forked from len0rd/rockbox
Patch #1421483 - AIFF codec by Jvo Studer
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8524 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Dave Chapman
parent
6479e4c95e
commit
fbd8e5d29c
9 changed files with 407 additions and 2 deletions
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@ -63,6 +63,7 @@ $(OBJDIR)/wavpack.elf: $(OBJDIR)/wavpack.o $(CODECDEPS) $(BUILDDIR)/libwavpack.a
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$(OBJDIR)/alac.elf: $(OBJDIR)/alac.o $(CODECDEPS) $(BUILDDIR)/libalac.a $(BUILDDIR)/libm4a.a
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$(OBJDIR)/aac.elf: $(OBJDIR)/aac.o $(CODECDEPS) $(BUILDDIR)/libfaad.a $(BUILDDIR)/libm4a.a
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$(OBJDIR)/shorten.elf: $(OBJDIR)/shorten.o $(CODECDEPS) $(BUILDDIR)/libffmpegFLAC.a
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$(OBJDIR)/aiff.elf: $(OBJDIR)/aiff.o $(CODECDEPS)
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$(OBJDIR)/%.elf: $(OBJDIR)/%.o $(CODECDEPS)
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$(ELFIT)
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@ -11,4 +11,5 @@ alac.c
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aac.c
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#endif
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shorten.c
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aiff.c
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#endif
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314
apps/codecs/aiff.c
Normal file
314
apps/codecs/aiff.c
Normal file
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@ -0,0 +1,314 @@
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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) 2005 Jvo Studer
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*
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* All files in this archive are subject to the GNU General Public License.
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* See the file COPYING in the source tree root for full license agreement.
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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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struct codec_api* rb;
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/* This codec supports AIFF files with the following formats:
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* - PCM, 8 and 16 bits, mono or stereo
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*/
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enum
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{
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AIFF_FORMAT_PCM = 0x0001, /* AIFF PCM Format (big endian) */
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IEEE_FORMAT_FLOAT = 0x0003, /* IEEE Float */
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AIFF_FORMAT_ALAW = 0x0004, /* AIFC ALaw compressed */
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AIFF_FORMAT_ULAW = 0x0005 /* AIFC uLaw compressed */
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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 AIF_CHUNK_SIZE (1024*2)
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#ifdef USE_IRAM
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extern char iramcopy[];
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extern char iramstart[];
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extern char iramend[];
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extern char iedata[];
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extern char iend[];
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#endif
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static int16_t int16_samples[AIF_CHUNK_SIZE] IBSS_ATTR;
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/* this is the codec entry point */
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enum codec_status codec_start(struct codec_api* api)
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{
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struct codec_api* ci;
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uint32_t numbytes, bytesdone;
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uint16_t numChannels = 0;
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uint32_t numSampleFrames = 0;
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uint16_t sampleSize = 0;
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uint32_t sampleRate = 0;
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uint32_t i;
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size_t n, aifbufsize;
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int endofstream;
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unsigned char* buf;
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uint16_t* aifbuf;
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long chunksize;
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uint32_t offset2snd = 0;
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uint16_t blockSize = 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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int shortorlong = 1; /* do we output shorts (1) or longs (2)? */
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int32_t * const int32_samples = (int32_t*)int16_samples;
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/* Generic codec initialisation */
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rb = api;
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ci = api;
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#ifdef USE_IRAM
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ci->memcpy(iramstart, iramcopy, iramend-iramstart);
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ci->memset(iedata, 0, iend - iedata);
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#endif
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ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
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ci->configure(DSP_DITHER, (bool *)false);
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next_track:
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if (codec_init(api)) {
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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)
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ci->yield();
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/* assume the AIFF header is less than 1024 bytes */
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buf=ci->request_buffer((long *)&n,1024);
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if (n<44) {
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i = CODEC_ERROR;
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goto exit;
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}
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if ((memcmp(buf,"FORM",4)!=0) || (memcmp(&buf[8],"AIFF",4)!=0)) {
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i = CODEC_ERROR;
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goto exit;
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}
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buf += 12;
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n -= 12;
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numbytes = 0;
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/* read until 'SSND' chunk, which typically is last */
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while(numbytes == 0 && n >= 8) {
