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rockbox/apps/codecs/mpa.c
Miika Pekkarinen 6d6ca6b6a5 Muting trick to prevent tiny pops and glitchless mp3 seeking. 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8416 a1c6a512-1295-4272-9138-f99709370657
2006-01-22 10:25:07 +00:00

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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 "codeclib.h"
#include <codecs/libmad/mad.h>
#include <inttypes.h>
CODEC_HEADER
struct mad_stream stream IBSS_ATTR;
struct mad_frame frame IBSS_ATTR;
struct mad_synth synth IBSS_ATTR;
/* The following function is used inside libmad - let's hope it's never
called.
*/
void abort(void) {
}
#define INPUT_CHUNK_SIZE 8192
mad_fixed_t mad_frame_overlap[2][32][18] IBSS_ATTR;
unsigned char mad_main_data[MAD_BUFFER_MDLEN] IBSS_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[];
extern char iedata[];
extern char iend[];
#endif
struct codec_api *ci;
int64_t samplecount;
int64_t samplesdone;
int stop_skip, start_skip;
int current_stereo_mode = -1;
unsigned int current_frequency = 0;
void recalc_samplecount(void)
{
/* NOTE: currently this doesn't work, the below calculated samples_count
seems to be right, but sometimes we just don't have all the data we
need... */
if (ci->id3->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 = ((int64_t)ci->id3->frame_count) * 1152;
} else {
samplecount = ((int64_t)ci->id3->length) * current_frequency / 1000;
}
samplecount -= start_skip + stop_skip;
}
void init_mad(void)
{
ci->memset(&stream, 0, sizeof(struct mad_stream));
ci->memset(&frame, 0, sizeof(struct mad_frame));
ci->memset(&synth, 0, sizeof(struct mad_synth));
mad_stream_init(&stream);
mad_frame_init(&frame);
mad_synth_init(&synth);
/* We do this so libmad doesn't try to call codec_calloc() */
ci->memset(mad_frame_overlap, 0, sizeof(mad_frame_overlap));
frame.overlap = &mad_frame_overlap;
stream.main_data = &mad_main_data;
}
/* this is the codec entry point */
enum codec_status codec_start(struct codec_api *api)
{
int status = CODEC_OK;
long size;
int file_end;
int frame_skip; /* samples to skip current frame */
int samples_to_skip; /* samples to skip in total for this file (at start) */
char *inputbuffer;
ci = api;
#ifdef USE_IRAM
ci->memcpy(iramstart, iramcopy, iramend - iramstart);
ci->memset(iedata, 0, iend - iedata);
#endif
if (codec_init(api))
return CODEC_ERROR;
/* Create a decoder instance */
ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(MAD_F_FRACBITS));
ci->configure(DSP_SET_CLIP_MIN, (int *)-MAD_F_ONE);
ci->configure(DSP_SET_CLIP_MAX, (int *)(MAD_F_ONE - 1));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
/** 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.
* Reinitializing seems to be necessary to avoid playback quircks when seeking. */
next_track:
init_mad();
file_end = 0;
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
ci->configure(DSP_SET_FREQUENCY, (int *)ci->id3->frequency);
current_frequency = ci->id3->frequency;
codec_set_replaygain(ci->id3);
ci->request_buffer(&size, ci->id3->first_frame_offset);
ci->advance_buffer(size);
if (ci->id3->lead_trim >= 0 && ci->id3->tail_trim >= 0) {
stop_skip = ci->id3->tail_trim - mpeg_latency[ci->id3->layer];
if (stop_skip < 0) stop_skip = 0;
start_skip = ci->id3->lead_trim + mpeg_latency[ci->id3->layer];
} else {
stop_skip = 0;
/* We want to skip this amount anyway */
start_skip = mpeg_latency[ci->id3->layer];
}
samplesdone = ((int64_t)ci->id3->elapsed) * current_frequency / 1000;
samples_to_skip = start_skip;
recalc_samplecount();
/* This is the decoding loop. */
while (1) {
int framelength;
ci->yield();
if (ci->stop_codec || ci->reload_codec)
break;
if (ci->seek_time) {
int newpos;
samplesdone = ((int64_t) (ci->seek_time - 1))
* current_frequency / 1000;
if (ci->seek_time-1 == 0)
newpos = 0;
else
newpos = ci->mp3_get_filepos(ci->seek_time-1) +
ci->id3->first_frame_offset;
if (!ci->seek_buffer(newpos))
goto next_track;
ci->seek_complete();
init_mad();
}
/* Lock buffers */
if (stream.error == 0) {
inputbuffer = ci->request_buffer(&size, INPUT_CHUNK_SIZE);
if (size == 0 || inputbuffer == NULL)
break;
mad_stream_buffer(&stream, (unsigned char *)inputbuffer, size);
}
if (mad_frame_decode(&frame, &stream)) {
if (stream.error == MAD_FLAG_INCOMPLETE
|| stream.error == MAD_ERROR_BUFLEN) {
/* This makes the codec support partially corrupted files */
if (file_end == 30)
break;
/* Fill the buffer */
if (stream.next_frame)
ci->advance_buffer_loc((void *)stream.next_frame);
else
ci->advance_buffer(size);
stream.error = 0;
file_end++;
continue;
} else if (MAD_RECOVERABLE(stream.error)) {
continue;
} else {
/* Some other unrecoverable error */
status = CODEC_ERROR;
break;
}
break;
}
file_end = 0;
mad_synth_frame(&synth, &frame);
/* We need to skip samples_to_skip samples from the start of every file
to properly support LAME style gapless MP3 files. samples_to_skip
might be larger than one frame. */
if (samples_to_skip < synth.pcm.length) {
/* skip just part of the frame */
frame_skip = samples_to_skip;
samples_to_skip = 0;
} else {
/* we need to skip an entire frame */
frame_skip = synth.pcm.length;
samples_to_skip -= synth.pcm.length;
}
framelength = synth.pcm.length - frame_skip;
if (stop_skip > 0) {
int64_t max = samplecount - samplesdone;
if (max < 0) max = 0;
if (max < framelength) framelength = (int)max;
if (framelength == 0 && frame_skip == 0) break;
}
/* Check if sample rate and stereo settings changed in this frame. */
if (frame.header.samplerate != current_frequency) {
current_frequency = frame.header.samplerate;
ci->configure(DSP_SWITCH_FREQUENCY, (int *)current_frequency);
recalc_samplecount();
}
if (MAD_NCHANNELS(&frame.header) == 2) {
if (current_stereo_mode != STEREO_NONINTERLEAVED) {
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
current_stereo_mode = STEREO_NONINTERLEAVED;
}
} else {
if (current_stereo_mode != STEREO_MONO) {
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
current_stereo_mode = STEREO_MONO;
}
}
/* Check if we can just skip the entire frame. */
if (frame_skip < synth.pcm.length) {
/* In case of a mono file, the second array will be ignored. */
ci->pcmbuf_insert_split(&synth.pcm.samples[0][frame_skip],
&synth.pcm.samples[1][frame_skip],
framelength * 4);
}
if (stream.next_frame)
ci->advance_buffer_loc((void *)stream.next_frame);
else
ci->advance_buffer(size);
samplesdone += framelength;
ci->set_elapsed(samplesdone / (current_frequency / 1000));
}
stream.error = 0;
if (ci->request_next_track())
goto next_track;
return status;
}
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