2acc0ac542
later. We still need to hunt down snippets used that are not. 1324 modified files... http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
727 lines
28 KiB
C
727 lines
28 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2005 Dave Chapman
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*
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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 "inttypes.h"
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CODEC_HEADER
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/* Macro that sign extends an unsigned byte */
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#define SE(x) ((int32_t)((int8_t)(x)))
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/* This codec support WAVE files with the following formats:
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* - PCM, up to 32 bits, supporting 32 bits playback when useful.
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* - ALAW and MULAW (16 bits compressed on 8 bits).
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* - DVI_ADPCM (16 bits compressed on 3 or 4 bits).
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*
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* For a good documentation on WAVE files, see:
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* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/WAVE.html
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* and
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* http://www.sonicspot.com/guide/wavefiles.html
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*
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* For sample WAV files, see:
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* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/Samples.html
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*
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* The most common formats seem to be PCM, ADPCM, DVI_ADPCM, IEEE_FLOAT,
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* ALAW and MULAW
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*/
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/* These constants are from RFC 2361. */
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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_VSELP = 0x0004, /* Compaq Computer's VSELP */
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WAVE_FORMAT_IBM_CVSD = 0x0005, /* IBM CVSD */
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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_OKI_ADPCM = 0x0010, /* OKI ADPCM */
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WAVE_FORMAT_DVI_ADPCM = 0x0011, /* Intel's DVI ADPCM */
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WAVE_FORMAT_MEDIASPACE_ADPCM = 0x0012, /* Videologic's MediaSpace ADPCM */
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WAVE_FORMAT_SIERRA_ADPCM = 0x0013, /* Sierra ADPCM */
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WAVE_FORMAT_G723_ADPCM = 0x0014, /* G.723 ADPCM */
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WAVE_FORMAT_DIGISTD = 0x0015, /* DSP Solutions' DIGISTD */
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WAVE_FORMAT_DIGIFIX = 0x0016, /* DSP Solutions' DIGIFIX */
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WAVE_FORMAT_DIALOGIC_OKI_ADPCM = 0x0017, /* Dialogic OKI ADPCM */
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WAVE_FORMAT_MEDIAVISION_ADPCM = 0x0018, /* MediaVision ADPCM */
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WAVE_FORMAT_CU_CODEC = 0x0019, /* HP CU */
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WAVE_FORMAT_YAMAHA_ADPCM = 0x0020, /* Yamaha ADPCM */
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WAVE_FORMAT_SONARC = 0x0021, /* Speech Compression's Sonarc */
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WAVE_FORMAT_DSP_TRUESPEECH = 0x0022, /* DSP Group's True Speech */
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WAVE_FORMAT_ECHOSC1 = 0x0023, /* Echo Speech's EchoSC1 */
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WAVE_FORMAT_AUDIOFILE_AF36 = 0x0024, /* Audiofile AF36 */
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WAVE_FORMAT_APTX = 0x0025, /* APTX */
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WAVE_FORMAT_DOLBY_AC2 = 0x0030, /* Dolby AC2 */
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WAVE_FORMAT_GSM610 = 0x0031, /* GSM610 */
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WAVE_FORMAT_MSNAUDIO = 0x0032, /* MSNAudio */
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WAVE_FORMAT_ANTEX_ADPCME = 0x0033, /* Antex ADPCME */
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WAVE_FORMAT_MPEG = 0x0050, /* MPEG */
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WAVE_FORMAT_MPEGLAYER3 = 0x0055, /* MPEG layer 3 */
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WAVE_FORMAT_LUCENT_G723 = 0x0059, /* Lucent G.723 */
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WAVE_FORMAT_G726_ADPCM = 0x0064, /* G.726 ADPCM */
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WAVE_FORMAT_G722_ADPCM = 0x0065, /* G.722 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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IBM_FORMAT_ADPCM = 0x0103,
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WAVE_FORMAT_CREATIVE_ADPCM = 0x0200,
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WAVE_FORMAT_EXTENSIBLE = 0xFFFE
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};
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/* Maximum number of bytes to process in one iteration */
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/* for 44.1kHz stereo 16bits, this represents 0.023s ~= 1/50s */
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#define WAV_CHUNK_SIZE (1024*2)
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static const int16_t alaw2linear16[256] ICONST_ATTR = {
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-5504, -5248, -6016, -5760, -4480, -4224, -4992,
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-4736, -7552, -7296, -8064, -7808, -6528, -6272,
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-7040, -6784, -2752, -2624, -3008, -2880, -2240,
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-2112, -2496, -2368, -3776, -3648, -4032, -3904,
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-3264, -3136, -3520, -3392, -22016, -20992, -24064,
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-23040, -17920, -16896, -19968, -18944, -30208, -29184,
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-32256, -31232, -26112, -25088, -28160, -27136, -11008,
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-10496, -12032, -11520, -8960, -8448, -9984, -9472,
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-15104, -14592, -16128, -15616, -13056, -12544, -14080,
