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 /* Global buffers to be used in the mdct synthesis. This way the arrays can
  36  * be moved to IRAM for some targets */
  37 #define GB_BUF_SIZE 1024
  38 static real_t gb_time_buffer[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM MEM_ALIGN_ATTR;
  39 static real_t gb_fb_intermed[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM MEM_ALIGN_ATTR;
  40
  41 /* this is the codec entry point */
  42 enum codec_status codec_main(void)
  43 {
  44     /* Note that when dealing with QuickTime/MPEG4 files, terminology is
  45      * a bit confusing. Files with sound are split up in chunks, where
  46      * each chunk contains one or more samples. Each sample in turn
  47      * contains a number of "sound samples" (the kind you refer to with
  48      * the sampling frequency).
  49      */
  50     size_t n;
  51     demux_res_t demux_res;
  52     stream_t input_stream;
  53     uint32_t sound_samples_done;
  54     uint32_t elapsed_time;
  55     int file_offset;
  56     int framelength;
  57     int lead_trim = 0;
  58     int needed_bufsize;
  59     unsigned int i;
  60     unsigned char* buffer;
  61     NeAACDecFrameInfo frame_info;
  62     NeAACDecHandle decoder;
  63     int err;
  64     uint32_t seek_idx = 0;
  65     uint32_t s = 0;
  66     uint32_t sbr_fac = 1;
  67     unsigned char c = 0;
  68     void *ret;
  69
  70     /* Generic codec initialisation */
  71     ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
  72     ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
  73
  74 next_track:
  75     err = CODEC_OK;
  76
  77     /* Clean and initialize decoder structures */
  78     memset(&demux_res , 0, sizeof(demux_res));
  79     if (codec_init()) {
  80         LOGF("FAAD: Codec init error\n");
  81         err = CODEC_ERROR;
  82         goto exit;
  83     }
  84
  85     if (codec_wait_taginfo() != 0)
  86         goto done;
  87
  88     file_offset = ci->id3->offset;
  89
  90     ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
  91     codec_set_replaygain(ci->id3);
  92
  93     stream_create(&input_stream,ci);
  94
  95     /* if qtmovie_read returns successfully, the stream is up to
  96      * the movie data, which can be used directly by the decoder */
  97     if (!qtmovie_read(&input_stream, &demux_res)) {
  98         LOGF("FAAD: File init error\n");
  99         err = CODEC_ERROR;
 100         goto done;
 101     }
 102
 103     /* initialise the sound converter */
 104     decoder = NeAACDecOpen();
 105
 106     if (!decoder) {
 107         LOGF("FAAD: Decode open error\n");
 108         err = CODEC_ERROR;
 109         goto done;
 110     }
 111
 112     NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
 113     conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
 114     NeAACDecSetConfiguration(decoder, conf);
 115
 116     err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
 117     if (err) {
 118         LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
 119         err = CODEC_ERROR;
 120         goto done;
 121     }
 122
 123     /* Set pointer to be able to use IRAM an to avoid alloc in decoder. Must
 124      * be called after NeAACDecOpen(). */
 125     /* A buffer of framelength or 2*frameLenght size must be allocated for
 126      * time_out. If frameLength is too big or SBR/forceUpSampling is active,
 127      * we do not use the IRAM buffer and keep faad's internal allocation (see
 128      * specrec.c). */
 129     needed_bufsize = decoder->frameLength;
 130 #ifdef SBR_DEC
 131     if ((decoder->sbr_present_flag == 1) || (decoder->forceUpSampling == 1))
 132     {
 133         needed_bufsize *= 2;
 134     }
 135 #endif
 136     if (needed_bufsize <= GB_BUF_SIZE)
 137     {
 138         decoder->time_out[0]    = &gb_time_buffer[0][0];
 139         decoder->time_out[1]    = &gb_time_buffer[1][0];
 140     }
 141     /* A buffer of with frameLength elements must be allocated for fb_intermed.