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/* chunkSize */
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i = ((buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7]);
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if (memcmp(buf,"COMM",4)==0) {
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if (i != 18) {
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DEBUGF("CODEC_ERROR: 'COMM' chunk size=%lu != 18\n",i);
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i = CODEC_ERROR;
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goto exit;
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}
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/* numChannels */
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numChannels = ((buf[8]<<8)|buf[9]);
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/* numSampleFrames */
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numSampleFrames = ((buf[10]<<24)|(buf[11]<<16)|(buf[12]<<8)|buf[13]);
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/* sampleSize */
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sampleSize = ((buf[14]<<8)|buf[15]);
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/* sampleRate (don't use last 4 bytes, only integer fs) */
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if (buf[16] != 0x40) {
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DEBUGF("CODEC_ERROR: wierd sampling rate (no @)\n",i);
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i = CODEC_ERROR;
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goto exit;
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}
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sampleRate = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21])+1;
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sampleRate = sampleRate >> (16+14-buf[17]);
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/* calc average bytes per second */
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avgbytespersec = sampleRate*numChannels*sampleSize/8;
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}
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else if (memcmp(buf,"SSND",4)==0) {
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if (sampleSize == 0) {
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DEBUGF("CODEC_ERROR: unsupported chunk order\n");
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i = CODEC_ERROR;
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goto exit;
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}
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/* offset2snd */
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offset2snd = ((buf[8]<<8)|buf[9]);
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/* blockSize */
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blockSize = ((buf[10]<<8)|buf[11]);
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if (blockSize == 0)
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blockSize = numChannels*sampleSize;
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numbytes = i-8-offset2snd;
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i = 8+offset2snd; /* advance to the beginning of data */
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}
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else {
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DEBUGF("unsupported AIFF chunk: '%c%c%c%c', size=%lu\n",
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buf[0], buf[1], buf[2], buf[3], i);
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}
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if (i & 0x01) /* odd chunk sizes must be padded */
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i++;
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buf += i+8;
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if (n < (i+8)) {
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DEBUGF("CODEC_ERROR: AIFF header size > 1024\n");
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i = CODEC_ERROR;
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goto exit;
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}
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n -= i+8;
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} /* while 'SSND' */
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if (numChannels == 0) {
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DEBUGF("CODEC_ERROR: 'COMM' chunk not found or 0-channels file\n");
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i = CODEC_ERROR;
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goto exit;
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}
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if (numbytes == 0) {
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DEBUGF("CODEC_ERROR: 'SSND' chunk not found or has zero length\n");
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i = CODEC_ERROR;
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goto exit;
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}
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if (sampleSize > 24) {
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DEBUGF("CODEC_ERROR: PCM with more than 24 bits per sample "
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"is unsupported\n");
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i = CODEC_ERROR;
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goto exit;
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}
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
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if (sampleSize <= 16) {
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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} else {
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shortorlong = 2;
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ci->configure(DSP_DITHER, (bool *)false);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (long *) (32));
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ci->configure(DSP_SET_CLIP_MAX, (long *) (2147483647));
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ci->configure(DSP_SET_CLIP_MIN, (long *) (-2147483647-1));
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}
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if (numChannels == 2) {
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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} else if (numChannels == 1) {
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
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} else {
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DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
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i = CODEC_ERROR;
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goto exit;
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}
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firstblockposn = (1024-n);
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ci->advance_buffer(firstblockposn);
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/* The main decoder loop */
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bytesdone=0;
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ci->set_elapsed(0);
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endofstream=0;
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/* chunksize is computed so that one chunk is about 1/50s.
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* this make 4096 for 44.1kHz 16bits stereo.
