frame, don't trust XCH frame size
if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of XCH frame\n");
return AVERROR_INVALIDDATA;
}
return 0;
}
static int parse_xxch_frame(DCACoreDecoder *s)
{
int xxch_nchsets, xxch_frame_size;
int ret, mask, header_size, header_pos = get_bits_count(&s->gb);
// XXCH sync word
if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XXCH) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH sync word\n");
return AVERROR_INVALIDDATA;
}
// XXCH frame header length
header_size = get_bits(&s->gb, 6) + 1;
// Check XXCH frame header CRC
if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
&& ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH frame header checksum\n");
return AVERROR_INVALIDDATA;
}
// CRC presence flag for channel set header
s->xxch_crc_present = get_bits1(&s->gb);
// Number of bits for loudspeaker mask
s->xxch_mask_nbits = get_bits(&s->gb, 5) + 1;
if (s->xxch_mask_nbits <= DCA_SPEAKER_Cs) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XXCH speaker mask (%d)\n", s->xxch_mask_nbits);
return AVERROR_INVALIDDATA;
}
// Number of channel sets
xxch_nchsets = get_bits(&s->gb, 2) + 1;
if (xxch_nchsets > 1) {
avpriv_request_sample(s->avctx, "%d XXCH channel sets", xxch_nchsets);
return AVERROR_PATCHWELCOME;
}
// Channel set 0 data byte size
xxch_frame_size = get_bits(&s->gb, 14) + 1;
// Core loudspeaker activity mask
s->xxch_core_mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
// Validate the core mask
mask = s->ch_mask;
if ((mask & DCA_SPEAKER_MASK_Ls) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
mask = (mask & ~DCA_SPEAKER_MASK_Ls) | DCA_SPEAKER_MASK_Lss;
if ((mask & DCA_SPEAKER_MASK_Rs) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
mask = (mask & ~DCA_SPEAKER_MASK_Rs) | DCA_SPEAKER_MASK_Rss;
if (mask != s->xxch_core_mask) {
av_log(s->avctx, AV_LOG_ERROR, "XXCH core speaker activity mask (%#x) disagrees with core (%#x)\n", s->xxch_core_mask, mask);
return AVERROR_INVALIDDATA;
}
// Reserved
// Byte align
// CRC16 of XXCH frame header
if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH frame header\n");
return AVERROR_INVALIDDATA;
}
// Parse XXCH channel set 0
if ((ret = parse_frame_data(s, HEADER_XXCH, s->nchannels)) < 0)
return ret;
if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8 + xxch_frame_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set\n");
return AVERROR_INVALIDDATA;
}
return 0;
}
static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchannels,
int *xbr_nsubbands, int xbr_transition_mode, int sf, int *sub_pos)
{
int xbr_nabits[DCA_CHANNELS];
int xbr_bit_allocation[DCA_CHANNELS][DCA_SUBBANDS];
int xbr_scale_nbits[DCA_CHANNELS];
int32_t xbr_scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2];
int ssf, ch, band, ofs;
// Check number of subband samples in this subframe
if (*sub_pos + s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES > s->npcmblocks) {
av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
return AVERROR_INVALIDDATA;
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
// Number of bits for XBR bit allocation index
for (ch = xbr_base_ch; ch < xbr_nchannels; ch++)
xbr_nabits[ch] = get_bits(&s->gb, 2) + 2;
// XBR bit allocation index
for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
for (band = 0; band < xbr_nsubbands[ch]; band++) {
xbr_bit_allocation[ch][band] = get_bits(&s->gb, xbr_nabits[ch]);
if (xbr_bit_allocation[ch][band] > DCA_ABITS_MAX) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR bit allocation index\n");
return AVERROR_INVALIDDATA;
}
}
}
// Number of bits for scale indices
for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
xbr_scale_nbits[ch] = get_bits(&s->gb, 3);
if (!xbr_scale_nbits[ch]) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XBR scale factor index\n");
return AVERROR_INVALIDDATA;
}
}
// XBR scale factors
for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
const uint32_t *scale_table;
int scale_size;
// Select the root square table
if (s->scale_factor_sel[ch] > 5) {
scale_table = ff_dca_scale_factor_quant7;
scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
} else {
scale_table = ff_dca_scale_factor_quant6;
scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
}
// Parse scale factor indices and look up scale factors from the root
// square table
for (band = 0; band < xbr_nsubbands[ch]; band++) {
if (xbr_bit_allocation[ch][band]) {
int scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
if (scale_index >= scale_size) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
return AVERROR_INVALIDDATA;
}
xbr_scale_factors[ch][band][0] = scale_table[scale_index];
if (xbr_transition_mode && s->transition_mode[sf][ch][band]) {
scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
if (scale_index >= scale_size) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
return AVERROR_INVALIDDATA;
}
xbr_scale_factors[ch][band][1] = scale_table[scale_index];
}
}
}
}
// Audio data
for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
for (band = 0; band < xbr_nsubbands[ch]; band++) {
int ret, trans_ssf, abits = xbr_bit_allocation[ch][band];
int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;
// Extract bits from the bit stream
if (abits > 7) {
// No further encoding
get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
} else if (abits > 0) {
// Block codes
if ((ret = parse_block_codes(s, audio, abits)) < 0)
return ret;
} else {
// No bits allocated
continue;
}
// Look up quantization step size
step_size = ff_dca_lossless_quant[abits];
// Identify transient location
if (xbr_transition_mode)
trans_ssf = s->transition_mode[sf][ch][band];
else
trans_ssf = 0;
// Determine proper scale factor
if (trans_ssf == 0 || ssf < trans_ssf)
scale = xbr_scale_factors[ch][band][0];
