apps/codecs/aac.c

   1 /***************************************************************************
   2  *             __________               __   ___.
   3  *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
   4  *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
   5  *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
   6  *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
   7  *                     \/            \/     \/    \/            \/
   8  * $Id$
   9  *
  10  * Copyright (C) 2005 Dave Chapman
  11  *
  12  * This program is free software; you can redistribute it and/or
  13  * modify it under the terms of the GNU General Public License
  14  * as published by the Free Software Foundation; either version 2
  15  * of the License, or (at your option) any later version.
  16  *
  17  * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
  18  * KIND, either express or implied.
  19  *
  20  ****************************************************************************/
  21
  22 #include "codeclib.h"
  23 #include "libm4a/m4a.h"
  24 #include "libfaad/common.h"
  25 #include "libfaad/structs.h"
  26 #include "libfaad/decoder.h"
  27
  28 CODEC_HEADER
  29
  30 /* this is the codec entry point */
  31 enum codec_status codec_main(void)
  32 {
  33     /* Note that when dealing with QuickTime/MPEG4 files, terminology is
  34      * a bit confusing. Files with sound are split up in chunks, where
  35      * each chunk contains one or more samples. Each sample in turn
  36      * contains a number of "sound samples" (the kind you refer to with
  37      * the sampling frequency).
  38      */
  39     size_t n;
  40     static demux_res_t demux_res;
  41     stream_t input_stream;
  42     uint32_t sound_samples_done;
  43     uint32_t elapsed_time;
  44     uint32_t sample_duration;
  45     uint32_t sample_byte_size;
  46     int file_offset;
  47     int framelength;
  48     int lead_trim = 0;
  49     unsigned int i;
  50     unsigned char* buffer;
  51     static NeAACDecFrameInfo frame_info;
  52     NeAACDecHandle decoder;
  53     int err;
  54     uint32_t s = 0;
  55     unsigned char c = 0;
  56     void *ret;
  57
  58     /* Generic codec initialisation */
  59     ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  60     ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  61
  62 next_track:
  63     err = CODEC_OK;
  64
  65     if (codec_init()) {
  66         LOGF("FAAD: Codec init error\n");
  67         err = CODEC_ERROR;
  68         goto exit;
  69     }
  70
  71     while (!*ci->taginfo_ready && !ci->stop_codec)
  72         ci->sleep(1);
  73
  74     sound_samples_done = ci->id3->offset;
  75
  76     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  77     codec_set_replaygain(ci->id3);
  78
  79     stream_create(&input_stream,ci);
  80
  81     /* if qtmovie_read returns successfully, the stream is up to
  82      * the movie data, which can be used directly by the decoder */
  83     if (!qtmovie_read(&input_stream, &demux_res)) {
  84         LOGF("FAAD: File init error\n");
  85         err = CODEC_ERROR;
  86         goto done;
  87     }
  88
  89     /* initialise the sound converter */
  90     decoder = NeAACDecOpen();
  91
  92     if (!decoder) {
  93         LOGF("FAAD: Decode open error\n");
  94         err = CODEC_ERROR;
  95         goto done;
  96     }
  97
  98     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
  99     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
 100     NeAACDecSetConfiguration(decoder, conf);
 101
 102     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 103     if (err) {
 104         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 105         err = CODEC_ERROR;
 106         goto done;
 107     }
 108
 109     ci->id3->frequency = s;
 110
 111     i = 0;
 112
 113     if (sound_samples_done > 0) {
 114         if (alac_seek_raw(&demux_res, &input_stream, sound_samples_done,
 115                           &sound_samples_done, (int*) &i)) {
 116             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 117             ci->set_elapsed(elapsed_time);
 118         } else {
 119             sound_samples_done = 0;
 120         }
 121     }
 122
 123     if (i == 0)
 124     {
 125         lead_trim = ci->id3->lead_trim;
 126     }
 127
 128     /* The main decoding loop */
 129     while (i < demux_res.num_sample_byte_sizes) {
 130         ci->yield();
 131
 132         if (ci->stop_codec || ci->new_track) {
 133             break;
 134         }
