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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "codeclib.h"
#include "libm4a/m4a.h"
#include "libfaad/common.h"
#include "libfaad/structs.h"
#include "libfaad/decoder.h"
CODEC_HEADER
/* this is the codec entry point */
enum codec_status codec_main(void)
{
/* Note that when dealing with QuickTime/MPEG4 files, terminology is
* a bit confusing. Files with sound are split up in chunks, where
* each chunk contains one or more samples. Each sample in turn
* contains a number of "sound samples" (the kind you refer to with
* the sampling frequency).
*/
size_t n;
static demux_res_t demux_res;
stream_t input_stream;
uint32_t sound_samples_done;
uint32_t elapsed_time;
uint32_t sample_duration;
uint32_t sample_byte_size;
int file_offset;
unsigned int i;
unsigned char* buffer;
static NeAACDecFrameInfo frame_info;
NeAACDecHandle decoder;
int err;
uint32_t s = 0;
unsigned char c = 0;
/* Generic codec initialisation */
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, 1024*16);
ci->configure(CODEC_SET_FILEBUF_WATERMARK, 1024*512);
ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
next_track:
err = CODEC_OK;
if (codec_init()) {
LOGF("FAAD: Codec init error\n");
err = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
sound_samples_done = ci->id3->offset;
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
stream_create(&input_stream,ci);
/* if qtmovie_read returns successfully, the stream is up to
* the movie data, which can be used directly by the decoder */
if (!qtmovie_read(&input_stream, &demux_res)) {
LOGF("FAAD: File init error\n");
err = CODEC_ERROR;
goto done;
}
/* initialise the sound converter */
decoder = NeAACDecOpen();
if (!decoder) {
LOGF("FAAD: Decode open error\n");
err = CODEC_ERROR;
goto done;
}
NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
NeAACDecSetConfiguration(decoder, conf);
err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
if (err) {
LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
err = CODEC_ERROR;
goto done;
}
ci->id3->frequency = s;
i = 0;
if (sound_samples_done > 0) {
if (alac_seek_raw(&demux_res, &input_stream, sound_samples_done,
&sound_samples_done, (int*) &i)) {
elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsed_time);
} else {
sound_samples_done = 0;
}
}
/* The main decoding loop */
while (i < demux_res.num_sample_byte_sizes) {
ci->yield();
if (ci->stop_codec || ci->new_track) {
break;
}
/* Deal with any pending seek requests */
if (ci->seek_time) {
if (alac_seek(&demux_res, &input_stream,
((ci->seek_time-1)/10)*(ci->id3->frequency/100),
&sound_samples_done, (int*) &i)) {
elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsed_time);
}
ci->seek_complete();
}
/* Lookup the length (in samples and bytes) of block i */
if (!get_sample_info(&demux_res, i, &sample_duration,
&sample_byte_size)) {
LOGF("AAC: get_sample_info error\n");
err = CODEC_ERROR;
goto done;
}
/* There can be gaps between chunks, so skip ahead if needed. It
* doesn't seem to happen much, but it probably means that a
* "proper" file can have chunks out of order. Why one would want
* that an good question (but files with gaps do exist, so who
* knows?), so we don't support that - for now, at least.
*/
file_offset = get_sample_offset(&demux_res, i);
if (file_offset > ci->curpos)
{
ci->advance_buffer(file_offset - ci->curpos);
}
else if (file_offset == 0)
{
LOGF("AAC: get_sample_offset error\n");
err = CODEC_ERROR;
goto done;
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n,sample_byte_size);
/* Decode one block - returned samples will be host-endian */
NeAACDecDecode(decoder, &frame_info, buffer, n);
/* Ignore return value, we access samples in the decoder struct
* directly.
*/
if (frame_info.error > 0) {
LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
err = CODEC_ERROR;
goto done;
}
/* Advance codec buffer */
ci->advance_buffer(n);
/* Output the audio */
ci->yield();
ci->pcmbuf_insert(decoder->time_out[0], decoder->time_out[1],
frame_info.samples >> 1);
/* Update the elapsed-time indicator */
sound_samples_done += sample_duration;
elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsed_time);
/* Keep track of current position - for resuming */
ci->set_offset(elapsed_time);
i++;
}
err = CODEC_OK;
done:
LOGF("AAC: Decoded %lu samples\n", sound_samples_done);
if (ci->request_next_track())
goto next_track;
exit:
return err;
}
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