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rockbox/apps/codecs/mpa.c
Thom Johansen 10426dbe22 Cleaned up code a bit, removed all rb references. 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6825 a1c6a512-1295-4272-9138-f99709370657
2005-06-22 21:18:05 +00:00

508 lines
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C
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/***************************************************************************
* __________ __ ___.
* 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 "codec.h"
#include <codecs/libmad/mad.h>
#include "playback.h"
#include "mp3data.h"
#include "lib/codeclib.h"
struct mad_stream Stream IDATA_ATTR;
struct mad_frame Frame IDATA_ATTR;
struct mad_synth Synth IDATA_ATTR;
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
*/
inline 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;
}
inline int detect_silence(mad_fixed_t sample)
{
unsigned int scalebits;
mad_fixed_t output, mask;
enum {
MIN = -MAD_F_ONE,
MAX = MAD_F_ONE - 1
};
/* bias */
output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
mask = (1L << scalebits) - 1;
/* 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;
/* scale */
output >>= scalebits + 4;
if (output == 0x00 || output == 0xff)
return 1;
return 0;
}
#define INPUT_CHUNK_SIZE 8192
#define OUTPUT_BUFFER_SIZE 65536 /* Must be an integer multiple of 4. */
unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
unsigned char *OutputPtr;
unsigned char *GuardPtr = NULL;
const unsigned char *OutputBufferEnd = OutputBuffer + OUTPUT_BUFFER_SIZE;
long resampled_data[2][5000]; /* enough to cope with 11khz upsampling */
mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR;
/* TODO: what latency does layer 1 have? */
int mpeg_latency[3] = { 0, 481, 529 };
#ifdef USE_IRAM
extern char iramcopy[];
extern char iramstart[];
extern char iramend[];
#endif
#undef DEBUG_GAPLESS
struct resampler {
long last_sample, phase, delta;
};
#if CONFIG_CPU==MCF5249 && !defined(SIMULATOR)
#define INIT() asm volatile ("move.l #0xb0, %macsr") /* frac, round, clip */
#define FRACMUL(x, y) \
({ \
long t; \
asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \
"movclr.l %%acc0, %[t]\n\t" \
: [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \
t; \
})
#else
#define INIT()
#define FRACMUL(x, y) (long)(((long long)(x)*(long long)(y)) << 1)
#endif
/* linear resampling, introduces one sample delay, because of our inability to
look into the future at the end of a frame */
long downsample(long *in, long *out, int num, struct resampler *s)
{
long i = 1, pos;
long last = s->last_sample;
INIT();
pos = s->phase >> 16;
/* check if we need last sample of previous frame for interpolation */
if (pos > 0)
last = in[pos - 1];
out[0] = last + FRACMUL((s->phase & 0xffff) << 15, in[pos] - last);
s->phase += s->delta;
while ((pos = s->phase >> 16) < num) {
out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
s->phase += s->delta;
}
/* wrap phase accumulator back to start of next frame */
s->phase -= num << 16;
s->last_sample = in[num - 1];
return i;
}
long upsample(long *in, long *out, int num, struct resampler *s)
{
long i = 0, pos;
INIT();
while ((pos = s->phase >> 16) == 0) {
out[i++] = s->last_sample + FRACMUL((s->phase & 0xffff) << 15, in[pos] - s->last_sample);
s->phase += s->delta;
}
while ((pos = s->phase >> 16) < num) {
out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
