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ffmpeg-4.2.2/libavcodec/mpc.c 3.13 KB
aac5773f   hucm   功能基本完成,接口待打磨
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  /*
   * Musepack decoder core
   * Copyright (c) 2006 Konstantin Shishkov
   *
   * 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
   */
  
  /**
   * @file
   * Musepack decoder core
   * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
   * divided into 32 subbands.
   */
  
  #include "libavutil/attributes.h"
  #include "avcodec.h"
  #include "mpegaudiodsp.h"
  #include "mpegaudio.h"
  
  #include "mpc.h"
  #include "mpcdata.h"
  
  av_cold void ff_mpc_init(void)
  {
      ff_mpa_synth_init_fixed(ff_mpa_synth_window_fixed);
  }
  
  /**
   * Process decoded Musepack data and produce PCM
   */
  static void mpc_synth(MPCContext *c, int16_t **out, int channels)
  {
      int dither_state = 0;
      int i, ch;
  
      for(ch = 0;  ch < channels; ch++){
          for(i = 0; i < SAMPLES_PER_BAND; i++) {
              ff_mpa_synth_filter_fixed(&c->mpadsp,
                                  c->synth_buf[ch], &(c->synth_buf_offset[ch]),
                                  ff_mpa_synth_window_fixed, &dither_state,
                                  out[ch] + 32 * i, 1,
                                  c->sb_samples[ch][i]);
          }
      }
  }
  
  void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, int16_t **out,
                                   int channels)
  {
      int i, j, ch;
      Band *bands = c->bands;
      int off;
      float mul;
  
      /* dequantize */
      memset(c->sb_samples, 0, sizeof(c->sb_samples));
      off = 0;
      for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
          for(ch = 0; ch < 2; ch++){
              if(bands[i].res[ch]){
                  j = 0;
                  mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0] & 0xFF];
                  for(; j < 12; j++)
                      c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
                  mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1] & 0xFF];
                  for(; j < 24; j++)
                      c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
                  mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2] & 0xFF];
                  for(; j < 36; j++)
                      c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
              }
          }
          if(bands[i].msf){
              int t1, t2;
              for(j = 0; j < SAMPLES_PER_BAND; j++){
                  t1 = c->sb_samples[0][j][i];
                  t2 = c->sb_samples[1][j][i];
                  c->sb_samples[0][j][i] = t1 + t2;
                  c->sb_samples[1][j][i] = t1 - t2;
              }
          }
      }
  
      mpc_synth(c, out, channels);
  }