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 static ALIGN real_t gb_time_buffer[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM;
  34 static 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     /* Clean and initialize decoder structures */
  73     memset(&demux_res , 0, sizeof(demux_res));
  74     if (codec_init()) {
  75         LOGF("FAAD: Codec init error\n");
  76         err = CODEC_ERROR;
  77         goto exit;
  78     }
  79
  80     while (!*ci->taginfo_ready && !ci->stop_codec)
  81         ci->sleep(1);
  82
  83     file_offset = ci->id3->offset;
  84
  85     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  86     codec_set_replaygain(ci->id3);
  87
  88     stream_create(&input_stream,ci);
  89
  90     /* if qtmovie_read returns successfully, the stream is up to
  91      * the movie data, which can be used directly by the decoder */
  92     if (!qtmovie_read(&input_stream, &demux_res)) {
  93         LOGF("FAAD: File init error\n");
  94         err = CODEC_ERROR;
  95         goto done;
  96     }
  97
  98     /* initialise the sound converter */
  99     decoder = NeAACDecOpen();
 100
 101     if (!decoder) {
 102         LOGF("FAAD: Decode open error\n");
 103         err = CODEC_ERROR;
 104         goto done;
 105     }
 106
 107     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
 108     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
 109     NeAACDecSetConfiguration(decoder, conf);
 110
 111     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 112     if (err) {
 113         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 114         err = CODEC_ERROR;
 115         goto done;
 116     }
 117
 118     /* Set pointer to be able to use IRAM an to avoid alloc in decoder. Must
 119      * be called after NeAACDecOpen(). */
 120     /* A buffer of framelength or 2*frameLenght size must be allocated for
 121      * time_out. If frameLength is too big or SBR/forceUpSampling is active,
 122      * we do not use the IRAM buffer and keep faad's internal allocation (see
 123      * specrec.c). */
 124     needed_bufsize = decoder->frameLength;
 125 #ifdef SBR_DEC
 126     if ((decoder->sbr_present_flag == 1) || (decoder->forceUpSampling == 1))
 127     {
 128         needed_bufsize *= 2;
 129     }
 130 #endif
 131     if (needed_bufsize <= GB_BUF_SIZE)
 132     {
 133         decoder->time_out[0]    = &gb_time_buffer[0][0];
 134         decoder->time_out[1]    = &gb_time_buffer[1][0];
 135     }
 136     /* A buffer of with frameLength elements must be allocated for fb_intermed.
 137      * If frameLength is too big, we do not use the IRAM buffer and keep faad's
 138      * internal allocation (see specrec.c). */
 139     needed_bufsize = decoder->frameLength;
 140     if (needed_bufsize <= GB_BUF_SIZE)
 141     {
 142         decoder->fb_intermed[0] = &gb_fb_intermed[0][0];
 143         decoder->fb_intermed[1] = &gb_fb_intermed[1][0];
 144     }
 145
 146     ci->id3->frequency = s;
 147
 148     i = 0;
 149
 150     if (file_offset > 0) {
 151         if (alac_seek_raw(&demux_res, &input_stream, file_offset,
 152                           &sound_samples_done, (int*) &i)) {
 153             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 154             ci->set_elapsed(elapsed_time);
 155         } else {
 156             sound_samples_done = 0;
 157         }
 158     } else {
 159         sound_samples_done = 0;
 160     }
 161
 162     if (i == 0)
 163     {
 164         lead_trim = ci->id3->lead_trim;
 165     }
 166
 167     /* The main decoding loop */
 168     while (i < demux_res.num_sample_byte_sizes) {
 169         ci->yield();
 170
 171         if (ci->stop_codec || ci->new_track) {
 172             break;
 173         }
 174
 175         /* Deal with any pending seek requests */
 176         if (ci->seek_time) {
 177             if (alac_seek(&demux_res, &input_stream,
 178                           ((ci->seek_time-1)/10)*(ci->id3->frequency/100),
 179                           &sound_samples_done, (int*) &i)) {
 180                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 181                 ci->set_elapsed(elapsed_time);
 182
 183                 if (i == 0)
 184                 {
 185                     lead_trim = ci->id3->lead_trim;
 186                 }
 187             }
 188             ci->seek_complete();
 189         }
 190
 191         /* Lookup the length (in samples and bytes) of block i */
 192         if (!get_sample_info(&demux_res, i, &sample_duration,
 193                              &sample_byte_size)) {
 194             LOGF("AAC: get_sample_info error\n");
 195             err = CODEC_ERROR;
 196             goto done;
 197         }
 198
 199         /* There can be gaps between chunks, so skip ahead if needed. It
 200          * doesn't seem to happen much, but it probably means that a
 201          * "proper" file can have chunks out of order. Why one would want
 202          * that an good question (but files with gaps do exist, so who
 203          * knows?), so we don't support that - for now, at least.
 204          */
 205         file_offset = get_sample_offset(&demux_res, i);
 206
 207         if (file_offset > ci->curpos)
 208         {
 209             ci->advance_buffer(file_offset - ci->curpos);
 210         }
 211         else if (file_offset == 0)
 212         {
 213             LOGF("AAC: get_sample_offset error\n");
 214             err = CODEC_ERROR;
 215             goto done;
 216         }
 217
 218         /* Request the required number of bytes from the input buffer */
 219         buffer=ci->request_buffer(&n,sample_byte_size);
 220
 221         /* Decode one block - returned samples will be host-endian */
 222         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 223
 224         /* NeAACDecDecode may sometimes return NULL without setting error. */
 225         if (ret == NULL || frame_info.error > 0) {
 226             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 227             err = CODEC_ERROR;
 228             goto done;
 229         }
 230
 231         /* Advance codec buffer (no need to call set_offset because of this) */
 232         ci->advance_buffer(n);
 233
 234         /* Output the audio */
 235         ci->yield();
 236
 237         framelength = (frame_info.samples >> 1) - lead_trim;
 238
 239         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 240         {
 241             /* Currently limited to at most one frame of tail_trim.
 242              * Seems to be enough.
 243              */
 244             if (ci->id3->tail_trim == 0
 245                 && sample_duration < (frame_info.samples >> 1))
 246             {
 247                 /* Subtract lead_trim just in case we decode a file with
 248                  * only one audio frame with actual data.
 249                  */
 250                 framelength = sample_duration - lead_trim;
 251             }
 252             else
 253             {
 254                 framelength -= ci->id3->tail_trim;
 255             }
 256         }
 257
 258         if (framelength > 0)
 259         {
 260             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 261                               &decoder->time_out[1][lead_trim],
 262                               framelength);
 263         }
 264
 265         if (lead_trim > 0)
 266         {
 267             /* frame_info.samples can be 0 for the first frame */
 268             lead_trim -= (i > 0 || frame_info.samples)
 269                 ? (frame_info.samples >> 1) : sample_duration;
 270
 271             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 272             {
 273                 lead_trim = 0;
 274             }
 275         }
 276
 277         /* Update the elapsed-time indicator */
 278         sound_samples_done += sample_duration;
 279         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 280         ci->set_elapsed(elapsed_time);
 281         i++;
 282     }
 283
 284     err = CODEC_OK;
 285
 286 done:
 287     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
 288
 289     if (ci->request_next_track())
 290         goto next_track;
 291
 292 exit:
 293     return err;
 294 }