259 lines
11 KiB
C
259 lines
11 KiB
C
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/*
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* Copyright (C) 2016 foo86
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVCODEC_DCA_CORE_H
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#define AVCODEC_DCA_CORE_H
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/fixed_dsp.h"
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#include "libavutil/mem.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "get_bits.h"
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#include "dca.h"
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#include "dca_exss.h"
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#include "dcadsp.h"
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#include "dcadct.h"
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#include "dcamath.h"
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#include "dcahuff.h"
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#include "fft.h"
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#include "synth_filter.h"
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#define DCA_CHANNELS 7
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#define DCA_SUBBANDS 32
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#define DCA_SUBBANDS_X96 64
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#define DCA_SUBFRAMES 16
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#define DCA_SUBBAND_SAMPLES 8
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#define DCA_PCMBLOCK_SAMPLES 32
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#define DCA_LFE_HISTORY 8
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#define DCA_ABITS_MAX 26
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#define DCA_CORE_CHANNELS_MAX 6
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#define DCA_DMIX_CHANNELS_MAX 4
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#define DCA_XXCH_CHANNELS_MAX 2
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#define DCA_EXSS_CHANNELS_MAX 8
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#define DCA_EXSS_CHSETS_MAX 4
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#define DCA_FILTER_MODE_X96 0x01
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#define DCA_FILTER_MODE_FIXED 0x02
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enum DCACoreAudioMode {
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DCA_AMODE_MONO, // Mode 0: A (mono)
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DCA_AMODE_MONO_DUAL, // Mode 1: A + B (dual mono)
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DCA_AMODE_STEREO, // Mode 2: L + R (stereo)
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DCA_AMODE_STEREO_SUMDIFF, // Mode 3: (L+R) + (L-R) (sum-diff)
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DCA_AMODE_STEREO_TOTAL, // Mode 4: LT + RT (left and right total)
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DCA_AMODE_3F, // Mode 5: C + L + R
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DCA_AMODE_2F1R, // Mode 6: L + R + S
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DCA_AMODE_3F1R, // Mode 7: C + L + R + S
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DCA_AMODE_2F2R, // Mode 8: L + R + SL + SR
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DCA_AMODE_3F2R, // Mode 9: C + L + R + SL + SR
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DCA_AMODE_COUNT
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};
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enum DCACoreExtAudioType {
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DCA_EXT_AUDIO_XCH = 0,
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DCA_EXT_AUDIO_X96 = 2,
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DCA_EXT_AUDIO_XXCH = 6
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};
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enum DCACoreLFEFlag {
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DCA_LFE_FLAG_NONE,
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DCA_LFE_FLAG_128,
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DCA_LFE_FLAG_64,
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DCA_LFE_FLAG_INVALID
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};
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typedef struct DCADSPData {
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union {
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struct {
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DECLARE_ALIGNED(32, float, hist1)[1024];
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DECLARE_ALIGNED(32, float, hist2)[64];
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} flt;
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struct {
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DECLARE_ALIGNED(32, int32_t, hist1)[1024];
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DECLARE_ALIGNED(32, int32_t, hist2)[64];
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} fix;
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} u;
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int offset;
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} DCADSPData;
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typedef struct DCACoreDecoder {
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AVCodecContext *avctx;
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GetBitContext gb;
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GetBitContext gb_in;
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// Bit stream header
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int crc_present; ///< CRC present flag
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int npcmblocks; ///< Number of PCM sample blocks
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int frame_size; ///< Primary frame byte size
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int audio_mode; ///< Audio channel arrangement
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int sample_rate; ///< Core audio sampling frequency
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int bit_rate; ///< Transmission bit rate
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int drc_present; ///< Embedded dynamic range flag
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int ts_present; ///< Embedded time stamp flag
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int aux_present; ///< Auxiliary data flag
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int ext_audio_type; ///< Extension audio descriptor flag
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int ext_audio_present; ///< Extended coding flag
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int sync_ssf; ///< Audio sync word insertion flag
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int lfe_present; ///< Low frequency effects flag
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int predictor_history; ///< Predictor history flag switch
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int filter_perfect; ///< Multirate interpolator switch
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int source_pcm_res; ///< Source PCM resolution
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int es_format; ///< Extended surround (ES) mastering flag
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int sumdiff_front; ///< Front sum/difference flag
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int sumdiff_surround; ///< Surround sum/difference flag
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// Primary audio coding header
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int nsubframes; ///< Number of subframes
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int nchannels; ///< Number of primary audio channels (incl. extension channels)
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int ch_mask; ///< Speaker layout mask (incl. LFE and extension channels)
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int8_t nsubbands[DCA_CHANNELS]; ///< Subband activity count
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int8_t subband_vq_start[DCA_CHANNELS]; ///< High frequency VQ start subband
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int8_t joint_intensity_index[DCA_CHANNELS]; ///< Joint intensity coding index
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int8_t transition_mode_sel[DCA_CHANNELS]; ///< Transient mode code book
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int8_t scale_factor_sel[DCA_CHANNELS]; ///< Scale factor code book
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int8_t bit_allocation_sel[DCA_CHANNELS]; ///< Bit allocation quantizer select
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int8_t quant_index_sel[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Quantization index codebook select
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int32_t scale_factor_adj[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Scale factor adjustment
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// Primary audio coding side information
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int8_t nsubsubframes[DCA_SUBFRAMES]; ///< Subsubframe count for each subframe
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int8_t prediction_mode[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction mode
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int16_t prediction_vq_index[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction coefficients VQ address
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int8_t bit_allocation[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Bit allocation index
