Nyaaori/rockbox
1
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity
rockbox/apps/codecs/aac.c
Nils Wallménius 74cc5c77e3 aac: put two local structs on the stack as they are small and the codec uses little stack anyway ( < 20% on h300) 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@28872 a1c6a512-1295-4272-9138-f99709370657
2010-12-21 13:35:02 +00:00

294 lines
9.5 KiB
C
Raw Blame History

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* 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
/* Global buffers to be used in the mdct synthesis. This way the arrays can
* be moved to IRAM for some targets */
#define GB_BUF_SIZE 1024
static ALIGN real_t gb_time_buffer[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM;
static ALIGN real_t gb_fb_intermed[2][GB_BUF_SIZE] IBSS_ATTR_FAAD_LARGE_IRAM;
/* 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;
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;
int framelength;
int lead_trim = 0;
int needed_bufsize;
unsigned int i;
unsigned char* buffer;
NeAACDecFrameInfo frame_info;
NeAACDecHandle decoder;
int err;
uint32_t s = 0;
unsigned char c = 0;
void *ret;
/* Generic codec initialisation */
ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
next_track:
err = CODEC_OK;
/* Clean and initialize decoder structures */
memset(&demux_res , 0, sizeof(demux_res));
if (codec_init()) {
LOGF("FAAD: Codec init error\n");
err = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
file_offset = 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;
}
/* Set pointer to be able to use IRAM an to avoid alloc in decoder. Must
* be called after NeAACDecOpen(). */
/* A buffer of framelength or 2*frameLenght size must be allocated for
* time_out. If frameLength is too big or SBR/forceUpSampling is active,
* we do not use the IRAM buffer and keep faad's internal allocation (see
* specrec.c). */
needed_bufsize = decoder->frameLength;
#ifdef SBR_DEC
if ((decoder->sbr_present_flag == 1) || (decoder->forceUpSampling == 1))
{
needed_bufsize *= 2;
}
#endif
if (needed_bufsize <= GB_BUF_SIZE)
{
decoder->time_out[0] = &gb_time_buffer[0][0];
decoder->time_out[1] = &gb_time_buffer[1][0];
}
/* A buffer of with frameLength elements must be allocated for fb_intermed.
* If frameLength is too big, we do not use the IRAM buffer and keep faad's
* internal allocation (see specrec.c). */
needed_bufsize = decoder->frameLength;
if (needed_bufsize <= GB_BUF_SIZE)
{
decoder->fb_intermed[0] = &gb_fb_intermed[0][0];
decoder->fb_intermed[1] = &gb_fb_intermed[1][0];
}
ci->id3->frequency = s;
i = 0;
if (file_offset > 0) {
if (alac_seek_raw(&demux_res, &input_stream, file_offset,
&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;
}
} else {
sound_samples_done = 0;
}
if (i == 0)
{
lead_trim = ci->id3->lead_trim;
}
/* 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);
if (i == 0)
{
lead_trim = ci->id3->lead_trim;
}
}
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 */
ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
/* NeAACDecDecode may sometimes return NULL without setting error. */
if (ret == NULL || frame_info.error > 0) {
LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
err = CODEC_ERROR;
goto done;
}
/* Advance codec buffer (no need to call set_offset because of this) */
ci->advance_buffer(n);
/* Output the audio */
ci->yield();
framelength = (frame_info.samples >> 1) - lead_trim;
if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
{
/* Currently limited to at most one frame of tail_trim.
* Seems to be enough.
*/
if (ci->id3->tail_trim == 0
&& sample_duration < (frame_info.samples >> 1))
{
/* Subtract lead_trim just in case we decode a file with
* only one audio frame with actual data.
*/
framelength = sample_duration - lead_trim;
}
else
{
framelength -= ci->id3->tail_trim;
}
}
if (framelength > 0)
{
ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
&decoder->time_out[1][lead_trim],
framelength);
}
if (lead_trim > 0)
{
/* frame_info.samples can be 0 for the first frame */
lead_trim -= (i > 0 || frame_info.samples)
? (frame_info.samples >> 1) : sample_duration;
if (lead_trim < 0 || ci->id3->lead_trim == 0)
{
lead_trim = 0;
}
}
/* 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);
i++;
}
err = CODEC_OK;
done:
LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
if (ci->request_next_track())
goto next_track;
exit:
return err;
}
Reference in a new issue View git blame Copy permalink