7ad2cad173
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
437 lines
14 KiB
C
437 lines
14 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2005 Dave Chapman
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* Copyright (C) 2009 Yoshihisa Uchida
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "codeclib.h"
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#include "codecs/libpcm/support_formats.h"
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CODEC_HEADER
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/* WAVE (RIFF) codec:
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*
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* For a good documentation on WAVE files, see:
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* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/WAVE.html
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* and
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* http://www.sonicspot.com/guide/wavefiles.html
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*
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* For sample WAV files, see:
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* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/Samples.html
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*
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*/
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#define PCM_SAMPLE_SIZE (4096*2)
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static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
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/* This codec support WAVE files with the following formats: */
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enum
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{
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WAVE_FORMAT_UNKNOWN = 0x0000, /* Microsoft Unknown Wave Format */
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WAVE_FORMAT_PCM = 0x0001, /* Microsoft PCM Format */
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WAVE_FORMAT_ADPCM = 0x0002, /* Microsoft ADPCM Format */
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WAVE_FORMAT_IEEE_FLOAT = 0x0003, /* IEEE Float */
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WAVE_FORMAT_ALAW = 0x0006, /* Microsoft ALAW */
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WAVE_FORMAT_MULAW = 0x0007, /* Microsoft MULAW */
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WAVE_FORMAT_DVI_ADPCM = 0x0011, /* Intel's DVI ADPCM */
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WAVE_FORMAT_DIALOGIC_OKI_ADPCM = 0x0017, /* Dialogic OKI ADPCM */
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WAVE_FORMAT_YAMAHA_ADPCM = 0x0020, /* Yamaha ADPCM */
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WAVE_FORMAT_XBOX_ADPCM = 0x0069, /* XBOX ADPCM */
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IBM_FORMAT_MULAW = 0x0101, /* same as WAVE_FORMAT_MULAW */
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IBM_FORMAT_ALAW = 0x0102, /* same as WAVE_FORMAT_ALAW */
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WAVE_FORMAT_SWF_ADPCM = 0x5346, /* Adobe SWF ADPCM */
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WAVE_FORMAT_EXTENSIBLE = 0xFFFE
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};
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static const struct pcm_entry wave_codecs[] = {
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{ WAVE_FORMAT_UNKNOWN, 0 },
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{ WAVE_FORMAT_PCM, get_linear_pcm_codec },
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{ WAVE_FORMAT_ADPCM, get_ms_adpcm_codec },
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{ WAVE_FORMAT_IEEE_FLOAT, get_ieee_float_codec },
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{ WAVE_FORMAT_ALAW, get_itut_g711_alaw_codec },
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{ WAVE_FORMAT_MULAW, get_itut_g711_mulaw_codec },
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{ WAVE_FORMAT_DVI_ADPCM, get_dvi_adpcm_codec },
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{ WAVE_FORMAT_DIALOGIC_OKI_ADPCM, get_dialogic_oki_adpcm_codec },
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{ WAVE_FORMAT_YAMAHA_ADPCM, get_yamaha_adpcm_codec },
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{ WAVE_FORMAT_XBOX_ADPCM, get_dvi_adpcm_codec },
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{ IBM_FORMAT_MULAW, get_itut_g711_mulaw_codec },
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{ IBM_FORMAT_ALAW, get_itut_g711_alaw_codec },
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{ WAVE_FORMAT_SWF_ADPCM, get_swf_adpcm_codec },
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};
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#define NUM_FORMATS 13
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static const struct pcm_codec *get_wave_codec(uint32_t formattag)
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{
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int i;
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for (i = 0; i < NUM_FORMATS; i++)
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{
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if (wave_codecs[i].format_tag == formattag)
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{
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if (wave_codecs[i].get_codec)
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return wave_codecs[i].get_codec();
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return 0;
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}
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}
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return 0;
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}
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static struct pcm_format format;
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static uint32_t bytesdone;
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static bool set_msadpcm_coeffs(const uint8_t *buf)
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{
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int i;
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int num;
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int size;
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buf += 4; /* skip 'fmt ' */
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size = buf[0] | (buf[1] << 8) | (buf[1] << 16) | (buf[1] << 24);
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if (size < 50)
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{
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DEBUGF("CODEC_ERROR: microsoft adpcm 'fmt ' chunk size=%lu < 50\n",
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(unsigned long)size);
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return false;
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}
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/* get nNumCoef */
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buf += 24;
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num = buf[0] | (buf[1] << 8);
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/*
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* In many case, nNumCoef is 7.
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* Depending upon the encoder, as for this value there is a possibility of
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* increasing more.
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* If you found the file where this value exceeds 7, please report.