 142      * If frameLength is too big, we do not use the IRAM buffer and keep faad's
 143      * internal allocation (see specrec.c). */
 144     needed_bufsize = decoder->frameLength;
 145     if (needed_bufsize <= GB_BUF_SIZE)
 146     {
 147         decoder->fb_intermed[0] = &gb_fb_intermed[0][0];
 148         decoder->fb_intermed[1] = &gb_fb_intermed[1][0];
 149     }
 150
 151 #ifdef SBR_DEC
 152     /* Check for need of special handling for seek/resume and elapsed time. */
 153     if (ci->id3->needs_upsampling_correction) {
 154         sbr_fac = 2;
 155     } else {
 156         sbr_fac = 1;
 157     }
 158 #endif
 159
 160     ci->id3->frequency = s;
 161
 162     i = 0;
 163
 164     if (file_offset > 0) {
 165         /* Resume the desired (byte) position. Important: When resuming SBR
 166          * upsampling files the resulting sound_samples_done must be expanded
 167          * by a factor of 2. This is done via using sbr_fac. */
 168         if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
 169                           &sound_samples_done, (int*) &i)) {
 170             sound_samples_done *= sbr_fac;
 171             elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 172             ci->set_elapsed(elapsed_time);
 173         } else {
 174             sound_samples_done = 0;
 175         }
 176         NeAACDecPostSeekReset(decoder, i);
 177     } else {
 178         sound_samples_done = 0;
 179     }
 180
 181     if (i == 0)
 182     {
 183         lead_trim = ci->id3->lead_trim;
 184     }
 185
 186     /* The main decoding loop */
 187     while (i < demux_res.num_sample_byte_sizes) {
 188         ci->yield();
 189
 190         if (ci->stop_codec || ci->new_track) {
 191             break;
 192         }
 193
 194         /* Deal with any pending seek requests */
 195         if (ci->seek_time) {
 196             /* Seek to the desired time position. Important: When seeking in SBR
 197              * upsampling files the seek_time must be divided by 2 when calling
 198              * m4a_seek and the resulting sound_samples_done must be expanded
 199              * by a factor 2. This is done via using sbr_fac. */
 200             if (m4a_seek(&demux_res, &input_stream,
 201                           ((ci->seek_time-1)/10/sbr_fac)*(ci->id3->frequency/100),
 202                           &sound_samples_done, (int*) &i)) {
 203                 sound_samples_done *= sbr_fac;
 204                 elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 205                 ci->set_elapsed(elapsed_time);
 206                 seek_idx = 0;
 207
 208                 if (i == 0)
 209                 {
 210                     lead_trim = ci->id3->lead_trim;
 211                 }
 212             }
 213             NeAACDecPostSeekReset(decoder, i);
 214             ci->seek_complete();
 215         }
 216
 217         /* There can be gaps between chunks, so skip ahead if needed. It
 218          * doesn't seem to happen much, but it probably means that a
 219          * "proper" file can have chunks out of order. Why one would want
 220          * that an good question (but files with gaps do exist, so who
 221          * knows?), so we don't support that - for now, at least.
 222          */
 223         file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);
 224
 225         if (file_offset > ci->curpos)
 226         {
 227             ci->advance_buffer(file_offset - ci->curpos);
 228         }
 229         else if (file_offset == 0)
 230         {
 231             LOGF("AAC: get_sample_offset error\n");
 232             err = CODEC_ERROR;
 233             goto done;
 234         }
 235
 236         /* Request the required number of bytes from the input buffer */
 237         buffer=ci->request_buffer(&n, FAAD_BYTE_BUFFER_SIZE);
 238
 239         /* Decode one block - returned samples will be host-endian */
 240         ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
 241
 242         /* NeAACDecDecode may sometimes return NULL without setting error. */
 243         if (ret == NULL || frame_info.error > 0) {
 244             LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
 245             err = CODEC_ERROR;
 246             goto done;
 247         }
 248
 249         /* Advance codec buffer (no need to call set_offset because of this) */
 250         ci->advance_buffer(frame_info.bytesconsumed);
 251
 252         /* Output the audio */
 253         ci->yield();
 254
 255         /* Gather number of samples for the decoded frame. */
 256         framelength = (frame_info.samples >> 1) - lead_trim;
 257
 258         if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
 259         {
 260             framelength -= ci->id3->tail_trim;
 261         }
 262
 263         if (framelength > 0)
 264         {
 265             ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
 266                               &decoder->time_out[1][lead_trim],
 267                               framelength);
 268         }
 269
 270         if (lead_trim > 0)
 271         {
 272             /* frame_info.samples can be 0 for the first frame */
 273             lead_trim -= (i > 0 || frame_info.samples)
 274                 ? (frame_info.samples >> 1) : (uint32_t)framelength;
 275
 276             if (lead_trim < 0 || ci->id3->lead_trim == 0)
 277             {
 278                 lead_trim = 0;
 279             }
 280         }
 281
 282         /* Update the elapsed-time indicator */
 283         sound_samples_done += framelength;
 284         elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
 285         ci->set_elapsed(elapsed_time);
 286         i++;
 287     }
 288
 289 done:
 290     LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
 291
 292     if (ci->request_next_track())
 293         goto next_track;
 294
 295 exit:
 296     return err;
 297 }