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-13568, -344, -328, -376, -360, -280, -264,
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-312, -296, -472, -456, -504, -488, -408,
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-392, -440, -424, -88, -72, -120, -104,
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-24, -8, -56, -40, -216, -200, -248,
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-232, -152, -136, -184, -168, -1376, -1312,
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-1504, -1440, -1120, -1056, -1248, -1184, -1888,
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-1824, -2016, -1952, -1632, -1568, -1760, -1696,
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-688, -656, -752, -720, -560, -528, -624,
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-592, -944, -912, -1008, -976, -816, -784,
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-880, -848, 5504, 5248, 6016, 5760, 4480,
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4224, 4992, 4736, 7552, 7296, 8064, 7808,
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6528, 6272, 7040, 6784, 2752, 2624, 3008,
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2880, 2240, 2112, 2496, 2368, 3776, 3648,
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4032, 3904, 3264, 3136, 3520, 3392, 22016,
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20992, 24064, 23040, 17920, 16896, 19968, 18944,
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30208, 29184, 32256, 31232, 26112, 25088, 28160,
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27136, 11008, 10496, 12032, 11520, 8960, 8448,
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9984, 9472, 15104, 14592, 16128, 15616, 13056,
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12544, 14080, 13568, 344, 328, 376, 360,
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280, 264, 312, 296, 472, 456, 504,
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488, 408, 392, 440, 424, 88, 72,
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120, 104, 24, 8, 56, 40, 216,
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200, 248, 232, 152, 136, 184, 168,
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1376, 1312, 1504, 1440, 1120, 1056, 1248,
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1184, 1888, 1824, 2016, 1952, 1632, 1568,
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1760, 1696, 688, 656, 752, 720, 560,
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528, 624, 592, 944, 912, 1008, 976,
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816, 784, 880, 848
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};
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static const int16_t ulaw2linear16[256] ICONST_ATTR = {
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-32124, -31100, -30076, -29052, -28028, -27004, -25980,
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-24956, -23932, -22908, -21884, -20860, -19836, -18812,
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-17788, -16764, -15996, -15484, -14972, -14460, -13948,
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-13436, -12924, -12412, -11900, -11388, -10876, -10364,
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-9852, -9340, -8828, -8316, -7932, -7676, -7420,
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-7164, -6908, -6652, -6396, -6140, -5884, -5628,
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-5372, -5116, -4860, -4604, -4348, -4092, -3900,
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-3772, -3644, -3516, -3388, -3260, -3132, -3004,
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-2876, -2748, -2620, -2492, -2364, -2236, -2108,
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-1980, -1884, -1820, -1756, -1692, -1628, -1564,
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-1500, -1436, -1372, -1308, -1244, -1180, -1116,
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-1052, -988, -924, -876, -844, -812, -780,
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-748, -716, -684, -652, -620, -588, -556,
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-524, -492, -460, -428, -396, -372, -356,
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-340, -324, -308, -292, -276, -260, -244,
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-228, -212, -196, -180, -164, -148, -132,
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-120, -112, -104, -96, -88, -80, -72,
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-64, -56, -48, -40, -32, -24, -16,
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-8, 0, 32124, 31100, 30076, 29052, 28028,
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27004, 25980, 24956, 23932, 22908, 21884, 20860,
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19836, 18812, 17788, 16764, 15996, 15484, 14972,
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14460, 13948, 13436, 12924, 12412, 11900, 11388,
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10876, 10364, 9852, 9340, 8828, 8316, 7932,
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7676, 7420, 7164, 6908, 6652, 6396, 6140,
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5884, 5628, 5372, 5116, 4860, 4604, 4348,
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4092, 3900, 3772, 3644, 3516, 3388, 3260,
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3132, 3004, 2876, 2748, 2620, 2492, 2364,
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2236, 2108, 1980, 1884, 1820, 1756, 1692,
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1628, 1564, 1500, 1436, 1372, 1308, 1244,
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1180, 1116, 1052, 988, 924, 876, 844,
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812, 780, 748, 716, 684, 652, 620,
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588, 556, 524, 492, 460, 428, 396,
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372, 356, 340, 324, 308, 292, 276,
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260, 244, 228, 212, 196, 180, 164,
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148, 132, 120, 112, 104, 96, 88,
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80, 72, 64, 56, 48, 40, 32,
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24, 16, 8, 0
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};
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static const uint16_t dvi_adpcm_steptab[89] ICONST_ATTR = {
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7, 8, 9, 10, 11, 12, 13, 14,
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16, 17, 19, 21, 23, 25, 28, 31,
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34, 37, 41, 45, 50, 55, 60, 66,
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73, 80, 88, 97, 107, 118, 130, 143,