else
scale = xbr_scale_factors[ch][band][1];
dequantize(s->subband_samples[ch][band] + ofs,
audio, step_size, scale, 1);
}
}
// DSYNC
if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
av_log(s->avctx, AV_LOG_ERROR, "XBR-DSYNC check failed\n");
return AVERROR_INVALIDDATA;
}
ofs += DCA_SUBBAND_SAMPLES;
}
// Advance subband sample pointer for the next subframe
*sub_pos = ofs;
return 0;
}
static int parse_xbr_frame(DCACoreDecoder *s)
{
int xbr_frame_size[DCA_EXSS_CHSETS_MAX];
int xbr_nchannels[DCA_EXSS_CHSETS_MAX];
int xbr_nsubbands[DCA_EXSS_CHSETS_MAX * DCA_EXSS_CHANNELS_MAX];
int xbr_nchsets, xbr_transition_mode, xbr_band_nbits, xbr_base_ch;
int i, ch1, ch2, ret, header_size, header_pos = get_bits_count(&s->gb);
// XBR sync word
if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XBR) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR sync word\n");
return AVERROR_INVALIDDATA;
}
// XBR frame header length
header_size = get_bits(&s->gb, 6) + 1;
// Check XBR frame header CRC
if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
&& ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR frame header checksum\n");
return AVERROR_INVALIDDATA;
}
// Number of channel sets
xbr_nchsets = get_bits(&s->gb, 2) + 1;
// Channel set data byte size
for (i = 0; i < xbr_nchsets; i++)
xbr_frame_size[i] = get_bits(&s->gb, 14) + 1;
// Transition mode flag
xbr_transition_mode = get_bits1(&s->gb);
// Channel set headers
for (i = 0, ch2 = 0; i < xbr_nchsets; i++) {
xbr_nchannels[i] = get_bits(&s->gb, 3) + 1;
xbr_band_nbits = get_bits(&s->gb, 2) + 5;
for (ch1 = 0; ch1 < xbr_nchannels[i]; ch1++, ch2++) {
xbr_nsubbands[ch2] = get_bits(&s->gb, xbr_band_nbits) + 1;
if (xbr_nsubbands[ch2] > DCA_SUBBANDS) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid number of active XBR subbands (%d)\n", xbr_nsubbands[ch2]);
return AVERROR_INVALIDDATA;
}
}
}
// Reserved
// Byte align
// CRC16 of XBR frame header
if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR frame header\n");
return AVERROR_INVALIDDATA;
}
// Channel set data
for (i = 0, xbr_base_ch = 0; i < xbr_nchsets; i++) {
header_pos = get_bits_count(&s->gb);
if (xbr_base_ch + xbr_nchannels[i] <= s->nchannels) {
int sf, sub_pos;
for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
if ((ret = parse_xbr_subframe(s, xbr_base_ch,
xbr_base_ch + xbr_nchannels[i],
xbr_nsubbands, xbr_transition_mode,
sf, &sub_pos)) < 0)
return ret;
}
}
xbr_base_ch += xbr_nchannels[i];
if (ff_dca_seek_bits(&s->gb, header_pos + xbr_frame_size[i] * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR channel set\n");
return AVERROR_INVALIDDATA;
}
}
return 0;
}
// Modified ISO/IEC 9899 linear congruential generator
// Returns pseudorandom integer in range [-2^30, 2^30 - 1]
static int rand_x96(DCACoreDecoder *s)
{
s->x96_rand = 1103515245U * s->x96_rand + 12345U;
return (s->x96_rand & 0x7fffffff) - 0x40000000;
}
static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int *sub_pos)
{
int n, ssf, ch, band, ofs;
// Check number of subband samples in this subframe
int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
if (*sub_pos + nsamples > s->npcmblocks) {
av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
return AVERROR_INVALIDDATA;
}
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
// VQ encoded or unallocated subbands
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
// Get the sample pointer and scale factor
int32_t *samples = s->x96_subband_samples[ch][band] + *sub_pos;
int32_t scale = s->scale_factors[ch][band >> 1][band & 1];
switch (s->bit_allocation[ch][band]) {
case 0: // No bits allocated for subband
if (scale <= 1)
memset(samples, 0, nsamples * sizeof(int32_t));
else for (n = 0; n < nsamples; n++)
// Generate scaled random samples
samples[n] = mul31(rand_x96(s), scale);
break;
case 1: // VQ encoded subband
for (ssf = 0; ssf < (s->nsubsubframes[sf] + 1) / 2; ssf++) {
// Extract the VQ address from the bit stream and look up
// the VQ code book for up to 16 subband samples
const int8_t *vq_samples = ff_dca_high_freq_vq[get_bits(&s->gb, 10)];
// Scale and take the samples
for (n = 0; n < FFMIN(nsamples - ssf * 16, 16); n++)
*samples++ = clip23(vq_samples[n] * scale + (1 << 3) >> 4);
}
break;
}
}
}
// Audio data
for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
int ret, abits = s->bit_allocation[ch][band] - 1;
int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;
// Not VQ encoded or unallocated subbands
if (abits < 1)
continue;
// Extract bits from the bit stream
if ((ret = extract_audio(s, audio, abits, ch)) < 0)
return ret;
// Select quantization step size table and look up quantization
// step size
if (s->bit_rate == 3)
step_size = ff_dca_lossless_quant[abits];
else
step_size = ff_dca_lossy_quant[abits];
// Get the scale factor
scale = s->scale_factors[ch][band >> 1][band & 1];
dequantize(s->x96_subband_samples[ch][band] + ofs,
audio, step_size, scale, 0);
}
}
// DSYNC
if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
av_log(s->avctx, AV_LOG_ERROR, "X96-DSYNC check failed\n");
return AVERROR_INVALIDDATA;
}
ofs += DCA_SUBBAND_SAMPLES;
}
// Inverse ADPCM
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
inverse_adpcm(s->x96_subband_samples[ch], s->prediction_vq_index[ch],
s->prediction_mode[ch], s->x96_subband_start, s->nsubbands[ch],
*sub_pos, nsamples);
}
// Joint subband coding
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
int src_ch = s->joint_intensity_index[ch] - 1;
if (src_ch >= 0) {
s->dcadsp->decode_joint(s->x96_subband_samples[ch], s->x96_subband_samples[src_ch],
s->joint_scale_factors[ch], s->nsubbands[ch],
s->nsubbands[src_ch], *sub_pos, nsamples);
}
}
// Advance subband sample pointer for the next subframe
*sub_pos = ofs;
return 0;
}
