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 /* Global buffers to be used in the mdct synthesis. This way the arrays can
  31  * be moved to IRAM for some targets */
  32 #define GB_BUF_SIZE 1024
  33 ALIGN real_t gb_time_buffer[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM;
  34 ALIGN real_t gb_fb_intermed[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM;
  35
  36 /* this is the codec entry point */
  37 enum codec_status codec_main(void)
  38 {
  39     /* Note that when dealing with QuickTime/MPEG4 files, terminology is
  40      * a bit confusing. Files with sound are split up in chunks, where
  41      * each chunk contains one or more samples. Each sample in turn
  42      * contains a number of "sound samples" (the kind you refer to with
  43      * the sampling frequency).
  44      */
  45     size_t n;
  46     static demux_res_t demux_res;
  47     stream_t input_stream;
  48     uint32_t sound_samples_done;
  49     uint32_t elapsed_time;
  50     uint32_t sample_duration;
  51     uint32_t sample_byte_size;
  52     int file_offset;
  53     int framelength;
  54     int lead_trim = 0;
  55     int needed_bufsize;
  56     unsigned int i;
  57     unsigned char* buffer;
  58     static NeAACDecFrameInfo frame_info;
  59     NeAACDecHandle decoder;
  60     int err;
  61     uint32_t s = 0;
  62     unsigned char c = 0;
  63     void *ret;
  64
  65     /* Generic codec initialisation */
  66     ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  67     ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  68
  69 next_track:
  70     err = CODEC_OK;
  71
  72     if (codec_init()) {
  73         LOGF("FAAD: Codec init error\n");
  74         err = CODEC_ERROR;
  75         goto exit;
  76     }
  77
  78     while (!*ci->taginfo_ready && !ci->stop_codec)
  79         ci->sleep(1);
  80
  81     file_offset = ci->id3->offset;
  82
  83     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  84     codec_set_replaygain(ci->id3);
  85
  86     stream_create(&input_stream,ci);
  87
  88     /* if qtmovie_read returns successfully, the stream is up to
  89      * the movie data, which can be used directly by the decoder */
  90     if (!qtmovie_read(&input_stream, &demux_res)) {
  91         LOGF("FAAD: File init error\n");
  92         err = CODEC_ERROR;
  93         goto done;
  94     }
  95
  96     /* initialise the sound converter */
  97     decoder = NeAACDecOpen();
  98
  99     if (!decoder) {
 100         LOGF("FAAD: Decode open error\n");
 101         err = CODEC_ERROR;
 102         goto done;
 103     }
 104
 105     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
 106     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
 107     NeAACDecSetConfiguration(decoder, conf);
 108
 109     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 110     if (err) {
 111         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 112         err = CODEC_ERROR;
 113         goto done;
 114     }
 115
 116     /* Set pointer to be able to use IRAM an to avoid alloc in decoder. Must
 117      * be called after NeAACDecOpen(). */
 118     /* A buffer of framelength or 2*frameLenght size must be allocated for
 119      * time_out. If frameLength is too big or SBR/forceUpSampling is active,
 120      * we do not use the IRAM buffer and keep faad's internal allocation (see
 121      * specrec.c). */
 122     needed_bufsize = decoder->frameLength;
 123 #ifdef SBR_DEC
 124     if ((decoder->sbr_present_flag == 1) || (decoder->forceUpSampling == 1))
 125     {
 126         needed_bufsize *= 2;
 127     }
 128 #endif
 129     if (needed_bufsize <= GB_BUF_SIZE)
 130     {
 131         decoder->time_out[0]    = &gb_time_buffer[0][0];
 132         decoder->time_out[1]    = &gb_time_buffer[1][0];
 133     }
 134     /* A buffer of with frameLength elements must be allocated for fb_intermed.
