2007-02-27 06:30:40 +00:00
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/*
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* DCA compatible decoder
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* Copyright (C) 2004 Gildas Bazin
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* Copyright (C) 2004 Benjamin Zores
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* Copyright (C) 2006 Benjamin Larsson
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* Copyright (C) 2007 Konstantin Shishkov
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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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/**
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* @file dca.c
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*/
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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#include "avcodec.h"
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#include "dsputil.h"
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#include "bitstream.h"
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#include "dcadata.h"
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#include "dcahuff.h"
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2007-05-03 23:50:24 +00:00
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#include "dca.h"
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2007-02-27 06:30:40 +00:00
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//#define TRACE
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#define DCA_PRIM_CHANNELS_MAX (5)
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#define DCA_SUBBANDS (32)
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#define DCA_ABITS_MAX (32) /* Should be 28 */
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#define DCA_SUBSUBFAMES_MAX (4)
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#define DCA_LFE_MAX (3)
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enum DCAMode {
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DCA_MONO = 0,
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DCA_CHANNEL,
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DCA_STEREO,
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DCA_STEREO_SUMDIFF,
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DCA_STEREO_TOTAL,
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DCA_3F,
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DCA_2F1R,
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DCA_3F1R,
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DCA_2F2R,
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DCA_3F2R,
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DCA_4F2R
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};
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#define DCA_DOLBY 101 /* FIXME */
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#define DCA_CHANNEL_BITS 6
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#define DCA_CHANNEL_MASK 0x3F
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#define DCA_LFE 0x80
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#define HEADER_SIZE 14
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#define CONVERT_BIAS 384
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2008-08-23 13:29:13 +00:00
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#define DCA_MAX_FRAME_SIZE 16384
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2007-02-27 06:30:40 +00:00
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/** Bit allocation */
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typedef struct {
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int offset; ///< code values offset
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int maxbits[8]; ///< max bits in VLC
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int wrap; ///< wrap for get_vlc2()
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VLC vlc[8]; ///< actual codes
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} BitAlloc;
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static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select
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static BitAlloc dca_tmode; ///< transition mode VLCs
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static BitAlloc dca_scalefactor; ///< scalefactor VLCs
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static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs
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/** Pre-calculated cosine modulation coefs for the QMF */
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static float cos_mod[544];
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2007-05-05 12:18:14 +00:00
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static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx)
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2007-02-27 06:30:40 +00:00
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{
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return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset;
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}
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typedef struct {
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AVCodecContext *avctx;
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/* Frame header */
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int frame_type; ///< type of the current frame
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int samples_deficit; ///< deficit sample count
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int crc_present; ///< crc is present in the bitstream
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int sample_blocks; ///< number of PCM sample blocks
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int frame_size; ///< primary frame byte size
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int amode; ///< audio channels arrangement
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int sample_rate; ///< audio sampling rate
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int bit_rate; ///< transmission bit rate
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int downmix; ///< embedded downmix enabled
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int dynrange; ///< embedded dynamic range flag
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int timestamp; ///< embedded time stamp flag
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int aux_data; ///< auxiliary data flag
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int hdcd; ///< source material is mastered in HDCD
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int ext_descr; ///< extension audio descriptor flag
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int ext_coding; ///< extended coding flag
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int aspf; ///< audio sync word insertion flag
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int lfe; ///< low frequency effects flag
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int predictor_history; ///< predictor history flag
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int header_crc; ///< header crc check bytes
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int multirate_inter; ///< multirate interpolator switch
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int version; ///< encoder software revision
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int copy_history; ///< copy history
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int source_pcm_res; ///< source pcm resolution
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int front_sum; ///< front sum/difference flag
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int surround_sum; ///< surround sum/difference flag
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int dialog_norm; ///< dialog normalisation parameter
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/* Primary audio coding header */
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int subframes; ///< number of subframes
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2008-03-06 14:22:02 +00:00
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int total_channels; ///< number of channels including extensions
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2007-02-27 06:30:40 +00:00
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int prim_channels; ///< number of primary audio channels
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int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count
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int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband
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int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index
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int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book
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int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book
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int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select
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int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select
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float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
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/* Primary audio coding side information */