static void erase_x96_adpcm_history(DCACoreDecoder *s)
{
int ch, band;
// Erase ADPCM history from previous frame if
// predictor history switch was disabled
for (ch = 0; ch < DCA_CHANNELS; ch++)
for (band = 0; band < DCA_SUBBANDS_X96; band++)
AV_ZERO128(s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS);
emms_c();
}
static int alloc_x96_sample_buffer(DCACoreDecoder *s)
{
int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS_X96;
unsigned int size = s->x96_subband_size;
int ch, band;
// Reallocate subband sample buffer
av_fast_mallocz(&s->x96_subband_buffer, &s->x96_subband_size,
nframesamples * sizeof(int32_t));
if (!s->x96_subband_buffer)
return AVERROR(ENOMEM);
if (size != s->x96_subband_size) {
for (ch = 0; ch < DCA_CHANNELS; ch++)
for (band = 0; band < DCA_SUBBANDS_X96; band++)
s->x96_subband_samples[ch][band] = s->x96_subband_buffer +
(ch * DCA_SUBBANDS_X96 + band) * nchsamples + DCA_ADPCM_COEFFS;
}
if (!s->predictor_history)
erase_x96_adpcm_history(s);
return 0;
}
static int parse_x96_subframe_header(DCACoreDecoder *s, int xch_base)
{
int ch, band, ret;
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
// Prediction mode
for (ch = xch_base; ch < s->x96_nchannels; ch++)
for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
s->prediction_mode[ch][band] = get_bits1(&s->gb);
// Prediction coefficients VQ address
for (ch = xch_base; ch < s->x96_nchannels; ch++)
for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
if (s->prediction_mode[ch][band])
s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);
// Bit allocation index
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
int sel = s->bit_allocation_sel[ch];
int abits = 0;
for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
// If Huffman code was used, the difference of abits was encoded
if (sel < 7)
abits += dca_get_vlc(&s->gb, &vlc_quant_index[5 + 2 * s->x96_high_res], sel);
else
abits = get_bits(&s->gb, 3 + s->x96_high_res);
if (abits < 0 || abits > 7 + 8 * s->x96_high_res) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 bit allocation index\n");
return AVERROR_INVALIDDATA;
}
s->bit_allocation[ch][band] = abits;
}
}
// Scale factors
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
int sel = s->scale_factor_sel[ch];
int scale_index = 0;
// Extract scales for subbands which are transmitted even for
// unallocated subbands
for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
if ((ret = parse_scale(s, &scale_index, sel)) < 0)
return ret;
s->scale_factors[ch][band >> 1][band & 1] = ret;
}
}
// Joint subband codebook select
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
if (s->joint_intensity_index[ch]) {
s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
if (s->joint_scale_sel[ch] == 7) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint scale factor code book\n");
return AVERROR_INVALIDDATA;
}
}
}
// Scale factors for joint subband coding
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
int src_ch = s->joint_intensity_index[ch] - 1;
if (src_ch >= 0) {
int sel = s->joint_scale_sel[ch];
for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
if ((ret = parse_joint_scale(s, sel)) < 0)
return ret;
s->joint_scale_factors[ch][band] = ret;
}
}
}
// Side information CRC check word
if (s->crc_present)
skip_bits(&s->gb, 16);
return 0;
}
static int parse_x96_coding_header(DCACoreDecoder *s, int exss, int xch_base)
{
int n, ch, header_size = 0, header_pos = get_bits_count(&s->gb);
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
if (exss) {
// Channel set header length
header_size = get_bits(&s->gb, 7) + 1;
// Check CRC
if (s->x96_crc_present
&& (s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
&& ff_dca_check_crc(&s->gb, header_pos, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 channel set header checksum\n");
return AVERROR_INVALIDDATA;
}
}
// High resolution flag
s->x96_high_res = get_bits1(&s->gb);
// First encoded subband
if (s->x96_rev_no < 8) {
s->x96_subband_start = get_bits(&s->gb, 5);
if (s->x96_subband_start > 27) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband start index (%d)\n", s->x96_subband_start);
return AVERROR_INVALIDDATA;
}
} else {
s->x96_subband_start = DCA_SUBBANDS;
}
// Subband activity count
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
s->nsubbands[ch] = get_bits(&s->gb, 6) + 1;
if (s->nsubbands[ch] < DCA_SUBBANDS) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband activity count (%d)\n", s->nsubbands[ch]);
return AVERROR_INVALIDDATA;
}
}
// Joint intensity coding index
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
if ((n = get_bits(&s->gb, 3)) && xch_base)
n += xch_base - 1;
if (n > s->x96_nchannels) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint intensity coding index\n");
return AVERROR_INVALIDDATA;
}
s->joint_intensity_index[ch] = n;
}
// Scale factor code book
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
if (s->scale_factor_sel[ch] >= 6) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 scale factor code book\n");
return AVERROR_INVALIDDATA;
}
}
// Bit allocation quantizer select
for (ch = xch_base; ch < s->x96_nchannels; ch++)
s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
// Quantization index codebook select
for (n = 0; n < 6 + 4 * s->x96_high_res; n++)
for (ch = xch_base; ch < s->x96_nchannels; ch++)
s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);
if (exss) {
// Reserved
// Byte align
// CRC16 of channel set header
if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set header\n");
return AVERROR_INVALIDDATA;
}
} else {
if (s->crc_present)
skip_bits(&s->gb, 16);
}
return 0;
}
static int parse_x96_frame_data(DCACoreDecoder *s, int exss, int xch_base)
{
int sf, ch, ret, band, sub_pos;
if ((ret = parse_x96_coding_header(s, exss, xch_base)) < 0)
