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