6c868dd48f
Just use long so the compiler potentially doesn't complain about use of other values not in the enum. It's also the type used around the system for event ids. Increase min codec API version. No functional changes. Change-Id: If4419b42912f5e4ef673adcdeb69313e503f94cc
325 lines
9.6 KiB
C
325 lines
9.6 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2010 Yoshihisa Uchida
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "codeclib.h"
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#include "codecs/libpcm/support_formats.h"
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CODEC_HEADER
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/* Sun Audio file (Au file format) codec
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*
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* References
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* [1] Sun Microsystems, Inc., Header file for Audio, .au, 1992
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* URL http://www.opengroup.org/public/pubs/external/auformat.html
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* [2] Wikipedia, Au file format, URL: http://en.wikipedia.org/wiki/Sun_Audio
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*/
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#define PCM_SAMPLE_SIZE (1024*2)
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static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
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enum
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{
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AU_FORMAT_UNSUPPORT = 0, /* unsupported format */
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AU_FORMAT_MULAW, /* G.711 MULAW */
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AU_FORMAT_PCM, /* Linear PCM */
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AU_FORMAT_IEEE_FLOAT, /* IEEE float */
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AU_FORMAT_ALAW, /* G.711 ALAW */
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};
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static const char support_formats[9][2] = {
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{ AU_FORMAT_UNSUPPORT, 0 }, /* encoding */
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{ AU_FORMAT_MULAW, 8 }, /* 1: G.711 MULAW */
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{ AU_FORMAT_PCM, 8 }, /* 2: Linear PCM 8bit (signed) */
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{ AU_FORMAT_PCM, 16 }, /* 3: Linear PCM 16bit (signed, big endian) */
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{ AU_FORMAT_PCM, 24 }, /* 4: Linear PCM 24bit (signed, big endian) */
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{ AU_FORMAT_PCM, 32 }, /* 5: Linear PCM 32bit (signed, big endian) */
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{ AU_FORMAT_IEEE_FLOAT, 32 }, /* 6: Linear PCM float 32bit (signed, big endian) */
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{ AU_FORMAT_IEEE_FLOAT, 64 }, /* 7: Linear PCM float 64bit (signed, big endian) */
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/* encoding 8 - 26 unsupported. */
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{ AU_FORMAT_ALAW, 8 }, /* 27: G.711 ALAW */
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};
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static const struct pcm_entry au_codecs[] = {
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{ AU_FORMAT_MULAW, get_itut_g711_mulaw_codec },
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{ AU_FORMAT_PCM, get_linear_pcm_codec },
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{ AU_FORMAT_IEEE_FLOAT, get_ieee_float_codec },
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{ AU_FORMAT_ALAW, get_itut_g711_alaw_codec },
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};
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#define NUM_FORMATS 4
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static const struct pcm_codec *get_au_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 (au_codecs[i].format_tag == formattag)
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{
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if (au_codecs[i].get_codec)
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return au_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_au_format(unsigned int encoding, struct libpcm_pcm_format *fmt)
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{
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fmt->formattag = AU_FORMAT_UNSUPPORT;
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if (encoding < 8)
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{
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fmt->formattag = support_formats[encoding][0];
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fmt->bitspersample = support_formats[encoding][1];
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}
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else if (encoding == 27)
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{
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fmt->formattag = support_formats[8][0];
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fmt->bitspersample = support_formats[8][1];
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}
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return fmt->formattag;
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}
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/* this is the codec entry point */
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enum codec_status codec_main(enum codec_entry_call_reason reason)
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{
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if (reason == CODEC_LOAD) {
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/* Generic codec initialisation */
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ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
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}
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return CODEC_OK;
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}
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/* this is called for each file to process */
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enum codec_status codec_run(void)
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{
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struct libpcm_pcm_format format;
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uint32_t bytesdone, decodedsamples;
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size_t n;
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int bufcount;
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int endofstream;
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unsigned char *buf;
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uint8_t *aubuf;
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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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int offset = 0;
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intptr_t param;
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if (codec_init()) {
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DEBUGF("codec_init() error\n");
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return CODEC_ERROR;
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}
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codec_set_replaygain(ci->id3);
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/* Need to save offset for later use (cleared indirectly by advance_buffer) */
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param = ci->id3->elapsed;
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bytesdone = ci->id3->offset;
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ci->memset(&format, 0, sizeof(struct libpcm_pcm_format));
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format.is_signed = true;
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format.is_little_endian = false;
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/* set format */
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ci->seek_buffer(0);
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buf = ci->request_buffer(&n, 24);
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if (n < 24 || (memcmp(buf, ".snd", 4) != 0))
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{
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/*
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* headerless sun audio file
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* It is decoded under conditions.