return ret;
for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
if ((ret = parse_x96_subframe_header(s, xch_base)) < 0)
return ret;
if ((ret = parse_x96_subframe_audio(s, sf, xch_base, &sub_pos)) < 0)
return ret;
}
for (ch = xch_base; ch < s->x96_nchannels; ch++) {
// Determine number of active subbands for this channel
int nsubbands = s->nsubbands[ch];
if (s->joint_intensity_index[ch])
nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);
// Update history for ADPCM and clear inactive subbands
for (band = 0; band < DCA_SUBBANDS_X96; band++) {
int32_t *samples = s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS;
if (band >= s->x96_subband_start && band < nsubbands)
AV_COPY128(samples, samples + s->npcmblocks);
else
memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
}
}
emms_c();
return 0;
}
static int parse_x96_frame(DCACoreDecoder *s)
{
int ret;
// Revision number
s->x96_rev_no = get_bits(&s->gb, 4);
if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
return AVERROR_INVALIDDATA;
}
s->x96_crc_present = 0;
s->x96_nchannels = s->nchannels;
if ((ret = alloc_x96_sample_buffer(s)) < 0)
return ret;
if ((ret = parse_x96_frame_data(s, 0, 0)) < 0)
return ret;
// Seek to the end of core frame
if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame\n");
return AVERROR_INVALIDDATA;
}
return 0;
}
static int parse_x96_frame_exss(DCACoreDecoder *s)
{
int x96_frame_size[DCA_EXSS_CHSETS_MAX];
int x96_nchannels[DCA_EXSS_CHSETS_MAX];
int x96_nchsets, x96_base_ch;
int i, ret, header_size, header_pos = get_bits_count(&s->gb);
// X96 sync word
if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_X96) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 sync word\n");
return AVERROR_INVALIDDATA;
}
// X96 frame header length
header_size = get_bits(&s->gb, 6) + 1;
// Check X96 frame header CRC
if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
&& ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 frame header checksum\n");
return AVERROR_INVALIDDATA;
}
// Revision number
s->x96_rev_no = get_bits(&s->gb, 4);
if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
return AVERROR_INVALIDDATA;
}
// CRC presence flag for channel set header
s->x96_crc_present = get_bits1(&s->gb);
// Number of channel sets
x96_nchsets = get_bits(&s->gb, 2) + 1;
// Channel set data byte size
for (i = 0; i < x96_nchsets; i++)
x96_frame_size[i] = get_bits(&s->gb, 12) + 1;
// Number of channels in channel set
for (i = 0; i < x96_nchsets; i++)
x96_nchannels[i] = get_bits(&s->gb, 3) + 1;
// Reserved
// Byte align
// CRC16 of X96 frame header
if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame header\n");
return AVERROR_INVALIDDATA;
}
if ((ret = alloc_x96_sample_buffer(s)) < 0)
return ret;
// Channel set data
for (i = 0, x96_base_ch = 0; i < x96_nchsets; i++) {
header_pos = get_bits_count(&s->gb);
if (x96_base_ch + x96_nchannels[i] <= s->nchannels) {
s->x96_nchannels = x96_base_ch + x96_nchannels[i];
if ((ret = parse_x96_frame_data(s, 1, x96_base_ch)) < 0)
return ret;
}
x96_base_ch += x96_nchannels[i];
if (ff_dca_seek_bits(&s->gb, header_pos + x96_frame_size[i] * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set\n");
return AVERROR_INVALIDDATA;
}
}
return 0;
}
static int parse_aux_data(DCACoreDecoder *s)
{
int aux_pos;
if (get_bits_left(&s->gb) < 0)
return AVERROR_INVALIDDATA;
// Auxiliary data byte count (can't be trusted)
skip_bits(&s->gb, 6);
// 4-byte align
skip_bits_long(&s->gb, -get_bits_count(&s->gb) & 31);
// Auxiliary data sync word
if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_REV1AUX) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data sync word\n");
return AVERROR_INVALIDDATA;
}
aux_pos = get_bits_count(&s->gb);
// Auxiliary decode time stamp flag
if (get_bits1(&s->gb))
skip_bits_long(&s->gb, 47);
// Auxiliary dynamic downmix flag
if (s->prim_dmix_embedded = get_bits1(&s->gb)) {
int i, m, n;
// Auxiliary primary channel downmix type
s->prim_dmix_type = get_bits(&s->gb, 3);
if (s->prim_dmix_type >= DCA_DMIX_TYPE_COUNT) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid primary channel set downmix type\n");
return AVERROR_INVALIDDATA;
}
// Size of downmix coefficients matrix
m = ff_dca_dmix_primary_nch[s->prim_dmix_type];
n = ff_dca_channels[s->audio_mode] + !!s->lfe_present;
// Dynamic downmix code coefficients
for (i = 0; i < m * n; i++) {
int code = get_bits(&s->gb, 9);
int sign = (code >> 8) - 1;
unsigned int index = code & 0xff;
if (index >= FF_DCA_DMIXTABLE_SIZE) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid downmix coefficient index\n");
return AVERROR_INVALIDDATA;
}
s->prim_dmix_coeff[i] = (ff_dca_dmixtable[index] ^ sign) - sign;
}
}
// Byte align
skip_bits(&s->gb, -get_bits_count(&s->gb) & 7);
// CRC16 of auxiliary data
skip_bits(&s->gb, 16);
// Check CRC
if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL))
&& ff_dca_check_crc(&s->gb, aux_pos, get_bits_count(&s->gb))) {
av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data checksum\n");
return AVERROR_INVALIDDATA;
}
return 0;
}
static int parse_optional_info(DCACoreDecoder *s)
{
DCAContext *dca = s->avctx->priv_data;
int ret = -1;
// Time code stamp
if (s->ts_present)
skip_bits_long(&s->gb, 32);
// Auxiliary data
if (s->aux_present && (ret = parse_aux_data(s)) < 0
&& (s->avctx->err_recognition & AV_EF_EXPLODE))
return ret;
if (ret < 0)
s->prim_dmix_embedded = 0;
// Core extensions
if (s->ext_audio_present && !dca->core_only) {
int sync_pos = FFMIN(s->frame_size / 4, s->gb.size_in_bits / 32) - 1;
int last_pos = get_bits_count(&s->gb) / 32;
int size, dist;
// Search for extension sync words aligned on 4-byte boundary. Search
// must be done backwards from the end of core frame to work around
// sync word aliasing issues.
