forked from len0rd/rockbox
Add wave64(.w64) codec (FS#11022)
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24959 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
parent
6434e760fc
commit
561cb2c401
10 changed files with 600 additions and 19 deletions
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@ -30,6 +30,7 @@ adx.c
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smaf.c
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au.c
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vox.c
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wav64.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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@ -93,6 +93,7 @@ $(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/au.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/vox.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/wav64.codec : $(CODECDIR)/libpcm.a
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$(CODECS): $(CODECLIB) # this must be last in codec dependency list
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447
apps/codecs/wav64.c
Normal file
447
apps/codecs/wav64.c
Normal file
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@ -0,0 +1,447 @@
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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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/* Wave64 codec
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*
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* References
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* [1] VCS Aktiengesellschaft, Sony Wave64, Informations_about_Sony_Wave64.pdf
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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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/* Wave64 GUIDs */
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#define WAVE64_GUID_RIFF "riff\x2e\x91\xcf\x11\xa5\xd6\x28\xdb\x04\xc1\x00\x00"
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#define WAVE64_GUID_WAVE "wave\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
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#define WAVE64_GUID_FMT "fmt \xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
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#define WAVE64_GUID_FACT "fact\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
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#define WAVE64_GUID_DATA "data\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
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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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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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/* Read an unaligned 64-bit little endian unsigned integer from buffer. */
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static uint64_t get_uint64_le(void* buf)
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{
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unsigned char* p = (unsigned char*) buf;
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return p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24) | ((uint64_t)p[4] << 32) |
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((uint64_t)p[5] << 40) | ((uint64_t)p[6] << 48) | ((uint64_t)p[7] << 56);
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}
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static bool set_msadpcm_coeffs(unsigned char *buf)
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{
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int i;
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int num;
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uint64_t size;
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buf += 16; /* skip 'fmt ' GUID */
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size = get_uint64_le(buf);
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if (size < 50)
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{
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DEBUGF("CODEC_ERROR: microsoft adpcm 'fmt ' chunk size=%d < 50\n", (int)size);
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return false;
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}
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/* get nNumCoef */
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buf += 28;
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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(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;
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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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uint64_t size;
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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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DEBUGF("codec_init() error\n");
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status = 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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/* 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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buf = ci->request_buffer(&n, 40);
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if (n < 40) {
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DEBUGF("request_buffer error\n");
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status = CODEC_ERROR;
