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     file_offset = 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 (file_offset > 0) {
 114         if (alac_seek_raw(&demux_res, &input_stream, file_offset,
 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     } else {
 122         sound_samples_done = 0;
 123     }
 124
 125     if (i == 0)
 126     {
 127         lead_trim = ci->id3->lead_trim;
 128     }
 129
 130     /* The main decoding loop */
 131     while (i < demux_res.num_sample_byte_sizes) {
 132         ci->yield();
 133
 134         if (ci->stop_codec || ci->new_track) {
 135             break;
 136         }
 137
 138         /* Deal with any pending seek requests */
 139         if (ci->seek_time) {
 140             if (alac_seek(&demux_res, &input_stream,
 141                           ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
 142                           &sound_samples_done, (int*) &i)) {
 143                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 144                 ci->set_elapsed(elapsed_time);
 145
 146                 if (i == 0)
 147                 {
 148                     lead_trim = ci->id3->lead_trim;
 149                 }
 150             }
 151             ci->seek_complete();
 152         }
 153
 154         /* Lookup the length (in samples and bytes) of block i */
 155         if (!get_sample_info(&demux_res, i, &sample_duration,
 156                              &sample_byte_size)) {
 157             LOGF("AAC: get_sample_info error\n");
 158             err = CODEC_ERROR;
 159             goto done;
 160         }
 161
 162         /* There can be gaps between chunks, so skip ahead if needed. It
 163          * doesn't seem to happen much, but it probably means that a
 164          * "proper" file can have chunks out of order. Why one would want
 165          * that an good question (but files with gaps do exist, so who
 166          * knows?), so we don't support that - for now, at least.
 167          */
 168         file_offset = get_sample_offset(&demux_res, i);
 169
 170         if (file_offset > ci->curpos)
 171         {
 172             ci->advance_buffer(file_offset - ci->curpos);
 173         }
 174         else if (file_offset == 0)
 175         {
 176             LOGF("AAC: get_sample_offset error\n");
 177             err = CODEC_ERROR;
 178             goto done;
 179         }
 180
 181         /* Request the required number of bytes from the input buffer */
 182         buffer=ci->request_buffer(&n,sample_byte_size);
 183
 184         /* Decode one block - returned samples will be host-endian */
 185         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 186
 187         /* NeAACDecDecode may sometimes return NULL without setting error. */
 188         if (ret == NULL || 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 (no need to call set_offset because of this) */
 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         i++;
 245     }
 246
 247     err = CODEC_OK;
 248
 249 done:
 250     LOGF("AAC: Decoded %lu samples\n", sound_samples_done);
 251
 252     if (ci->request_next_track())
 253         goto next_track;
 254
 255 exit:
 256     return err;
 257 }