 135
 136         /* Deal with any pending seek requests */
 137         if (ci->seek_time) {
 138             if (alac_seek(&demux_res, &input_stream,
 139                           ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
 140                           &sound_samples_done, (int*) &i)) {
 141                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 142                 ci->set_elapsed(elapsed_time);
 143
 144                 if (i == 0)
 145                 {
 146                     lead_trim = ci->id3->lead_trim;
 147                 }
 148             }
 149             ci->seek_complete();
 150         }
 151
 152         /* Lookup the length (in samples and bytes) of block i */
 153         if (!get_sample_info(&demux_res, i, &sample_duration,
 154                              &sample_byte_size)) {
 155             LOGF("AAC: get_sample_info error\n");
 156             err = CODEC_ERROR;
 157             goto done;
 158         }
 159
 160         /* There can be gaps between chunks, so skip ahead if needed. It
 161          * doesn't seem to happen much, but it probably means that a
 162          * "proper" file can have chunks out of order. Why one would want
 163          * that an good question (but files with gaps do exist, so who
 164          * knows?), so we don't support that - for now, at least.
 165          */
 166         file_offset = get_sample_offset(&demux_res, i);
 167
 168         if (file_offset > ci->curpos)
 169         {
 170             ci->advance_buffer(file_offset - ci->curpos);
 171         }
 172         else if (file_offset == 0)
 173         {
 174             LOGF("AAC: get_sample_offset error\n");
 175             err = CODEC_ERROR;
 176             goto done;
 177         }
 178
 179         /* Request the required number of bytes from the input buffer */
 180         buffer=ci->request_buffer(&n,sample_byte_size);
 181
 182         /* Decode one block - returned samples will be host-endian */
 183         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 184
 185         /* NeAACDecDecode may sometimes return NULL without setting error. */
 186         if (ret == NULL || frame_info.error > 0) {
 187             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 188             err = CODEC_ERROR;
 189             goto done;
 190         }
 191
 192         /* Advance codec buffer */
 193         ci->advance_buffer(n);
 194
 195         /* Output the audio */
 196         ci->yield();
 197
 198         framelength = (frame_info.samples >> 1) - lead_trim;
 199
 200         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 201         {
 202             /* Currently limited to at most one frame of tail_trim.
 203              * Seems to be enough.
 204              */
 205             if (ci->id3->tail_trim == 0
 206                 && sample_duration < (frame_info.samples >> 1))
 207             {
 208                 /* Subtract lead_trim just in case we decode a file with
 209                  * only one audio frame with actual data.
 210                  */
 211                 framelength = sample_duration - lead_trim;
 212             }
 213             else
 214             {
 215                 framelength -= ci->id3->tail_trim;
 216             }
 217         }
 218
 219         if (framelength > 0)
 220         {
 221             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 222                               &decoder->time_out[1][lead_trim],
 223                               framelength);
 224         }
 225
 226         if (lead_trim > 0)
 227         {
 228             /* frame_info.samples can be 0 for the first frame */
 229             lead_trim -= (i > 0 || frame_info.samples)
 230                 ? (frame_info.samples >> 1) : sample_duration;
 231
 232             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 233             {
 234                 lead_trim = 0;
 235             }
 236         }
 237
 238         /* Update the elapsed-time indicator */
 239         sound_samples_done += sample_duration;
 240         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 241         ci->set_elapsed(elapsed_time);
 242
 243         /* Keep track of current position - for resuming */
 244         ci->set_offset(elapsed_time);
 245
 246         i++;
 247     }
 248
 249     err = CODEC_OK;
 250
 251 done:
 252     LOGF("AAC: Decoded %lu samples\n", sound_samples_done);
 253
 254     if (ci->request_next_track())
 255         goto next_track;
 256
 257 exit:
 258     return err;
 259 }