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*/
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if (num != MSADPCM_NUM_COEFF)
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{
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DEBUGF("CODEC_ERROR: microsoft adpcm nNumCoef=%d != 7\n", num);
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return false;
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}
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/* get aCoeffs */
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buf += 2;
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for (i = 0; i < MSADPCM_NUM_COEFF; i++)
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{
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format.coeffs[i][0] = buf[0] | (SE(buf[1]) << 8);
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format.coeffs[i][1] = buf[2] | (SE(buf[3]) << 8);
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buf += 4;
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}
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return true;
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}
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static uint8_t *read_buffer(size_t *realsize)
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{
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uint8_t *buffer = (uint8_t *)ci->request_buffer(realsize, format.chunksize);
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if (bytesdone + (*realsize) > format.numbytes)
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*realsize = format.numbytes - bytesdone;
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bytesdone += *realsize;
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ci->advance_buffer(*realsize);
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return buffer;
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}
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/* this is the codec entry point */
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enum codec_status codec_main(enum codec_entry_call_reason reason)
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{
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if (reason == CODEC_LOAD) {
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/* Generic codec initialisation */
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ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
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}
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return CODEC_OK;
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}
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/* this is called for each file to process */
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enum codec_status codec_run(void)
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{
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uint32_t decodedsamples;
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size_t n;
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int bufcount;
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int endofstream;
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unsigned char *buf;
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uint8_t *wavbuf;
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off_t firstblockposn; /* position of the first block in file */
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const struct pcm_codec *codec;
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uint32_t size;
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intptr_t param;
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if (codec_init()) {
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DEBUGF("codec_init() error\n");
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return CODEC_ERROR;
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}
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codec_set_replaygain(ci->id3);
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/* Need to save offset for later use (cleared indirectly by advance_buffer) */
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bytesdone = ci->id3->offset;
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/* get RIFF chunk header */
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ci->seek_buffer(0);
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buf = ci->request_buffer(&n, 12);
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if (n < 12) {
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DEBUGF("request_buffer error\n");
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return CODEC_ERROR;
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}
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if ((memcmp(buf, "RIFF", 4) != 0) || (memcmp(&buf[8], "WAVE", 4) != 0)) {
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DEBUGF("CODEC_ERROR: missing riff header\n");
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return CODEC_ERROR;
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}
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/* advance to first WAVE chunk */
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ci->advance_buffer(12);
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firstblockposn = 12;
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ci->memset(&format, 0, sizeof(struct pcm_format));
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format.is_signed = true;
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format.is_little_endian = true;
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decodedsamples = 0;
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codec = 0;
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/* iterate over WAVE chunks until the 'data' chunk, which should be after the 'fmt ' chunk */
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while (true) {
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/* get WAVE chunk header */
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buf = ci->request_buffer(&n, 1024);
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if (n < 8) {
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DEBUGF("data chunk request_buffer error\n");
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/* no more chunks, 'data' chunk must not have been found */
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return CODEC_ERROR;
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}
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/* chunkSize */
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size = (buf[4]|(buf[5]<<8)|(buf[6]<<16)|(buf[7]<<24));
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if (memcmp(buf, "fmt ", 4) == 0) {
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if (size < 16) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%lu < 16\n",
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(unsigned long)size);
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return CODEC_ERROR;
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}
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/* wFormatTag */
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format.formattag=buf[8]|(buf[9]<<8);
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/* wChannels */
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format.channels=buf[10]|(buf[11]<<8);
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/* skipping dwSamplesPerSec */
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/* skipping dwAvgBytesPerSec */
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/* wBlockAlign */
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format.blockalign=buf[20]|(buf[21]<<8);
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/* wBitsPerSample */
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format.bitspersample=buf[22]|(buf[23]<<8);
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if (format.formattag != WAVE_FORMAT_PCM) {
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if (size < 18) {
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/* this is not a fatal error with some formats,
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* we'll see later if we can't decode it */
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DEBUGF("CODEC_WARNING: non-PCM WAVE (formattag=0x%x) "
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"doesn't have ext. fmt descr (chunksize=%d<18).\n",
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(unsigned int)format.formattag, (int)size);
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}
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else
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{
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if (format.formattag != WAVE_FORMAT_EXTENSIBLE)
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format.samplesperblock = buf[26]|(buf[27]<<8);
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else
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{
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format.size = buf[24]|(buf[25]<<8);
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if (format.size < 22) {
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DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
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"missing extension\n");
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return CODEC_ERROR;
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}
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/* wValidBitsPerSample */
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format.bitspersample = buf[26]|(buf[27]<<8);
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/* skipping dwChannelMask (4bytes) */
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/* SubFormat (only get the first two bytes) */
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format.formattag = buf[32]|(buf[33]<<8);
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}
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}
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}
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/* msadpcm specific */
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if (format.formattag == WAVE_FORMAT_ADPCM)
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{
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if (!set_msadpcm_coeffs(buf))
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{
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return CODEC_ERROR;
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}
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}
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/* get codec */
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codec = get_wave_codec(format.formattag);