switch (s->ext_audio_type) {
case EXT_AUDIO_XCH:
if (dca->request_channel_layout)
break;
// The distance between XCH sync word and end of the core frame
// must be equal to XCH frame size. Off by one error is allowed for
// compatibility with legacy bitstreams. Minimum XCH frame size is
// 96 bytes. AMODE and PCHS are further checked to reduce
// probability of alias sync detection.
for (; sync_pos >= last_pos; sync_pos--) {
if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XCH) {
s->gb.index = (sync_pos + 1) * 32;
size = get_bits(&s->gb, 10) + 1;
dist = s->frame_size - sync_pos * 4;
if (size >= 96
&& (size == dist || size - 1 == dist)
&& get_bits(&s->gb, 7) == 0x08) {
s->xch_pos = get_bits_count(&s->gb);
break;
}
}
}
if (s->avctx->err_recognition & AV_EF_EXPLODE) {
av_log(s->avctx, AV_LOG_ERROR, "XCH sync word not found\n");
return AVERROR_INVALIDDATA;
}
break;
case EXT_AUDIO_X96:
// The distance between X96 sync word and end of the core frame
// must be equal to X96 frame size. Minimum X96 frame size is 96
// bytes.
for (; sync_pos >= last_pos; sync_pos--) {
if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_X96) {
s->gb.index = (sync_pos + 1) * 32;
size = get_bits(&s->gb, 12) + 1;
dist = s->frame_size - sync_pos * 4;
if (size >= 96 && size == dist) {
s->x96_pos = get_bits_count(&s->gb);
break;
}
}
}
if (s->avctx->err_recognition & AV_EF_EXPLODE) {
av_log(s->avctx, AV_LOG_ERROR, "X96 sync word not found\n");
return AVERROR_INVALIDDATA;
}
break;
case EXT_AUDIO_XXCH:
if (dca->request_channel_layout)
break;
// XXCH frame header CRC must be valid. Minimum XXCH frame header
// size is 11 bytes.
for (; sync_pos >= last_pos; sync_pos--) {
if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XXCH) {
s->gb.index = (sync_pos + 1) * 32;
size = get_bits(&s->gb, 6) + 1;
if (size >= 11 &&
!ff_dca_check_crc(&s->gb, (sync_pos + 1) * 32,
sync_pos * 32 + size * 8)) {
s->xxch_pos = sync_pos * 32;
break;
}
}
}
if (s->avctx->err_recognition & AV_EF_EXPLODE) {
av_log(s->avctx, AV_LOG_ERROR, "XXCH sync word not found\n");
return AVERROR_INVALIDDATA;
}
break;
}
}
return 0;
}
int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size)
{
int ret;
s->ext_audio_mask = 0;
s->xch_pos = s->xxch_pos = s->x96_pos = 0;
if ((ret = init_get_bits8(&s->gb, data, size)) < 0)
return ret;
skip_bits_long(&s->gb, 32);
if ((ret = parse_frame_header(s)) < 0)
return ret;
if ((ret = alloc_sample_buffer(s)) < 0)
return ret;
if ((ret = parse_frame_data(s, HEADER_CORE, 0)) < 0)
return ret;
if ((ret = parse_optional_info(s)) < 0)
return ret;
// Workaround for DTS in WAV
if (s->frame_size > size && s->frame_size < size + 4) {
av_log(s->avctx, AV_LOG_DEBUG, "Working around excessive core frame size (%d > %d)\n", s->frame_size, size);
s->frame_size = size;
}
if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
av_log(s->avctx, AV_LOG_ERROR, "Read past end of core frame\n");
if (s->avctx->err_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
return 0;
}
int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset)
{
AVCodecContext *avctx = s->avctx;
DCAContext *dca = avctx->priv_data;
GetBitContext gb = s->gb;
int exss_mask = asset ? asset->extension_mask : 0;
int ret = 0, ext = 0;
// Parse (X)XCH unless downmixing
if (!dca->request_channel_layout) {
if (exss_mask & DCA_EXSS_XXCH) {
if ((ret = init_get_bits8(&s->gb, data + asset->xxch_offset, asset->xxch_size)) < 0)
return ret;
ret = parse_xxch_frame(s);
ext = DCA_EXSS_XXCH;
} else if (s->xxch_pos) {
s->gb.index = s->xxch_pos;
ret = parse_xxch_frame(s);
ext = DCA_CSS_XXCH;
} else if (s->xch_pos) {
s->gb.index = s->xch_pos;
ret = parse_xch_frame(s);
ext = DCA_CSS_XCH;
}
// Revert to primary channel set in case (X)XCH parsing fails
if (ret < 0) {
if (avctx->err_recognition & AV_EF_EXPLODE)
return ret;
s->nchannels = ff_dca_channels[s->audio_mode];
s->ch_mask = audio_mode_ch_mask[s->audio_mode];
if (s->lfe_present)
s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
} else {
s->ext_audio_mask |= ext;
}
}
// Parse XBR
if (exss_mask & DCA_EXSS_XBR) {
if ((ret = init_get_bits8(&s->gb, data + asset->xbr_offset, asset->xbr_size)) < 0)
return ret;
if ((ret = parse_xbr_frame(s)) < 0) {
if (avctx->err_recognition & AV_EF_EXPLODE)
return ret;
} else {
s->ext_audio_mask |= DCA_EXSS_XBR;
}
}
// Parse X96 unless decoding XLL
if (!(dca->packet & DCA_PACKET_XLL)) {
if (exss_mask & DCA_EXSS_X96) {
if ((ret = init_get_bits8(&s->gb, data + asset->x96_offset, asset->x96_size)) < 0)
return ret;
if ((ret = parse_x96_frame_exss(s)) < 0) {
if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
return ret;
} else {
s->ext_audio_mask |= DCA_EXSS_X96;
}
} else if (s->x96_pos) {