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157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658,
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724, 796, 876, 963, 1060, 1166, 1282, 1411,
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1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
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3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
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7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
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32767 };
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static const int dvi_adpcm_indextab4[8] ICONST_ATTR = {
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-1, -1, -1, -1, 2, 4, 6, 8 };
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static const int dvi_adpcm_indextab3[4] ICONST_ATTR = { -1, -1, 1, 2 };
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static int32_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
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static enum codec_status
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decode_dvi_adpcm(struct codec_api *ci,
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const uint8_t *buf,
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int n,
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uint16_t channels, uint16_t bitspersample,
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int32_t *pcmout,
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size_t *pcmoutsize);
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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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uint32_t numbytes, bytesdone;
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uint32_t totalsamples = 0;
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uint16_t channels = 0;
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uint16_t samplesperblock = 0;
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int bytespersample = 0;
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uint16_t bitspersample;
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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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long chunksize;
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uint16_t formattag = 0;
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uint16_t blockalign = 0;
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uint32_t avgbytespersec = 0;
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off_t firstblockposn; /* position of the first block in file */
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/* Generic codec initialisation */
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ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
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ci->configure(CODEC_SET_FILEBUF_WATERMARK, 1024*512);
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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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/* 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, 12);
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if (n < 12) {
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i = CODEC_ERROR;
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goto done;
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}
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if ((memcmp(buf, "RIFF", 4) != 0) || (memcmp(&buf[8], "WAVE", 4) != 0)) {
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i = 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(12);
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firstblockposn = 12;
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bitspersample = 0;
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numbytes = 0;
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totalsamples = 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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/* no more chunks, 'data' chunk must not have been found */
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i = CODEC_ERROR;
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goto done;
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}
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/* chunkSize */
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i = (buf[4]|(buf[5]<<8)|(buf[6]<<16)|(buf[7]<<24));
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if (memcmp(buf, "fmt ", 4) == 0) {
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if (i < 16) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%lu < 16\n",
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(unsigned long)i);
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i = CODEC_ERROR;
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goto done;
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}
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/* wFormatTag */
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formattag=buf[8]|(buf[9]<<8);
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/* wChannels */
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channels=buf[10]|(buf[11]<<8);
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/* skipping dwSamplesPerSec */
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/* dwAvgBytesPerSec */
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avgbytespersec = buf[16]|(buf[17]<<8)|(buf[18]<<16)|(buf[19]<<24);
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/* wBlockAlign */
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blockalign=buf[20]|(buf[21]<<8);
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/* wBitsPerSample */
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bitspersample=buf[22]|(buf[23]<<8);
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if (formattag != WAVE_FORMAT_PCM) {
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uint16_t size;
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if (i < 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=%ld<18).\n",
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formattag, (long)i);
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}
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size = buf[24]|(buf[25]<<8);
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if (formattag == WAVE_FORMAT_DVI_ADPCM) {
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if (size < 2) {
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DEBUGF("CODEC_ERROR: dvi_adpcm is missing "
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"SamplesPerBlock value\n");
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i = CODEC_ERROR;
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goto done;