 135      * If frameLength is too big, we do not use the IRAM buffer and keep faad's
 136      * internal allocation (see specrec.c). */
 137     needed_bufsize = decoder->frameLength;
 138     if (needed_bufsize <= GB_BUF_SIZE)
 139     {
 140         decoder->fb_intermed[0] = &gb_fb_intermed[0][0];
 141         decoder->fb_intermed[1] = &gb_fb_intermed[1][0];
 142     }
 143
 144     ci->id3->frequency = s;
 145
 146     i = 0;
 147
 148     if (file_offset > 0) {
 149         if (alac_seek_raw(&demux_res, &input_stream, file_offset,
 150                           &sound_samples_done, (int*) &i)) {
 151             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 152             ci->set_elapsed(elapsed_time);
 153         } else {
 154             sound_samples_done = 0;
 155         }
 156     } else {
 157         sound_samples_done = 0;
 158     }
 159
 160     if (i == 0)
 161     {
 162         lead_trim = ci->id3->lead_trim;
 163     }
 164
 165     /* The main decoding loop */
 166     while (i < demux_res.num_sample_byte_sizes) {
 167         ci->yield();
 168
 169         if (ci->stop_codec || ci->new_track) {
 170             break;
 171         }
 172
 173         /* Deal with any pending seek requests */
 174         if (ci->seek_time) {
 175             if (alac_seek(&demux_res, &input_stream,
 176                           ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
 177                           &sound_samples_done, (int*) &i)) {
 178                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 179                 ci->set_elapsed(elapsed_time);
 180
 181                 if (i == 0)
 182                 {
 183                     lead_trim = ci->id3->lead_trim;
 184                 }
 185             }
 186             ci->seek_complete();
 187         }
 188
 189         /* Lookup the length (in samples and bytes) of block i */
 190         if (!get_sample_info(&demux_res, i, &sample_duration,
 191                              &sample_byte_size)) {
 192             LOGF("AAC: get_sample_info error\n");
 193             err = CODEC_ERROR;
 194             goto done;
 195         }
 196
 197         /* There can be gaps between chunks, so skip ahead if needed. It
 198          * doesn't seem to happen much, but it probably means that a
 199          * "proper" file can have chunks out of order. Why one would want
 200          * that an good question (but files with gaps do exist, so who
 201          * knows?), so we don't support that - for now, at least.
 202          */
 203         file_offset = get_sample_offset(&demux_res, i);
 204
 205         if (file_offset > ci->curpos)
 206         {
 207             ci->advance_buffer(file_offset - ci->curpos);
 208         }
 209         else if (file_offset == 0)
 210         {
 211             LOGF("AAC: get_sample_offset error\n");
 212             err = CODEC_ERROR;
 213             goto done;
 214         }
 215
 216         /* Request the required number of bytes from the input buffer */
 217         buffer=ci->request_buffer(&n,sample_byte_size);
 218
 219         /* Decode one block - returned samples will be host-endian */
 220         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 221
 222         /* NeAACDecDecode may sometimes return NULL without setting error. */
 223         if (ret == NULL || frame_info.error > 0) {
 224             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 225             err = CODEC_ERROR;
 226             goto done;
 227         }
 228
 229         /* Advance codec buffer (no need to call set_offset because of this) */
 230         ci->advance_buffer(n);
 231
 232         /* Output the audio */
 233         ci->yield();
 234
 235         framelength = (frame_info.samples >> 1) - lead_trim;
 236
 237         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 238         {
 239             /* Currently limited to at most one frame of tail_trim.
 240              * Seems to be enough.
 241              */
 242             if (ci->id3->tail_trim == 0
 243                 && sample_duration < (frame_info.samples >> 1))
 244             {
 245                 /* Subtract lead_trim just in case we decode a file with
 246                  * only one audio frame with actual data.
 247                  */
 248                 framelength = sample_duration - lead_trim;
 249             }
 250             else
 251             {
 252                 framelength -= ci->id3->tail_trim;
 253             }
 254         }
 255
 256         if (framelength > 0)
 257         {
 258             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 259                               &decoder->time_out[1][lead_trim],
 260                               framelength);
 261         }
 262
 263         if (lead_trim > 0)
 264         {
 265             /* frame_info.samples can be 0 for the first frame */
 266             lead_trim -= (i > 0 || frame_info.samples)
 267                 ? (frame_info.samples >> 1) : sample_duration;
 268
 269             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 270             {
 271                 lead_trim = 0;
 272             }
 273         }
 274
 275         /* Update the elapsed-time indicator */
 276         sound_samples_done += sample_duration;
 277         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 278         ci->set_elapsed(elapsed_time);
 279         i++;
 280     }
 281
 282     err = CODEC_OK;
 283
 284 done:
 285     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
 286
 287     if (ci->request_next_track())
 288         goto next_track;
 289
 290 exit:
 291     return err;
 292 }