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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
* 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 "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/fixed_dsp.h"
#include "libavutil/mem_internal.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "dca.h"
#include "dca_exss.h"
#include "dcadsp.h"
#include "dcadct.h"
#include "dcamath.h"
#include "dcahuff.h"
#include "fft.h"
#include "synth_filter.h"
#define DCA_CHANNELS 7
#define DCA_SUBBANDS 32
#define DCA_SUBBANDS_X96 64
#define DCA_SUBFRAMES 16
#define DCA_ABITS_MAX 26
#define DCA_FILTER_MODE_X96 0x01
enum DCACoreAudioMode {
DCA_AMODE_MONO, // Mode 0: A (mono)
DCA_AMODE_MONO_DUAL, // Mode 1: A + B (dual mono)
DCA_AMODE_STEREO, // Mode 2: L + R (stereo)
DCA_AMODE_STEREO_SUMDIFF, // Mode 3: (L+R) + (L-R) (sum-diff)
DCA_AMODE_STEREO_TOTAL, // Mode 4: LT + RT (left and right total)
DCA_AMODE_3F, // Mode 5: C + L + R
DCA_AMODE_2F1R, // Mode 6: L + R + S
DCA_AMODE_3F1R, // Mode 7: C + L + R + S
DCA_AMODE_2F2R, // Mode 8: L + R + SL + SR
DCA_AMODE_3F2R, // Mode 9: C + L + R + SL + SR
enum DCACoreExtAudioType {
enum DCACoreLFEFlag {
typedef struct DCADSPData {
union {
struct {
DECLARE_ALIGNED(32, float, hist1)[1024];
DECLARE_ALIGNED(32, float, hist2)[64];
} flt;
struct {
DECLARE_ALIGNED(32, int32_t, hist1)[1024];
DECLARE_ALIGNED(32, int32_t, hist2)[64];
} fix;
} u;
int offset;
typedef struct DCACoreDecoder {
AVCodecContext *avctx;
GetBitContext gb;
GetBitContext gb_in;
// Bit stream header
int crc_present; ///< CRC present flag
int npcmblocks; ///< Number of PCM sample blocks
int frame_size; ///< Primary frame byte size
int audio_mode; ///< Audio channel arrangement
int sample_rate; ///< Core audio sampling frequency
int bit_rate; ///< Transmission bit rate
int drc_present; ///< Embedded dynamic range flag
int ts_present; ///< Embedded time stamp flag
int aux_present; ///< Auxiliary data flag
int ext_audio_type; ///< Extension audio descriptor flag
int ext_audio_present; ///< Extended coding flag
int sync_ssf; ///< Audio sync word insertion flag
int lfe_present; ///< Low frequency effects flag
int predictor_history; ///< Predictor history flag switch
int filter_perfect; ///< Multirate interpolator switch
int source_pcm_res; ///< Source PCM resolution
int es_format; ///< Extended surround (ES) mastering flag
int sumdiff_front; ///< Front sum/difference flag
int sumdiff_surround; ///< Surround sum/difference flag
// Primary audio coding header
int nsubframes; ///< Number of subframes
int nchannels; ///< Number of primary audio channels (incl. extension channels)
int ch_mask; ///< Speaker layout mask (incl. LFE and extension channels)
int8_t nsubbands[DCA_CHANNELS]; ///< Subband activity count
int8_t subband_vq_start[DCA_CHANNELS]; ///< High frequency VQ start subband
int8_t joint_intensity_index[DCA_CHANNELS]; ///< Joint intensity coding index
int8_t transition_mode_sel[DCA_CHANNELS]; ///< Transient mode code book
int8_t scale_factor_sel[DCA_CHANNELS]; ///< Scale factor code book
int8_t bit_allocation_sel[DCA_CHANNELS]; ///< Bit allocation quantizer select
int8_t quant_index_sel[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Quantization index codebook select
int32_t scale_factor_adj[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Scale factor adjustment
// Primary audio coding side information
int8_t nsubsubframes[DCA_SUBFRAMES]; ///< Subsubframe count for each subframe
int8_t prediction_mode[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction mode
int16_t prediction_vq_index[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction coefficients VQ address
int8_t bit_allocation[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Bit allocation index
int8_t transition_mode[DCA_SUBFRAMES][DCA_CHANNELS][DCA_SUBBANDS]; ///< Transition mode
int32_t scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2]; ///< Scale factors (2x for transients and X96)
int8_t joint_scale_sel[DCA_CHANNELS]; ///< Joint subband codebook select
