rockbox/apps/codecs/alac.c
Michael Sevakis 7ad2cad173 Commit work started in FS#12153 to put timing/position information in PCM
buffer chunks.

* Samples and position indication is closely associated with audio data
  instead of compensating by a latency constant. Alleviates problems with
  using the elapsed as a track indicator where it could be off by several
  steps.

* Timing is accurate throughout track even if resampling for pitch shift,
  whereas before it updated during transition latency at the normal 1:1 rate.

* Simpler PCM buffer with a constant chunk size, no linked lists.

In converting crossfade, a minor change was made to not change the WPS until
the fade-in of the incoming track, whereas before it would change upon the
start of the fade-out of the outgoing track possibly having the WPS change
with far too much lead time.

Codec changes are to set elapsed times *before* writing next PCM frame because
 time and position data last set are saved in the next committed PCM chunk. 


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657
2011-08-28 07:45:35 +00:00

146 lines
4.8 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "codeclib.h"
#include "libm4a/m4a.h"
#include "libalac/decomp.h"
CODEC_HEADER
/* The maximum buffer size handled. This amount of bytes is buffered for each
* frame. */
#define ALAC_BYTE_BUFFER_SIZE 32768
static int32_t outputbuffer[ALAC_MAX_CHANNELS][ALAC_BLOCKSIZE] IBSS_ATTR;
/* this is the codec entry point */
enum codec_status codec_main(enum codec_entry_call_reason reason)
{
if (reason == CODEC_LOAD) {
/* Generic codec initialisation */
ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, ALAC_OUTPUT_DEPTH-1);
}
return CODEC_OK;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
size_t n;
demux_res_t demux_res;
stream_t input_stream;
uint32_t samplesdone;
uint32_t elapsedtime = 0;
int samplesdecoded;
unsigned int i;
unsigned char* buffer;
alac_file alac;
intptr_t param;
/* Clean and initialize decoder structures */
memset(&demux_res , 0, sizeof(demux_res));
if (codec_init()) {
LOGF("ALAC: Error initialising codec\n");
return CODEC_ERROR;
}
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
ci->seek_buffer(0);
stream_create(&input_stream,ci);
/* Read from ci->id3->offset before calling qtmovie_read. */
samplesdone = (uint32_t)(((uint64_t)(ci->id3->offset) * ci->id3->frequency) /
(ci->id3->bitrate*128));
/* if qtmovie_read returns successfully, the stream is up to
* the movie data, which can be used directly by the decoder */
if (!qtmovie_read(&input_stream, &demux_res)) {
LOGF("ALAC: Error initialising file\n");
return CODEC_ERROR;
}
/* initialise the sound converter */
alac_set_info(&alac, demux_res.codecdata);
/* Set i for first frame, seek to desired sample position for resuming. */
i=0;
if (samplesdone > 0) {
if (m4a_seek(&demux_res, &input_stream, samplesdone,
&samplesdone, (int*) &i)) {
elapsedtime = (samplesdone * 10) / (ci->id3->frequency / 100);
ci->set_elapsed(elapsedtime);
} else {
samplesdone = 0;
}
}
ci->set_elapsed(elapsedtime);
/* The main decoding loop */
while (i < demux_res.num_sample_byte_sizes) {
enum codec_command_action action = ci->get_command(&param);
if (action == CODEC_ACTION_HALT)
break;
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
/* Deal with any pending seek requests */
if (action == CODEC_ACTION_SEEK_TIME) {
if (m4a_seek(&demux_res, &input_stream,
(param/10) * (ci->id3->frequency/100),
&samplesdone, (int *)&i)) {
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
}
ci->set_elapsed(elapsedtime);
ci->seek_complete();
}
/* Request the required number of bytes from the input buffer */
buffer=ci->request_buffer(&n, ALAC_BYTE_BUFFER_SIZE);
/* Decode one block - returned samples will be host-endian */
samplesdecoded=alac_decode_frame(&alac, buffer, outputbuffer, ci->yield);
ci->yield();
/* Advance codec buffer by amount of consumed bytes */
ci->advance_buffer(alac.bytes_consumed);
/* Output the audio */
ci->pcmbuf_insert(outputbuffer[0], outputbuffer[1], samplesdecoded);
/* Update the elapsed-time indicator */
samplesdone+=samplesdecoded;
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
ci->set_elapsed(elapsedtime);
i++;
}
LOGF("ALAC: Decoded %lu samples\n",(unsigned long)samplesdone);
return CODEC_OK;
}