d8cb703b1e
codecs (currently works corrently only with mp3's, somebody should fix that). git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6877 a1c6a512-1295-4272-9138-f99709370657
151 lines
4.1 KiB
C
151 lines
4.1 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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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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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 "codec.h"
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#include "playback.h"
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#include "lib/codeclib.h"
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#include "dsp.h"
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#define BYTESWAP(x) (((x>>8) & 0xff) | ((x<<8) & 0xff00))
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/* Number of bytes to process in one iteration */
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#define WAV_CHUNK_SIZE (1024*4)
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#ifndef SIMULATOR
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extern char iramcopy[];
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extern char iramstart[];
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extern char iramend[];
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#endif
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/* this is the codec entry point */
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enum codec_status codec_start(struct codec_api* api)
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{
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struct codec_api* rb = api;
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struct codec_api* ci = api;
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unsigned long samplerate,numbytes,totalsamples,samplesdone,nsamples;
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int channels,bytespersample,bitspersample;
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unsigned int i;
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size_t n;
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int endofstream;
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unsigned char* header;
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unsigned short* wavbuf;
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/* Generic codec initialisation */
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TEST_CODEC_API(api);
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/* if you are using a global api pointer, don't forget to copy it!
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otherwise you will get lovely "I04: IllInstr" errors... :-) */
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rb = api;
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#ifndef SIMULATOR
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rb->memcpy(iramstart, iramcopy, iramend-iramstart);
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#endif
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ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*10));
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ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
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ci->configure(DSP_DITHER, (bool *)false);
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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next_track:
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if (codec_init(api)) {
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return CODEC_ERROR;
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}
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while (!rb->taginfo_ready)
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rb->yield();
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if (rb->id3->frequency != NATIVE_FREQUENCY) {
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rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
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rb->configure(CODEC_DSP_ENABLE, (bool *)true);
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} else {
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rb->configure(CODEC_DSP_ENABLE, (bool *)false);
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}
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/* FIX: Correctly parse WAV header - we assume canonical 44-byte header */
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header=ci->request_buffer(&n,44);
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if (n!=44) {
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return CODEC_ERROR;
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}
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if ((memcmp(header,"RIFF",4)!=0) || (memcmp(&header[8],"WAVEfmt",7)!=0)) {
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return CODEC_ERROR;
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}
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samplerate=header[24]|(header[25]<<8)|(header[26]<<16)|(header[27]<<24);
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bitspersample=header[34];
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channels=header[22];
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bytespersample=((bitspersample/8)*channels);
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numbytes=(header[40]|(header[41]<<8)|(header[42]<<16)|(header[43]<<24));
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totalsamples=numbytes/bytespersample;
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if ((bitspersample!=16) || (channels != 2)) {
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return CODEC_ERROR;
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}
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ci->advance_buffer(44);
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/* The main decoder loop */
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samplesdone=0;
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ci->set_elapsed(0);
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endofstream=0;
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while (!endofstream) {
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if (ci->stop_codec || ci->reload_codec) {
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break;
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}
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wavbuf=ci->request_buffer(&n,WAV_CHUNK_SIZE);
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if (n==0) break; /* End of stream */
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nsamples=(n/bytespersample);
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/* WAV files can contain extra data at the end - so we can't just
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process until the end of the file */
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if (samplesdone+nsamples > totalsamples) {
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nsamples=(totalsamples-samplesdone);
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n=nsamples*bytespersample;
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endofstream=1;
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}
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/* Byte-swap data */
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for (i=0;i<n/2;i++) {
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wavbuf[i]=SWAB16(wavbuf[i]);
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}
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samplesdone+=nsamples;
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ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
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rb->yield();
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while (!ci->audiobuffer_insert((unsigned char*)wavbuf, n))
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rb->yield();
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ci->advance_buffer(n);
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}
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if (ci->request_next_track())
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goto next_track;
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return CODEC_OK;
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}
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