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* It also has to be a multiple of blockalign */
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chunksize = (1 + avgbytespersec / (50*blockSize)) * blockSize;
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/* check that the output buffer is big enough (convert to samplespersec,
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then round to the blockSize multiple below) */
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if (((uint64_t)chunksize*ci->id3->frequency*numChannels*shortorlong)
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/ (uint64_t)avgbytespersec >= AIF_CHUNK_SIZE) {
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chunksize = ((uint64_t)AIF_CHUNK_SIZE * avgbytespersec
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/ ((uint64_t)ci->id3->frequency * numChannels * shortorlong
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* blockSize)) * blockSize;
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}
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while (!endofstream) {
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uint8_t *aifbuf8;
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ci->yield();
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if (ci->stop_codec || ci->reload_codec) {
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break;
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}
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if (ci->seek_time) {
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uint32_t newpos;
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/* use avgbytespersec to round to the closest blockalign multiple,
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add firstblockposn. 64-bit casts to avoid overflows. */
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newpos = (((uint64_t)avgbytespersec * (ci->seek_time - 1))
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/ (1000LL*blockSize)) * blockSize;
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if (newpos > numbytes)
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break;
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if (ci->seek_buffer(firstblockposn + newpos)) {
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bytesdone = newpos;
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}
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ci->seek_complete();
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}
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aifbuf=ci->request_buffer((long *)&n,chunksize);
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aifbuf8 = (uint8_t*)aifbuf;
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if (n==0)
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break; /* End of stream */
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if (bytesdone + n > numbytes) {
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n = numbytes - bytesdone;
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endofstream = 1;
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}
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aifbufsize = sizeof(int16_samples);
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if (sampleSize > 24) {
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for (i=0;i<n;i+=4) {
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int32_samples[i/4]=(int32_t)((aifbuf8[i]<<24)|
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(aifbuf8[i+1]<<16)|(aifbuf8[i+2]<<8)|aifbuf8[i+3]);
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}
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aifbufsize = n;
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} else if (sampleSize > 16) {
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for (i=0;i<n;i+=3) {
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int32_samples[i/3]=(int32_t)((aifbuf8[i]<<24)|
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(aifbuf8[i+1]<<16)|(aifbuf8[i+2]<<8));
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}
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aifbufsize = n*4/3;
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} else if (sampleSize > 8) {
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/* copy data. */
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for (i=0;i<n;i+=2) {
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int16_samples[i/2]=(int16_t)((aifbuf8[i]<<8)|aifbuf8[i+1]);
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}
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aifbufsize = n;
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} else {
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for (i=0;i<n;i++) {
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int16_samples[i] = (aifbuf8[i]<<8) - 0x8000;
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}
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aifbufsize = n*2;
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}
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while (!ci->pcmbuf_insert((char*)int16_samples, aifbufsize)) {
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ci->yield();
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}
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ci->advance_buffer(n);
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bytesdone += n;
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if (bytesdone >= numbytes) {
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endofstream=1;
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}
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ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
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}
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if (ci->request_next_track())
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goto next_track;
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i = CODEC_OK;
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exit:
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return i;
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}
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@ -78,6 +78,8 @@ static const struct format_list formats[] =
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{ AFMT_ALAC, "m4a" },
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{ AFMT_AAC, "mp4" },
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{ AFMT_SHN, "shn" },
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{ AFMT_AIFF, "aif" },
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{ AFMT_AIFF, "aiff" },
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};
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static const unsigned short a52_bitrates[] =
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@ -894,7 +896,6 @@ static bool get_wave_metadata(int fd, struct mp3entry* id3)
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}
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static bool get_m4a_metadata(int fd, struct mp3entry* id3)
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{
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unsigned char* buf;
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@ -1245,6 +1246,76 @@ static bool get_musepack_metadata(int fd, struct mp3entry *id3)
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return true;
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}
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static bool get_aiff_metadata(int fd, struct mp3entry* id3)
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{
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/* Use the trackname part of the id3 structure as a temporary buffer */
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unsigned char* buf = id3->path;
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unsigned long numChannels = 0;
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unsigned long numSampleFrames = 0;
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unsigned long sampleSize = 0;
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unsigned long sampleRate = 0;