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int subsubframes; ///< number of subsubframes
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int partial_samples; ///< partial subsubframe samples count
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int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
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int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs
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int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index
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int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients)
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int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient)
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int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook
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int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors
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int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients
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int dynrange_coef; ///< dynamic range coefficient
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int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
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float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX *
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2 /*history */ ]; ///< Low frequency effect data
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int lfe_scale_factor;
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/* Subband samples history (for ADPCM) */
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float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
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float subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512];
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float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][64];
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int output; ///< type of output
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int bias; ///< output bias
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DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
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DECLARE_ALIGNED_16(int16_t, tsamples[1536]);
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uint8_t dca_buffer[DCA_MAX_FRAME_SIZE];
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int dca_buffer_size; ///< how much data is in the dca_buffer
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GetBitContext gb;
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/* Current position in DCA frame */
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int current_subframe;
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int current_subsubframe;
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int debug_flag; ///< used for suppressing repeated error messages output
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DSPContext dsp;
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} DCAContext;
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2008-03-21 03:11:20 +00:00
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static av_cold void dca_init_vlcs(void)
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2007-02-27 06:30:40 +00:00
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{
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2008-02-13 09:26:10 +00:00
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static int vlcs_initialized = 0;
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2007-02-27 06:30:40 +00:00
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int i, j;
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2008-02-13 09:26:10 +00:00
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if (vlcs_initialized)
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2007-02-27 06:30:40 +00:00
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return;
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dca_bitalloc_index.offset = 1;
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2007-05-24 04:08:48 +00:00
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dca_bitalloc_index.wrap = 2;
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2007-02-27 06:30:40 +00:00
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for (i = 0; i < 5; i++)
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init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12,
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bitalloc_12_bits[i], 1, 1,
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bitalloc_12_codes[i], 2, 2, 1);
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dca_scalefactor.offset = -64;
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dca_scalefactor.wrap = 2;
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for (i = 0; i < 5; i++)
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init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129,
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scales_bits[i], 1, 1,
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scales_codes[i], 2, 2, 1);
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dca_tmode.offset = 0;
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dca_tmode.wrap = 1;
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for (i = 0; i < 4; i++)
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init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4,
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tmode_bits[i], 1, 1,
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tmode_codes[i], 2, 2, 1);
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for(i = 0; i < 10; i++)
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for(j = 0; j < 7; j++){
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if(!bitalloc_codes[i][j]) break;
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dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i];
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dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4);
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init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j],
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bitalloc_sizes[i],
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bitalloc_bits[i][j], 1, 1,
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bitalloc_codes[i][j], 2, 2, 1);
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}
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2008-02-13 09:26:10 +00:00
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vlcs_initialized = 1;
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2007-02-27 06:30:40 +00:00
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}
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static inline void get_array(GetBitContext *gb, int *dst, int len, int bits)
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{
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while(len--)
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*dst++ = get_bits(gb, bits);
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}
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static int dca_parse_frame_header(DCAContext * s)
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{
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int i, j;
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static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 };
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static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
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static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
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s->bias = CONVERT_BIAS;
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init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
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/* Sync code */
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get_bits(&s->gb, 32);
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/* Frame header */
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s->frame_type = get_bits(&s->gb, 1);
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s->samples_deficit = get_bits(&s->gb, 5) + 1;
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s->crc_present = get_bits(&s->gb, 1);
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s->sample_blocks = get_bits(&s->gb, 7) + 1;
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s->frame_size = get_bits(&s->gb, 14) + 1;
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if (s->frame_size < 95)
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return -1;
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s->amode = get_bits(&s->gb, 6);
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s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)];
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if (!s->sample_rate)
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return -1;
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s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)];
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if (!s->bit_rate)
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return -1;
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s->downmix = get_bits(&s->gb, 1);