int32_t joint_scale_factors[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Scale factors for joint subband coding
// Auxiliary data
int prim_dmix_embedded; ///< Auxiliary dynamic downmix flag
int prim_dmix_type; ///< Auxiliary primary channel downmix type
int prim_dmix_coeff[DCA_DMIX_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Dynamic downmix code coefficients
// Core extensions
int ext_audio_mask; ///< Bit mask of fully decoded core extensions
// XCH extension data
int xch_pos; ///< Bit position of XCH frame in core substream
// XXCH extension data
int xxch_crc_present; ///< CRC presence flag for XXCH channel set header
int xxch_mask_nbits; ///< Number of bits for loudspeaker mask
int xxch_core_mask; ///< Core loudspeaker activity mask
int xxch_spkr_mask; ///< Loudspeaker layout mask
int xxch_dmix_embedded; ///< Downmix already performed by encoder
int xxch_dmix_scale_inv; ///< Downmix scale factor
int xxch_dmix_mask[DCA_XXCH_CHANNELS_MAX]; ///< Downmix channel mapping mask
int xxch_dmix_coeff[DCA_XXCH_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Downmix coefficients
int xxch_pos; ///< Bit position of XXCH frame in core substream
// X96 extension data
int x96_rev_no; ///< X96 revision number
int x96_crc_present; ///< CRC presence flag for X96 channel set header
int x96_nchannels; ///< Number of primary channels in X96 extension
int x96_high_res; ///< X96 high resolution flag
int x96_subband_start; ///< First encoded subband in X96 extension
int x96_rand; ///< Random seed for generating samples for unallocated X96 subbands
int x96_pos; ///< Bit position of X96 frame in core substream
// Sample buffers
unsigned int x96_subband_size;
int32_t *x96_subband_buffer; ///< X96 subband sample buffer base
int32_t *x96_subband_samples[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< X96 subband samples
unsigned int subband_size;
int32_t *subband_buffer; ///< Subband sample buffer base
int32_t *subband_samples[DCA_CHANNELS][DCA_SUBBANDS]; ///< Subband samples
int32_t *lfe_samples; ///< Decimated LFE samples
// DSP contexts
DCADSPData dcadsp_data[DCA_CHANNELS]; ///< FIR history buffers
DCADSPContext *dcadsp;
DCADCTContext dcadct;
FFTContext imdct[2];
SynthFilterContext synth;
AVFloatDSPContext *float_dsp;
AVFixedDSPContext *fixed_dsp;
// PCM output data
unsigned int output_size;
void *output_buffer; ///< PCM output buffer base
int32_t *output_samples[DCA_SPEAKER_COUNT]; ///< PCM output for fixed point mode
int32_t output_history_lfe_fixed; ///< LFE PCM history for X96 filter
float output_history_lfe_float; ///< LFE PCM history for X96 filter
int ch_remap[DCA_SPEAKER_COUNT]; ///< Channel to speaker map
int request_mask; ///< Requested channel layout (for stereo downmix)
int npcmsamples; ///< Number of PCM samples per channel
int output_rate; ///< Output sample rate (1x or 2x header rate)
int filter_mode; ///< Previous filtering mode for detecting changes
} DCACoreDecoder;
static inline int ff_dca_core_map_spkr(DCACoreDecoder *core, int spkr)
if (core->ch_mask & (1U << spkr))
return spkr;
if (spkr == DCA_SPEAKER_Lss && (core->ch_mask & DCA_SPEAKER_MASK_Ls))
return DCA_SPEAKER_Ls;
if (spkr == DCA_SPEAKER_Rss && (core->ch_mask & DCA_SPEAKER_MASK_Rs))
return DCA_SPEAKER_Rs;
return -1;
static inline void ff_dca_core_dequantize(int32_t *output, const int32_t *input,
int32_t step_size, int32_t scale, int residual, int len)
// 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 < len; n++)
output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
} else {
for (n = 0; n < len; n++)
output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size);
int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset);
int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth);
int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame);
av_cold void ff_dca_core_flush(DCACoreDecoder *s);
av_cold int ff_dca_core_init(DCACoreDecoder *s);
av_cold void ff_dca_core_close(DCACoreDecoder *s);