s->gb = gb;
s->gb.index = s->x96_pos;
if ((ret = parse_x96_frame(s)) < 0) {
if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
return ret;
} else {
s->ext_audio_mask |= DCA_CSS_X96;
}
}
}
return 0;
}
static int map_prm_ch_to_spkr(DCACoreDecoder *s, int ch)
{
int pos, spkr;
// Try to map this channel to core first
pos = ff_dca_channels[s->audio_mode];
if (ch < pos) {
spkr = prm_ch_to_spkr_map[s->audio_mode][ch];
if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
if (s->xxch_core_mask & (1U << spkr))
return spkr;
if (spkr == DCA_SPEAKER_Ls && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
return DCA_SPEAKER_Lss;
if (spkr == DCA_SPEAKER_Rs && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
return DCA_SPEAKER_Rss;
return -1;
}
return spkr;
}
// Then XCH
if ((s->ext_audio_mask & DCA_CSS_XCH) && ch == pos)
return DCA_SPEAKER_Cs;
// Then XXCH
if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
for (spkr = DCA_SPEAKER_Cs; spkr < s->xxch_mask_nbits; spkr++)
if (s->xxch_spkr_mask & (1U << spkr))
if (pos++ == ch)
return spkr;
}
// No mapping
return -1;
}
static void erase_dsp_history(DCACoreDecoder *s)
{
memset(s->dcadsp_data, 0, sizeof(s->dcadsp_data));
s->output_history_lfe_fixed = 0;
s->output_history_lfe_float = 0;
}
static void set_filter_mode(DCACoreDecoder *s, int mode)
{
if (s->filter_mode != mode) {
erase_dsp_history(s);
s->filter_mode = mode;
}
}
int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth)
{
int n, ch, spkr, nsamples, x96_nchannels = 0;
const int32_t *filter_coeff;
int32_t *ptr;
// Externally set x96_synth flag implies that X96 synthesis should be
// enabled, yet actual X96 subband data should be discarded. This is a
// special case for lossless residual decoder that ignores X96 data if
// present.
if (!x96_synth && (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96))) {
x96_nchannels = s->x96_nchannels;
x96_synth = 1;
}
if (x96_synth < 0)
x96_synth = 0;
s->output_rate = s->sample_rate << x96_synth;
s->npcmsamples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;
// Reallocate PCM output buffer
av_fast_malloc(&s->output_buffer, &s->output_size,
nsamples * av_popcount(s->ch_mask) * sizeof(int32_t));
if (!s->output_buffer)
return AVERROR(ENOMEM);
ptr = (int32_t *)s->output_buffer;
for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
if (s->ch_mask & (1U << spkr)) {
s->output_samples[spkr] = ptr;
ptr += nsamples;
} else {
s->output_samples[spkr] = NULL;
}
}
// Handle change of filtering mode
set_filter_mode(s, x96_synth | DCA_FILTER_MODE_FIXED);
// Select filter
if (x96_synth)
filter_coeff = ff_dca_fir_64bands_fixed;
else if (s->filter_perfect)
filter_coeff = ff_dca_fir_32bands_perfect_fixed;
else
filter_coeff = ff_dca_fir_32bands_nonperfect_fixed;
// Filter primary channels
for (ch = 0; ch < s->nchannels; ch++) {
// Map this primary channel to speaker
spkr = map_prm_ch_to_spkr(s, ch);
if (spkr < 0)
return AVERROR(EINVAL);
// Filter bank reconstruction
s->dcadsp->sub_qmf_fixed[x96_synth](
&s->synth,
&s->dcadct,
s->output_samples[spkr],
s->subband_samples[ch],
ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
s->dcadsp_data[ch].u.fix.hist1,
&s->dcadsp_data[ch].offset,
s->dcadsp_data[ch].u.fix.hist2,
filter_coeff,
s->npcmblocks);
}
// Filter LFE channel
if (s->lfe_present) {
int32_t *samples = s->output_samples[DCA_SPEAKER_LFE1];
int nlfesamples = s->npcmblocks >> 1;
// Check LFF
if (s->lfe_present == LFE_FLAG_128) {
av_log(s->avctx, AV_LOG_ERROR, "Fixed point mode doesn't support LFF=1\n");
return AVERROR(EINVAL);
}
// Offset intermediate buffer for X96
if (x96_synth)
samples += nsamples / 2;
// Interpolate LFE channel
s->dcadsp->lfe_fir_fixed(samples, s->lfe_samples + DCA_LFE_HISTORY,
ff_dca_lfe_fir_64_fixed, s->npcmblocks);
if (x96_synth) {
// Filter 96 kHz oversampled LFE PCM to attenuate high frequency
// (47.6 - 48.0 kHz) components of interpolation image
s->dcadsp->lfe_x96_fixed(s->output_samples[DCA_SPEAKER_LFE1],
samples, &s->output_history_lfe_fixed,
nsamples / 2);
}
// Update LFE history
for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
}
return 0;
}
static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame)
{
AVCodecContext *avctx = s->avctx;
DCAContext *dca = avctx->priv_data;
int i, n, ch, ret, spkr, nsamples;
// Don't filter twice when falling back from XLL
if (!(dca->packet & DCA_PACKET_XLL) && (ret = ff_dca_core_filter_fixed(s, 0)) < 0)
return ret;
avctx->sample_rate = s->output_rate;
avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
avctx->bits_per_raw_sample = 24;
frame->nb_samples = nsamples = s->npcmsamples;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
// Undo embedded XCH downmix