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s->dynrange = get_bits(&s->gb, 1);
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s->timestamp = get_bits(&s->gb, 1);
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s->aux_data = get_bits(&s->gb, 1);
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s->hdcd = get_bits(&s->gb, 1);
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s->ext_descr = get_bits(&s->gb, 3);
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s->ext_coding = get_bits(&s->gb, 1);
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s->aspf = get_bits(&s->gb, 1);
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s->lfe = get_bits(&s->gb, 2);
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s->predictor_history = get_bits(&s->gb, 1);
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/* TODO: check CRC */
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if (s->crc_present)
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s->header_crc = get_bits(&s->gb, 16);
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s->multirate_inter = get_bits(&s->gb, 1);
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s->version = get_bits(&s->gb, 4);
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s->copy_history = get_bits(&s->gb, 2);
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s->source_pcm_res = get_bits(&s->gb, 3);
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s->front_sum = get_bits(&s->gb, 1);
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s->surround_sum = get_bits(&s->gb, 1);
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s->dialog_norm = get_bits(&s->gb, 4);
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/* FIXME: channels mixing levels */
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2007-05-02 20:31:32 +00:00
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s->output = s->amode;
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if(s->lfe) s->output |= DCA_LFE;
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2007-02-27 06:30:40 +00:00
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#ifdef TRACE
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av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type);
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av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit);
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av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present);
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av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n",
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s->sample_blocks, s->sample_blocks * 32);
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av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size);
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av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n",
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s->amode, dca_channels[s->amode]);
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av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n",
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s->sample_rate, dca_sample_rates[s->sample_rate]);
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av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n",
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s->bit_rate, dca_bit_rates[s->bit_rate]);
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av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix);
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av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange);
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av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp);
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av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data);
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av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd);
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av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr);
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av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding);
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av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf);
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av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe);
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av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n",
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|
s->predictor_history);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n",
|
|
|
|
s->multirate_inter);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG,
|
|
|
|
"source pcm resolution: %i (%i bits/sample)\n",
|
|
|
|
s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
#endif
|
|
|
|
|
|
|
|
/* Primary audio coding header */
|
|
|
|
s->subframes = get_bits(&s->gb, 4) + 1;
|
2008-03-06 14:22:02 +00:00
|
|
|
s->total_channels = get_bits(&s->gb, 3) + 1;
|
|
|
|
s->prim_channels = s->total_channels;
|
|
|
|
if (s->prim_channels > DCA_PRIM_CHANNELS_MAX)
|
|
|
|
s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */
|
2007-02-27 06:30:40 +00:00
|
|
|
|
|
|
|
|
|
|
|
for (i = 0; i < s->prim_channels; i++) {
|
|
|
|
s->subband_activity[i] = get_bits(&s->gb, 5) + 2;
|
|
|
|
if (s->subband_activity[i] > DCA_SUBBANDS)
|
|
|
|
s->subband_activity[i] = DCA_SUBBANDS;
|
|
|
|
}
|
|
|
|
for (i = 0; i < s->prim_channels; i++) {
|
|
|
|
s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
|
|
|
|
if (s->vq_start_subband[i] > DCA_SUBBANDS)
|
|
|
|
s->vq_start_subband[i] = DCA_SUBBANDS;
|
|
|
|
}
|
|
|
|
get_array(&s->gb, s->joint_intensity, s->prim_channels, 3);
|
|
|
|
get_array(&s->gb, s->transient_huffman, s->prim_channels, 2);
|
|
|
|
get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3);
|
|
|
|
get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3);
|
|
|
|
|
|
|
|
/* Get codebooks quantization indexes */
|
|
|
|
memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman));
|
|
|
|
for (j = 1; j < 11; j++)
|
|
|
|
for (i = 0; i < s->prim_channels; i++)
|
|
|
|
s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]);
|
|
|
|
|
|
|
|
/* Get scale factor adjustment */
|
|
|
|
for (j = 0; j < 11; j++)
|
|
|
|
for (i = 0; i < s->prim_channels; i++)
|
|
|
|
s->scalefactor_adj[i][j] = 1;
|
|
|
|
|
|
|
|
for (j = 1; j < 11; j++)
|
|
|
|
for (i = 0; i < s->prim_channels; i++)
|
|
|
|
if (s->quant_index_huffman[i][j] < thr[j])
|
|
|
|
s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)];
|
|
|
|
|
|
|
|
if (s->crc_present) {
|
|
|
|
/* Audio header CRC check */
|
|
|
|
get_bits(&s->gb, 16);
|
|
|
|
}
|
|
|
|
|
|
|
|
s->current_subframe = 0;
|
|
|
|
s->current_subsubframe = 0;
|
|
|
|
|
|
|
|
#ifdef TRACE
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels);
|
|
|
|
for(i = 0; i < s->prim_channels; i++){
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:");
|
|
|
|
for (j = 0; j < 11; j++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i",
|
|
|
|
s->quant_index_huffman[i][j]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:");
|
|
|
|
for (j = 0; j < 11; j++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
#endif
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
|
2007-04-24 17:31:39 +00:00
|
|
|
static inline int get_scale(GetBitContext *gb, int level, int value)
|
2007-02-27 06:30:40 +00:00
|
|
|
{
|
|
|
|
if (level < 5) {
|
|
|
|
/* huffman encoded */
|
2007-04-24 17:31:39 +00:00
|
|
|
value += get_bitalloc(gb, &dca_scalefactor, level);
|
2007-02-27 06:30:40 +00:00
|
|
|
} else if(level < 8)
|
|
|
|
value = get_bits(gb, level + 1);
|
|
|
|
return value;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int dca_subframe_header(DCAContext * s)
|
|
|
|
{
|
|
|
|
/* Primary audio coding side information */
|
|
|
|
int j, k;
|
|
|
|
|
|
|
|
s->subsubframes = get_bits(&s->gb, 2) + 1;
|
|
|
|
s->partial_samples = get_bits(&s->gb, 3);
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
|
|
s->prediction_mode[j][k] = get_bits(&s->gb, 1);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Get prediction codebook */
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
|
|
if (s->prediction_mode[j][k] > 0) {
|
|
|
|
/* (Prediction coefficient VQ address) */
|
|
|
|
s->prediction_vq[j][k] = get_bits(&s->gb, 12);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Bit allocation index */
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
for (k = 0; k < s->vq_start_subband[j]; k++) {
|
|
|
|
if (s->bitalloc_huffman[j] == 6)
|
|
|
|
s->bitalloc[j][k] = get_bits(&s->gb, 5);
|
|
|
|
else if (s->bitalloc_huffman[j] == 5)
|
|
|
|
s->bitalloc[j][k] = get_bits(&s->gb, 4);
|
2008-03-06 14:22:02 +00:00
|
|
|
else if (s->bitalloc_huffman[j] == 7) {
|
|
|
|
av_log(s->avctx, AV_LOG_ERROR,
|
|
|
|
"Invalid bit allocation index\n");
|
|
|
|
return -1;
|
|
|
|
} else {
|
2007-02-27 06:30:40 +00:00
|
|
|
s->bitalloc[j][k] =
|
2007-04-24 17:31:39 +00:00
|
|
|
get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]);
|
2007-02-27 06:30:40 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
if (s->bitalloc[j][k] > 26) {
|
|
|
|
// av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n",
|
|
|
|
// j, k, s->bitalloc[j][k]);
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Transition mode */
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
|
|
s->transition_mode[j][k] = 0;
|
|
|
|
if (s->subsubframes > 1 &&
|
|
|
|
k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) {
