rockbox/apps/codecs/mpa.c
Miika Pekkarinen d8cb703b1e Initial DSP implementation. DSP supports resampling audio stream from
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
2005-06-26 19:41:29 +00:00

253 lines
8.6 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 "codec.h"
#include <codecs/libmad/mad.h>
#include "playback.h"
#include "dsp.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;
/* The following function is used inside libmad - let's hope it's never
called.
*/
void abort(void) {
}
#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;
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
/*
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;
char *InputBuffer;
unsigned int samplecount;
unsigned int samplesdone;
bool first_frame;
int stop_skip, start_skip;
// struct resampler lr = { 0, 0, 0 }, rr = { 0, 0, 0 };
/* 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->configure(DSP_SET_CLIP_MIN, (int *)-MAD_F_ONE);
ci->configure(DSP_SET_CLIP_MAX, (int *)(MAD_F_ONE - 1));
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(MAD_F_FRACBITS));
ci->configure(DSP_DITHER, (bool *)true);
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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:
info = ci->mp3data;
first_frame = false;
file_end = 0;
OutputPtr = OutputBuffer;
while (!*ci->taginfo_ready)
ci->yield();
ci->configure(DSP_SET_FREQUENCY, (int *)ci->id3->frequency);
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;
}
/* 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(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);
/* 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);
ci->audiobuffer_insert_split(&Synth.pcm.samples[0][start_skip],
&Synth.pcm.samples[1][start_skip],
(Synth.pcm.length - start_skip) * 4);
start_skip = 0; /* not very elegant, and might want to keep this value */
samplesdone += Synth.pcm.length;
samplecount -= Synth.pcm.length;
ci->set_elapsed(samplesdone / (ci->id3->frequency/1000));
}
Stream.error = 0;
if (ci->request_next_track())
goto next_track;
return CODEC_OK;
}