s->phase += s->delta;
}
/* wrap phase accumulator back to start of next frame */
s->phase -= num << 16;
s->last_sample = in[num - 1];
return i;
}
long resample(long *in, long *out, int num, struct resampler *s)
{
if (s->delta >= (1 << 16))
return downsample(in, out, num, s);
else
return upsample(in, out, num, s);
}
/* this is the codec entry point */
enum codec_status codec_start(struct codec_api* api)
{
struct codec_api *ci = api;
struct mp3info *info;
int Status = 0;
size_t size;
int file_end;
unsigned short Sample;
char *InputBuffer;
unsigned int samplecount;
unsigned int samplesdone;
bool first_frame;
#ifdef DEBUG_GAPLESS
bool first = true;
int fd;
#endif
int i;
int yieldcounter = 0;
int stop_skip, start_skip;
struct resampler lr = { 0, 0, 0 }, rr = { 0, 0, 0 };
long length;
/* Generic codec inititialisation */
TEST_CODEC_API(api);
#ifdef USE_IRAM
ci->memcpy(iramstart, iramcopy, iramend - iramstart);
#endif
/* This function sets up the buffers and reads the file into RAM */
if (codec_init(api)) {
return CODEC_ERROR;
}
/* Create a decoder instance */
ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
ci->memset(&Stream, 0, sizeof(struct mad_stream));
ci->memset(&Frame, 0, sizeof(struct mad_frame));
ci->memset(&Synth, 0, sizeof(struct mad_synth));
ci->memset(&Timer, 0, sizeof(mad_timer_t));
mad_stream_init(&Stream);
mad_frame_init(&Frame);
mad_synth_init(&Synth);
mad_timer_reset(&Timer);
/* We do this so libmad doesn't try to call codec_calloc() */
memset(mad_frame_overlap, 0, sizeof(mad_frame_overlap));
Frame.overlap = &mad_frame_overlap;
Stream.main_data = &mad_main_data;
/* This label might need to be moved above all the init code, but I don't
think reiniting the codec is necessary for MPEG. It might even be unwanted
for gapless playback */
next_track:
#ifdef DEBUG_GAPLESS
if (first)
fd = ci->open("/first.pcm", O_WRONLY | O_CREAT);
else
fd = ci->open("/second.pcm", O_WRONLY | O_CREAT);
first = false;
#endif
info = ci->mp3data;
first_frame = false;
file_end = 0;
OutputPtr = OutputBuffer;
while (!*ci->taginfo_ready)
ci->yield();
ci->request_buffer(&size, ci->id3->first_frame_offset);
ci->advance_buffer(size);
if (info->enc_delay >= 0 && info->enc_padding >= 0) {
stop_skip = info->enc_padding - mpeg_latency[info->layer];
if (stop_skip < 0) stop_skip = 0;
start_skip = info->enc_delay + mpeg_latency[info->layer];
} else {
stop_skip = 0;
/* We want to skip this amount anyway */
start_skip = mpeg_latency[info->layer];
}
/* NOTE: currently this doesn't work, the below calculated samples_count
seems to be right, but sometimes libmad just can't supply us with
all the data we need... */
if (info->frame_count) {
/* TODO: 1152 is the frame size in samples for MPEG1 layer 2 and layer 3,
it's probably not correct at all for MPEG2 and layer 1 */
samplecount = info->frame_count*1152 - (start_skip + stop_skip);
samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
} else {
samplecount = ci->id3->length * (ci->id3->frequency / 100) / 10;
samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
}
/* rb->snprintf(buf2, sizeof(buf2), "sc: %d", samplecount);
rb->splash(0, true, buf2);
rb->snprintf(buf2, sizeof(buf2), "length: %d", ci->id3->length);
rb->splash(HZ*5, true, buf2);
rb->snprintf(buf2, sizeof(buf2), "frequency: %d", ci->id3->frequency);
rb->splash(HZ*5, true, buf2); */
lr.delta = rr.delta = ci->id3->frequency*65536/44100;