if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
&& s->audio_mode >= AMODE_2F2R) {
s->dcadsp->dmix_sub_xch(s->output_samples[DCA_SPEAKER_Ls],
s->output_samples[DCA_SPEAKER_Rs],
s->output_samples[DCA_SPEAKER_Cs],
nsamples);
}
// Undo embedded XXCH downmix
if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
&& s->xxch_dmix_embedded) {
int scale_inv = s->xxch_dmix_scale_inv;
int *coeff_ptr = s->xxch_dmix_coeff;
int xch_base = ff_dca_channels[s->audio_mode];
av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);
// Undo embedded core downmix pre-scaling
for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
if (s->xxch_core_mask & (1U << spkr)) {
s->dcadsp->dmix_scale_inv(s->output_samples[spkr],
scale_inv, nsamples);
}
}
// Undo downmix
for (ch = xch_base; ch < s->nchannels; ch++) {
int src_spkr = map_prm_ch_to_spkr(s, ch);
if (src_spkr < 0)
return AVERROR(EINVAL);
for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
int coeff = mul16(*coeff_ptr++, scale_inv);
if (coeff) {
s->dcadsp->dmix_sub(s->output_samples[spkr ],
s->output_samples[src_spkr],
coeff, nsamples);
}
}
}
}
}
if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
// Front sum/difference decoding
if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
|| s->audio_mode == AMODE_STEREO_SUMDIFF) {
s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_L],
s->output_samples[DCA_SPEAKER_R],
nsamples);
}
// Surround sum/difference decoding
if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_Ls],
s->output_samples[DCA_SPEAKER_Rs],
nsamples);
}
}
// Downmix primary channel set to stereo
if (s->request_mask != s->ch_mask) {
ff_dca_downmix_to_stereo_fixed(s->dcadsp,
s->output_samples,
s->prim_dmix_coeff,
nsamples, s->ch_mask);
}
for (i = 0; i < avctx->channels; i++) {
int32_t *samples = s->output_samples[s->ch_remap[i]];
int32_t *plane = (int32_t *)frame->extended_data[i];
for (n = 0; n < nsamples; n++)
plane[n] = clip23(samples[n]) * (1 << 8);
}
return 0;
}
static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame)
{
AVCodecContext *avctx = s->avctx;
int x96_nchannels = 0, x96_synth = 0;
int i, n, ch, ret, spkr, nsamples, nchannels;
float *output_samples[DCA_SPEAKER_COUNT] = { NULL }, *ptr;
const float *filter_coeff;
if (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96)) {
x96_nchannels = s->x96_nchannels;
x96_synth = 1;
}
avctx->sample_rate = s->sample_rate << x96_synth;
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
avctx->bits_per_raw_sample = 0;
frame->nb_samples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
// Build reverse speaker to channel mapping
for (i = 0; i < avctx->channels; i++)
output_samples[s->ch_remap[i]] = (float *)frame->extended_data[i];
// Allocate space for extra channels
nchannels = av_popcount(s->ch_mask) - avctx->channels;
if (nchannels > 0) {
av_fast_malloc(&s->output_buffer, &s->output_size,
nsamples * nchannels * sizeof(float));
if (!s->output_buffer)
return AVERROR(ENOMEM);
ptr = (float *)s->output_buffer;
for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
if (!(s->ch_mask & (1U << spkr)))
continue;
if (output_samples[spkr])
continue;
output_samples[spkr] = ptr;
ptr += nsamples;
}
}
// Handle change of filtering mode
set_filter_mode(s, x96_synth);
// Select filter
if (x96_synth)
filter_coeff = ff_dca_fir_64bands;
else if (s->filter_perfect)
filter_coeff = ff_dca_fir_32bands_perfect;
else
filter_coeff = ff_dca_fir_32bands_nonperfect;
// Filter primary channels
for (ch = 0; ch < s->nchannels; ch++) {
// Map this primary channel to speaker
spkr = map_prm_ch_to_spkr(s, ch);
if (spkr < 0)
return AVERROR(EINVAL);
// Filter bank reconstruction
s->dcadsp->sub_qmf_float[x96_synth](
&s->synth,
&s->imdct[x96_synth],
output_samples[spkr],
s->subband_samples[ch],
ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
s->dcadsp_data[ch].u.flt.hist1,
&s->dcadsp_data[ch].offset,
s->dcadsp_data[ch].u.flt.hist2,
filter_coeff,
s->npcmblocks,
1.0f / (1 << (17 - x96_synth)));
}
// Filter LFE channel
if (s->lfe_present) {
int dec_select = (s->lfe_present == LFE_FLAG_128);
float *samples = output_samples[DCA_SPEAKER_LFE1];
int nlfesamples = s->npcmblocks >> (dec_select + 1);
// Offset intermediate buffer for X96
if (x96_synth)
samples += nsamples / 2;
// Select filter
if (dec_select)
filter_coeff = ff_dca_lfe_fir_128;
else
filter_coeff = ff_dca_lfe_fir_64;
// Interpolate LFE channel
s->dcadsp->lfe_fir_float[dec_select](
samples, s->lfe_samples + DCA_LFE_HISTORY,
filter_coeff, s->npcmblocks);
if (x96_synth) {
// Filter 96 kHz oversampled LFE PCM to attenuate high frequency
// (47.6 - 48.0 kHz) components of interpolation image
s->dcadsp->lfe_x96_float(output_samples[DCA_SPEAKER_LFE1],