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}
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samplesperblock = buf[26]|(buf[27]<<8);
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} else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
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if (size < 22) {
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DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
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"missing extension\n");
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i = CODEC_ERROR;
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goto done;
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}
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/* wValidBitsPerSample */
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bitspersample = buf[26]|(buf[27]<<8);
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/* skipping dwChannelMask (4bytes) */
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/* SubFormat (only get the first two bytes) */
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formattag = buf[32]|(buf[33]<<8);
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}
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}
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} else if (memcmp(buf, "data", 4) == 0) {
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numbytes = i;
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/* advance to start of data */
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ci->advance_buffer(8);
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firstblockposn += 8;
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break;
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} else if (memcmp(buf, "fact", 4) == 0) {
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/* dwSampleLength */
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if (i >= 4)
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totalsamples = (buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24));
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} else {
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DEBUGF("unknown WAVE chunk: '%c%c%c%c', size=%lu\n",
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buf[0], buf[1], buf[2], buf[3], (unsigned long)i);
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}
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/* go to next chunk (even chunk sizes must be padded) */
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if (i & 0x01)
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i++;
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ci->advance_buffer(i+8);
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firstblockposn += i + 8;
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}
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if (channels == 0) {
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DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n");
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i = CODEC_ERROR;
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goto done;
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}
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if (numbytes == 0) {
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DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
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i = CODEC_ERROR;
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goto done;
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}
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if (formattag != WAVE_FORMAT_PCM && totalsamples == 0) {
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/* This is non-fatal for some formats */
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DEBUGF("CODEC_WARNING: non-PCM WAVE doesn't have a 'fact' chunk\n");
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}
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if (formattag == WAVE_FORMAT_ALAW || formattag == WAVE_FORMAT_MULAW ||
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formattag == IBM_FORMAT_ALAW || formattag == IBM_FORMAT_MULAW) {
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if (bitspersample != 8) {
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DEBUGF("CODEC_ERROR: alaw and mulaw must have 8 bitspersample\n");
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i = CODEC_ERROR;
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goto done;
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}
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bytespersample = channels;
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}
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if (formattag == WAVE_FORMAT_DVI_ADPCM
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&& bitspersample != 4 && bitspersample != 3) {
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DEBUGF("CODEC_ERROR: dvi_adpcm must have 3 or 4 bitspersample\n");
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i = CODEC_ERROR;
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goto done;
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}
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if (formattag == WAVE_FORMAT_PCM && bitspersample > 32) {
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DEBUGF("CODEC_ERROR: pcm with more than 32 bitspersample "
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"is unsupported\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 (channels == 2) {
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|
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
|
|
} else if (channels == 1) {
|
|
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
|
|
} else {
|
|
DEBUGF("CODEC_ERROR: more than 2 channels\n");
|
|
i = CODEC_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
if (totalsamples == 0) {
|
|
if (formattag == WAVE_FORMAT_PCM ||
|
|
formattag == WAVE_FORMAT_ALAW || formattag == WAVE_FORMAT_MULAW ||
|
|
formattag == IBM_FORMAT_ALAW || formattag == IBM_FORMAT_MULAW) {
|
|
/* for PCM and derived formats only */
|
|
bytespersample = (((bitspersample - 1)/8 + 1)*channels);
|
|
totalsamples = numbytes/bytespersample;
|
|
} else {
|
|
DEBUGF("CODEC_ERROR: cannot compute totalsamples\n");
|
|
i = CODEC_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* make sure we're at the correct offset */
|
|
if (bytesdone > (uint32_t) firstblockposn) {
|
|
/* Round down to previous block */
|
|
uint32_t offset = bytesdone - bytesdone % blockalign;
|
|
|
|
ci->advance_buffer(offset-firstblockposn);
|
|
bytesdone = offset - firstblockposn;
|
|
} else {
|
|
/* already where we need to be */
|
|
bytesdone = 0;
|
|
}
|
|
|
|
/* The main decoder loop */
|
|
endofstream = 0;
|
|
/* chunksize is computed so that one chunk is about 1/50s.