/* This is the decoding loop. */
while (1) {
ci->yield();
if (ci->stop_codec || ci->reload_codec) {
break ;
}
if (ci->seek_time) {
unsigned int sample_loc;
int newpos;
sample_loc = ci->seek_time/1000 * ci->id3->frequency;
newpos = ci->mp3_get_filepos(ci->seek_time-1);
if (ci->seek_buffer(newpos)) {
if (sample_loc >= samplecount + samplesdone)
break ;
samplecount += samplesdone - sample_loc;
samplesdone = sample_loc;
}
ci->seek_time = 0;
}
/* Lock buffers */
if (Stream.error == 0) {
InputBuffer = ci->request_buffer(&size, INPUT_CHUNK_SIZE);
if (size == 0 || InputBuffer == NULL)
break ;
mad_stream_buffer(&Stream, InputBuffer, size);
}
//if ((int)ci->curpos >= ci->id3->first_frame_offset)
//first_frame = true;
if(mad_frame_decode(&Frame,&Stream))
{
if (Stream.error == MAD_FLAG_INCOMPLETE || Stream.error == MAD_ERROR_BUFLEN) {
// ci->splash(HZ*1, true, "Incomplete");
/* This makes the codec to support partially corrupted files too. */
if (file_end == 30)
break ;
/* Fill the buffer */
Stream.error = 0;
file_end++;
continue ;
}
else 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) {
//rb->splash(HZ*1, true, "Buflen error");
break ;
} else {
//rb->splash(HZ*1, true, "Unrecoverable error");
Status=1;
break;
}
}
if (Stream.next_frame)
ci->advance_buffer_loc((void *)Stream.next_frame);
file_end = false;
/* ?? Do we need the timer module? */
// mad_timer_add(&Timer,Frame.header.duration);
mad_synth_frame(&Synth,&Frame);
//if (!first_frame) {
//samplecount -= Synth.pcm.length;
//continue ;
//}
/* Convert MAD's numbers to an array of 16-bit LE signed integers */
/* We skip start_skip number of samples here, this should only happen for
very first frame in the stream. */
/* TODO: possible for start_skip to exceed one frames worth of samples? */
length = resample((long *)&Synth.pcm.samples[0][start_skip], resampled_data[0], Synth.pcm.length, &lr);
if (MAD_NCHANNELS(&Frame.header) == 2)
resample((long *)&Synth.pcm.samples[1][start_skip], resampled_data[1], Synth.pcm.length, &rr);
for (i = 0; i < length; i++)
{
start_skip = 0; /* not very elegant, and might want to keep this value */
samplesdone++;
//if (ci->mp3data->padding > 0) {
// ci->mp3data->padding--;
// continue ;
//}
/*if (!first_frame) {
if (detect_silence(Synth.pcm.samples[0][i]))
continue ;
first_frame = true;
}*/
/* Left channel */
Sample = scale(resampled_data[0][i], &d0);
*(OutputPtr++) = Sample >> 8;
*(OutputPtr++) = Sample & 0xff;
/* 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(resampled_data[1][i], &d1);
*(OutputPtr++) = Sample >> 8;
*(OutputPtr++) = Sample & 0xff;
samplecount--;
if (samplecount == 0) {
#ifdef DEBUG_GAPLESS
ci->write(fd, OutputBuffer, (int)OutputPtr - (int)OutputBuffer);
#endif
while (!ci->audiobuffer_insert(OutputBuffer, (int)OutputPtr - (int)OutputBuffer))
ci->yield();
goto song_end;
}
if (yieldcounter++ == 200) {
ci->yield();
yieldcounter = 0;
}
/* Flush the buffer if it is full. */
if (OutputPtr == OutputBufferEnd)
{
#ifdef DEBUG_GAPLESS
ci->write(fd, OutputBuffer, OUTPUT_BUFFER_SIZE);
#endif
while (!ci->audiobuffer_insert(OutputBuffer, OUTPUT_BUFFER_SIZE))
ci->yield();
OutputPtr = OutputBuffer;
}
}
ci->set_elapsed(samplesdone / (ci->id3->frequency/1000));
}
song_end:
#ifdef DEBUG_GAPLESS
ci->close(fd);
#endif
Stream.error = 0;
if (ci->request_next_track())
goto next_track;
return CODEC_OK;
}
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