|
|
|
|
s->transition_mode[j][k] =
|
|
|
|
get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
2008-02-01 01:40:46 +00:00
|
|
|
const uint32_t *scale_table;
|
2007-02-27 06:30:40 +00:00
|
|
|
int scale_sum;
|
|
|
|
|
|
|
|
memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2);
|
|
|
|
|
|
|
|
if (s->scalefactor_huffman[j] == 6)
|
2008-02-01 01:40:46 +00:00
|
|
|
scale_table = scale_factor_quant7;
|
2007-02-27 06:30:40 +00:00
|
|
|
else
|
2008-02-01 01:40:46 +00:00
|
|
|
scale_table = scale_factor_quant6;
|
2007-02-27 06:30:40 +00:00
|
|
|
|
|
|
|
/* When huffman coded, only the difference is encoded */
|
|
|
|
scale_sum = 0;
|
|
|
|
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
|
|
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) {
|
2007-04-24 17:31:39 +00:00
|
|
|
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
|
2007-02-27 06:30:40 +00:00
|
|
|
s->scale_factor[j][k][0] = scale_table[scale_sum];
|
|
|
|
}
|
|
|
|
|
|
|
|
if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) {
|
|
|
|
/* Get second scale factor */
|
2007-04-24 17:31:39 +00:00
|
|
|
scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum);
|
2007-02-27 06:30:40 +00:00
|
|
|
s->scale_factor[j][k][1] = scale_table[scale_sum];
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Joint subband scale factor codebook select */
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
/* Transmitted only if joint subband coding enabled */
|
|
|
|
if (s->joint_intensity[j] > 0)
|
|
|
|
s->joint_huff[j] = get_bits(&s->gb, 3);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Scale factors for joint subband coding */
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
int source_channel;
|
|
|
|
|
|
|
|
/* Transmitted only if joint subband coding enabled */
|
|
|
|
if (s->joint_intensity[j] > 0) {
|
|
|
|
int scale = 0;
|
|
|
|
source_channel = s->joint_intensity[j] - 1;
|
|
|
|
|
|
|
|
/* When huffman coded, only the difference is encoded
|
|
|
|
* (is this valid as well for joint scales ???) */
|
|
|
|
|
|
|
|
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) {
|
2007-04-24 17:31:39 +00:00
|
|
|
scale = get_scale(&s->gb, s->joint_huff[j], 0);
|
2007-02-27 06:30:40 +00:00
|
|
|
scale += 64; /* bias */
|
|
|
|
s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!s->debug_flag & 0x02) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG,
|
|
|
|
"Joint stereo coding not supported\n");
|
|
|
|
s->debug_flag |= 0x02;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Stereo downmix coefficients */
|
2007-05-02 20:33:32 +00:00
|
|
|
if (s->prim_channels > 2) {
|
|
|
|
if(s->downmix) {
|
2007-05-02 20:42:44 +00:00
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
s->downmix_coef[j][0] = get_bits(&s->gb, 7);
|
|
|
|
s->downmix_coef[j][1] = get_bits(&s->gb, 7);
|
|
|
|
}
|
2007-05-02 20:33:32 +00:00
|
|
|
} else {
|
|
|
|
int am = s->amode & DCA_CHANNEL_MASK;
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
s->downmix_coef[j][0] = dca_default_coeffs[am][j][0];
|
|
|
|
s->downmix_coef[j][1] = dca_default_coeffs[am][j][1];
|
|
|
|
}
|
|
|
|
}
|
2007-02-27 06:30:40 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
/* Dynamic range coefficient */
|
|
|
|
if (s->dynrange)
|
|
|
|
s->dynrange_coef = get_bits(&s->gb, 8);
|
|
|
|
|
|
|
|
/* Side information CRC check word */
|
|
|
|
if (s->crc_present) {
|
|
|
|
get_bits(&s->gb, 16);
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Primary audio data arrays
|
|
|
|
*/
|
|
|
|
|
|
|
|
/* VQ encoded high frequency subbands */
|
|
|
|
for (j = 0; j < s->prim_channels; j++)
|
|
|
|
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
|
|
|
|
/* 1 vector -> 32 samples */
|
|
|
|
s->high_freq_vq[j][k] = get_bits(&s->gb, 10);
|
|
|
|
|
|
|
|
/* Low frequency effect data */
|
|
|
|
if (s->lfe) {
|
|
|
|
/* LFE samples */
|
|
|
|
int lfe_samples = 2 * s->lfe * s->subsubframes;
|
|
|
|
float lfe_scale;
|
|
|
|
|
|
|
|
for (j = lfe_samples; j < lfe_samples * 2; j++) {
|
|
|
|
/* Signed 8 bits int */
|
|
|
|
s->lfe_data[j] = get_sbits(&s->gb, 8);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Scale factor index */
|
|
|
|
s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)];
|
|
|
|
|
|
|
|
/* Quantization step size * scale factor */
|
|
|
|
lfe_scale = 0.035 * s->lfe_scale_factor;
|
|
|
|
|
|
|
|
for (j = lfe_samples; j < lfe_samples * 2; j++)
|
|
|
|
s->lfe_data[j] *= lfe_scale;
|
|
|
|
}
|
|
|
|
|
|
|
|
#ifdef TRACE
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n",
|
|
|
|
s->partial_samples);
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:");
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG,
|
|
|
|
"prediction coefs: %f, %f, %f, %f\n",
|
|
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192,
|
|
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192,
|
|
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192,
|
|
|
|
(float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192);
|
|
|
|
}
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: ");
|
|
|
|
for (k = 0; k < s->vq_start_subband[j]; k++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:");
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:");
|
|
|
|
for (k = 0; k < s->subband_activity[j]; k++) {
|
|
|
|
if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]);
|
|
|
|
if (k < s->vq_start_subband[j] && s->transition_mode[j][k])
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]);
|
|
|
|
}
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
if (s->joint_intensity[j] > 0) {
|
2007-05-24 04:02:47 +00:00
|
|
|
int source_channel = s->joint_intensity[j] - 1;
|
2007-02-27 06:30:40 +00:00
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n");
|
|
|
|
for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
}
|
|
|
|
if (s->prim_channels > 2 && s->downmix) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n");
|
|
|
|
for (j = 0; j < s->prim_channels; j++) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]);
|
|
|
|
}
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
for (j = 0; j < s->prim_channels; j++)
|
|
|
|
for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]);
|
|
|
|
if(s->lfe){
|
2007-05-24 04:02:47 +00:00
|
|
|
int lfe_samples = 2 * s->lfe * s->subsubframes;
|
2007-02-27 06:30:40 +00:00
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n");
|
|
|
|
for (j = lfe_samples; j < lfe_samples * 2; j++)
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]);
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "\n");
|
|
|
|
}
|
|
|
|
#endif
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static void qmf_32_subbands(DCAContext * s, int chans,
|
|
|
|
float samples_in[32][8], float *samples_out,
|
|
|
|
float scale, float bias)
|
|
|
|
{
|
2007-12-02 21:12:43 +00:00
|
|
|
const float *prCoeff;
|
2007-02-27 06:30:40 +00:00
|
|
|
int i, j, k;
|
|
|
|
float praXin[33], *raXin = &praXin[1];
|
|
|
|
|
|
|
|
float *subband_fir_hist = s->subband_fir_hist[chans];
|
|
|
|
float *subband_fir_hist2 = s->subband_fir_noidea[chans];
|
|
|
|
|
|
|
|
int chindex = 0, subindex;
|
|
|
|
|
|
|
|
praXin[0] = 0.0;
|
|
|
|
|
|
|
|
/* Select filter */
|
|
|
|
if (!s->multirate_inter) /* Non-perfect reconstruction */
|
2007-12-02 21:12:43 +00:00
|
|
|
prCoeff = fir_32bands_nonperfect;
|
2007-02-27 06:30:40 +00:00
|
|
|
else /* Perfect reconstruction */
|
2007-12-02 21:12:43 +00:00
|
|
|
prCoeff = fir_32bands_perfect;
|
2007-02-27 06:30:40 +00:00
|
|
|
|
|
|
|
/* Reconstructed channel sample index */
|
|
|
|
for (subindex = 0; subindex < 8; subindex++) {
|
|
|
|
float t1, t2, sum[16], diff[16];
|
|
|
|
|
|
|
|
/* Load in one sample from each subband and clear inactive subbands */
|
|
|
|
for (i = 0; i < s->subband_activity[chans]; i++)
|
|
|
|
raXin[i] = samples_in[i][subindex];
|
|
|
|
for (; i < 32; i++)
|
|
|
|
raXin[i] = 0.0;
|
|
|
|
|
|
|
|
/* Multiply by cosine modulation coefficients and
|
|
|
|
* create temporary arrays SUM and DIFF */
|
|
|
|
for (j = 0, k = 0; k < 16; k++) {
|
|
|
|
t1 = 0.0;
|
|
|
|
t2 = 0.0;
|
|
|
|
for (i = 0; i < 16; i++, j++){
|
|
|
|
t1 += (raXin[2 * i] + raXin[2 * i + 1]) * cos_mod[j];
|
|
|
|
t2 += (raXin[2 * i] + raXin[2 * i - 1]) * cos_mod[j + 256];
|
|
|
|
}
|
|
|
|
sum[k] = t1 + t2;
|
|
|
|
diff[k] = t1 - t2;
|
|
|
|
}
|
|
|
|
|
|
|
|
j = 512;
|
|
|
|
/* Store history */
|
|
|
|
for (k = 0; k < 16; k++)
|
|
|
|
subband_fir_hist[k] = cos_mod[j++] * sum[k];
|
|
|
|
for (k = 0; k < 16; k++)
|
|
|
|
subband_fir_hist[32-k-1] = cos_mod[j++] * diff[k];
|
|
|
|
|
|
|
|
/* Multiply by filter coefficients */
|
|
|
|
for (k = 31, i = 0; i < 32; i++, k--)
|
|
|
|
for (j = 0; j < 512; j += 64){
|
|
|
|
subband_fir_hist2[i] += prCoeff[i+j] * ( subband_fir_hist[i+j] - subband_fir_hist[j+k]);
|
|
|
|
subband_fir_hist2[i+32] += prCoeff[i+j+32]*(-subband_fir_hist[i+j] - subband_fir_hist[j+k]);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Create 32 PCM output samples */
|
|
|
|
for (i = 0; i < 32; i++)
|
|
|
|
samples_out[chindex++] = subband_fir_hist2[i] * scale + bias;
|
|
|
|
|
|
|
|
/* Update working arrays */
|
|
|
|
memmove(&subband_fir_hist[32], &subband_fir_hist[0], (512 - 32) * sizeof(float));
|
|
|
|
memmove(&subband_fir_hist2[0], &subband_fir_hist2[32], 32 * sizeof(float));
|
|
|
|
memset(&subband_fir_hist2[32], 0, 32 * sizeof(float));
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static void lfe_interpolation_fir(int decimation_select,
|
|
|
|
int num_deci_sample, float *samples_in,
|
|
|
|
float *samples_out, float scale,
|
|
|
|
float bias)
|
|
|
|
{
|
|
|
|
/* samples_in: An array holding decimated samples.
