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 real_t gb_time_buffer[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM MEM_ALIGN_ATTR;
  34 static real_t gb_fb_intermed[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM MEM_ALIGN_ATTR;
  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     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     NeAACDecFrameInfo frame_info;
  59     NeAACDecHandle decoder;
  60     int err;
  61     uint32_t s = 0;
  62     uint32_t sbr_fac = 1;
  63     unsigned char c = 0;
  64     void *ret;
  65
  66     /* Generic codec initialisation */
  67     ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  68     ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  69
  70 next_track:
  71     err = CODEC_OK;
  72
  73     /* Clean and initialize decoder structures */
  74     memset(&demux_res , 0, sizeof(demux_res));
  75     if (codec_init()) {
  76         LOGF("FAAD: Codec init error\n");
  77         err = CODEC_ERROR;
  78         goto exit;
  79     }
  80
  81     while (!*ci->taginfo_ready && !ci->stop_codec)
  82         ci->sleep(1);
  83
  84     file_offset = ci->id3->offset;
  85
  86     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  87     codec_set_replaygain(ci->id3);
  88
  89     stream_create(&input_stream,ci);
  90
  91     /* if qtmovie_read returns successfully, the stream is up to
  92      * the movie data, which can be used directly by the decoder */
  93     if (!qtmovie_read(&input_stream, &demux_res)) {
  94         LOGF("FAAD: File init error\n");
  95         err = CODEC_ERROR;
  96         goto done;
  97     }
  98
  99     /* initialise the sound converter */
 100     decoder = NeAACDecOpen();
 101
 102     if (!decoder) {
 103         LOGF("FAAD: Decode open error\n");
 104         err = CODEC_ERROR;
 105         goto done;
 106     }
 107
 108     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
 109     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
 110     NeAACDecSetConfiguration(decoder, conf);
 111
 112     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 113     if (err) {
 114         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 115         err = CODEC_ERROR;
 116         goto done;
 117     }
 118
 119     /* Set pointer to be able to use IRAM an to avoid alloc in decoder. Must
 120      * be called after NeAACDecOpen(). */
 121     /* A buffer of framelength or 2*frameLenght size must be allocated for
 122      * time_out. If frameLength is too big or SBR/forceUpSampling is active,
 123      * we do not use the IRAM buffer and keep faad's internal allocation (see
 124      * specrec.c). */
 125     needed_bufsize = decoder->frameLength;
 126 #ifdef SBR_DEC
 127     if ((decoder->sbr_present_flag == 1) || (decoder->forceUpSampling == 1))
 128     {
 129         needed_bufsize *= 2;
 130     }
 131 #endif
 132     if (needed_bufsize <= GB_BUF_SIZE)
 133     {
 134         decoder->time_out[0]    = &gb_time_buffer[0][0];
 135         decoder->time_out[1]    = &gb_time_buffer[1][0];
 136     }
 137     /* A buffer of with frameLength elements must be allocated for fb_intermed.
 138      * If frameLength is too big, we do not use the IRAM buffer and keep faad's
 139      * internal allocation (see specrec.c). */
 140     needed_bufsize = decoder->frameLength;
 141     if (needed_bufsize <= GB_BUF_SIZE)
 142     {
 143         decoder->fb_intermed[0] = &gb_fb_intermed[0][0];
 144         decoder->fb_intermed[1] = &gb_fb_intermed[1][0];
 145     }
 146
 147 #ifdef SBR_DEC
 148     /* Check for need of special handling for seek/resume and elapsed time. */
 149     if (ci->id3->needs_upsampling_correction) {
 150         sbr_fac = 2;
 151     } else {
 152         sbr_fac = 1;
 153     }
 154 #endif
 155
 156     ci->id3->frequency = s;
 157
 158     i = 0;
 159
 160     if (file_offset > 0) {
 161         /* Resume the desired (byte) position. Important: When resuming SBR
 162          * upsampling files the resulting sound_samples_done must be expanded
 163          * by a factor of 2. This is done via using sbr_fac. */
 164         if (alac_seek_raw(&demux_res, &input_stream, file_offset,
 165                           &sound_samples_done, (int*) &i)) {
 166             sound_samples_done *= sbr_fac;
 167             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 168             ci->set_elapsed(elapsed_time);
 169         } else {
 170             sound_samples_done = 0;
 171         }
 172     } else {
 173         sound_samples_done = 0;
 174     }
 175
 176     if (i == 0)
 177     {
 178         lead_trim = ci->id3->lead_trim;
 179     }
 180
 181     /* The main decoding loop */
 182     while (i < demux_res.num_sample_byte_sizes) {
 183         ci->yield();
 184
 185         if (ci->stop_codec || ci->new_track) {
 186             break;
 187         }
 188
 189         /* Deal with any pending seek requests */
 190         if (ci->seek_time) {
