2acc0ac542
later. We still need to hunt down snippets used that are not. 1324 modified files... http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
261 lines
8 KiB
C
261 lines
8 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2005 Dave Chapman
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "codeclib.h"
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#include "libm4a/m4a.h"
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#include "libfaad/common.h"
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#include "libfaad/structs.h"
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#include "libfaad/decoder.h"
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CODEC_HEADER
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/* this is the codec entry point */
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enum codec_status codec_main(void)
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{
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/* Note that when dealing with QuickTime/MPEG4 files, terminology is
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* a bit confusing. Files with sound are split up in chunks, where
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* each chunk contains one or more samples. Each sample in turn
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* contains a number of "sound samples" (the kind you refer to with
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* the sampling frequency).
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*/
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size_t n;
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static demux_res_t demux_res;
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stream_t input_stream;
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uint32_t sound_samples_done;
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uint32_t elapsed_time;
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uint32_t sample_duration;
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uint32_t sample_byte_size;
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int file_offset;
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int framelength;
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int lead_trim = 0;
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unsigned int i;
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unsigned char* buffer;
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static NeAACDecFrameInfo frame_info;
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NeAACDecHandle decoder;
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int err;
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uint32_t s = 0;
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unsigned char c = 0;
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/* Generic codec initialisation */
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ci->configure(CODEC_SET_FILEBUF_WATERMARK, 1024*512);
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ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
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ci->configure(DSP_SET_SAMPLE_DEPTH, 29);
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next_track:
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err = CODEC_OK;
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if (codec_init()) {
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LOGF("FAAD: Codec init error\n");
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err = CODEC_ERROR;
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goto exit;
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}
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while (!*ci->taginfo_ready && !ci->stop_codec)
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ci->sleep(1);
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sound_samples_done = ci->id3->offset;
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ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
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codec_set_replaygain(ci->id3);
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stream_create(&input_stream,ci);
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/* if qtmovie_read returns successfully, the stream is up to
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* the movie data, which can be used directly by the decoder */
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if (!qtmovie_read(&input_stream, &demux_res)) {
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LOGF("FAAD: File init error\n");
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err = CODEC_ERROR;
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goto done;
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}
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/* initialise the sound converter */
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decoder = NeAACDecOpen();
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if (!decoder) {
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LOGF("FAAD: Decode open error\n");
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err = CODEC_ERROR;
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goto done;
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}
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NeAACDecConfigurationPtr conf = NeAACDecGetCurrentConfiguration(decoder);
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conf->outputFormat = FAAD_FMT_24BIT; /* irrelevant, we don't convert */
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NeAACDecSetConfiguration(decoder, conf);
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err = NeAACDecInit2(decoder, demux_res.codecdata, demux_res.codecdata_len, &s, &c);
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if (err) {
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LOGF("FAAD: DecInit: %d, %d\n", err, decoder->object_type);
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err = CODEC_ERROR;
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goto done;
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}
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ci->id3->frequency = s;
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i = 0;
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if (sound_samples_done > 0) {
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if (alac_seek_raw(&demux_res, &input_stream, sound_samples_done,
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&sound_samples_done, (int*) &i)) {
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elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
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ci->set_elapsed(elapsed_time);
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} else {
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sound_samples_done = 0;
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}
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}
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if (i == 0)
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{
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lead_trim = ci->id3->lead_trim;
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}
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/* The main decoding loop */
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while (i < demux_res.num_sample_byte_sizes) {
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ci->yield();
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if (ci->stop_codec || ci->new_track) {
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break;
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}
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/* Deal with any pending seek requests */
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if (ci->seek_time) {
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if (alac_seek(&demux_res, &input_stream,
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((ci->seek_time-1)/10)*(ci->id3->frequency/100),
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&sound_samples_done, (int*) &i)) {
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elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
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ci->set_elapsed(elapsed_time);
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if (i == 0)
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{
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lead_trim = ci->id3->lead_trim;
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}
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}
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ci->seek_complete();
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}
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/* Lookup the length (in samples and bytes) of block i */
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if (!get_sample_info(&demux_res, i, &sample_duration,
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&sample_byte_size)) {
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LOGF("AAC: get_sample_info error\n");
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err = CODEC_ERROR;
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goto done;
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}
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/* There can be gaps between chunks, so skip ahead if needed. It
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* doesn't seem to happen much, but it probably means that a
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* "proper" file can have chunks out of order. Why one would want
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* that an good question (but files with gaps do exist, so who
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* knows?), so we don't support that - for now, at least.
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*/
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file_offset = get_sample_offset(&demux_res, i);
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if (file_offset > ci->curpos)
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{
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ci->advance_buffer(file_offset - ci->curpos);
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}
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else if (file_offset == 0)
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{
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LOGF("AAC: get_sample_offset error\n");
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err = CODEC_ERROR;
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goto done;
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}
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/* Request the required number of bytes from the input buffer */
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buffer=ci->request_buffer(&n,sample_byte_size);
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/* Decode one block - returned samples will be host-endian */
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NeAACDecDecode(decoder, &frame_info, buffer, n);
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/* Ignore return value, we access samples in the decoder struct
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* directly.
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*/
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if (frame_info.error > 0) {
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LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
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err = CODEC_ERROR;
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goto done;
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}
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/* Advance codec buffer */
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ci->advance_buffer(n);
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/* Output the audio */
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ci->yield();
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framelength = (frame_info.samples >> 1) - lead_trim;
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if (i == demux_res.num_sample_byte_sizes - 1 && framelength > 0)
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{
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/* Currently limited to at most one frame of tail_trim.
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* Seems to be enough.
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*/
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if (ci->id3->tail_trim == 0
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&& sample_duration < (frame_info.samples >> 1))
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{
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/* Subtract lead_trim just in case we decode a file with
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* only one audio frame with actual data.
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*/
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framelength = sample_duration - lead_trim;
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}
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else
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{
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framelength -= ci->id3->tail_trim;
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}
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}
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if (framelength > 0)
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{
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ci->pcmbuf_insert(&decoder->time_out[0][lead_trim],
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&decoder->time_out[1][lead_trim],
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framelength);
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}
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if (lead_trim > 0)
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{
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/* frame_info.samples can be 0 for the first frame */
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lead_trim -= (i > 0 || frame_info.samples)
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? (frame_info.samples >> 1) : sample_duration;
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if (lead_trim < 0 || ci->id3->lead_trim == 0)
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{
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lead_trim = 0;
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}
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}
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/* Update the elapsed-time indicator */
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sound_samples_done += sample_duration;
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elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
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ci->set_elapsed(elapsed_time);
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/* Keep track of current position - for resuming */
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ci->set_offset(elapsed_time);
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i++;
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}
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err = CODEC_OK;
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done:
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LOGF("AAC: Decoded %lu samples\n", sound_samples_done);
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if (ci->request_next_track())
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goto next_track;
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exit:
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return err;
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}
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