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
  57     /* Generic codec initialisation */
  58     ci->configure(CODEC_SET_FILEBUF_WATERMARK, 1024*512);
  59
  60     ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  61     ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  62
  63 next_track:
  64     err = CODEC_OK;
  65
  66     if (codec_init()) {
  67         LOGF("FAAD: Codec init error\n");
  68         err = CODEC_ERROR;
  69         goto exit;
  70     }
  71
  72     while (!*ci->taginfo_ready && !ci->stop_codec)
  73         ci->sleep(1);
  74
  75     sound_samples_done = ci->id3->offset;
  76
  77     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  78     codec_set_replaygain(ci->id3);
  79
  80     stream_create(&input_stream,ci);
  81
  82     /* if qtmovie_read returns successfully, the stream is up to
  83      * the movie data, which can be used directly by the decoder */
  84     if (!qtmovie_read(&input_stream, &demux_res)) {
  85         LOGF("FAAD: File init error\n");
  86         err = CODEC_ERROR;
  87         goto done;
  88     }
  89
  90     /* initialise the sound converter */
  91     decoder = NeAACDecOpen();
  92
  93     if (!decoder) {
  94         LOGF("FAAD: Decode open error\n");
  95         err = CODEC_ERROR;
  96         goto done;
  97     }
  98
  99     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
 100     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
 101     NeAACDecSetConfiguration(decoder, conf);
 102
 103     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 104     if (err) {
 105         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 106         err = CODEC_ERROR;
 107         goto done;
 108     }
 109
 110     ci->id3->frequency = s;
 111
 112     i = 0;
 113
 114     if (sound_samples_done > 0) {
 115         if (alac_seek_raw(&demux_res, &input_stream, sound_samples_done,
 116                           &sound_samples_done, (int*) &i)) {
 117             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 118             ci->set_elapsed(elapsed_time);
 119         } else {
 120             sound_samples_done = 0;
 121         }
 122     }
 123
 124     if (i == 0)
 125     {
 126         lead_trim = ci->id3->lead_trim;
 127     }
 128
 129     /* The main decoding loop */
 130     while (i < demux_res.num_sample_byte_sizes) {
 131         ci->yield();
 132
 133         if (ci->stop_codec || ci->new_track) {
 134             break;
 135         }
 136
 137         /* Deal with any pending seek requests */
 138         if (ci->seek_time) {
 139             if (alac_seek(&demux_res, &input_stream,
 140                           ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
 141                           &sound_samples_done, (int*) &i)) {
 142                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 143                 ci->set_elapsed(elapsed_time);
 144
 145                 if (i == 0)
 146                 {
 147                     lead_trim = ci->id3->lead_trim;
 148                 }
 149             }
 150             ci->seek_complete();
 151         }
 152
 153         /* Lookup the length (in samples and bytes) of block i */
 154         if (!get_sample_info(&demux_res, i, &sample_duration,
 155                              &sample_byte_size)) {
 156             LOGF("AAC: get_sample_info error\n");
 157             err = CODEC_ERROR;
 158             goto done;
 159         }
 160
 161         /* There can be gaps between chunks, so skip ahead if needed. It
 162          * doesn't seem to happen much, but it probably means that a
 163          * "proper" file can have chunks out of order. Why one would want
 164          * that an good question (but files with gaps do exist, so who
 165          * knows?), so we don't support that - for now, at least.
 166          */
 167         file_offset = get_sample_offset(&demux_res, i);
 168
 169         if (file_offset > ci->curpos)
 170         {
 171             ci->advance_buffer(file_offset - ci->curpos);
 172         }
 173         else if (file_offset == 0)
 174         {
 175             LOGF("AAC: get_sample_offset error\n");
 176             err = CODEC_ERROR;
 177             goto done;
 178         }
 179
 180         /* Request the required number of bytes from the input buffer */
 181         buffer=ci->request_buffer(&n,sample_byte_size);
 182
 183         /* Decode one block - returned samples will be host-endian */
 184         NeAACDecDecode(decoder, &frame_info, buffer, n);
 185         /* Ignore return value, we access samples in the decoder struct
 186          * directly.
 187          */
 188         if (frame_info.error > 0) {
 189             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 190             err = CODEC_ERROR;
 191             goto done;
 192         }
 193
 194         /* Advance codec buffer */
 195         ci->advance_buffer(n);
 196
 197         /* Output the audio */
 198         ci->yield();
 199
 200         framelength = (frame_info.samples >> 1) - lead_trim;
 201
 202         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 203         {
 204             /* Currently limited to at most one frame of tail_trim.
 205              * Seems to be enough.
 206              */
 207             if (ci->id3->tail_trim == 0
 208                 && sample_duration < (frame_info.samples >> 1))
 209             {
 210                 /* Subtract lead_trim just in case we decode a file with
 211                  * only one audio frame with actual data.
 212                  */
 213                 framelength = sample_duration - lead_trim;
 214             }
 215             else
 216             {
 217                 framelength -= ci->id3->tail_trim;
 218             }
 219         }
 220
 221         if (framelength > 0)
 222         {
 223             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 224                               &decoder->time_out[1][lead_trim],
 225                               framelength);
 226         }
 227
 228         if (lead_trim > 0)
 229         {
 230             /* frame_info.samples can be 0 for the first frame */
 231             lead_trim -= (i > 0 || frame_info.samples)
 232                 ? (frame_info.samples >> 1) : sample_duration;
 233
 234             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 235             {
 236                 lead_trim = 0;
 237             }
 238         }
 239
 240         /* Update the elapsed-time indicator */
 241         sound_samples_done += sample_duration;
 242         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 243         ci->set_elapsed(elapsed_time);
 244
 245         /* Keep track of current position - for resuming */
 246         ci->set_offset(elapsed_time);
 247
 248         i++;
 249     }
 250
 251     err = CODEC_OK;
 252
 253 done:
 254     LOGF("AAC: Decoded %lu samples\n", sound_samples_done);
 255
 256     if (ci->request_next_track())
 257         goto next_track;
 258
 259 exit:
 260     return err;
 261 }