4909e09267
Some of these were found with http://www.samba.org/junkcode/#findstatic. Changes of note: * The old MDCT has been removed. * Makefile.test files that create test programs for libatrac, libcook, and libffmpegFLAC have been removed, as they don't work. My project will have a replacement that works with all codecs. * I've tried not to remove anything useful. CLIP_TO_15 was removed from libtremor because there's another copy (also commented) in codeclib. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29945 a1c6a512-1295-4272-9138-f99709370657
144 lines
4.7 KiB
C
144 lines
4.7 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 "libalac/decomp.h"
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CODEC_HEADER
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/* The maximum buffer size handled. This amount of bytes is buffered for each
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* frame. */
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#define ALAC_BYTE_BUFFER_SIZE 32768
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static int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE] IBSS_ATTR;
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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, ALAC_OUTPUT_DEPTH-1);
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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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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 samplesdone;
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uint32_t elapsedtime = 0;
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int samplesdecoded;
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unsigned int i;
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unsigned char* buffer;
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alac_file alac;
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intptr_t param;
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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("ALAC: Error initialising codec\n");
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return CODEC_ERROR;
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}
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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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ci->seek_buffer(0);
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stream_create(&input_stream,ci);
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/* Read from ci->id3->offset before calling qtmovie_read. */
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samplesdone = (uint32_t)(((uint64_t)(ci->id3->offset) * ci->id3->frequency) /
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(ci->id3->bitrate*128));
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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("ALAC: Error initialising file\n");
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return CODEC_ERROR;
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}
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/* initialise the sound converter */
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alac_set_info(&alac, demux_res.codecdata);
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/* Set i for first frame, seek to desired sample position for resuming. */
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i=0;
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if (samplesdone > 0) {
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if (m4a_seek(&demux_res, &input_stream, samplesdone,
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&samplesdone, (int*) &i)) {
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elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
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ci->set_elapsed(elapsedtime);
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} else {
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samplesdone = 0;
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}
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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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/* Request the required number of bytes from the input buffer */
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buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
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/* Deal with any pending seek requests */
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if (action == CODEC_ACTION_SEEK_TIME) {
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if (m4a_seek(&demux_res, &input_stream,
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(param/10) * (ci->id3->frequency/100),
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&samplesdone, (int *)&i)) {
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elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
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}
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ci->set_elapsed(elapsedtime);
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ci->seek_complete();
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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, ALAC_BYTE_BUFFER_SIZE);
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/* Decode one block - returned samples will be host-endian */
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samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, ci->yield);
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ci->yield();
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/* Advance codec buffer by amount of consumed bytes */
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ci->advance_buffer(alac.bytes_consumed);
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/* Output the audio */
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ci->pcmbuf_insert(outputbuffer[0], outputbuffer[1], samplesdecoded);
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/* Update the elapsed-time indicator */
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samplesdone+=samplesdecoded;
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elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
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ci->set_elapsed(elapsedtime);
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i++;
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}
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LOGF("ALAC: Decoded %lu samples\n",(unsigned long)samplesdone);
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return CODEC_OK;
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}
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