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(enum codec_entry_call_reason reason)
  37 {
  38     if (reason == CODEC_LOAD) {
  39         /* Generic codec initialisation */
  40         ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  41         ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  42     }
  43
  44     return CODEC_OK;
  45 }
  46
  47 /* this is called for each file to process */
  48 enum codec_status codec_run(void)
  49 {
  50     /* Note that when dealing with QuickTime/MPEG4 files, terminology is
  51      * a bit confusing. Files with sound are split up in chunks, where
  52      * each chunk contains one or more samples. Each sample in turn
  53      * contains a number of "sound samples" (the kind you refer to with
  54      * the sampling frequency).
  55      */
  56     size_t n;
  57     demux_res_t demux_res;
  58     stream_t input_stream;
  59     uint32_t sound_samples_done;
  60     uint32_t elapsed_time;
  61     int file_offset;
  62     int framelength;
  63     int lead_trim = 0;
  64     unsigned int i;
  65     unsigned char* buffer;
  66     NeAACDecFrameInfo frame_info;
  67     NeAACDecHandle decoder;
  68     int err;
  69     uint32_t seek_idx = 0;
  70     uint32_t s = 0;
  71     uint32_t sbr_fac = 1;
  72     unsigned char c = 0;
  73     void *ret;
  74     intptr_t param;
  75
  76     /* Clean and initialize decoder structures */
  77     memset(&demux_res , 0, sizeof(demux_res));
  78     if (codec_init()) {
  79         LOGF("FAAD: Codec init error\n");
  80         return CODEC_ERROR;
  81     }
  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     ci->seek_buffer(ci->id3->first_frame_offset);
  91
  92     /* if qtmovie_read returns successfully, the stream is up to
  93      * the movie data, which can be used directly by the decoder */
  94     if (!qtmovie_read(&input_stream, &demux_res)) {
  95         LOGF("FAAD: File init error\n");
  96         return CODEC_ERROR;
  97     }
  98
  99     /* initialise the sound converter */
 100     decoder = NeAACDecOpen();
 101
 102     if (!decoder) {
 103         LOGF("FAAD: Decode open error\n");
 104         return CODEC_ERROR;
 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         return CODEC_ERROR;
 115     }
 116
 117 #ifdef SBR_DEC
 118     /* Check for need of special handling for seek/resume and elapsed time. */
 119     if (ci->id3->needs_upsampling_correction) {
 120         sbr_fac = 2;
 121     } else {
 122         sbr_fac = 1;
 123     }
 124 #endif
 125
 126     i = 0;
 127
 128     if (file_offset > 0) {
 129         /* Resume the desired (byte) position. Important: When resuming SBR
 130          * upsampling files the resulting sound_samples_done must be expanded
 131          * by a factor of 2. This is done via using sbr_fac. */
 132         if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
 133                           &sound_samples_done, (int*) &i)) {
 134             sound_samples_done *= sbr_fac;
 135             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 136             ci->set_elapsed(elapsed_time);
 137         } else {
 138             sound_samples_done = 0;
 139         }
 140         NeAACDecPostSeekReset(decoder, i);
 141     } else {
 142         sound_samples_done = 0;
 143     }
 144
 145     if (i == 0)
 146     {
 147         lead_trim = ci->id3->lead_trim;
 148     }
 149
 150     /* The main decoding loop */
 151     while (i < demux_res.num_sample_byte_sizes) {
 152         enum codec_command_action action = ci->get_command(&param);
 153
 154         if (action == CODEC_ACTION_HALT)
 155             break;
 156
 157         /* Deal with any pending seek requests */
 158         if (action == CODEC_ACTION_SEEK_TIME) {
 159             /* Seek to the desired time position. Important: When seeking in SBR
 160              * upsampling files the seek_time must be divided by 2 when calling
 161              * m4a_seek and the resulting sound_samples_done must be expanded
 162              * by a factor 2. This is done via using sbr_fac. */
 163             if (m4a_seek(&demux_res, &input_stream,
 164                           (param/10/sbr_fac)*(ci->id3->frequency/100),
 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                 seek_idx = 0;
 170
 171                 if (i == 0)
 172                 {
 173                     lead_trim = ci->id3->lead_trim;
 174                 }
 175             }
 176             NeAACDecPostSeekReset(decoder, i);
 177             ci->seek_complete();
 178         }
 179
 180         /* There can be gaps between chunks, so skip ahead if needed. It
 181          * doesn't seem to happen much, but it probably means that a
 182          * "proper" file can have chunks out of order. Why one would want
 183          * that an good question (but files with gaps do exist, so who
 184          * knows?), so we don't support that - for now, at least.
 185          */
 186         file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);
 187
 188         if (file_offset > ci->curpos)
 189         {
 190             ci->advance_buffer(file_offset - ci->curpos);
 191         }
 192         else if (file_offset == 0)
 193         {
 194             LOGF("AAC: get_sample_offset error\n");
 195             return CODEC_ERROR;
 196         }
 197
 198         /* Request the required number of bytes from the input buffer */
 199         buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
 200
 201         /* Decode one block - returned samples will be host-endian */
 202         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 203
 204         /* NeAACDecDecode may sometimes return NULL without setting error. */
 205         if (ret == NULL || frame_info.error > 0) {
 206             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 207             return CODEC_ERROR;
 208         }
 209
 210         /* Advance codec buffer (no need to call set_offset because of this) */
 211         ci->advance_buffer(frame_info.bytesconsumed);
 212
 213         /* Output the audio */
 214         ci->yield();
 215
 216         /* Gather number of samples for the decoded frame. */
 217         framelength = (frame_info.samples >> 1) - lead_trim;
 218
 219         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 220         {
 221             framelength -= ci->id3->tail_trim;
 222         }
 223
 224         if (framelength > 0)
 225         {
 226             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 227                               &decoder->time_out[1][lead_trim],
 228                               framelength);
 229         }
 230
 231         if (lead_trim > 0)
 232         {
 233             /* frame_info.samples can be 0 for the first frame */
 234             lead_trim -= (i > 0 || frame_info.samples)
 235                 ? (frame_info.samples >> 1) : (uint32_t)framelength;
 236
 237             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 238             {
 239                 lead_trim = 0;
 240             }
 241         }
 242
 243         /* Update the elapsed-time indicator */
 244         sound_samples_done += framelength;
 245         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 246         ci->set_elapsed(elapsed_time);
 247         i++;
 248     }
 249
 250     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
 251     return CODEC_OK;
 252 }