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 /* The maximum buffer size handled by faad. 12 bytes are required by libfaad
  31  * as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered
  32  * for each frame. */
  33 #define FAAD_BYTE_BUFFER_SIZE (2048-12)
  34
  35 /* this is the codec entry point */
  36 enum codec_status codec_main(void)
  37 {
  38     /* Note that when dealing with QuickTime/MPEG4 files, terminology is
  39      * a bit confusing. Files with sound are split up in chunks, where
  40      * each chunk contains one or more samples. Each sample in turn
  41      * contains a number of "sound samples" (the kind you refer to with
  42      * the sampling frequency).
  43      */
  44     size_t n;
  45     demux_res_t demux_res;
  46     stream_t input_stream;
  47     uint32_t sound_samples_done;
  48     uint32_t elapsed_time;
  49     int file_offset;
  50     int framelength;
  51     int lead_trim = 0;
  52     unsigned int i;
  53     unsigned char* buffer;
  54     NeAACDecFrameInfo frame_info;
  55     NeAACDecHandle decoder;
  56     int err;
  57     uint32_t seek_idx = 0;
  58     uint32_t s = 0;
  59     uint32_t sbr_fac = 1;
  60     unsigned char c = 0;
  61     void *ret;
  62
  63     /* Generic codec initialisation */
  64     ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  65     ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  66
  67 next_track:
  68     err = CODEC_OK;
  69
  70     /* Clean and initialize decoder structures */
  71     memset(&demux_res , 0, sizeof(demux_res));
  72     if (codec_init()) {
  73         LOGF("FAAD: Codec init error\n");
  74         err = CODEC_ERROR;
  75         goto exit;
  76     }
  77
  78     if (codec_wait_taginfo() != 0)
  79         goto done;
  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 #ifdef SBR_DEC
 117     /* Check for need of special handling for seek/resume and elapsed time. */
 118     if (ci->id3->needs_upsampling_correction) {
 119         sbr_fac = 2;
 120     } else {
 121         sbr_fac = 1;
 122     }
 123 #endif
 124
 125     ci->id3->frequency = s;
 126
 127     i = 0;
 128
 129     if (file_offset > 0) {
 130         /* Resume the desired (byte) position. Important: When resuming SBR
 131          * upsampling files the resulting sound_samples_done must be expanded
 132          * by a factor of 2. This is done via using sbr_fac. */
 133         if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
 134                           &sound_samples_done, (int*) &i)) {
 135             sound_samples_done *= sbr_fac;
 136             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 137             ci->set_elapsed(elapsed_time);
 138         } else {
 139             sound_samples_done = 0;
 140         }
 141         NeAACDecPostSeekReset(decoder, i);
 142     } else {
 143         sound_samples_done = 0;
 144     }
 145
 146     if (i == 0)
 147     {
 148         lead_trim = ci->id3->lead_trim;
 149     }
 150
 151     /* The main decoding loop */
 152     while (i < demux_res.num_sample_byte_sizes) {
 153         ci->yield();
 154
 155         if (ci->stop_codec || ci->new_track) {
 156             break;
 157         }
 158
 159         /* Deal with any pending seek requests */
 160         if (ci->seek_time) {
 161             /* Seek to the desired time position. Important: When seeking in SBR
 162              * upsampling files the seek_time must be divided by 2 when calling
 163              * m4a_seek and the resulting sound_samples_done must be expanded
 164              * by a factor 2. This is done via using sbr_fac. */
 165             if (m4a_seek(&demux_res, &input_stream,
 166                           ((ci->seek_time-1)/10/sbr_fac)*(ci->id3->frequency/100),
 167                           &sound_samples_done, (int*) &i)) {
 168                 sound_samples_done *= sbr_fac;
 169                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 170                 ci->set_elapsed(elapsed_time);
 171                 seek_idx = 0;
 172
 173                 if (i == 0)
 174                 {
 175                     lead_trim = ci->id3->lead_trim;
 176                 }
 177             }
 178             NeAACDecPostSeekReset(decoder, i);
 179             ci->seek_complete();
 180         }
 181
 182         /* There can be gaps between chunks, so skip ahead if needed. It
 183          * doesn't seem to happen much, but it probably means that a
 184          * "proper" file can have chunks out of order. Why one would want
 185          * that an good question (but files with gaps do exist, so who
 186          * knows?), so we don't support that - for now, at least.
 187          */
 188         file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);
 189
 190         if (file_offset > ci->curpos)
 191         {
 192             ci->advance_buffer(file_offset - ci->curpos);
 193         }
 194         else if (file_offset == 0)
 195         {
 196             LOGF("AAC: get_sample_offset error\n");
 197             err = CODEC_ERROR;
 198             goto done;
 199         }
 200
 201         /* Request the required number of bytes from the input buffer */
 202         buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
 203
 204         /* Decode one block - returned samples will be host-endian */
 205         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 206
 207         /* NeAACDecDecode may sometimes return NULL without setting error. */
 208         if (ret == NULL || frame_info.error > 0) {
 209             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 210             err = CODEC_ERROR;
 211             goto done;
 212         }
 213
 214         /* Advance codec buffer (no need to call set_offset because of this) */
 215         ci->advance_buffer(frame_info.bytesconsumed);
 216
 217         /* Output the audio */
 218         ci->yield();
 219
 220         /* Gather number of samples for the decoded frame. */
 221         framelength = (frame_info.samples >> 1) - lead_trim;
 222
 223         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 224         {
 225             framelength -= ci->id3->tail_trim;
 226         }
 227
 228         if (framelength > 0)
 229         {
 230             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 231                               &decoder->time_out[1][lead_trim],
 232                               framelength);
 233         }
 234
 235         if (lead_trim > 0)
 236         {
 237             /* frame_info.samples can be 0 for the first frame */
 238             lead_trim -= (i > 0 || frame_info.samples)
 239                 ? (frame_info.samples >> 1) : (uint32_t)framelength;
 240
 241             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 242             {
 243                 lead_trim = 0;
 244             }
 245         }
 246
 247         /* Update the elapsed-time indicator */
 248         sound_samples_done += framelength;
 249         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 250         ci->set_elapsed(elapsed_time);
 251         i++;
 252     }
 253
 254 done:
 255     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
 256
 257     if (ci->request_next_track())
 258         goto next_track;
 259
 260 exit:
 261     return err;
 262 }