|
|
* this make 4096 for 44.1kHz 16bits stereo.
|
|
* It also has to be a multiple of blockalign */
|
|
chunksize = (1 + avgbytespersec / (50*blockalign))*blockalign;
|
|
/* check that the output buffer is big enough (convert to samplespersec,
|
|
then round to the blockalign multiple below) */
|
|
if (((uint64_t)chunksize*ci->id3->frequency*channels*2)
|
|
/(uint64_t)avgbytespersec >= WAV_CHUNK_SIZE) {
|
|
chunksize = ((uint64_t)WAV_CHUNK_SIZE*avgbytespersec
|
|
/((uint64_t)ci->id3->frequency*channels*2
|
|
*blockalign))*blockalign;
|
|
}
|
|
|
|
while (!endofstream) {
|
|
ci->yield();
|
|
if (ci->stop_codec || ci->new_track) {
|
|
break;
|
|
}
|
|
|
|
if (ci->seek_time) {
|
|
uint32_t newpos;
|
|
|
|
/* use avgbytespersec to round to the closest blockalign multiple,
|
|
add firstblockposn. 64-bit casts to avoid overflows. */
|
|
newpos = (((uint64_t)avgbytespersec*(ci->seek_time - 1))
|
|
/ (1000LL*blockalign))*blockalign;
|
|
if (newpos > numbytes)
|
|
break;
|
|
if (ci->seek_buffer(firstblockposn + newpos))
|
|
bytesdone = newpos;
|
|
ci->seek_complete();
|
|
}
|
|
wavbuf = (uint8_t *)ci->request_buffer(&n, chunksize);
|
|
|
|
if (n == 0)
|
|
break; /* End of stream */
|
|
|
|
if (bytesdone + n > numbytes) {
|
|
n = numbytes - bytesdone;
|
|
endofstream = 1;
|
|
}
|
|
|
|
if (formattag == WAVE_FORMAT_PCM) {
|
|
if (bitspersample > 24) {
|
|
for (i = 0; i < n; i += 4) {
|
|
samples[i/4] = (wavbuf[i] >> 3)|
|
|
(wavbuf[i + 1]<<5)|(wavbuf[i + 2]<<13)|
|
|
(SE(wavbuf[i + 3])<<21);
|
|
}
|
|
bufcount = n >> 2;
|
|
} else if (bitspersample > 16) {
|
|
for (i = 0; i < n; i += 3) {
|
|
samples[i/3] = (wavbuf[i]<<5)|
|
|
(wavbuf[i + 1]<<13)|(SE(wavbuf[i + 2])<<21);
|
|
}
|
|
bufcount = n/3;
|
|
} else if (bitspersample > 8) {
|
|
for (i = 0; i < n; i += 2) {
|
|
samples[i/2] = (wavbuf[i]<<13)|(SE(wavbuf[i + 1])<<21);
|
|
}
|
|
bufcount = n >> 1;
|
|
} else {
|
|
for (i = 0; i < n; i++) {
|
|
samples[i] = (wavbuf[i] - 0x80)<<21;
|
|
}
|
|
bufcount = n;
|
|
}
|
|
|
|
if (channels == 2)
|
|
bufcount >>= 1;
|
|
} else if (formattag == WAVE_FORMAT_ALAW
|
|
|| formattag == IBM_FORMAT_ALAW) {
|
|
for (i = 0; i < n; i++)
|
|
samples[i] = alaw2linear16[wavbuf[i]] << 13;
|
|
|
|
bufcount = (channels == 2) ? (n >> 1) : n;
|
|
} else if (formattag == WAVE_FORMAT_MULAW
|
|
|| formattag == IBM_FORMAT_MULAW) {
|
|
for (i = 0; i < n; i++)
|
|
samples[i] = ulaw2linear16[wavbuf[i]] << 13;
|
|
|
|
bufcount = (channels == 2) ? (n >> 1) : n;
|
|
}
|
|
else if (formattag == WAVE_FORMAT_DVI_ADPCM) {
|
|
unsigned int nblocks = chunksize/blockalign;
|
|
|
|
for (i = 0; i < nblocks; i++) {
|
|
size_t decodedsize = samplesperblock*channels;
|
|