|
|
|
|
* Samples in current subframe starts from samples_in[0],
|
|
|
|
* while samples_in[-1], samples_in[-2], ..., stores samples
|
|
|
|
* from last subframe as history.
|
|
|
|
*
|
|
|
|
* samples_out: An array holding interpolated samples
|
|
|
|
*/
|
|
|
|
|
|
|
|
int decifactor, k, j;
|
|
|
|
const float *prCoeff;
|
|
|
|
|
|
|
|
int interp_index = 0; /* Index to the interpolated samples */
|
|
|
|
int deciindex;
|
|
|
|
|
|
|
|
/* Select decimation filter */
|
|
|
|
if (decimation_select == 1) {
|
|
|
|
decifactor = 128;
|
|
|
|
prCoeff = lfe_fir_128;
|
|
|
|
} else {
|
|
|
|
decifactor = 64;
|
|
|
|
prCoeff = lfe_fir_64;
|
|
|
|
}
|
|
|
|
/* Interpolation */
|
|
|
|
for (deciindex = 0; deciindex < num_deci_sample; deciindex++) {
|
|
|
|
/* One decimated sample generates decifactor interpolated ones */
|
|
|
|
for (k = 0; k < decifactor; k++) {
|
|
|
|
float rTmp = 0.0;
|
|
|
|
//FIXME the coeffs are symetric, fix that
|
|
|
|
for (j = 0; j < 512 / decifactor; j++)
|
|
|
|
rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
|
|
|
|
samples_out[interp_index++] = rTmp / scale + bias;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* downmixing routines */
|
2007-05-02 20:33:32 +00:00
|
|
|
#define MIX_REAR1(samples, si1, rs, coef) \
|
|
|
|
samples[i] += samples[si1] * coef[rs][0]; \
|
|
|
|
samples[i+256] += samples[si1] * coef[rs][1];
|
2007-02-27 06:30:40 +00:00
|
|
|
|
2007-05-02 20:33:32 +00:00
|
|
|
#define MIX_REAR2(samples, si1, si2, rs, coef) \
|
|
|
|
samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \
|
|
|
|
samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1];
|
2007-02-27 06:30:40 +00:00
|
|
|
|
2007-05-02 20:33:32 +00:00
|
|
|
#define MIX_FRONT3(samples, coef) \
|
2007-02-27 06:30:40 +00:00
|
|
|
t = samples[i]; \
|
2007-05-02 20:33:32 +00:00
|
|
|
samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \
|
|
|
|
samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1];
|
2007-02-27 06:30:40 +00:00
|
|
|
|
|
|
|
#define DOWNMIX_TO_STEREO(op1, op2) \
|
|
|
|
for(i = 0; i < 256; i++){ \
|
|
|
|
op1 \
|
|
|
|
op2 \
|
|
|
|
}
|
|
|
|
|
2007-05-02 20:33:32 +00:00
|
|
|
static void dca_downmix(float *samples, int srcfmt,
|
|
|
|
int downmix_coef[DCA_PRIM_CHANNELS_MAX][2])
|
2007-02-27 06:30:40 +00:00
|
|
|
{
|
|
|
|
int i;
|
|
|
|
float t;
|
2007-05-02 20:33:32 +00:00
|
|
|
float coef[DCA_PRIM_CHANNELS_MAX][2];
|
|
|
|
|
|
|
|
for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) {
|
|
|
|
coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]];
|
|
|
|
coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]];
|
|
|
|
}
|
2007-02-27 06:30:40 +00:00
|
|
|
|
|
|
|
switch (srcfmt) {
|
|
|
|
case DCA_MONO:
|
|
|
|
case DCA_CHANNEL:
|
|
|
|
case DCA_STEREO_TOTAL:
|
|
|
|
case DCA_STEREO_SUMDIFF:
|
|
|
|
case DCA_4F2R:
|
|
|
|
av_log(NULL, 0, "Not implemented!\n");
|
|
|
|
break;
|
|
|
|
case DCA_STEREO:
|
|
|
|
break;
|
|
|
|
case DCA_3F:
|
2007-05-02 20:33:32 +00:00
|
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),);
|
2007-02-27 06:30:40 +00:00
|
|
|
break;
|
|
|
|
case DCA_2F1R:
|
2007-05-02 20:33:32 +00:00
|
|
|
DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),);
|
2007-02-27 06:30:40 +00:00
|
|
|
break;
|
|
|
|
case DCA_3F1R:
|
2007-05-02 20:33:32 +00:00
|
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
|
|
|
|
MIX_REAR1(samples, i + 768, 3, coef));
|
2007-02-27 06:30:40 +00:00
|
|
|
break;
|
|
|
|
case DCA_2F2R:
|
2007-05-02 20:33:32 +00:00
|
|
|
DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),);
|
2007-02-27 06:30:40 +00:00
|
|
|
break;
|
|
|
|
case DCA_3F2R:
|
2007-05-02 20:33:32 +00:00
|
|
|
DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),
|
|
|
|
MIX_REAR2(samples, i + 768, i + 1024, 3, coef));
|
2007-02-27 06:30:40 +00:00
|
|
|
break;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
/* Very compact version of the block code decoder that does not use table
|
|
|
|
* look-up but is slightly slower */
|
|
|
|
static int decode_blockcode(int code, int levels, int *values)
|
|
|
|
{
|
|
|
|
int i;
|
|
|
|
int offset = (levels - 1) >> 1;
|
|
|
|
|
|
|
|
for (i = 0; i < 4; i++) {
|
|
|
|
values[i] = (code % levels) - offset;
|
|
|
|
code /= levels;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (code == 0)
|
|
|
|
return 0;
|
|
|
|
else {
|
|
|
|
av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n");
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
|
|
|
|
static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
|
|
|
|
|
|
|
|
static int dca_subsubframe(DCAContext * s)
|
|
|
|
{
|
|
|
|
int k, l;
|
|
|
|
int subsubframe = s->current_subsubframe;
|
|
|
|
|
2008-02-01 01:40:46 +00:00
|
|
|
