/* * 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/common.h" #include "libavutil/float_dsp.h" #include "libavutil/fixed_dsp.h" #include "libavutil/mem.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_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; 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); #endif