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* format: G.711 mu-law
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* channel: mono
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* frequency: 8000 kHz
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*/
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offset = 0;
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format.formattag = AU_FORMAT_MULAW;
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format.channels = 1;
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format.bitspersample = 8;
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format.numbytes = ci->id3->filesize;
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}
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else
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{
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/* parse header */
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/* data offset */
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offset = get_be32(buf + 4);
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if (offset < 24)
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{
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DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
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return CODEC_ERROR;
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}
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/* data size */
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format.numbytes = get_be32(buf + 8);
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if (format.numbytes == (uint32_t)0xffffffff)
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format.numbytes = ci->id3->filesize - offset;
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/* encoding */
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format.formattag = convert_au_format(get_be32(buf + 12), &format);
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if (format.formattag == AU_FORMAT_UNSUPPORT)
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{
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DEBUGF("CODEC_ERROR: sun audio unsupport format: %d\n", get_be32(buf + 12));
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return CODEC_ERROR;
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}
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/* skip sample rate */
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format.channels = get_be32(buf + 20);
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}
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/* advance to first WAVE chunk */
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ci->advance_buffer(offset);
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firstblockposn = offset;
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decodedsamples = 0;
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codec = 0;
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/* get codec */
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codec = get_au_codec(format.formattag);
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if (!codec)
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{
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DEBUGF("CODEC_ERROR: unsupport sun audio format: %x\n", (int)format.formattag);
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return CODEC_ERROR;
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}
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if (!codec->set_format(&format))
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{
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return CODEC_ERROR;
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}
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if (format.numbytes == 0) {
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DEBUGF("CODEC_ERROR: data size is 0\n");
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return CODEC_ERROR;
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}
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/* check chunksize */
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if ((format.chunksize / format.blockalign) * format.samplesperblock * format.channels
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> PCM_SAMPLE_SIZE)
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format.chunksize = (PCM_SAMPLE_SIZE / format.blockalign) * format.blockalign;
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if (format.chunksize == 0)
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{
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DEBUGF("CODEC_ERROR: chunksize is 0\n");
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return CODEC_ERROR;
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}
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ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
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if (format.channels == 2) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
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} else if (format.channels == 1) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
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} else {
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DEBUGF("CODEC_ERROR: more than 2 channels\n");
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return CODEC_ERROR;
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}
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/* make sure we're at the correct offset */
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if (bytesdone > (uint32_t) firstblockposn || param) {
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uint32_t seek_val;
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int seek_mode;
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if (bytesdone) {
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seek_val = bytesdone - MIN((uint32_t) firstblockposn, bytesdone);
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seek_mode = PCM_SEEK_POS;
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} else {
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seek_val = param;
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seek_mode = PCM_SEEK_TIME;
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}
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/* Round down to previous block */
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struct pcm_pos *newpos = codec->get_seek_pos(seek_val, seek_mode, NULL);
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if (newpos->pos > format.numbytes)
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goto done;
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if (ci->seek_buffer(firstblockposn + newpos->pos))
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{
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bytesdone = newpos->pos;
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decodedsamples = newpos->samples;
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}
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} else {
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/* already where we need to be */
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bytesdone = 0;
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}
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ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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/* The main decoder loop */
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endofstream = 0;
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while (!endofstream) {
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long action = ci->get_command(¶m);
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if (action == CODEC_ACTION_HALT)
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break;
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if (action == CODEC_ACTION_SEEK_TIME) {
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/* 3rd args(read_buffer) is unnecessary in the format which Sun Audio supports. */
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struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME, NULL);
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if (newpos->pos > format.numbytes)
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{
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ci->set_elapsed(ci->id3->length);
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ci->seek_complete();
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break;
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}
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if (ci->seek_buffer(firstblockposn + newpos->pos))
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{
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bytesdone = newpos->pos;
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decodedsamples = newpos->samples;
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}
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ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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ci->seek_complete();
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}
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aubuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
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if (n == 0)
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break; /* End of stream */
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if (bytesdone + n > format.numbytes) {
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n = format.numbytes - bytesdone;
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endofstream = 1;
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}
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if (codec->decode(aubuf, n, samples, &bufcount) == CODEC_ERROR)
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{
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DEBUGF("codec error\n");
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return CODEC_ERROR;
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}
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ci->pcmbuf_insert(samples, NULL, bufcount);
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ci->advance_buffer(n);
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bytesdone += n;
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decodedsamples += bufcount;
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if (bytesdone >= format.numbytes)
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endofstream = 1;
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ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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}
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done:
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return CODEC_OK;
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}
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