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int8_t transition_mode[DCA_SUBFRAMES][DCA_CHANNELS][DCA_SUBBANDS]; ///< Transition mode
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int32_t scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2]; ///< Scale factors (2x for transients and X96)
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int8_t joint_scale_sel[DCA_CHANNELS]; ///< Joint subband codebook select
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int32_t joint_scale_factors[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Scale factors for joint subband coding
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// Auxiliary data
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int prim_dmix_embedded; ///< Auxiliary dynamic downmix flag
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int prim_dmix_type; ///< Auxiliary primary channel downmix type
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int prim_dmix_coeff[DCA_DMIX_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Dynamic downmix code coefficients
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// Core extensions
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int ext_audio_mask; ///< Bit mask of fully decoded core extensions
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// XCH extension data
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int xch_pos; ///< Bit position of XCH frame in core substream
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// XXCH extension data
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int xxch_crc_present; ///< CRC presence flag for XXCH channel set header
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int xxch_mask_nbits; ///< Number of bits for loudspeaker mask
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int xxch_core_mask; ///< Core loudspeaker activity mask
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int xxch_spkr_mask; ///< Loudspeaker layout mask
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int xxch_dmix_embedded; ///< Downmix already performed by encoder
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int xxch_dmix_scale_inv; ///< Downmix scale factor
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int xxch_dmix_mask[DCA_XXCH_CHANNELS_MAX]; ///< Downmix channel mapping mask
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int xxch_dmix_coeff[DCA_XXCH_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Downmix coefficients
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int xxch_pos; ///< Bit position of XXCH frame in core substream
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// X96 extension data
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int x96_rev_no; ///< X96 revision number
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int x96_crc_present; ///< CRC presence flag for X96 channel set header
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int x96_nchannels; ///< Number of primary channels in X96 extension
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int x96_high_res; ///< X96 high resolution flag
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int x96_subband_start; ///< First encoded subband in X96 extension
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int x96_rand; ///< Random seed for generating samples for unallocated X96 subbands
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int x96_pos; ///< Bit position of X96 frame in core substream
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// Sample buffers
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unsigned int x96_subband_size;
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int32_t *x96_subband_buffer; ///< X96 subband sample buffer base
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int32_t *x96_subband_samples[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< X96 subband samples
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unsigned int subband_size;
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int32_t *subband_buffer; ///< Subband sample buffer base
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int32_t *subband_samples[DCA_CHANNELS][DCA_SUBBANDS]; ///< Subband samples
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int32_t *lfe_samples; ///< Decimated LFE samples
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// DSP contexts
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DCADSPData dcadsp_data[DCA_CHANNELS]; ///< FIR history buffers
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DCADSPContext *dcadsp;
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DCADCTContext dcadct;
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FFTContext imdct[2];
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SynthFilterContext synth;
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AVFloatDSPContext *float_dsp;
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AVFixedDSPContext *fixed_dsp;
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// PCM output data
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unsigned int output_size;
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void *output_buffer; ///< PCM output buffer base
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int32_t *output_samples[DCA_SPEAKER_COUNT]; ///< PCM output for fixed point mode
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int32_t output_history_lfe_fixed; ///< LFE PCM history for X96 filter
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float output_history_lfe_float; ///< LFE PCM history for X96 filter
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int ch_remap[DCA_SPEAKER_COUNT]; ///< Channel to speaker map
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int request_mask; ///< Requested channel layout (for stereo downmix)
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int npcmsamples; ///< Number of PCM samples per channel
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int output_rate; ///< Output sample rate (1x or 2x header rate)
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int filter_mode; ///< Previous filtering mode for detecting changes
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} DCACoreDecoder;
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static inline int ff_dca_core_map_spkr(DCACoreDecoder *core, int spkr)
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{
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if (core->ch_mask & (1U << spkr))
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return spkr;
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if (spkr == DCA_SPEAKER_Lss && (core->ch_mask & DCA_SPEAKER_MASK_Ls))
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return DCA_SPEAKER_Ls;
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if (spkr == DCA_SPEAKER_Rss && (core->ch_mask & DCA_SPEAKER_MASK_Rs))
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return DCA_SPEAKER_Rs;
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return -1;
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}
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static inline void ff_dca_core_dequantize(int32_t *output, const int32_t *input,
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int32_t step_size, int32_t scale, int residual, int len)
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{
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// Account for quantizer step size
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int64_t step_scale = (int64_t)step_size * scale;
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int n, shift = 0;
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// Limit scale factor resolution to 22 bits
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if (step_scale > (1 << 23)) {
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shift = av_log2(step_scale >> 23) + 1;
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step_scale >>= shift;
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}
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// Scale the samples
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if (residual) {
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for (n = 0; n < len; n++)
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output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
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} else {
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for (n = 0; n < len; n++)
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output[n] = clip23(norm__(input[n] * step_scale, 22 - shift));
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}
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}
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int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size);
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int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset);
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int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth);
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int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame);
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av_cold void ff_dca_core_flush(DCACoreDecoder *s);
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av_cold int ff_dca_core_init(DCACoreDecoder *s);
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av_cold void ff_dca_core_close(DCACoreDecoder *s);
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#endif
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