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FFmpeg/libavcodec/dca_core.c
foo86 db44b59980 avcodec/dca: clear X96 channels if nothing was decoded 
The first X96 channel set can have more channels than core, causing X96
decoding to be skipped. Clear the number of decoded X96 channels to zero
in this rudimentary case.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-03-06 18:24:20 +01:00

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/*
* 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
s->x96_nchannels = 0;
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;
}
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