forked from len0rd/rockbox
add SMAF codec (.mmf extension)(FS#10432)
This codec supports only wave data (ADPCM and PCM). It does not support MIDI, picture, and movie. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24878 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
parent
aa58715a54
commit
45e009a364
9 changed files with 821 additions and 1 deletions
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@ -27,6 +27,7 @@ shorten.c
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aiff.c
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speex.c
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adx.c
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smaf.c
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#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
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/* encoders */
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aiff_enc.c
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@ -90,6 +90,7 @@ $(CODECDIR)/atrac3_rm.codec : $(CODECDIR)/libatrac.a $(CODECDIR)/librm.a
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$(CODECDIR)/atrac3_oma.codec : $(CODECDIR)/libatrac.a
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$(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a
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$(CODECS): $(CODECLIB) # this must be last in codec dependency list
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435
apps/codecs/smaf.c
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435
apps/codecs/smaf.c
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@ -0,0 +1,435 @@
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/***************************************************************************
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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_TRACK_CHUNK_SCORE = 0, /* Score Track */
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SMAF_TRACK_CHUNK_AUDIO, /* PCM Audio 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 int support_formats[2][3] = {
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{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_UNSIGNED_PCM, SMAF_FORMAT_ADPCM },
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{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_ADPCM, SMAF_FORMAT_UNSUPPORT },
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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 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(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_audio_format(int track_chunk, unsigned int audio_format)
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{
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if (audio_format > 3)
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return SMAF_FORMAT_UNSUPPORT;
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return support_formats[track_chunk][audio_format];
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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 0;
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return basebits[basebit];
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}
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static bool parse_audio_track(struct pcm_format *fmt,
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unsigned char **stbuf, unsigned char *endbuf)
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{
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unsigned char *buf = *stbuf;
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int chunksize;
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buf += 8;
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fmt->channels = ((buf[2] & 0x80) >> 7) + 1;
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fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_AUDIO,
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(buf[2] >> 4) & 0x07);
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if (fmt->formattag == SMAF_FORMAT_UNSUPPORT)
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{
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DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n", (buf[2] >> 4) & 0x07);
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return false;
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}
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fmt->bitspersample = convert_smaf_audio_basebit(buf[3] >> 4);
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if (fmt->bitspersample == 0)
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{
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DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n", buf[3] >> 4);
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return false;
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}
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buf += 6;
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while (buf < endbuf)
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{
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chunksize = get_be32(buf + 4) + 8;
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if (memcmp(buf, "Awa", 3) == 0)
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{
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fmt->numbytes = get_be32(buf + 4);
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buf += 8;
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return true;
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}
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buf += chunksize;
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}
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DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
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return false;
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}
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static bool parse_score_track(struct pcm_format *fmt,
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unsigned char **stbuf, unsigned char *endbuf)
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{
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unsigned char *buf = *stbuf;
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int chunksize;
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if (buf[9] != 0x00)
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{
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DEBUGF("CODEC_ERROR: score track chunk unsupport sequence type %d\n", buf[9]);
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return false;
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}
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/*
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* skip to the next chunk.