const float *quant_step_table;
|
2007-02-27 06:30:40 +00:00
|
|
|
|
|
|
|
/* FIXME */
|
|
|
|
float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8];
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Audio data
|
|
|
|
*/
|
|
|
|
|
|
|
|
/* Select quantization step size table */
|
|
|
|
if (s->bit_rate == 0x1f)
|
2008-02-01 01:40:46 +00:00
|
|
|
quant_step_table = lossless_quant_d;
|
2007-02-27 06:30:40 +00:00
|
|
|
else
|
2008-02-01 01:40:46 +00:00
|
|
|
quant_step_table = lossy_quant_d;
|
2007-02-27 06:30:40 +00:00
|
|
|
|
|
|
|
for (k = 0; k < s->prim_channels; k++) {
|
|
|
|
for (l = 0; l < s->vq_start_subband[k]; l++) {
|
|
|
|
int m;
|
|
|
|
|
|
|
|
/* Select the mid-tread linear quantizer */
|
|
|
|
int abits = s->bitalloc[k][l];
|
|
|
|
|
|
|
|
float quant_step_size = quant_step_table[abits];
|
|
|
|
float rscale;
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Determine quantization index code book and its type
|
|
|
|
*/
|
|
|
|
|
|
|
|
/* Select quantization index code book */
|
|
|
|
int sel = s->quant_index_huffman[k][abits];
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Extract bits from the bit stream
|
|
|
|
*/
|
|
|
|
if(!abits){
|
|
|
|
memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0]));
|
|
|
|
}else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){
|
|
|
|
if(abits <= 7){
|
|
|
|
/* Block code */
|
|
|
|
int block_code1, block_code2, size, levels;
|
|
|
|
int block[8];
|
|
|
|
|
|
|
|
size = abits_sizes[abits-1];
|
|
|
|
levels = abits_levels[abits-1];
|
|
|
|
|
|
|
|
block_code1 = get_bits(&s->gb, size);
|
|
|
|
/* FIXME Should test return value */
|
|
|
|
decode_blockcode(block_code1, levels, block);
|
|
|
|
block_code2 = get_bits(&s->gb, size);
|
|
|
|
decode_blockcode(block_code2, levels, &block[4]);
|
|
|
|
for (m = 0; m < 8; m++)
|
|
|
|
subband_samples[k][l][m] = block[m];
|
|
|
|
}else{
|
|
|
|
/* no coding */
|
|
|
|
for (m = 0; m < 8; m++)
|
|
|
|
subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3);
|
|
|
|
}
|
|
|
|
}else{
|
|
|
|
/* Huffman coded */
|
|
|
|
for (m = 0; m < 8; m++)
|
|
|
|
subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel);
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Deal with transients */
|
|
|
|
if (s->transition_mode[k][l] &&
|
|
|
|
subsubframe >= s->transition_mode[k][l])
|
|
|
|
rscale = quant_step_size * s->scale_factor[k][l][1];
|
|
|
|
else
|
|
|
|
rscale = quant_step_size * s->scale_factor[k][l][0];
|
|
|
|
|
|
|
|
rscale *= s->scalefactor_adj[k][sel];
|
|
|
|
|
|
|
|
for (m = 0; m < 8; m++)
|
|
|
|
subband_samples[k][l][m] *= rscale;
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Inverse ADPCM if in prediction mode
|
|
|
|
*/
|
|
|
|
if (s->prediction_mode[k][l]) {
|
|
|
|
int n;
|
|
|
|
for (m = 0; m < 8; m++) {
|
|
|
|
for (n = 1; n <= 4; n++)
|
|
|
|
if (m >= n)
|
|
|
|
subband_samples[k][l][m] +=
|
|
|
|
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
|
|
|
|
subband_samples[k][l][m - n] / 8192);
|
|
|
|
else if (s->predictor_history)
|
|
|
|
subband_samples[k][l][m] +=
|
|
|
|
(adpcm_vb[s->prediction_vq[k][l]][n - 1] *
|
|
|
|
s->subband_samples_hist[k][l][m - n +
|
|
|
|
4] / 8192);
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Decode VQ encoded high frequencies
|
|
|
|
*/
|
|
|
|
for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) {
|
|
|
|
/* 1 vector -> 32 samples but we only need the 8 samples
|
|
|
|
* for this subsubframe. */
|
|
|
|
int m;
|
|
|
|
|
|
|
|
if (!s->debug_flag & 0x01) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
|
|
|
|
s->debug_flag |= 0x01;
|
|
|
|
}
|
|
|
|
|
|
|
|
for (m = 0; m < 8; m++) {
|
|
|
|
subband_samples[k][l][m] =
|
|
|
|
high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
|
|
|
|
m]
|
|
|
|
* (float) s->scale_factor[k][l][0] / 16.0;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Check for DSYNC after subsubframe */
|
|
|
|
if (s->aspf || subsubframe == s->subsubframes - 1) {
|
|
|
|
if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */
|
|
|
|
#ifdef TRACE
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n");
|
|
|
|
#endif
|
|
|
|
} else {
|
|
|
|
av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Backup predictor history for adpcm */
|
|
|
|
for (k = 0; k < s->prim_channels; k++)
|
|
|
|
for (l = 0; l < s->vq_start_subband[k]; l++)
|
|
|
|
memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4],
|
|
|
|
4 * sizeof(subband_samples[0][0][0]));
|
|
|
|
|
|
|
|
/* 32 subbands QMF */
|
|
|
|
for (k = 0; k < s->prim_channels; k++) {
|
|
|
|
/* static float pcm_to_double[8] =
|
|
|
|
{32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
|
|
|
|
qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k],
