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 frame_samples;
  65     unsigned int i;
  66     unsigned char* buffer;
  67     NeAACDecFrameInfo frame_info;
  68     NeAACDecHandle decoder;
  69     int err;
  70     uint32_t seek_idx = 0;
  71     uint32_t s = 0;
  72     uint32_t sbr_fac = 1;
  73     unsigned char c = 0;
  74     void *ret;
  75     intptr_t param;
  76     bool empty_first_frame = false;
  77
  78     /* Clean and initialize decoder structures */
  79     memset(&demux_res , 0, sizeof(demux_res));
  80     if (codec_init()) {
  81         LOGF("FAAD: Codec init error\n");
  82         return CODEC_ERROR;
  83     }
  84
  85     file_offset = ci->id3->offset;
  86
  87     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  88     codec_set_replaygain(ci->id3);
  89
  90     stream_create(&input_stream,ci);
  91
  92     ci->seek_buffer(ci->id3->first_frame_offset);
  93
  94     /* if qtmovie_read returns successfully, the stream is up to
  95      * the movie data, which can be used directly by the decoder */
  96     if (!qtmovie_read(&input_stream, &demux_res)) {
  97         LOGF("FAAD: File init error\n");
  98         return CODEC_ERROR;
  99     }
 100
 101     /* initialise the sound converter */
 102     decoder = NeAACDecOpen();
 103
 104     if (!decoder) {
 105         LOGF("FAAD: Decode open error\n");
 106         return CODEC_ERROR;
 107     }
 108
 109     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
 110     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
 111     NeAACDecSetConfiguration(decoder, conf);
 112
 113     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 114     if (err) {
 115         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 116         return CODEC_ERROR;
 117     }
 118
 119 #ifdef SBR_DEC
 120     /* Check for need of special handling for seek/resume and elapsed time. */
 121     if (ci->id3->needs_upsampling_correction) {
 122         sbr_fac = 2;
 123     } else {
 124         sbr_fac = 1;
 125     }
 126 #endif
 127
 128     i = 0;
 129
 130     if (file_offset > 0) {
 131         /* Resume the desired (byte) position. Important: When resuming SBR
 132          * upsampling files the resulting sound_samples_done must be expanded
 133          * by a factor of 2. This is done via using sbr_fac. */
 134         if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
 135                           &sound_samples_done, (int*) &i)) {
 136             sound_samples_done *= sbr_fac;
 137             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 138             ci->set_elapsed(elapsed_time);
 139         } else {
 140             sound_samples_done = 0;
 141         }
 142         NeAACDecPostSeekReset(decoder, i);
 143     } else {
 144         sound_samples_done = 0;
 145     }
 146
 147     if (i == 0)
 148     {
 149         lead_trim = ci->id3->lead_trim;
 150     }
 151
 152     /* The main decoding loop */
 153     while (i < demux_res.num_sample_byte_sizes) {
 154         enum codec_command_action action = ci->get_command(&param);
 155
 156         if (action == CODEC_ACTION_HALT)
 157             break;
 158
 159         /* Deal with any pending seek requests */
 160         if (action == CODEC_ACTION_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                           (param/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             return CODEC_ERROR;
 198         }
 199
 200         /* Request the required number of bytes from the input buffer */
 201         buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
 202
 203         /* Decode one block - returned samples will be host-endian */
 204         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 205
 206         /* NeAACDecDecode may sometimes return NULL without setting error. */
 207         if (ret == NULL || frame_info.error > 0) {
 208             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 209             return CODEC_ERROR;
 210         }
 211
 212         /* Advance codec buffer (no need to call set_offset because of this) */
 213         ci->advance_buffer(frame_info.bytesconsumed);
 214
 215         /* Output the audio */
 216         ci->yield();
 217
 218         frame_samples = frame_info.samples >> 1;
 219
 220         if (empty_first_frame)
 221         {
 222             /* Remove the first frame from lead_trim, under the assumption
 223              * that it had the same size as this frame
 224              */
 225             empty_first_frame = false;
 226             lead_trim -= frame_samples;
 227
 228             if (lead_trim < 0)
 229             {
 230                 lead_trim = 0;
 231             }
 232         }
 233
 234         /* Gather number of samples for the decoded frame. */
 235         framelength = frame_samples - lead_trim;
 236
 237         if (i == demux_res.num_sample_byte_sizes - 1)
 238         {
 239             // Size of the last frame
 240             const uint32_t sample_duration = (demux_res.num_time_to_samples > 0) ?
 241                 demux_res.time_to_sample[demux_res.num_time_to_samples - 1].sample_duration :
 242                 frame_samples;
 243
 244             /* Currently limited to at most one frame of tail_trim.
 245              * Seems to be enough.
 246              */
 247             if (ci->id3->tail_trim == 0 && sample_duration < frame_samples)
 248             {
 249                 /* Subtract lead_trim just in case we decode a file with only
 250                  * one audio frame with actual data (lead_trim is usually zero
 251                  * here).
 252                  */
 253                 framelength = sample_duration - lead_trim;
 254             }
 255             else
 256             {
 257                 framelength -= ci->id3->tail_trim;
 258             }
 259         }
 260
 261         if (framelength > 0)
 262         {
 263             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 264                               &decoder->time_out[1][lead_trim],
 265                               framelength);
 266             sound_samples_done += framelength;
 267             /* Update the elapsed-time indicator */
 268             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 269             ci->set_elapsed(elapsed_time);
 270         }
 271
 272         if (lead_trim > 0)
 273         {
 274             /* frame_info.samples can be 0 for frame 0. We still want to
 275              * remove it from lead_trim, so do that during frame 1.
 276              */
 277             if (0 == i && 0 == frame_info.samples)
 278             {
 279                 empty_first_frame = true;
 280             }
 281
 282             lead_trim -= frame_samples;
 283
 284             if (lead_trim < 0)
 285             {
 286                 lead_trim = 0;
 287             }
 288         }
 289
 290         ++i;
 291     }
 292
 293     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
 294     return CODEC_OK;
 295 }