if (decode_dvi_adpcm(ci, wavbuf + i*blockalign,
|
|
blockalign, channels, bitspersample,
|
|
samples + i*samplesperblock*channels,
|
|
&decodedsize) != CODEC_OK) {
|
|
i = CODEC_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
bufcount = nblocks*samplesperblock;
|
|
} else {
|
|
DEBUGF("CODEC_ERROR: unsupported format %x\n", formattag);
|
|
i = CODEC_ERROR;
|
|
goto done;
|
|
}
|
|
|
|
ci->pcmbuf_insert(samples, NULL, bufcount);
|
|
|
|
ci->advance_buffer(n);
|
|
bytesdone += n;
|
|
if (bytesdone >= numbytes)
|
|
endofstream = 1;
|
|
ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
|
|
}
|
|
i = CODEC_OK;
|
|
|
|
done:
|
|
if (ci->request_next_track())
|
|
goto next_track;
|
|
|
|
exit:
|
|
return i;
|
|
}
|
|
|
|
static enum codec_status
|
|
decode_dvi_adpcm(struct codec_api *ci,
|
|
const uint8_t *buf,
|
|
int n,
|
|
uint16_t channels, uint16_t bitspersample,
|
|
int32_t *pcmout,
|
|
size_t *pcmoutsize)
|
|
{
|
|
size_t nsamples = 0;
|
|
int sample[2];
|
|
int samplecode[32][2];
|
|
int i;
|
|
int stepindex[2];
|
|
int c;
|
|
int diff;
|
|
int step;
|
|
int codem;
|
|
int code;
|
|
|
|
(void)ci;
|
|
if (bitspersample != 4 && bitspersample != 3) {
|
|
DEBUGF("decode_dvi_adpcm: wrong bitspersample\n");
|
|
return CODEC_ERROR;
|
|
}
|
|
|
|
/* decode block header */
|
|
for (c = 0; c < channels && n >= 4; c++) {
|
|
/* decode + push first sample */
|
|
sample[c] = (short)(buf[0]|(buf[1]<<8));/* need cast for sign-extend */
|
|
pcmout[c] = sample[c] << 13;
|
|
nsamples++;
|
|
stepindex[c] = buf[2];
|
|
/* check for step table index overflow */
|
|
if (stepindex[c] > 88) {
|
|
DEBUGF("decode_dvi_adpcm: stepindex[%d]=%d>88\n",c,stepindex[c]);
|
|
return CODEC_ERROR;
|
|
}
|
|
|
|
buf += 4;
|
|
n -= 4;
|
|
}
|
|
if (bitspersample == 4) {
|
|
while (n>= channels*4 && (nsamples + 8*channels) <= *pcmoutsize) {
|
|
for (c = 0; c < channels; c++) {
|
|
samplecode[0][c] = buf[0]&0xf;
|
|
samplecode[1][c] = buf[0]>>4;
|
|
samplecode[2][c] = buf[1]&0xf;
|
|
samplecode[3][c] = buf[1]>>4;
|
|
samplecode[4][c] = buf[2]&0xf;
|
|
samplecode[5][c] = buf[2]>>4;
|
|
samplecode[6][c] = buf[3]&0xf;
|
|
samplecode[7][c] = buf[3]>>4;
|
|
buf += 4;
|
|
n -= 4;
|
|
}
|
|
for (i = 0; i < 8; i++) {
|
|
for (c = 0; c < channels; c++) {
|
|
step = dvi_adpcm_steptab[stepindex[c]];
|
|
codem = samplecode[i][c];
|
|
code = codem & 0x07;
|
|
|
|
/* adjust the step table index */
|
|
stepindex[c] += dvi_adpcm_indextab4[code];
|
|
/* check for step table index overflow and underflow */
|
|
if (stepindex[c] > 88)
|
|
stepindex[c] = 88;
|
|
else if (stepindex[c] < 0)
|
|
stepindex[c] = 0;
|
|
/* calculate the difference */
|
|
#ifdef STRICT_IMA
|
|
diff = 0;
|
|
if (code & 4)