|
|
|
|
2.0 / 3 /*pcm_to_double[s->source_pcm_res] */ ,
|
|
|
|
0 /*s->bias */ );
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Down mixing */
|
|
|
|
|
|
|
|
if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) {
|
2007-05-02 20:33:32 +00:00
|
|
|
dca_downmix(s->samples, s->amode, s->downmix_coef);
|
2007-02-27 06:30:40 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
/* Generate LFE samples for this subsubframe FIXME!!! */
|
|
|
|
if (s->output & DCA_LFE) {
|
|
|
|
int lfe_samples = 2 * s->lfe * s->subsubframes;
|
|
|
|
int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK];
|
|
|
|
|
|
|
|
lfe_interpolation_fir(s->lfe, 2 * s->lfe,
|
|
|
|
s->lfe_data + lfe_samples +
|
|
|
|
2 * s->lfe * subsubframe,
|
|
|
|
&s->samples[256 * i_channels],
|
2008-01-04 20:02:10 +00:00
|
|
|
256.0, 0 /* s->bias */);
|
2007-02-27 06:30:40 +00:00
|
|
|
/* Outputs 20bits pcm samples */
|
|
|
|
}
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
static int dca_subframe_footer(DCAContext * s)
|
|
|
|
{
|
|
|
|
int aux_data_count = 0, i;
|
|
|
|
int lfe_samples;
|
|
|
|
|
|
|
|
/*
|
|
|
|
* Unpack optional information
|
|
|
|
*/
|
|
|
|
|
|
|
|
if (s->timestamp)
|
|
|
|
get_bits(&s->gb, 32);
|
|
|
|
|
|
|
|
if (s->aux_data)
|
|
|
|
aux_data_count = get_bits(&s->gb, 6);
|
|
|
|
|
|
|
|
for (i = 0; i < aux_data_count; i++)
|
|
|
|
get_bits(&s->gb, 8);
|
|
|
|
|
|
|
|
if (s->crc_present && (s->downmix || s->dynrange))
|
|
|
|
get_bits(&s->gb, 16);
|
|
|
|
|
|
|
|
lfe_samples = 2 * s->lfe * s->subsubframes;
|
|
|
|
for (i = 0; i < lfe_samples; i++) {
|
|
|
|
s->lfe_data[i] = s->lfe_data[i + lfe_samples];
|
|
|
|
}
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
|
|
* Decode a dca frame block
|
|
|
|
*
|
|
|
|
* @param s pointer to the DCAContext
|
|
|
|
*/
|
|
|
|
|
|
|
|
static int dca_decode_block(DCAContext * s)
|
|
|
|
{
|
|
|
|
|
|
|
|
/* Sanity check */
|
|
|
|
if (s->current_subframe >= s->subframes) {
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
|
|
|
|
s->current_subframe, s->subframes);
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (!s->current_subsubframe) {
|
|
|
|
#ifdef TRACE
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n");
|
|
|
|
#endif
|
|
|
|
/* Read subframe header */
|
|
|
|
if (dca_subframe_header(s))
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Read subsubframe */
|
|
|
|
#ifdef TRACE
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n");
|
|
|
|
#endif
|
|
|
|
if (dca_subsubframe(s))
|
|
|
|
return -1;
|
|
|
|
|
|
|
|
/* Update state */
|
|
|
|
s->current_subsubframe++;
|
|
|
|
if (s->current_subsubframe >= s->subsubframes) {
|
|
|
|
s->current_subsubframe = 0;
|
|
|
|
s->current_subframe++;
|
|
|
|
}
|
|
|
|
if (s->current_subframe >= s->subframes) {
|
|
|
|
#ifdef TRACE
|
|
|
|
av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n");
|
|
|
|
#endif
|
|
|
|
/* Read subframe footer */
|
|
|
|
if (dca_subframe_footer(s))
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
|
|
* Convert bitstream to one representation based on sync marker
|
|
|
|
*/
|
2008-02-01 01:40:46 +00:00
|
|
|
static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst,
|
2007-02-27 06:30:40 +00:00
|
|
|
int max_size)
|
|
|
|
{
|
|
|
|
uint32_t mrk;
|
|
|
|
int i, tmp;
|
2008-02-01 01:40:46 +00:00
|
|
|
const uint16_t *ssrc = (const uint16_t *) src;
|
|
|
|
uint16_t *sdst = (uint16_t *) dst;
|
2007-02-27 06:30:40 +00:00
|
|
|
PutBitContext pb;
|
|
|
|
|
2007-05-18 10:06:07 +00:00
|
|
|
if((unsigned)src_size > (unsigned)max_size) {
|
|
|
|
av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n");
|
2007-04-27 12:41:55 +00:00
|
|
|
return -1;
|
2007-05-18 10:06:07 +00:00
|
|
|
}
|
2007-04-27 12:41:55 +00:00
|
|
|
|
2007-02-27 06:30:40 +00:00
|
|
|
mrk = AV_RB32(src);
|
|
|
|
switch (mrk) {
|
|
|
|
case DCA_MARKER_RAW_BE:
|
2008-08-23 13:35:12 +00:00
|
|
|
memcpy(dst, src, src_size);
|
|
|
|
return src_size;
|
2007-02-27 06:30:40 +00:00
|
|
|
case DCA_MARKER_RAW_LE:
|
2008-08-23 13:35:12 +00:00
|
|
|
for (i = 0; i < (src_size + 1) >> 1; i++)
|
2007-02-27 06:30:40 +00:00
|
|
|
*sdst++ = bswap_16(*ssrc++);
|
2008-08-23 13:35:12 +00:00
|
|
|
return src_size;
|
2007-02-27 06:30:40 +00:00
|
|
|
case DCA_MARKER_14B_BE:
|
|
|
|
case DCA_MARKER_14B_LE:
|
|
|
|
init_put_bits(&pb, dst, max_size);
|
|
|
|
for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) {
|
|
|
|
tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF;
|
|
|
|
put_bits(&pb, 14, tmp);
|
|
|
|
}
|
|
|
|
flush_put_bits(&pb);
|
|
|
|
return (put_bits_count(&pb) + 7) >> 3;
|
|
|
|
default:
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
|
|
* Main frame decoding function
|
|
|
|
* FIXME add arguments