 191             /* Seek to the desired time position. Important: When seeking in SBR
 192              * upsampling files the seek_time must be divided by 2 when calling
 193              * alac_seek and the resulting sound_samples_done must be expanded
 194              * by a factor 2. This is done via using sbr_fac. */
 195             if (alac_seek(&demux_res, &input_stream,
 196                           ((ci->seek_time-1)/10/sbr_fac)*(ci->id3->frequency/100),
 197                           &sound_samples_done, (int*) &i)) {
 198                 sound_samples_done *= sbr_fac;
 199                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 200                 ci->set_elapsed(elapsed_time);
 201
 202                 if (i == 0)
 203                 {
 204                     lead_trim = ci->id3->lead_trim;
 205                 }
 206             }
 207             ci->seek_complete();
 208         }
 209
 210         /* Lookup the length (in samples and bytes) of block i */
 211         if (!get_sample_info(&demux_res, i, &sample_duration,
 212                              &sample_byte_size)) {
 213             LOGF("AAC: get_sample_info error\n");
 214             err = CODEC_ERROR;
 215             goto done;
 216         }
 217
 218         /* There can be gaps between chunks, so skip ahead if needed. It
 219          * doesn't seem to happen much, but it probably means that a
 220          * "proper" file can have chunks out of order. Why one would want
 221          * that an good question (but files with gaps do exist, so who
 222          * knows?), so we don't support that - for now, at least.
 223          */
 224         file_offset = get_sample_offset(&demux_res, i);
 225
 226         if (file_offset > ci->curpos)
 227         {
 228             ci->advance_buffer(file_offset - ci->curpos);
 229         }
 230         else if (file_offset == 0)
 231         {
 232             LOGF("AAC: get_sample_offset error\n");
 233             err = CODEC_ERROR;
 234             goto done;
 235         }
 236
 237         /* Request the required number of bytes from the input buffer */
 238         buffer=ci->request_buffer(&n,sample_byte_size);
 239
 240         /* Decode one block - returned samples will be host-endian */
 241         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 242
 243         /* NeAACDecDecode may sometimes return NULL without setting error. */
 244         if (ret == NULL || frame_info.error > 0) {
 245             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 246             err = CODEC_ERROR;
 247             goto done;
 248         }
 249
 250         /* Advance codec buffer (no need to call set_offset because of this) */
 251         ci->advance_buffer(n);
 252
 253         /* Output the audio */
 254         ci->yield();
 255
 256         /* Ensure correct sample_duration is used. For SBR upsampling files
 257          * sample_duration is only half the size of real output frame size. */
 258         sample_duration *= sbr_fac;
 259
 260         framelength = (frame_info.samples >> 1) - lead_trim;
 261
 262         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 263         {
 264             /* Currently limited to at most one frame of tail_trim.
 265              * Seems to be enough.
 266              */
 267             if (ci->id3->tail_trim == 0
 268                 && sample_duration < (frame_info.samples >> 1))
 269             {
 270                 /* Subtract lead_trim just in case we decode a file with
 271                  * only one audio frame with actual data.
 272                  */
 273                 framelength = sample_duration - lead_trim;
 274             }
 275             else
 276             {
 277                 framelength -= ci->id3->tail_trim;
 278             }
 279         }
 280
 281         if (framelength > 0)
 282         {
 283             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 284                               &decoder->time_out[1][lead_trim],
 285                               framelength);
 286         }
 287
 288         if (lead_trim > 0)
 289         {
 290             /* frame_info.samples can be 0 for the first frame */
 291             lead_trim -= (i > 0 || frame_info.samples)
 292                 ? (frame_info.samples >> 1) : (uint32_t)framelength;
 293
 294             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 295             {
 296                 lead_trim = 0;
 297             }
 298         }
 299
 300         /* Update the elapsed-time indicator */
 301         sound_samples_done += framelength;
 302         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 303         ci->set_elapsed(elapsed_time);
 304         i++;
 305     }
 306
 307     err = CODEC_OK;
 308
 309 done:
 310     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
 311
 312     if (ci->request_next_track())
 313         goto next_track;
 314
 315 exit:
 316     return err;
 317 }