6c8ef19dfd
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30012 a1c6a512-1295-4272-9138-f99709370657
273 lines
8.9 KiB
C
273 lines
8.9 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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/* The maximum buffer size handled by faad. 12 bytes are required by libfaad
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* as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered
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* for each frame. */
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#define FAAD_BYTE_BUFFER_SIZE (2048-12)
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/* this is the codec entry point */
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enum codec_status codec_main(enum codec_entry_call_reason reason)
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{
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if (reason == CODEC_LOAD) {
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/* Generic codec initialisation */
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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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}
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return CODEC_OK;
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}
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/* this is called for each file to process */
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enum codec_status codec_run(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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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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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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NeAACDecFrameInfo frame_info;
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NeAACDecHandle decoder;
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int err;
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uint32_t seek_idx = 0;
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uint32_t s = 0;
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uint32_t sbr_fac = 1;
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unsigned char c = 0;
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void *ret;
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intptr_t param;
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bool empty_first_frame = false;
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/* Clean and initialize decoder structures */
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memset(&demux_res , 0, sizeof(demux_res));
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if (codec_init()) {
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LOGF("FAAD: Codec init error\n");
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return CODEC_ERROR;
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}
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file_offset = 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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ci->seek_buffer(ci->id3->first_frame_offset);
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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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return CODEC_ERROR;
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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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return CODEC_ERROR;
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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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return CODEC_ERROR;
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}
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#ifdef SBR_DEC
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/* Check for need of special handling for seek/resume and elapsed time. */
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if (ci->id3->needs_upsampling_correction) {
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sbr_fac = 2;
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} else {
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sbr_fac = 1;
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}
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#endif
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i = 0;
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if (file_offset > 0) {
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/* Resume the desired (byte) position. Important: When resuming SBR
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* upsampling files the resulting sound_samples_done must be expanded
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* by a factor of 2. This is done via using sbr_fac. */
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if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
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&sound_samples_done, (int*) &i)) {
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sound_samples_done *= sbr_fac;
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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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NeAACDecPostSeekReset(decoder, i);
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} else {
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sound_samples_done = 0;
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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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enum codec_command_action action = ci->get_command(¶m);
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if (action == CODEC_ACTION_HALT)
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break;
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/* Deal with any pending seek requests */
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if (action == CODEC_ACTION_SEEK_TIME) {
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/* Seek to the desired time position. Important: When seeking in SBR
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* upsampling files the seek_time must be divided by 2 when calling
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* m4a_seek and the resulting sound_samples_done must be expanded
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* by a factor 2. This is done via using sbr_fac. */
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if (m4a_seek(&demux_res, &input_stream,
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(param/10/sbr_fac)*(ci->id3->frequency/100),
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&sound_samples_done, (int*) &i)) {
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sound_samples_done *= sbr_fac;
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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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seek_idx = 0;
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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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NeAACDecPostSeekReset(decoder, i);
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ci->seek_complete();
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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 = m4a_check_sample_offset(&demux_res, i, &seek_idx);
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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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return CODEC_ERROR;
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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, FAAD_BYTE_BUFFER_SIZE);
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/* Decode one block - returned samples will be host-endian */
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ret = NeAACDecDecode(decoder, &frame_info, buffer, n);
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/* NeAACDecDecode may sometimes return NULL without setting error. */
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if (ret == NULL || frame_info.error > 0) {
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LOGF("FAAD: decode error '%s'\n", NeAACDecGetErrorMessage(frame_info.error));
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return CODEC_ERROR;
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}
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/* Advance codec buffer (no need to call set_offset because of this) */
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ci->advance_buffer(frame_info.bytesconsumed);
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/* Output the audio */
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ci->yield();
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if (empty_first_frame)
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{
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/* Remove the first frame from lead_trim, under the assumption
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* that it had the same size as this frame
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*/
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empty_first_frame = false;
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lead_trim -= (frame_info.samples >> 1);
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if (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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/* Gather number of samples for the decoded frame. */
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framelength = (frame_info.samples >> 1) - lead_trim;
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if (i == demux_res.num_sample_byte_sizes - 1)
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{
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framelength -= ci->id3->tail_trim;
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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 frame 0. We still want to
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* remove it from lead_trim, so do that during frame 1.
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*/
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if (0 == i && 0 == frame_info.samples)
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{
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empty_first_frame = true;
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}
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lead_trim -= (frame_info.samples >> 1);
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if (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 += framelength;
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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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i++;
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}
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LOGF("AAC: Decoded %lu samples\n", (unsigned long)sound_samples_done);
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return CODEC_OK;
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}
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