samples, &s->output_history_lfe_float,
nsamples / 2);
}
// Update LFE history
for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
}
// Undo embedded XCH downmix
if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
&& s->audio_mode >= AMODE_2F2R) {
s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Ls],
output_samples[DCA_SPEAKER_Cs],
-M_SQRT1_2, nsamples);
s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Rs],
output_samples[DCA_SPEAKER_Cs],
-M_SQRT1_2, nsamples);
}
// Undo embedded XXCH downmix
if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
&& s->xxch_dmix_embedded) {
float scale_inv = s->xxch_dmix_scale_inv * (1.0f / (1 << 16));
int *coeff_ptr = s->xxch_dmix_coeff;
int xch_base = ff_dca_channels[s->audio_mode];
av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);
// Undo downmix
for (ch = xch_base; ch < s->nchannels; ch++) {
int src_spkr = map_prm_ch_to_spkr(s, ch);
if (src_spkr < 0)
return AVERROR(EINVAL);
for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
int coeff = *coeff_ptr++;
if (coeff) {
s->float_dsp->vector_fmac_scalar(output_samples[ spkr],
output_samples[src_spkr],
coeff * (-1.0f / (1 << 15)),
nsamples);
}
}
}
}
// Undo embedded core downmix pre-scaling
for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
if (s->xxch_core_mask & (1U << spkr)) {
s->float_dsp->vector_fmul_scalar(output_samples[spkr],
output_samples[spkr],
scale_inv, nsamples);
}
}
}
if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
// Front sum/difference decoding
if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
|| s->audio_mode == AMODE_STEREO_SUMDIFF) {
s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_L],
output_samples[DCA_SPEAKER_R],
nsamples);
}
// Surround sum/difference decoding
if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_Ls],
output_samples[DCA_SPEAKER_Rs],
nsamples);
}
}
// Downmix primary channel set to stereo
if (s->request_mask != s->ch_mask) {
ff_dca_downmix_to_stereo_float(s->float_dsp, output_samples,
s->prim_dmix_coeff,
nsamples, s->ch_mask);
}
return 0;
}
int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame)
{
AVCodecContext *avctx = s->avctx;
DCAContext *dca = avctx->priv_data;
DCAExssAsset *asset = &dca->exss.assets[0];
enum AVMatrixEncoding matrix_encoding;
int ret;
// Handle downmixing to stereo request
if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO
&& s->audio_mode > AMODE_MONO && s->prim_dmix_embedded
&& (s->prim_dmix_type == DCA_DMIX_TYPE_LoRo ||
s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
s->request_mask = DCA_SPEAKER_LAYOUT_STEREO;
else
s->request_mask = s->ch_mask;
if (!ff_dca_set_channel_layout(avctx, s->ch_remap, s->request_mask))
return AVERROR(EINVAL);
// Force fixed point mode when falling back from XLL
if ((avctx->flags & AV_CODEC_FLAG_BITEXACT) || ((dca->packet & DCA_PACKET_EXSS)
&& (asset->extension_mask & DCA_EXSS_XLL)))
ret = filter_frame_fixed(s, frame);
else
ret = filter_frame_float(s, frame);
if (ret < 0)
return ret;
// Set profile, bit rate, etc
if (s->ext_audio_mask & DCA_EXSS_MASK)
avctx->profile = FF_PROFILE_DTS_HD_HRA;
else if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH))
avctx->profile = FF_PROFILE_DTS_ES;
else if (s->ext_audio_mask & DCA_CSS_X96)
avctx->profile = FF_PROFILE_DTS_96_24;
else
avctx->profile = FF_PROFILE_DTS;
if (s->bit_rate > 3 && !(s->ext_audio_mask & DCA_EXSS_MASK))
avctx->bit_rate = s->bit_rate;
else
avctx->bit_rate = 0;
if (s->audio_mode == AMODE_STEREO_TOTAL || (s->request_mask != s->ch_mask &&
s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
else
matrix_encoding = AV_MATRIX_ENCODING_NONE;
if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0)
return ret;
return 0;
}
av_cold void ff_dca_core_flush(DCACoreDecoder *s)
{
if (s->subband_buffer) {
erase_adpcm_history(s);
memset(s->lfe_samples, 0, DCA_LFE_HISTORY * sizeof(int32_t));
}
if (s->x96_subband_buffer)
erase_x96_adpcm_history(s);
erase_dsp_history(s);
}
av_cold int ff_dca_core_init(DCACoreDecoder *s)
{
dca_init_vlcs();
if (!(s->float_dsp = avpriv_float_dsp_alloc(0)))
return -1;
if (!(s->fixed_dsp = avpriv_alloc_fixed_dsp(0)))
return -1;
ff_dcadct_init(&s->dcadct);
if (ff_mdct_init(&s->imdct[0], 6, 1, 1.0) < 0)
return -1;
if (ff_mdct_init(&s->imdct[1], 7, 1, 1.0) < 0)
return -1;
ff_synth_filter_init(&s->synth);
s->x96_rand = 1;
return 0;
}
av_cold void ff_dca_core_close(DCACoreDecoder *s)
{
av_freep(&s->float_dsp);
av_freep(&s->fixed_dsp);
ff_mdct_end(&s->imdct[0]);
ff_mdct_end(&s->imdct[1]);
av_freep(&s->subband_buffer);
s->subband_size = 0;
av_freep(&s->x96_subband_buffer);
s->x96_subband_size = 0;
av_freep(&s->output_buffer);
s->output_size = 0;
}