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unsigned long numbytes = 0;
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int read_bytes;
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int i;
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if ((lseek(fd, 0, SEEK_SET) < 0)
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|| ((read_bytes = read(fd, buf, sizeof(id3->path))) < 44))
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{
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return false;
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}
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if ((memcmp(buf, "FORM",4) != 0)
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|| (memcmp(&buf[8], "AIFF", 4) !=0 ))
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{
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return false;
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}
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buf += 12;
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read_bytes -= 12;
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while ((numbytes == 0) && (read_bytes >= 8))
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{
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/* chunkSize */
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i = ((buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7]);
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if (memcmp(buf, "COMM", 4) == 0)
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{
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/* numChannels */
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numChannels = ((buf[8]<<8)|buf[9]);
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/* numSampleFrames */
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numSampleFrames =((buf[10]<<24)|(buf[11]<<16)|(buf[12]<<8)|buf[13]);
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/* sampleSize */
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sampleSize = ((buf[14]<<8)|buf[15]);
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/* sampleRate */
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sampleRate = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21]);
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sampleRate = sampleRate >> (16+14-buf[17]);
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/* save format infos */
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id3->bitrate = (sampleSize * numChannels * sampleRate) / 1000;
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id3->frequency = sampleRate;
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id3->length = (numSampleFrames / id3->frequency) * 1000;
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id3->vbr = false; /* AIFF files are CBR */
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id3->filesize = filesize(fd);
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}
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else if (memcmp(buf, "SSND", 4) == 0)
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{
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numbytes = i - 8;
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}
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if (i & 0x01)
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{
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i++; /* odd chunk sizes must be padded */
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}
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buf += i + 8;
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read_bytes -= i + 8;
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}
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if ((numbytes == 0) || (numChannels == 0))
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{
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return false;
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}
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return true;
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}
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/* Simple file type probing by looking at the filename extension. */
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static unsigned int probe_file_format(const char *filename)
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{
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@ -1448,6 +1519,14 @@ bool get_metadata(struct track_info* track, int fd, const char* trackname,
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/* TODO: read the id3v2 header if it exists */
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break;
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case AFMT_AIFF:
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if (!get_aiff_metadata(fd, &(track->id3)))
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{
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return false;
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}
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break;
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default:
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/* If we don't know how to read the metadata, assume we can't play
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the file */
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@ -81,6 +81,7 @@ static volatile bool paused;
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#define CODEC_ALAC "/.rockbox/codecs/alac.codec"
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#define CODEC_AAC "/.rockbox/codecs/aac.codec"
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#define CODEC_SHN "/.rockbox/codecs/shorten.codec"
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#define CODEC_AIFF "/.rockbox/codecs/aiff.codec"
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#define AUDIO_DEFAULT_FIRST_LIMIT (1024*1024*10)
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#define AUDIO_FILL_CYCLE (1024*256)
|
||||
|
|
@ -950,6 +951,10 @@ static bool loadcodec(bool start_play)
|
|||
logf("Codec: SHN");
|
||||
codec_path = CODEC_SHN;
|
||||
break;
|
||||
case AFMT_AIFF:
|
||||
logf("Codec: PCM AIFF");
|
||||
codec_path = CODEC_AIFF;
|
||||
break;
|
||||
default:
|
||||
logf("Codec: Unsupported");
|
||||
codec_path = NULL;
|
||||
|
|
|
|||
|
|
@ -85,7 +85,7 @@ const struct filetype filetypes[] = {
|
|||
{ "ogg", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "wma", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "wav", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "flac", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "flac",TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "ac3", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "a52", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "mpc", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
|
|
@ -93,6 +93,8 @@ const struct filetype filetypes[] = {
|
|||
{ "m4a", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "mp4", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "shn", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "aif", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
{ "aiff",TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
|
||||
#endif
|
||||
{ "m3u", TREE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },
|
||||
{ "cfg", TREE_ATTR_CFG, Icon_Config, VOICE_EXT_CFG },
|
||||
|
|
|
|||
|
|
@ -161,3 +161,4 @@ Tomasz Malesinski
|
|||
Andrew Pilley
|
||||
Matt v.d. Westhuizen
|
||||
Tim Crist
|
||||
Jvo Studer
|
||||
|
|
|
|||
|
|
@ -41,6 +41,7 @@ enum {
|
|||
AFMT_ALAC, /* Apple Lossless Audio Codec */
|
||||
AFMT_AAC, /* Advanced Audio Coding (AAC) in M4A container */
|
||||
AFMT_SHN, /* Shorten */
|
||||
AFMT_AIFF, /* Audio Interchange File Format */
|
||||
|
||||
/* New formats must be added to the end of this list */
|
||||
|
||||
|
|
|
|||
|
|
@ -102,6 +102,7 @@ static const char* const codec_labels[] = {
|
|||
"ALAC", /* Apple Lossless Audio Codec */
|
||||
"AAC", /* Advanced Audio Coding in M4A container */
|
||||
"SHN", /* Shorten */
|
||||
"AIFF", /* Audio Interchange File Format */
|
||||
#endif
|
||||
};
|
||||
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue