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
492 lines
13 KiB
C
492 lines
13 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) 2010 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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/*
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* SMAF (Synthetic music Mobile Application Format)
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*
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* References
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* [1] YAMAHA Corporation, Synthetic music Mobile Application Format Ver.3.05, 2002
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*/
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enum {
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SMAF_AUDIO_TRACK_CHUNK = 0, /* PCM Audio Track */
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SMAF_SCORE_TRACK_CHUNK, /* Score Track */
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};
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/* SMAF supported codec formats */
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enum {
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SMAF_FORMAT_UNSUPPORT = 0, /* unsupported format */
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SMAF_FORMAT_SIGNED_PCM, /* 2's complement PCM */
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SMAF_FORMAT_UNSIGNED_PCM, /* Offset Binary PCM */
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SMAF_FORMAT_ADPCM, /* YAMAHA ADPCM */
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};
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static const int support_formats[2][3] = {
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{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_ADPCM, SMAF_FORMAT_UNSUPPORT },
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{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_UNSIGNED_PCM, SMAF_FORMAT_ADPCM },
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};
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static const struct pcm_entry pcm_codecs[] = {
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{ SMAF_FORMAT_SIGNED_PCM, get_linear_pcm_codec },
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{ SMAF_FORMAT_UNSIGNED_PCM, get_linear_pcm_codec },
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{ SMAF_FORMAT_ADPCM, get_yamaha_adpcm_codec },
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};
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#define NUM_FORMATS 3
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static const int basebits[4] = { 4, 8, 12, 16 };
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#define PCM_SAMPLE_SIZE (2048*2)
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static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
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static const struct pcm_codec *get_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 (pcm_codecs[i].format_tag == formattag)
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{
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if (pcm_codecs[i].get_codec)
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return pcm_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 unsigned int get_be32(const uint8_t *buf)
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{
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return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
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}
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static int convert_smaf_channels(unsigned int ch)
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{
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return (ch >> 7) + 1;
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}
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static int convert_smaf_audio_format(unsigned int chunk, unsigned int audio_format)
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{
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int idx = (audio_format & 0x70) >> 4;
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if (idx < 3)
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return support_formats[chunk][idx];
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DEBUGF("CODEC_ERROR: unsupport audio format: %d\n", audio_format);
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return SMAF_FORMAT_UNSUPPORT;
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}
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static int convert_smaf_audio_basebit(unsigned int basebit)
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{
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if (basebit < 4)
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return basebits[basebit];
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DEBUGF("CODEC_ERROR: illegal basebit: %d\n", basebit);
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return 0;
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}
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static unsigned int search_chunk(const unsigned char *name, int nlen, off_t *pos)
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{
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const unsigned char *buf;
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unsigned int chunksize;
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size_t size;
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while (true)
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{
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buf = ci->request_buffer(&size, 8);
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if (size < 8)
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break;
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chunksize = get_be32(buf + 4);
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ci->advance_buffer(8);
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*pos += 8;
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if (memcmp(buf, name, nlen) == 0)
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return chunksize;
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ci->advance_buffer(chunksize);
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*pos += chunksize;
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}
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DEBUGF("CODEC_ERROR: missing '%s' chunk\n", name);
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return 0;
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}
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static bool parse_audio_track(struct pcm_format *fmt, unsigned int chunksize, off_t *pos)
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{
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const unsigned char *buf;
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size_t size;
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/* search PCM Audio Track Chunk */
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ci->advance_buffer(chunksize);
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*pos += chunksize;
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if (search_chunk("ATR", 3, pos) == 0)
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{
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DEBUGF("CODEC_ERROR: missing PCM Audio Track Chunk\n");
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return false;
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}
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/*
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* get format
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* buf
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* +0: Format Type
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* +1: Sequence Type
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* +2: bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: frequency
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* +3: bit 4-7: base bit
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* +4: TimeBase_D
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* +5: TimeBase_G
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*
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* Note: If PCM Audio Track does not include Sequence Data Chunk,
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* tmp+6 is the start position of Wave Data Chunk.
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*/
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buf = ci->request_buffer(&size, 6);
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if (size < 6)
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{
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DEBUGF("CODEC_ERROR: smaf is too small\n");
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return false;
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}
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fmt->formattag = convert_smaf_audio_format(SMAF_AUDIO_TRACK_CHUNK, buf[2]);
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fmt->channels = convert_smaf_channels(buf[2]);
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fmt->bitspersample = convert_smaf_audio_basebit(buf[3] >> 4);
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/* search Wave Data Chunk */
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ci->advance_buffer(6);
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*pos += 6;
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fmt->numbytes = search_chunk("Awa", 3, pos);
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if (fmt->numbytes == 0)
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{
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DEBUGF("CODEC_ERROR: missing Wave Data Chunk\n");
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return false;
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}
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return true;
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}
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static bool parse_score_track(struct pcm_format *fmt, off_t *pos)
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{
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const unsigned char *buf;
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unsigned int chunksize;
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size_t size;
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/* parse Optional Data Chunk */
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buf = ci->request_buffer(&size, 13);
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if (size < 13)
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{
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DEBUGF("CODEC_ERROR: smaf is too small\n");
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return false;
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}
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if (memcmp(buf + 5, "OPDA", 4) != 0)
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{
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DEBUGF("CODEC_ERROR: missing Optional Data Chunk\n");
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return false;
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}
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/* Optional Data Chunk size */
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chunksize = get_be32(buf + 9);
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/* search Score Track Chunk */
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ci->advance_buffer(13 + chunksize);
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*pos += (13 + chunksize);
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if (search_chunk("MTR", 3, pos) == 0)
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{
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DEBUGF("CODEC_ERROR: missing Score Track Chunk\n");
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return false;
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}
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/*
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* search next chunk
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* usually, next chunk ('M***') found within 40 bytes.
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*/
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buf = ci->request_buffer(&size, 40);
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if (size < 40)
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{
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DEBUGF("CODEC_ERROR: smaf is too small\n");
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return false;
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}
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size = 0;
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while (size < 40 && buf[size] != 'M')
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size++;
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if (size >= 40)
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{
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DEBUGF("CODEC_ERROR: missing Score Track Stream PCM Data Chunk");
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return false;
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}
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/* search Score Track Stream PCM Data Chunk */
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ci->advance_buffer(size);
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*pos += size;
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if (search_chunk("Mtsp", 4, pos) == 0)
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{
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DEBUGF("CODEC_ERROR: missing Score Track Stream PCM Data Chunk\n");
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return false;
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}
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/*
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* parse Score Track Stream Wave Data Chunk
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* buf
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* +4-7: chunk size (WaveType(3bytes) + wave data count)
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* +8: bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: base bit
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* +9: frequency (MSB)
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* +10: frequency (LSB)
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*/
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buf = ci->request_buffer(&size, 9);
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if (size < 9)
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{
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DEBUGF("CODEC_ERROR: smaf is too small\n");
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return false;
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}
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if (memcmp(buf, "Mwa", 3) != 0)
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{
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DEBUGF("CODEC_ERROR: missing Score Track Stream Wave Data Chunk\n");
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return false;
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}
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fmt->formattag = convert_smaf_audio_format(SMAF_SCORE_TRACK_CHUNK, buf[8]);
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fmt->channels = convert_smaf_channels(buf[8]);
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fmt->bitspersample = convert_smaf_audio_basebit(buf[8] & 0xf);
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fmt->numbytes = get_be32(buf + 4) - 3;
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*pos += 11;
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return true;
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}
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static bool parse_header(struct pcm_format *fmt, off_t *pos)
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{
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const unsigned char *buf;
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unsigned int chunksize;
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size_t size;
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ci->memset(fmt, 0, sizeof(struct pcm_format));
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/* check File Chunk and Contents Info Chunk */
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buf = ci->request_buffer(&size, 16);
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if (size < 16)
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{
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DEBUGF("CODEC_ERROR: smaf is too small\n");
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return false;
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}
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if ((memcmp(buf, "MMMD", 4) != 0) || (memcmp(buf + 8, "CNTI", 4) != 0))
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{
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DEBUGF("CODEC_ERROR: does not smaf format\n");
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return false;
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}
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chunksize = get_be32(buf + 12);
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ci->advance_buffer(16);
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*pos = 16;
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if (chunksize > 5)
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{
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if (!parse_audio_track(fmt, chunksize, pos))
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return false;
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}
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else if (!parse_score_track(fmt, pos))
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return false;
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/* data signess (default signed) */
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fmt->is_signed = (fmt->formattag != SMAF_FORMAT_UNSIGNED_PCM);
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/* data is always big endian */
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fmt->is_little_endian = false;
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return true;
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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 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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uint8_t *smafbuf;
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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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intptr_t param;
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if (codec_init())
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return CODEC_ERROR;
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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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decodedsamples = 0;
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codec = 0;
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ci->seek_buffer(0);
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if (!parse_header(&format, &firstblockposn))
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{
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return CODEC_ERROR;
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}
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codec = get_codec(format.formattag);
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if (codec == 0)
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{
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DEBUGF("CODEC_ERROR: unsupport audio format: 0x%x\n", (int)format.formattag);
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return CODEC_ERROR;
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}
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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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/* 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 unsupported\n");
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return CODEC_ERROR;
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}
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ci->seek_buffer(firstblockposn);
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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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{
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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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}
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else
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{
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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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smafbuf = (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(smafbuf, 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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