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Add first version of WAV playback to iRiver - only 16-bit Stereo 44.1KHz files supported.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6654 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Dave Chapman 2005-06-10 18:08:08 +00:00
parent d30f1100ec
commit 3c2c2f59f4
2 changed files with 183 additions and 1 deletions
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53
apps/playback.c
View file

@ -598,7 +598,7 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
int rc, i;
int copy_n;
/* Used by the FLAC metadata parser */
unsigned long totalsamples;
unsigned long totalsamples,bytespersample,channels,bitspersample,numbytes;
unsigned char* buf;
if (track_count >= MAX_TRACK)
@ -677,6 +677,57 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
tracks[track_widx].taginfo_ready = true;
break ;
case AFMT_PCM_WAV:
/* Use the trackname part of the id3 structure as a temporary buffer */
buf=tracks[track_widx].id3.path;
lseek(fd, 0, SEEK_SET);
rc = read(fd, buf, 44);
if (rc < 44) {
close(fd);
return false;
}
if ((memcmp(buf,"RIFF",4)!=0) ||
(memcmp(&buf[8],"WAVEfmt",7)!=0)) {
logf("%s is not a WAV file\n",trackname);
close(fd);
return(false);
}
/* FIX: Correctly parse WAV header - we assume canonical
44-byte header */
bitspersample=buf[34];
channels=buf[22];
if ((bitspersample!=16) || (channels != 2)) {
logf("Unsupported WAV file - %d bitspersample, %d channels\n",
bitspersample,channels);
close(fd);
return(false);
}
bytespersample=((bitspersample/8)*channels);
numbytes=(buf[40]|(buf[41]<<8)|(buf[42]<<16)|(buf[43]<<24));
totalsamples=numbytes/bytespersample;
tracks[track_widx].id3.vbr=false; /* All WAV files are CBR */
tracks[track_widx].id3.filesize=filesize(fd);
tracks[track_widx].id3.frequency=buf[24]|(buf[25]<<8)|(buf[26]<<16)|(buf[27]<<24);
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
tracks[track_widx].id3.length=(totalsamples/tracks[track_widx].id3.frequency)*1000;
tracks[track_widx].id3.bitrate=(tracks[track_widx].id3.frequency*bytespersample)/1024;
lseek(fd, 0, SEEK_SET);
strncpy(tracks[track_widx].id3.path,trackname,sizeof(tracks[track_widx].id3.path));
tracks[track_widx].taginfo_ready = true;
break;
case AFMT_FLAC:
/* A simple parser to read vital metadata from a FLAC file - length, frequency, bitrate etc. */
/* This code should either be moved to a seperate file, or discarded in favour of the libFLAC code */

131
apps/plugins/codecwav.c Normal file
View file

@ -0,0 +1,131 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "plugin.h"
#include "playback.h"
#include "lib/codeclib.h"
#ifndef SIMULATOR
extern char iramcopy[];
extern char iramstart[];
extern char iramend[];
#endif
/* This is probably a waste of IRAM, but why not? */
static unsigned char wavbuf[16384] IDATA_ATTR;
/* this is the plugin entry point */
enum plugin_status plugin_start(struct plugin_api* api, void* parm)
{
struct plugin_api* rb = (struct plugin_api*)api;
struct codec_api* ci = (struct codec_api*)parm;
unsigned long samplerate,numbytes,totalsamples,samplesdone,nsamples;
int channels,bytespersample,bitspersample;
unsigned int i,j,n;
int endofstream;
/* Generic plugin initialisation */
TEST_PLUGIN_API(api);
/* if you are using a global api pointer, don't forget to copy it!
otherwise you will get lovely "I04: IllInstr" errors... :-) */
rb = api;
#ifndef SIMULATOR
rb->memcpy(iramstart, iramcopy, iramend-iramstart);
#endif
ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*10));
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
next_track:
if (codec_init(api, ci)) {
return PLUGIN_ERROR;
}
/* FIX: Correctly parse WAV header - we assume canonical 44-byte header */
n=(unsigned)(ci->read_filebuf(wavbuf,44));
if (n!=44) {
return PLUGIN_ERROR;
}
if ((memcmp(wavbuf,"RIFF",4)!=0) || (memcmp(&wavbuf[8],"WAVEfmt",7)!=0)) {
return PLUGIN_ERROR;
}
samplerate=wavbuf[24]|(wavbuf[25]<<8)|(wavbuf[26]<<16)|(wavbuf[27]<<24);
bitspersample=wavbuf[34];
channels=wavbuf[22];
bytespersample=((bitspersample/8)*channels);
numbytes=(wavbuf[40]|(wavbuf[41]<<8)|(wavbuf[42]<<16)|(wavbuf[43]<<24));
totalsamples=numbytes/bytespersample;
if ((bitspersample!=16) || (channels != 2)) {
return PLUGIN_ERROR;
}
/* The main decoder loop */
samplesdone=0;
ci->set_elapsed(0);
endofstream=0;
while (!endofstream) {
rb->yield();
if (ci->stop_codec || ci->reload_codec) {
break;
}
n=(unsigned)(ci->read_filebuf(wavbuf,sizeof(wavbuf)));
rb->yield();
if (n==0) break; /* End of stream */
nsamples=(n/bytespersample);
/* WAV files can contain extra data at the end - so we can't just
process until the end of the file */
if (samplesdone+nsamples > totalsamples) {
nsamples=(totalsamples-samplesdone);
n=nsamples*bytespersample;
endofstream=1;
}
/* Byte-swap data */
for (i=0;i<n;i+=2) {
j=wavbuf[i];
wavbuf[i]=wavbuf[i+1];
wavbuf[i+1]=j;
}
samplesdone+=nsamples;
ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
rb->yield();
while (!ci->audiobuffer_insert(wavbuf, n))
rb->yield();
}
if (ci->request_next_track())
goto next_track;
return PLUGIN_OK;
}