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goto done;
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}
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if ((memcmp(buf , WAVE64_GUID_RIFF, 16) != 0) ||
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(memcmp(buf+24, WAVE64_GUID_WAVE, 16) != 0))
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{
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status = CODEC_ERROR;
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goto done;
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}
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/* advance to first WAVE chunk */
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ci->advance_buffer(40);
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firstblockposn = 40;
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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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status = CODEC_ERROR;
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goto done;
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}
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/* chunkSize */
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size = get_uint64_le(buf+16) - 24;
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if (memcmp(buf, WAVE64_GUID_FMT, 16) == 0) {
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if (size < 16) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%d < 16\n", (int)size);
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status = CODEC_ERROR;
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goto done;
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}
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/* wFormatTag */
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format.formattag=buf[24]|(buf[25]<<8);
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/* wChannels */
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format.channels=buf[26]|(buf[27]<<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[36]|(buf[37]<<8);
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/* wBitsPerSample */
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format.bitspersample=buf[38]|(buf[39]<<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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format.size = buf[40]|(buf[41]<<8);
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if (format.formattag != WAVE_FORMAT_EXTENSIBLE)
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format.samplesperblock = buf[42]|(buf[43]<<8);
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else {
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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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status = CODEC_ERROR;
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goto done;
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}
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/* wValidBitsPerSample */
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format.bitspersample = buf[42]|(buf[43]<<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[48]|(buf[49]<<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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status = CODEC_ERROR;
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goto done;
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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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status = CODEC_ERROR;
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goto done;
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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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/* check format, and calculate chunk size */
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if (!codec->set_format(&format))
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{
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status = CODEC_ERROR;
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goto done;
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}
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} else if (memcmp(buf, WAVE64_GUID_DATA, 16) == 0) {
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format.numbytes = size;
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/* advance to start of data */
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ci->advance_buffer(24);
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firstblockposn += 24;
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break;
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} else if (memcmp(buf, WAVE64_GUID_FACT, 16) == 0) {
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/* skip 'fact' chunk */
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} else {
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DEBUGF("unknown Wave64 chunk: "
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"'%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x'\n",
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buf[0], buf[1], buf[ 2], buf[ 3], buf[ 4], buf[ 5], buf[ 6], buf[ 7],
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buf[8], buf[9], buf[10], buf[11], buf[12], buf[13], buf[14], buf[15]);
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}
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/* go to next chunk (8byte bound) */
