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(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  59     ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  60
  61 next_track:
  62     err = CODEC_OK;
  63
  64     if (codec_init()) {
  65         LOGF("FAAD: Codec init error\n");
  66         err = CODEC_ERROR;
  67         goto exit;
  68     }
  69
  70     while (!*ci->taginfo_ready && !ci->stop_codec)
  71         ci->sleep(1);
  72
  73     sound_samples_done = ci->id3->offset;
  74
  75     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  76     codec_set_replaygain(ci->id3);
  77
  78     stream_create(&input_stream,ci);
  79
  80     /* if qtmovie_read returns successfully, the stream is up to
  81      * the movie data, which can be used directly by the decoder */
  82     if (!qtmovie_read(&input_stream, &demux_res)) {
  83         LOGF("FAAD: File init error\n");
  84         err = CODEC_ERROR;
  85         goto done;
  86     }
  87
  88     /* initialise the sound converter */
  89     decoder = NeAACDecOpen();
  90
  91     if (!decoder) {
  92         LOGF("FAAD: Decode open error\n");
  93         err = CODEC_ERROR;
  94         goto done;
  95     }
  96
  97     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
  98     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
  99     NeAACDecSetConfiguration(decoder, conf);
 100
 101     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 102     if (err) {
 103         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 104         err = CODEC_ERROR;
 105         goto done;
 106     }
 107
 108     ci->id3->frequency = s;
 109
 110     i = 0;
 111
 112     if (sound_samples_done > 0) {
 113         if (alac_seek_raw(&demux_res, &input_stream, sound_samples_done,
 114                           &sound_samples_done, (int*) &i)) {
 115             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 116             ci->set_elapsed(elapsed_time);
 117         } else {
 118             sound_samples_done = 0;
 119         }
 120     }
 121
 122     if (i == 0)
 123     {
 124         lead_trim = ci->id3->lead_trim;
 125     }
 126
 127     /* The main decoding loop */
 128     while (i < demux_res.num_sample_byte_sizes) {
 129         ci->yield();
 130
 131         if (ci->stop_codec || ci->new_track) {
 132             break;
 133         }
 134
 135         /* Deal with any pending seek requests */
 136         if (ci->seek_time) {
 137             if (alac_seek(&demux_res, &input_stream,
 138                           ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
 139                           &sound_samples_done, (int*) &i)) {
 140                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 141                 ci->set_elapsed(elapsed_time);
 142
 143                 if (i == 0)
 144                 {
 145                     lead_trim = ci->id3->lead_trim;
 146                 }
 147             }
 148             ci->seek_complete();
 149         }
 150
 151         /* Lookup the length (in samples and bytes) of block i */
 152         if (!get_sample_info(&demux_res, i, &sample_duration,
 153                              &sample_byte_size)) {
 154             LOGF("AAC: get_sample_info error\n");
 155             err = CODEC_ERROR;
 156             goto done;
 157         }
 158
 159         /* There can be gaps between chunks, so skip ahead if needed. It
 160          * doesn't seem to happen much, but it probably means that a
 161          * "proper" file can have chunks out of order. Why one would want
 162          * that an good question (but files with gaps do exist, so who
 163          * knows?), so we don't support that - for now, at least.
 164          */
 165         file_offset = get_sample_offset(&demux_res, i);
 166
 167         if (file_offset > ci->curpos)
 168         {
 169             ci->advance_buffer(file_offset - ci->curpos);
 170         }
 171         else if (file_offset == 0)
 172         {
 173             LOGF("AAC: get_sample_offset error\n");
 174             err = CODEC_ERROR;
 175             goto done;
 176         }
 177
 178         /* Request the required number of bytes from the input buffer */
 179         buffer=ci->request_buffer(&n,sample_byte_size);
 180
 181         /* Decode one block - returned samples will be host-endian */
 182         NeAACDecDecode(decoder, &frame_info, buffer, n);
 183         /* Ignore return value, we access samples in the decoder struct
 184          * directly.
 185          */
 186         if (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 }