rockbox/apps/plugins/mpa2wav.c
Dave Chapman 7b56110e5e Build the codec plugins in the simulator - only tested for X11
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6011 a1c6a512-1295-4272-9138-f99709370657
2005-02-18 16:28:52 +00:00

283 lines
7.2 KiB
C

/***************************************************************************
* __________ __ ___.
* 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"
#if (CONFIG_HWCODEC == MASNONE)
/* software codec platforms */
#include <codecs/libmad/mad.h>
#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
static struct plugin_api* rb;
struct mad_stream Stream;
struct mad_frame Frame;
struct mad_synth Synth;
mad_timer_t Timer;
struct dither d0, d1;
/* The following function is used inside libmad - let's hope it's never
called.
*/
void abort(void) {
}
/* The "dither" code to convert the 24-bit samples produced by libmad was
taken from the coolplayer project - coolplayer.sourceforge.net */
struct dither {
mad_fixed_t error[3];
mad_fixed_t random;
};
# define SAMPLE_DEPTH 16
# define scale(x, y) dither((x), (y))
/*
* NAME: prng()
* DESCRIPTION: 32-bit pseudo-random number generator
*/
static __inline
unsigned long prng(unsigned long state)
{
return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
}
/*
* NAME: dither()
* DESCRIPTION: dither and scale sample
*/
static __inline
signed int dither(mad_fixed_t sample, struct dither *dither)
{
unsigned int scalebits;
mad_fixed_t output, mask, random;
enum {
MIN = -MAD_F_ONE,
MAX = MAD_F_ONE - 1
};
/* noise shape */
sample += dither->error[0] - dither->error[1] + dither->error[2];
dither->error[2] = dither->error[1];
dither->error[1] = dither->error[0] / 2;
/* bias */
output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
mask = (1L << scalebits) - 1;
/* dither */
random = prng(dither->random);
output += (random & mask) - (dither->random & mask);
dither->random = random;
/* clip */
if (output > MAX) {
output = MAX;
if (sample > MAX)
sample = MAX;
}
else if (output < MIN) {
output = MIN;
if (sample < MIN)
sample = MIN;
}
/* quantize */
output &= ~mask;
/* error feedback */
dither->error[0] = sample - output;
/* scale */
return output >> scalebits;
}
#define SHRT_MAX 32767
#define INPUT_BUFFER_SIZE (5*8192)
#define OUTPUT_BUFFER_SIZE 8192 /* Must be an integer multiple of 4. */
unsigned char InputBuffer[INPUT_BUFFER_SIZE+MAD_BUFFER_GUARD];
unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
unsigned char *OutputPtr=OutputBuffer;
unsigned char *GuardPtr=NULL;
const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
/* this is the plugin entry point */
enum plugin_status plugin_start(struct plugin_api* api, void* file)
{
file_info_struct file_info;
int Status=0;
unsigned short Sample;
int i;
size_t ReadSize, Remaining;
unsigned char *ReadStart;
/* Generic plugin inititialisation */
TEST_PLUGIN_API(api);
rb = api;
/* This function sets up the buffers and reads the file into RAM */
if (local_init(file,"/libmadtest.wav",&file_info,api)) {
return PLUGIN_ERROR;
}
/* Create a decoder instance */
mad_stream_init(&Stream);
mad_frame_init(&Frame);
mad_synth_init(&Synth);
mad_timer_reset(&Timer);
//if error: return PLUGIN_ERROR;
file_info.curpos=0;
file_info.start_tick=*(rb->current_tick);
rb->button_clear_queue();
/* This is the decoding loop. */
while (file_info.curpos < file_info.filesize) {
if(Stream.buffer==NULL || Stream.error==MAD_ERROR_BUFLEN) {
if(Stream.next_frame!=NULL) {
Remaining=Stream.bufend-Stream.next_frame;
memmove(InputBuffer,Stream.next_frame,Remaining);
ReadStart=InputBuffer+Remaining;
ReadSize=INPUT_BUFFER_SIZE-Remaining;
} else {
ReadSize=INPUT_BUFFER_SIZE;
ReadStart=InputBuffer;
Remaining=0;
}
/* Fill-in the buffer. If an error occurs print a message
* and leave the decoding loop. If the end of stream is
* reached we also leave the loop but the return status is
* left untouched.
*/
if ((file_info.filesize-file_info.curpos) < (int) ReadSize) {
ReadSize=file_info.filesize-file_info.curpos;
}
memcpy(ReadStart,&filebuf[file_info.curpos],ReadSize);
file_info.curpos+=ReadSize;
if (file_info.curpos >= file_info.filesize)
{
GuardPtr=ReadStart+ReadSize;
memset(GuardPtr,0,MAD_BUFFER_GUARD);
ReadSize+=MAD_BUFFER_GUARD;
}
/* Pipe the new buffer content to libmad's stream decoder facility */
mad_stream_buffer(&Stream,InputBuffer,ReadSize+Remaining);
Stream.error=0;
}
if(mad_frame_decode(&Frame,&Stream))
{
if(MAD_RECOVERABLE(Stream.error))
{
if(Stream.error!=MAD_ERROR_LOSTSYNC || Stream.this_frame!=GuardPtr)
{
rb->splash(HZ*1, true, "Recoverable...!");
}
continue;
}
else
if(Stream.error==MAD_ERROR_BUFLEN)
continue;
else
{
rb->splash(HZ*1, true, "Recoverable...!");
//fprintf(stderr,"%s: unrecoverable frame level error.\n",ProgName);
Status=1;
break;
}
}
/* We assume all frames have same samplerate as the first */
if(file_info.frames_decoded==0) {
file_info.samplerate=Frame.header.samplerate;
}
file_info.frames_decoded++;
/* ?? Do we need the timer module? */
mad_timer_add(&Timer,Frame.header.duration);
/* DAVE: This can be used to attenuate the audio */
// if(DoFilter)
// ApplyFilter(&Frame);
mad_synth_frame(&Synth,&Frame);
/* Convert MAD's numbers to an array of 16-bit LE signed integers */
for(i=0;i<Synth.pcm.length;i++)
{
/* Left channel */
Sample=scale(Synth.pcm.samples[0][i],&d0);
*(OutputPtr++)=Sample&0xff;
*(OutputPtr++)=Sample>>8;
/* Right channel. If the decoded stream is monophonic then
* the right output channel is the same as the left one.
*/
if(MAD_NCHANNELS(&Frame.header)==2)
Sample=scale(Synth.pcm.samples[1][i],&d1);
*(OutputPtr++)=Sample&0xff;
*(OutputPtr++)=Sample>>8;
/* Flush the buffer if it is full. */
if(OutputPtr==OutputBufferEnd)
{
rb->write(file_info.outfile,OutputBuffer,OUTPUT_BUFFER_SIZE);
OutputPtr=OutputBuffer;
}
}
file_info.current_sample+=Synth.pcm.length;
display_status(&file_info);
if (rb->button_get(false)!=BUTTON_NONE) {
close_wav(&file_info);
return PLUGIN_OK;
}
}
close_wav(&file_info);
rb->splash(HZ*2, true, "FINISHED!");
return PLUGIN_OK;
}
#endif /* CONFIG_HWCODEC == MASNONE */