|
|
|
|
*/
|
|
|
|
static int dca_decode_frame(AVCodecContext * avctx,
|
|
|
|
void *data, int *data_size,
|
2008-02-01 01:40:46 +00:00
|
|
|
const uint8_t * buf, int buf_size)
|
2007-02-27 06:30:40 +00:00
|
|
|
{
|
|
|
|
|
|
|
|
int i, j, k;
|
|
|
|
int16_t *samples = data;
|
|
|
|
DCAContext *s = avctx->priv_data;
|
|
|
|
int channels;
|
|
|
|
|
|
|
|
|
|
|
|
s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE);
|
|
|
|
if (s->dca_buffer_size == -1) {
|
2007-05-18 10:06:07 +00:00
|
|
|
av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n");
|
2007-02-27 06:30:40 +00:00
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
|
|
|
|
if (dca_parse_frame_header(s) < 0) {
|
|
|
|
//seems like the frame is corrupt, try with the next one
|
2007-09-06 14:55:13 +00:00
|
|
|
*data_size=0;
|
2007-02-27 06:30:40 +00:00
|
|
|
return buf_size;
|
|
|
|
}
|
|
|
|
//set AVCodec values with parsed data
|
|
|
|
avctx->sample_rate = s->sample_rate;
|
|
|
|
avctx->bit_rate = s->bit_rate;
|
|
|
|
|
2007-05-02 20:31:32 +00:00
|
|
|
channels = s->prim_channels + !!s->lfe;
|
2007-12-15 07:01:33 +00:00
|
|
|
if(avctx->request_channels == 2 && s->prim_channels > 2) {
|
|
|
|
channels = 2;
|
2007-05-02 20:31:32 +00:00
|
|
|
s->output = DCA_STEREO;
|
|
|
|
}
|
|
|
|
|
2008-04-10 06:15:17 +00:00
|
|
|
/* There is nothing that prevents a dts frame to change channel configuration
|
|
|
|
but FFmpeg doesn't support that so only set the channels if it is previously
|
|
|
|
unset. Ideally during the first probe for channels the crc should be checked
|
|
|
|
and only set avctx->channels when the crc is ok. Right now the decoder could
|
|
|
|
set the channels based on a broken first frame.*/
|
|
|
|
if (!avctx->channels)
|
|
|
|
avctx->channels = channels;
|
|
|
|
|
2007-02-27 06:30:40 +00:00
|
|
|
if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
|
|
|
|
return -1;
|
|
|
|
*data_size = 0;
|
|
|
|
for (i = 0; i < (s->sample_blocks / 8); i++) {
|
|
|
|
dca_decode_block(s);
|
|
|
|
s->dsp.float_to_int16(s->tsamples, s->samples, 256 * channels);
|
|
|
|
/* interleave samples */
|
|
|
|
for (j = 0; j < 256; j++) {
|
|
|
|
for (k = 0; k < channels; k++)
|
|
|
|
samples[k] = s->tsamples[j + k * 256];
|
|
|
|
samples += channels;
|
|
|
|
}
|
|
|
|
*data_size += 256 * sizeof(int16_t) * channels;
|
|
|
|
}
|
|
|
|
|
|
|
|
return buf_size;
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
* Build the cosine modulation tables for the QMF
|
|
|
|
*
|
|
|
|
* @param s pointer to the DCAContext
|
|
|
|
*/
|
|
|
|
|
2008-03-21 03:11:20 +00:00
|
|
|
static av_cold void pre_calc_cosmod(DCAContext * s)
|
2007-02-27 06:30:40 +00:00
|
|
|
{
|
|
|
|
int i, j, k;
|
2008-02-13 09:26:10 +00:00
|
|
|
static int cosmod_initialized = 0;
|
2007-02-27 06:30:40 +00:00
|
|
|
|
2008-02-13 09:26:10 +00:00
|
|
|
if(cosmod_initialized) return;
|
2007-02-27 06:30:40 +00:00
|
|
|
for (j = 0, k = 0; k < 16; k++)
|
|
|
|
for (i = 0; i < 16; i++)
|
|
|
|
cos_mod[j++] = cos((2 * i + 1) * (2 * k + 1) * M_PI / 64);
|
|
|
|
|
|
|
|
for (k = 0; k < 16; k++)
|
|
|
|
for (i = 0; i < 16; i++)
|
|
|
|
cos_mod[j++] = cos((i) * (2 * k + 1) * M_PI / 32);
|
|
|
|
|
|
|
|
for (k = 0; k < 16; k++)
|
|
|
|
cos_mod[j++] = 0.25 / (2 * cos((2 * k + 1) * M_PI / 128));
|
|
|
|
|
|
|
|
for (k = 0; k < 16; k++)
|
|
|
|
cos_mod[j++] = -0.25 / (2.0 * sin((2 * k + 1) * M_PI / 128));
|
|
|
|
|
2008-02-13 09:26:10 +00:00
|
|
|
cosmod_initialized = 1;
|
2007-02-27 06:30:40 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
* DCA initialization
|
|
|
|
*
|
|
|
|
* @param avctx pointer to the AVCodecContext
|
|
|
|
*/
|
|
|
|
|
2008-03-21 03:11:20 +00:00
|
|
|
static av_cold int dca_decode_init(AVCodecContext * avctx)
|
2007-02-27 06:30:40 +00:00
|
|
|
{
|
|
|
|
DCAContext *s = avctx->priv_data;
|
|
|
|
|
|
|
|
s->avctx = avctx;
|
|
|
|
dca_init_vlcs();
|
|
|
|
pre_calc_cosmod(s);
|
|
|
|
|
|
|
|
dsputil_init(&s->dsp, avctx);
|
2008-01-05 21:23:57 +00:00
|
|
|
|
|
|
|
/* allow downmixing to stereo */
|
|
|
|
if (avctx->channels > 0 && avctx->request_channels < avctx->channels &&
|
|
|
|
avctx->request_channels == 2) {
|
|
|
|
avctx->channels = avctx->request_channels;
|
|
|
|
}
|
|
|
|
|
2008-07-31 10:47:31 +00:00
|
|
|
avctx->sample_fmt = SAMPLE_FMT_S16;
|
2007-02-27 06:30:40 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
AVCodec dca_decoder = {
|
|
|
|
.name = "dca",
|
|
|
|
.type = CODEC_TYPE_AUDIO,
|
|
|
|
.id = CODEC_ID_DTS,
|
|
|
|
.priv_data_size = sizeof(DCAContext),
|
|
|
|
.init = dca_decode_init,
|
|
|
|
.decode = dca_decode_frame,
|
2008-06-12 21:50:13 +00:00
|
|
|
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
|
2007-02-27 06:30:40 +00:00
|
|
|
};
|