|
|
diff += step;
|
|
step = step >> 1;
|
|
if (code & 2)
|
|
diff += step;
|
|
step = step >> 1;
|
|
if (code & 1)
|
|
diff += step;
|
|
step = step >> 1;
|
|
diff += step;
|
|
#else
|
|
diff = ((code + code + 1) * step) >> 3; /* faster */
|
|
#endif
|
|
/* check the sign bit */
|
|
/* check for overflow and underflow errors */
|
|
if (code != codem) {
|
|
sample[c] -= diff;
|
|
if (sample[c] < -32768)
|
|
sample[c] = -32768;
|
|
} else {
|
|
sample[c] += diff;
|
|
if (sample[c] > 32767)
|
|
sample[c] = 32767;
|
|
}
|
|
/* output the new sample */
|
|
pcmout[nsamples] = sample[c] << 13;
|
|
nsamples++;
|
|
}
|
|
}
|
|
}
|
|
} else { /* bitspersample == 3 */
|
|
while (n >= channels*12 && (nsamples + 32*channels) <= *pcmoutsize) {
|
|
for (c = 0; c < channels; c++) {
|
|
uint16_t bitstream = 0;
|
|
int bitsread = 0;
|
|
for (i = 0; i < 32 && n > 0; i++) {
|
|
if (bitsread < 3) {
|
|
/* read 8 more bits */
|
|
bitstream |= buf[0]<<bitsread;
|
|
bitsread += 8;
|
|
n--;
|
|
buf++;
|
|
}
|
|
samplecode[i][c] = bitstream & 7;
|
|
bitstream = bitstream>>3;
|
|
bitsread -= 3;
|
|
}
|
|
if (bitsread != 0) {
|
|
/* 32*3 = 3 words, so we should end with bitsread==0 */
|
|
DEBUGF("decode_dvi_adpcm: error in implementation\n");
|
|
return CODEC_ERROR;
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < 32; i++) {
|
|
for (c = 0; c < channels; c++) {
|
|
step = dvi_adpcm_steptab[stepindex[c]];
|
|
codem = samplecode[i][c];
|
|
code = codem & 0x03;
|
|
|
|
/* adjust the step table index */
|
|
stepindex[c] += dvi_adpcm_indextab3[code];
|
|
/* check for step table index overflow and underflow */
|
|
if (stepindex[c] > 88)
|
|
stepindex[c] = 88;
|
|
else if (stepindex[c] < 0)
|
|
stepindex[c] = 0;
|
|
/* calculate the difference */
|
|
#ifdef STRICT_IMA
|
|
diff = 0;
|
|
if (code & 2)
|
|
diff += step;
|
|
step = step >> 1;
|
|
if (code & 1)
|
|
diff += step;
|
|
step = step >> 1;
|
|
diff += step;
|
|
#else
|
|
diff = ((code + code + 1) * step) >> 3; /* faster */
|
|
#endif
|
|
/* check the sign bit */
|
|
/* check for overflow and underflow errors */
|
|
if (code != codem) {
|
|
sample[c] -= diff;
|
|
if (sample[c] < -32768)
|
|
sample[c] = -32768;
|
|
}
|
|
else {
|
|
sample[c] += diff;
|
|
if (sample[c] > 32767)
|
|
sample[c] = 32767;
|
|
}
|
|
/* output the new sample */
|
|
pcmout[nsamples] = sample[c] << 13;
|
|
nsamples++;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (nsamples > *pcmoutsize) {
|
|
DEBUGF("decode_dvi_adpcm: output buffer overflow!\n");
|
|
return CODEC_ERROR;
|
|
}
|
|
*pcmoutsize = nsamples;
|
|
if (n != 0) {
|
|
DEBUGF("decode_dvi_adpcm: n=%d unprocessed bytes\n", n);
|
|
}
|
|
return CODEC_OK;
|
|
}
|
|
|