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if (size & 0x07)
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size += 8 - (size & 0x7);
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ci->advance_buffer(size + 24);
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firstblockposn += size + 24;
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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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status = 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\n");
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status = CODEC_ERROR;
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goto done;
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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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uint32_t offset = bytesdone - bytesdone % format.blockalign;
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ci->advance_buffer(offset-firstblockposn);
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bytesdone = offset - firstblockposn;
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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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/* The main decoder loop */
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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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}
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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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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;
|
||||
endofstream = 1;
|
||||
}
|
||||
|
||||
status = codec->decode(wavbuf, n, samples, &bufcount);
|
||||
if (status == CODEC_ERROR)
|
||||
{
|
||||
DEBUGF("codec error\n");
|
||||
goto done;
|
||||
}
|
||||
|
||||
ci->pcmbuf_insert(samples, NULL, bufcount);
|
||||
ci->advance_buffer(n);
|
||||
bytesdone += n;
|
||||
decodedsamples += bufcount;
|
||||
|
||||
if (bytesdone >= format.numbytes)
|
||||
endofstream = 1;
|
||||
ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
|
||||
}
|
||||
status = CODEC_OK;
|
||||
|
||||
done:
|
||||
if (ci->request_next_track())
|
||||
goto next_track;
|
||||
|
||||
exit:
|
||||
return status;
|
||||
}
|
||||
|
|
@ -105,6 +105,7 @@ static const struct filetype inbuilt_filetypes[] = {
|
|||
{ "au", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "snd", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "vox", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "w64", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
|
||||
#endif
|
||||
{ "m3u", FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },
|
||||
{ "m3u8",FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },
|
||||
|
|
|
|||
|
|
@ -174,6 +174,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
|
|||
/* VOX (Dialogic telephony file formats) */
|
||||
[AFMT_VOX] =
|
||||
AFMT_ENTRY("VOX", "vox", NULL, "vox\0" ),
|
||||
/* Wave64 */
|
||||
[AFMT_WAVE64] =
|
||||
AFMT_ENTRY("WAVE64", "wav64", NULL, "w64\0" ),
|
||||
#endif
|
||||
};
|
||||
|
||||
|
|
@ -481,6 +484,14 @@ bool get_metadata(struct mp3entry* id3, int fd, const char* trackname)
|
|||
}
|
||||
break;
|
||||
|
||||
case AFMT_WAVE64:
|
||||
if (!get_wave64_metadata(fd, id3))
|
||||
{
|
||||
DEBUGF("get_wave64_metadata error\n");
|
||||
return false;
|
||||
}
|
||||
break;
|
||||
|
||||
#endif /* CONFIG_CODEC == SWCODEC */
|
||||
|
||||
default:
|
||||
|
|
|
|||
|
|
@ -81,6 +81,7 @@ enum
|
|||
AFMT_SMAF, /* SMAF */
|
||||
AFMT_AU, /* Sun Audio file */
|
||||
AFMT_VOX, /* VOX */
|
||||
AFMT_WAVE64, /* Wave64 */
|
||||
#endif
|
||||
|
||||
/* add new formats at any index above this line to have a sensible order -
|
||||
|
|
|
|||
|
|
@ -174,6 +174,15 @@ int read_uint64le(int fd, uint64_t* buf)
|
|||
}
|
||||
#endif
|
||||
|
||||
/* Read an unaligned 64-bit little endian unsigned integer from buffer. */
|
||||
uint64_t get_uint64_le(void* buf)
|
||||
{
|
||||
unsigned char* p = (unsigned char*) buf;
|
||||
|
||||
return p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24) | ((uint64_t)p[4] << 32) |
|
||||
((uint64_t)p[5] << 40) | ((uint64_t)p[6] << 48) | ((uint64_t)p[7] << 56);
|
||||
}
|
||||
|
||||
/* Read an unaligned 32-bit little endian long from buffer. */
|
||||
unsigned long get_long_le(void* buf)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -57,6 +57,7 @@ int read_uint64be(int fd, uint64_t* buf);
|
|||
#define read_uint64le(fd,buf) read((fd), (buf), 8)
|
||||
#endif
|
||||
|
||||
uint64_t get_uint64_le(void* buf);
|
||||
unsigned long get_long_le(void* buf);
|
||||
unsigned short get_short_le(void* buf);
|
||||
unsigned long get_long_be(void* buf);
|
||||
|
|
|
|||
|
|
@ -45,3 +45,4 @@ bool get_oma_metadata(int fd, struct mp3entry* id3);
|
|||
bool get_smaf_metadata(int fd, struct mp3entry* id3);
|
||||
bool get_au_metadata(int fd, struct mp3entry* id3);
|
||||
bool get_vox_metadata(int fd, struct mp3entry* id3);
|
||||
bool get_wave64_metadata(int fd, struct mp3entry* id3);
|
||||
|
|
|
|||
|
|
@ -8,6 +8,7 @@
|
|||
* $Id$
|
||||
*
|
||||
* Copyright (C) 2005 Dave Chapman
|
||||
* Copyright (C) 2010 Yoshihisa Uchida
|
||||
*
|
||||
* This program is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU General Public License
|
||||
|
|
@ -41,6 +42,15 @@
|
|||
((uint8_t*)(p))[1] = (d)>>8; \
|
||||
} while(0)
|
||||
|
||||
/* Wave(RIFF)/Wave64 format */
|
||||
|
||||
/* Wave64 GUIDs */
|
||||
