/* * 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 */ #ifndef AVCODEC_DCA_CORE_H#define AVCODEC_DCA_CORE_H #include "libavutil/float_dsp.h"#include "libavutil/fixed_dsp.h"#include "libavutil/mem_internal.h"#include "libavutil/tx.h" #include "avcodec.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 "synth_filter.h" #define DCA_CHANNELS            7#define DCA_SUBBANDS            32#define DCA_SUBBANDS_X96        64#define DCA_SUBFRAMES           16#define DCA_SUBBAND_SAMPLES     8#define DCA_PCMBLOCK_SAMPLES    32#define DCA_LFE_HISTORY         8#define DCA_ABITS_MAX           26 #define DCA_CORE_CHANNELS_MAX       6#define DCA_DMIX_CHANNELS_MAX       4#define DCA_XXCH_CHANNELS_MAX       2#define DCA_EXSS_CHANNELS_MAX       8#define DCA_EXSS_CHSETS_MAX         4 #define DCA_FILTER_MODE_X96     0x01#define DCA_FILTER_MODE_FIXED   0x02 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     DCA_AMODE_COUNT}; enum DCACoreExtAudioType {    DCA_EXT_AUDIO_XCH   = 0,    DCA_EXT_AUDIO_X96   = 2,    DCA_EXT_AUDIO_XXCH  = 6}; enum DCACoreLFEFlag {    DCA_LFE_FLAG_NONE,    DCA_LFE_FLAG_128,    DCA_LFE_FLAG_64,    DCA_LFE_FLAG_INVALID}; 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;} DCADSPData; 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;    AVTXContext            *imdct[2];    av_tx_fn                imdct_fn[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, const uint8_t *data, int size);int ff_dca_core_parse_exss(DCACoreDecoder *s, const 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); #endif