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if (!codec)
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{
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DEBUGF("CODEC_ERROR: unsupported wave format 0x%x\n",
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(unsigned int) format.formattag);
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return CODEC_ERROR;
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}
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/* riff 8bit linear pcm is unsigned */
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if (format.formattag == WAVE_FORMAT_PCM && format.bitspersample == 8)
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format.is_signed = false;
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/* set format, parse codec specific tag, check format, and calculate chunk size */
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if (!codec->set_format(&format))
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{
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return CODEC_ERROR;
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}
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} else if (memcmp(buf, "data", 4) == 0) {
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format.numbytes = size;
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/* advance to start of data */
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ci->advance_buffer(8);
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firstblockposn += 8;
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break;
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} else if (memcmp(buf, "fact", 4) == 0) {
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/* dwSampleLength */
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if (size >= 4)
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format.totalsamples =
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(buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24));
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} else {
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DEBUGF("unknown WAVE chunk: '%c%c%c%c', size=%lu\n",
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buf[0], buf[1], buf[2], buf[3], (unsigned long)size);
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}
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/* go to next chunk (even chunk sizes must be padded) */
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size += 8 + (size & 0x01);
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ci->advance_buffer(size);
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firstblockposn += size;
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}
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if (!codec)
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{
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DEBUGF("CODEC_ERROR: 'fmt ' chunk not found\n");
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return CODEC_ERROR;
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}
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/* common format check */
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if (format.channels == 0) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n");
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return CODEC_ERROR;
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}
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if (format.samplesperblock == 0) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-wSamplesPerBlock file\n");
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return CODEC_ERROR;
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}
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if (format.blockalign == 0)
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{
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DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-blockalign file\n");
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return CODEC_ERROR;
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}
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if (format.numbytes == 0) {
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DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
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return CODEC_ERROR;
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}
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/* check chunksize */
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if ((format.chunksize / format.blockalign) * format.samplesperblock * format.channels
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> PCM_SAMPLE_SIZE)
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format.chunksize = (PCM_SAMPLE_SIZE / format.blockalign) * format.blockalign;
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if (format.chunksize == 0)
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{
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DEBUGF("CODEC_ERROR: chunksize is 0\n");
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return CODEC_ERROR;
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}
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ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
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if (format.channels == 2) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
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} else if (format.channels == 1) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
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} else {
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DEBUGF("CODEC_ERROR: more than 2 channels\n");
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return CODEC_ERROR;
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}
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/* make sure we're at the correct offset */
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if (bytesdone > (uint32_t) firstblockposn) {
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/* Round down to previous block */
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struct pcm_pos *newpos = codec->get_seek_pos(bytesdone - firstblockposn,
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PCM_SEEK_POS, &read_buffer);
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if (newpos->pos > format.numbytes)
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return CODEC_OK;
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if (ci->seek_buffer(firstblockposn + newpos->pos))
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{
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bytesdone = newpos->pos;
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decodedsamples = newpos->samples;
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}
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} else {
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/* already where we need to be */
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bytesdone = 0;
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}
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ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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/* The main decoder loop */
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endofstream = 0;
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while (!endofstream) {
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enum codec_command_action action = ci->get_command(¶m);
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if (action == CODEC_ACTION_HALT)
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break;
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if (action == CODEC_ACTION_SEEK_TIME) {
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struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME,
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&read_buffer);
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if (newpos->pos > format.numbytes)
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{
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ci->set_elapsed(ci->id3->length);
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ci->seek_complete();
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break;
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}
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if (ci->seek_buffer(firstblockposn + newpos->pos))
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{
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bytesdone = newpos->pos;
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decodedsamples = newpos->samples;
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}
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ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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ci->seek_complete();
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}
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wavbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
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if (n == 0)
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break; /* End of stream */
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if (bytesdone + n > format.numbytes) {
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n = format.numbytes - bytesdone;
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endofstream = 1;
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}
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if (codec->decode(wavbuf, n, samples, &bufcount) == CODEC_ERROR)
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{
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DEBUGF("codec error\n");
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return CODEC_ERROR;
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}
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ci->pcmbuf_insert(samples, NULL, bufcount);
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ci->advance_buffer(n);
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bytesdone += n;
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decodedsamples += bufcount;
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if (bytesdone >= format.numbytes)
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endofstream = 1;
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ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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}
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return CODEC_OK;
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}
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