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* MA-2/MA-3/MA-5: padding 16 bytes
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* MA-7: padding 32 bytes
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*/
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if (buf[3] < 7)
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buf += 28;
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else
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buf += 44;
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while (buf < endbuf)
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{
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chunksize = get_be32(buf + 4) + 8;
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if (memcmp(buf, "Mtsp", 4) == 0)
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{
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buf += 8;
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if (memcmp(buf, "Mwa", 3) != 0)
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{
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DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
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return false;
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}
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fmt->numbytes = get_be32(buf + 4) - 3;
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fmt->channels = ((buf[8] & 0x80) >> 7) + 1;
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fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_SCORE,
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(buf[8] >> 4) & 0x07);
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if (fmt->formattag == SMAF_FORMAT_UNSUPPORT)
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{
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DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n",
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(buf[8] >> 4) & 0x07);
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return false;
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}
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fmt->bitspersample = convert_smaf_audio_basebit(buf[8] & 0x0f);
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if (fmt->bitspersample == 0)
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{
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DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n",
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buf[8] & 0x0f);
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return false;
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}
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buf += 11;
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return true;
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}
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buf += chunksize;
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}
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DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
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return false;
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}
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static bool parse_header(struct pcm_format *fmt, size_t *pos)
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{
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unsigned char *buf, *stbuf, *endbuf;
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size_t chunksize;
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ci->memset(fmt, 0, sizeof(struct pcm_format));
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/* assume the SMAF pcm data position is less than 1024 bytes */
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stbuf = ci->request_buffer(&chunksize, 1024);
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if (chunksize < 1024)
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return false;
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buf = stbuf;
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endbuf = stbuf + chunksize;
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if (memcmp(buf, "MMMD", 4) != 0)
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{
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DEBUGF("CODEC_ERROR: does not smaf format %c%c%c%c\n",
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buf[0], buf[1], buf[2], buf[3]);
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return false;
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}
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buf += 8;
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while (buf < endbuf)
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{
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chunksize = get_be32(buf + 4) + 8;
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if (memcmp(buf, "ATR", 3) == 0)
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{
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if (!parse_audio_track(fmt, &buf, endbuf))
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return false;
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break;
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}
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if (memcmp(buf, "MTR", 3) == 0)
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{
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if (!parse_score_track(fmt, &buf, endbuf))
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return false;
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break;
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}
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buf += chunksize;
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}
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if (buf >= endbuf)
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{
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DEBUGF("CODEC_ERROR: unsupported smaf format\n");
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return false;
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}
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/* blockalign */
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if (fmt->formattag == SMAF_FORMAT_SIGNED_PCM ||
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fmt->formattag == SMAF_FORMAT_UNSIGNED_PCM)
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fmt->blockalign = fmt->channels * fmt->bitspersample >> 3;
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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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fmt->is_little_endian = false;
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/* sets pcm data position */
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*pos = buf - stbuf;
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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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enum codec_status codec_main(void)
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{
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int status = CODEC_OK;
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uint32_t decodedsamples;
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uint32_t i = CODEC_OK;
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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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/* Generic codec initialisation */
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ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
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next_track:
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if (codec_init()) {
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i = CODEC_ERROR;
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goto exit;
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}
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while (!*ci->taginfo_ready && !ci->stop_codec)
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ci->sleep(1);
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codec_set_replaygain(ci->id3);
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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 = false;
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decodedsamples = 0;
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codec = 0;
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if (!parse_header(&format, &n))
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{
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i = CODEC_ERROR;
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goto done;
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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%lx\n", format.formattag);
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i = CODEC_ERROR;
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goto done;
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}
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if (!codec->set_format(&format))
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{
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i = CODEC_ERROR;
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goto done;
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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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status = CODEC_ERROR;
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goto done;
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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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status = CODEC_ERROR;
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goto done;
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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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i = CODEC_ERROR;
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goto done;
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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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status = CODEC_ERROR;
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goto done;
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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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i = CODEC_ERROR;
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goto done;
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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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i = CODEC_ERROR;
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goto done;
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}
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firstblockposn = 1024 - n;
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ci->advance_buffer(firstblockposn);
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/* The main decoder loop */
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bytesdone = 0;
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ci->set_elapsed(0);
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endofstream = 0;
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while (!endofstream) {
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ci->yield();
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if (ci->stop_codec || ci->new_track)
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break;
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if (ci->seek_time) {
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struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, &read_buffer);
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decodedsamples = newpos->samples;
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if (newpos->pos > format.numbytes)
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break;
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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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}
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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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status = codec->decode(smafbuf, n, samples, &bufcount);
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if (status == CODEC_ERROR)
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{
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DEBUGF("codec error\n");
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goto done;
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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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i = CODEC_OK;
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done:
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if (ci->request_next_track())
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goto next_track;
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exit:
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return i;
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}
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