#define WAVE64_GUID_RIFF "riff\x2e\x91\xcf\x11\xa5\xd6\x28\xdb\x04\xc1\x00\x00"
|
||||
#define WAVE64_GUID_WAVE "wave\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
|
||||
#define WAVE64_GUID_FMT "fmt \xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
|
||||
#define WAVE64_GUID_FACT "fact\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
|
||||
#define WAVE64_GUID_DATA "data\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
|
||||
|
||||
enum
|
||||
{
|
||||
WAVE_FORMAT_PCM = 0x0001, /* Microsoft PCM Format */
|
||||
|
|
@ -64,7 +74,7 @@ struct wave_fmt {
|
|||
unsigned int blockalign;
|
||||
unsigned long bitspersample;
|
||||
unsigned int samplesperblock;
|
||||
unsigned long numbytes;
|
||||
uint64_t numbytes;
|
||||
};
|
||||
|
||||
static unsigned long get_totalsamples(struct wave_fmt *fmt, struct mp3entry* id3)
|
||||
|
|
@ -114,6 +124,28 @@ static unsigned long get_totalsamples(struct wave_fmt *fmt, struct mp3entry* id3
|
|||
return totalsamples;
|
||||
}
|
||||
|
||||
static void parse_riff_format(unsigned char* buf, int fmtsize, struct wave_fmt *fmt,
|
||||
struct mp3entry* id3)
|
||||
{
|
||||
/* wFormatTag */
|
||||
fmt->formattag = buf[0] | (buf[1] << 8);
|
||||
/* wChannels */
|
||||
fmt->channels = buf[2] | (buf[3] << 8);
|
||||
/* dwSamplesPerSec */
|
||||
id3->frequency = get_long_le(&buf[4]);
|
||||
/* dwAvgBytesPerSec */
|
||||
id3->bitrate = (get_long_le(&buf[8]) * 8) / 1000;
|
||||
/* wBlockAlign */
|
||||
fmt->blockalign = buf[12] | (buf[13] << 8);
|
||||
/* wBitsPerSample */
|
||||
fmt->bitspersample = buf[14] | (buf[15] << 8);
|
||||
if (fmtsize > 19)
|
||||
{
|
||||
/* wSamplesPerBlock */
|
||||
fmt->samplesperblock = buf[18] | (buf[19] << 8);
|
||||
}
|
||||
}
|
||||
|
||||
bool get_wave_metadata(int fd, struct mp3entry* id3)
|
||||
{
|
||||
/* Use the trackname part of the id3 structure as a temporary buffer */
|
||||
|
|
@ -166,24 +198,7 @@ bool get_wave_metadata(int fd, struct mp3entry* id3)
|
|||
offset += read_bytes;
|
||||
i -= read_bytes;
|
||||
|
||||
/* wFormatTag */
|
||||
fmt.formattag = buf[0] | (buf[1] << 8);
|
||||
/* wChannels */
|
||||
fmt.channels = buf[2] | (buf[3] << 8);
|
||||
/* dwSamplesPerSec */
|
||||
id3->frequency = get_long_le(&buf[4]);
|
||||
/* dwAvgBytesPerSec */
|
||||
id3->bitrate = (get_long_le(&buf[8]) * 8) / 1000;
|
||||
/* wBlockAlign */
|
||||
fmt.blockalign = buf[12] | (buf[13] << 8);
|
||||
id3->bytesperframe = fmt.blockalign;
|
||||
/* wBitsPerSample */
|
||||
fmt.bitspersample = buf[14] | (buf[15] << 8);
|
||||
if (read_bytes > 19)
|
||||
{
|
||||
/* wSamplesPerBlock */
|
||||
fmt.samplesperblock = buf[18] | (buf[19] << 8);
|
||||
}
|
||||
parse_riff_format(buf, i, &fmt, id3);
|
||||
|
||||
/* Check for ATRAC3 stream */
|
||||
if (fmt.formattag == WAVE_FORMAT_ATRAC3)
|
||||
|
|
@ -256,3 +271,96 @@ bool get_wave_metadata(int fd, struct mp3entry* id3)
|
|||
|
||||
return true;
|
||||
}
|
||||
|
||||
bool get_wave64_metadata(int fd, struct mp3entry* id3)
|
||||
{
|
||||
/* Use the trackname part of the id3 structure as a temporary buffer */
|
||||
unsigned char* buf = (unsigned char *)id3->path;
|
||||
struct wave_fmt fmt;
|
||||
unsigned long totalsamples = 0;
|
||||
int read_bytes;
|
||||
uint64_t i;
|
||||
|
||||
memset(&fmt, 0, sizeof(struct wave_fmt));
|
||||
|
||||
/* get RIFF chunk header */
|
||||
if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, buf, 40) < 40))
|
||||
return false;
|
||||
|
||||
if ((memcmp(buf , WAVE64_GUID_RIFF, 16) != 0)||
|
||||
(memcmp(buf+24, WAVE64_GUID_WAVE, 16) != 0))
|
||||
{
|
||||
DEBUGF("metada error: does not wave64 file\n");
|
||||
return false;
|
||||
}
|
||||
|
||||
/* iterate over WAVE chunks until 'data' chunk */
|
||||
while (true)
|
||||
{
|
||||
/* get chunk header */
|
||||
if (read(fd, buf, 24) < 24)
|
||||
return false;
|
||||
|
||||
/* chunkSize (excludes GUID and size length) */
|
||||
i = get_uint64_le(&buf[16]) - 24;
|
||||
|
||||
if (memcmp(buf, WAVE64_GUID_FMT, 16) == 0)
|
||||
{
|
||||
DEBUGF("find 'fmt ' chunk\n");
|
||||
if (i < 16)
|
||||
return false;
|
||||
|
||||
read_bytes = 16;
|
||||
if (i > 16)
|
||||
{
|
||||
read_bytes = 24;
|
||||
if (i < 24)
|
||||
i = 24;
|
||||
}
|
||||
|
||||
/* get rest of chunk */
|
||||
if (read(fd, buf, read_bytes) < read_bytes)
|
||||
return false;
|
||||
|
||||
i -= read_bytes;
|
||||
parse_riff_format(buf, read_bytes, &fmt, id3);
|
||||
}
|
||||
else if (memcmp(buf, WAVE64_GUID_DATA, 16) == 0)
|
||||
{
|
||||
DEBUGF("find 'data' chunk\n");
|
||||
fmt.numbytes = i;
|
||||
break;
|
||||
}
|
||||
else if (memcmp(buf, WAVE64_GUID_FACT, 16) == 0)
|
||||
{
|
||||
/* Skip 'fact' chunk */
|
||||
DEBUGF("find 'fact' chunk\n");
|
||||
}
|
||||
|
||||
/* seek to next chunk (8byte bound) */
|
||||
if (i & 0x07)
|
||||
i += 8 - (i & 0x7);
|
||||
|
||||
if(lseek(fd, i, SEEK_CUR) < 0)
|
||||
return false;
|
||||
}
|
||||
|
||||
if ((fmt.numbytes == 0) || (fmt.channels == 0) || (fmt.blockalign == 0))
|
||||
{
|
||||
DEBUGF("metadata error: numbytes, channels, or blockalign is 0\n");
|
||||
return false;
|
||||
}
|
||||
|
||||
if (totalsamples == 0)
|
||||
{
|
||||
totalsamples = get_totalsamples(&fmt, id3);
|
||||
}
|
||||
|
||||
id3->vbr = false; /* All Wave64 files are CBR */
|
||||
id3->filesize = filesize(fd);
|
||||
|
||||
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
|
||||
id3->length = ((int64_t) totalsamples * 1000) / id3->frequency;
|
||||
|
||||
return true;
|
||||
}
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue