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New FLAC decoder from the ffmpeg project

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7656 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Dave Chapman 2005-10-26 12:35:58 +00:00
parent 48be8e6a8b
commit 273d2e81f7
17 changed files with 2202 additions and 301 deletions
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6
apps/FILES
View file

@ -34,11 +34,7 @@ codecs/SOURCES
codecs/*.[ch] codecs/*.[ch]
codecs/Makefile codecs/Makefile
codecs/liba52/* codecs/liba52/*
codecs/libFLAC/* codecs/libffmpegFLAC/*
codecs/libFLAC/include/private/*.h
codecs/libFLAC/include/protected/*.h
codecs/libFLAC/include/FLAC/*.h
codecs/libFLAC/include/*.h
codecs/libmad/* codecs/libmad/*
codecs/libwavpack/* codecs/libwavpack/*
codecs/Tremor/* codecs/Tremor/*

18
apps/codecs/Makefile
View file

@ -17,7 +17,7 @@ ifdef APPEXTRA
endif endif
ifdef SOFTWARECODECS ifdef SOFTWARECODECS
CODECLIBS = -lmad -la52 -lFLAC -lTremor -lwavpack -lmusepack -lalac CODECLIBS = -lmad -la52 -lffmpegFLAC -lTremor -lwavpack -lmusepack -lalac
endif endif
# we "borrow" the plugin LDS file # we "borrow" the plugin LDS file
@ -39,7 +39,7 @@ DIRS = .
CODECDEPS = $(LINKCODEC) $(BUILDDIR)/libcodec.a CODECDEPS = $(LINKCODEC) $(BUILDDIR)/libcodec.a
.PHONY: libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac .PHONY: libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac
OUTPUT = $(SOFTWARECODECS) OUTPUT = $(SOFTWARECODECS)
@ -55,7 +55,7 @@ ELFIT = $(SILENT)(file=`basename $@`; \
$(OBJDIR)/mpa.elf: $(OBJDIR)/mpa.o $(CODECDEPS) $(BUILDDIR)/libmad.a $(OBJDIR)/mpa.elf: $(OBJDIR)/mpa.o $(CODECDEPS) $(BUILDDIR)/libmad.a
$(OBJDIR)/a52.elf: $(OBJDIR)/a52.o $(CODECDEPS) $(BUILDDIR)/liba52.a $(OBJDIR)/a52.elf: $(OBJDIR)/a52.o $(CODECDEPS) $(BUILDDIR)/liba52.a
$(OBJDIR)/flac.elf: $(OBJDIR)/flac.o $(CODECDEPS) $(BUILDDIR)/libFLAC.a $(OBJDIR)/flac.elf: $(OBJDIR)/flac.o $(CODECDEPS) $(BUILDDIR)/libffmpegFLAC.a
$(OBJDIR)/vorbis.elf: $(OBJDIR)/vorbis.o $(CODECDEPS) $(BUILDDIR)/libTremor.a $(OBJDIR)/vorbis.elf: $(OBJDIR)/vorbis.o $(CODECDEPS) $(BUILDDIR)/libTremor.a
$(OBJDIR)/mpc.elf: $(OBJDIR)/mpc.o $(CODECDEPS) $(BUILDDIR)/libmusepack.a $(OBJDIR)/mpc.elf: $(OBJDIR)/mpc.o $(CODECDEPS) $(BUILDDIR)/libmusepack.a
$(OBJDIR)/wav.elf: $(OBJDIR)/wav.o $(CODECDEPS) $(OBJDIR)/wav.elf: $(OBJDIR)/wav.o $(CODECDEPS)
@ -128,10 +128,10 @@ liba52:
@mkdir -p $(OBJDIR)/liba52 @mkdir -p $(OBJDIR)/liba52
@$(MAKE) -C liba52 OBJDIR=$(OBJDIR)/liba52 OUTPUT=$(BUILDDIR)/liba52.a @$(MAKE) -C liba52 OBJDIR=$(OBJDIR)/liba52 OUTPUT=$(BUILDDIR)/liba52.a
libFLAC: libffmpegFLAC:
@echo "MAKE in libFLAC" @echo "MAKE in libffmpegFLAC"
@mkdir -p $(OBJDIR)/libFLAC @mkdir -p $(OBJDIR)/libffmpegFLAC
@$(MAKE) -C libFLAC OBJDIR=$(OBJDIR)/libFLAC OUTPUT=$(BUILDDIR)/libFLAC.a @$(MAKE) -C libffmpegFLAC OBJDIR=$(OBJDIR)/libffmpegFLAC OUTPUT=$(BUILDDIR)/libffmpegFLAC.a
libTremor: libTremor:
@echo "MAKE in Tremor" @echo "MAKE in Tremor"
@ -160,10 +160,10 @@ libalac:
clean: clean:
@echo "cleaning codecs" @echo "cleaning codecs"
$(SILENT)rm -fr $(OBJDIR)/libmad $(BUILDDIR)/libmad.a $(OBJDIR)/liba52 $(OBJDIR)/libFLAC $(OBJDIR)/Tremor $(OBJDIR)/libwavpack $(OBJDIR)/dumb $(BUILDDIR)/libdumb.a $(BUILDDIR)/libdumbd.a $(OBJDIR)/libmusepack $(BUILDDIR)/libmusepack.a $(OBJDIR)/libalac $(BUILDDIR)/libalac.a $(SILENT)rm -fr $(OBJDIR)/libmad $(BUILDDIR)/libmad.a $(OBJDIR)/liba52 $(OBJDIR)/libffmpegFLAC $(BUILDDIR)/libffmpegFLAC.a $(OBJDIR)/Tremor $(OBJDIR)/libwavpack $(OBJDIR)/dumb $(BUILDDIR)/libdumb.a $(BUILDDIR)/libdumbd.a $(OBJDIR)/libmusepack $(BUILDDIR)/libmusepack.a $(OBJDIR)/libalac $(BUILDDIR)/libalac.a
@$(MAKE) -C libmad clean OBJDIR=$(OBJDIR)/libmad @$(MAKE) -C libmad clean OBJDIR=$(OBJDIR)/libmad
@$(MAKE) -C liba52 clean OBJDIR=$(OBJDIR)/liba52 @$(MAKE) -C liba52 clean OBJDIR=$(OBJDIR)/liba52
@$(MAKE) -C libFLAC clean OBJDIR=$(OBJDIR)/libFLAC @$(MAKE) -C libffmpegFLAC clean OBJDIR=$(OBJDIR)/libffmpegFLAC
@$(MAKE) -C Tremor clean OBJDIR=$(OBJDIR)/Tremor @$(MAKE) -C Tremor clean OBJDIR=$(OBJDIR)/Tremor
@$(MAKE) -C libwavpack clean OBJDIR=$(OBJDIR)/libwavpack @$(MAKE) -C libwavpack clean OBJDIR=$(OBJDIR)/libwavpack
@$(MAKE) -C libmusepack clean OBJDIR=$(OBJDIR)/libmusepack @$(MAKE) -C libmusepack clean OBJDIR=$(OBJDIR)/libmusepack

405
apps/codecs/flac.c
View file

@ -7,7 +7,7 @@
* \/ \/ \/ \/ \/ * \/ \/ \/ \/ \/
* $Id$ * $Id$
* *
* Copyright (C) 2002 Björn Stenberg * Copyright (C) 2005 Dave Chapman
* *
* All files in this archive are subject to the GNU General Public License. * All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement. * See the file COPYING in the source tree root for full license agreement.
@ -18,156 +18,7 @@
****************************************************************************/ ****************************************************************************/
#include "codeclib.h" #include "codeclib.h"
#include <codecs/libFLAC/include/FLAC/seekable_stream_decoder.h> #include <codecs/libffmpegFLAC/decoder.h>
#include <codecs/libFLAC/include/FLAC/format.h>
#include <codecs/libFLAC/include/FLAC/metadata.h>
#define FLAC_MAX_SUPPORTED_BLOCKSIZE 4608
#define FLAC_MAX_SUPPORTED_CHANNELS 2
static uint32_t samplesdone;
static FLAC__StreamMetadata *stream_info;
static FLAC__StreamMetadata *seek_table;
unsigned int metadata_length;
/* Called when the FLAC decoder needs some FLAC data to decode */
FLAC__SeekableStreamDecoderReadStatus flac_read_handler(const FLAC__SeekableStreamDecoder *dec,
FLAC__byte buffer[], unsigned *bytes, void *data)
{
struct codec_api* ci = (struct codec_api*)data;
(void)dec;
*bytes=(unsigned)(ci->read_filebuf(buffer,*bytes));
if (*bytes==0) {
return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
} else {
return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
}
}
static unsigned char pcmbuf[FLAC_MAX_SUPPORTED_BLOCKSIZE*FLAC_MAX_SUPPORTED_CHANNELS*2] IDATA_ATTR;
/* Called when the FLAC decoder has some decoded PCM data to write */
FLAC__StreamDecoderWriteStatus flac_write_handler(const FLAC__SeekableStreamDecoder *dec,
const FLAC__Frame *frame,
const FLAC__int32 * const buf[],
void *data)
{
struct codec_api* ci = (struct codec_api*)data;
(void)dec;
unsigned int c_samp, c_chan, d_samp;
uint32_t data_size = frame->header.blocksize * frame->header.channels * 2; /* Assume 16-bit words */
uint32_t samples = frame->header.blocksize;
int yieldcounter = 0;
if (samples*frame->header.channels > (FLAC_MAX_SUPPORTED_BLOCKSIZE*FLAC_MAX_SUPPORTED_CHANNELS)) {
// ERROR!!!
// DEBUGF("ERROR: samples*frame->header.channels=%d\n",samples*frame->header.channels);
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
(void)dec;
for (c_samp = d_samp = 0; c_samp < samples; c_samp++) {
for(c_chan = 0; c_chan < frame->header.channels; c_chan++, d_samp++) {
pcmbuf[d_samp*2] = (buf[c_chan][c_samp]&0xff00)>>8;
pcmbuf[(d_samp*2)+1] = buf[c_chan][c_samp]&0xff;
if (yieldcounter++ == 100) {
ci->yield();
yieldcounter = 0;
}
}
}
samplesdone+=samples;
ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
ci->yield();
while (!ci->pcmbuf_insert(pcmbuf, data_size))
ci->yield();
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
void flac_metadata_handler(const FLAC__SeekableStreamDecoder *dec,
const FLAC__StreamMetadata *meta, void *data)
{
/* Ignore metadata for now... */
(void)dec;
(void)data;
metadata_length += meta->length;
if ( meta->type == FLAC__METADATA_TYPE_STREAMINFO ) {
stream_info = FLAC__metadata_object_clone( meta );
if ( stream_info == NULL ) {
//return CODEC_ERROR;
}
} else if ( meta->type == FLAC__METADATA_TYPE_SEEKTABLE ) {
seek_table = FLAC__metadata_object_clone( meta );
if ( seek_table == NULL ) {
//return CODEC_ERROR;
}
}
}
void flac_error_handler(const FLAC__SeekableStreamDecoder *dec,
FLAC__StreamDecoderErrorStatus status, void *data)
{
(void)dec;
(void)status;
(void)data;
}
FLAC__SeekableStreamDecoderSeekStatus flac_seek_handler(const FLAC__SeekableStreamDecoder *decoder,
FLAC__uint64 absolute_byte_offset,
void *client_data)
{
(void)decoder;
struct codec_api* ci = (struct codec_api*)client_data;
if (ci->seek_buffer(absolute_byte_offset)) {
return FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK;
} else {
return FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR;
}
}
FLAC__SeekableStreamDecoderTellStatus flac_tell_handler(const FLAC__SeekableStreamDecoder *decoder,
FLAC__uint64 *absolute_byte_offset, void *client_data)
{
struct codec_api* ci = (struct codec_api*)client_data;
(void)decoder;
*absolute_byte_offset = ci->curpos;
return FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK;
}
FLAC__SeekableStreamDecoderLengthStatus flac_length_handler(const FLAC__SeekableStreamDecoder *decoder,
FLAC__uint64 *stream_length, void *client_data)
{
struct codec_api* ci = (struct codec_api*)client_data;
(void)decoder;
*stream_length = ci->filesize;
return FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK;
}
FLAC__bool flac_eof_handler(const FLAC__SeekableStreamDecoder *decoder,
void *client_data)
{
struct codec_api* ci = (struct codec_api*)client_data;
(void)decoder;
if (ci->curpos >= ci->filesize) {
return true;
} else {
return false;
}
}
#ifndef SIMULATOR #ifndef SIMULATOR
extern char iramcopy[]; extern char iramcopy[];
@ -175,78 +26,101 @@ extern char iramstart[];
extern char iramend[]; extern char iramend[];
#endif #endif
FLAC__uint64 find_sample_number(struct codec_api *ci, size_t offset) struct codec_api* rb;
struct codec_api* ci;
/* The output buffers containing the decoded samples (channels 0 and 1) */
int32_t decoded0[MAX_BLOCKSIZE] IBSS_ATTR;
int32_t decoded1[MAX_BLOCKSIZE] IBSS_ATTR;
static bool flac_init(FLACContext* fc)
{ {
FLAC__StreamMetadata_SeekPoint *points; unsigned char buf[255];
FLAC__uint64 prev_sample, next_sample; bool found_streaminfo=false;
size_t prev_offset, next_offset; int endofmetadata=0;
int percent; int blocklength;
if (offset >= (ci->id3->filesize - metadata_length)) { if (ci->read_filebuf(buf, 4) < 4)
return stream_info->data.stream_info.total_samples; {
return false;
} }
prev_offset = 0; if (ci->memcmp(buf,"fLaC",4) != 0)
prev_sample = 0; {
next_offset = ci->id3->filesize - metadata_length; return false;
next_sample = stream_info->data.stream_info.total_samples; }
fc->metadatalength = 4;
if (seek_table) { while (!endofmetadata) {
int left, right, middle; if (ci->read_filebuf(buf, 4) < 4)
{
return false;
}
middle = 0; /* Silence compiler warnings */ endofmetadata=(buf[0]&0x80);
points = seek_table->data.seek_table.points; blocklength = (buf[1] << 16) | (buf[2] << 8) | buf[3];
left = 0; fc->metadatalength+=blocklength+4;
right = seek_table->data.seek_table.num_points - 1;
/* Do a binary search to find the matching seek point */ if ((buf[0] & 0x7f) == 0) /* 0 is the STREAMINFO block */
while (left <= right) { {
middle = (left + right) / 2; /* FIXME: Don't trust the value of blocklength */
if (ci->read_filebuf(buf, blocklength) < 0)
if ((FLAC__uint64)offset < points[middle].stream_offset) { {
right = middle - 1; return false;
} else if ((FLAC__uint64)offset > points[middle].stream_offset) {
left = middle + 1;
} else {
return points[middle].sample_number;
} }
}
/* Didn't find a matching seek point, so get the sample numbers of the fc->filesize = ci->filesize;
* seek points to the left and right of offset to make our guess more fc->min_blocksize = (buf[0] << 8) | buf[1];
* accurate. Accuracy depends on how close these sample numbers are to fc->max_blocksize = (buf[2] << 8) | buf[3];
* each other. fc->min_framesize = (buf[4] << 16) | (buf[5] << 8) | buf[6];
*/ fc->max_framesize = (buf[7] << 16) | (buf[8] << 8) | buf[9];
if ((unsigned)left >= seek_table->data.seek_table.num_points) { fc->samplerate = (buf[10] << 12) | (buf[11] << 4)
prev_offset = points[middle].stream_offset; | ((buf[12] & 0xf0) >> 4);
prev_sample = points[middle].sample_number; fc->channels = ((buf[12]&0x0e)>>1) + 1;
} else if (right < 0) { fc->bps = (((buf[12]&0x01) << 4) | ((buf[13]&0xf0)>>4) ) + 1;
next_offset = points[middle].stream_offset;
next_sample = points[middle].sample_number; /* totalsamples is a 36-bit field, but we assume <= 32 bits are
used */
fc->totalsamples = (buf[14] << 24) | (buf[15] << 16)
| (buf[16] << 8) | buf[17];
/* Calculate track length (in ms) and estimate the bitrate
(in kbit/s) */
fc->length = (fc->totalsamples / fc->samplerate) * 1000;
found_streaminfo=true;
} else if ((buf[0] & 0x7f) == 3) { /* 3 is the SEEKTABLE block */
ci->advance_buffer(blocklength);
} else { } else {
middle--; /* Skip to next metadata block */
prev_offset = points[middle].stream_offset; ci->advance_buffer(blocklength);
prev_sample = points[middle].sample_number;
next_offset = points[middle+1].stream_offset;
next_sample = points[middle+1].sample_number;
} }
} }
/* Either there's no seek table or we didn't find our seek point, so now we if (found_streaminfo) {
* have to guess. fc->bitrate = ((fc->filesize-fc->metadatalength) * 8) / fc->length;
*/ return true;
percent = ((offset - prev_offset) * 100) / (next_offset - prev_offset); } else {
return (FLAC__uint64)(percent * (next_sample - prev_sample) / 100 + prev_sample); return false;
}
} }
/* this is the codec entry point */ /* this is the codec entry point */
enum codec_status codec_start(struct codec_api* api) enum codec_status codec_start(struct codec_api* api)
{ {
struct codec_api* ci = api; size_t n;
FLAC__SeekableStreamDecoder* flacDecoder; static int8_t buf[MAX_FRAMESIZE];
FLAC__uint64 offset; FLACContext fc;
uint32_t samplesdone;
uint32_t elapsedtime;
int bytesleft;
int consumed;
TEST_CODEC_API(ci); /* Generic codec initialisation */
TEST_CODEC_API(api);
rb = api;
ci = (struct codec_api*)api;
#ifndef SIMULATOR #ifndef SIMULATOR
ci->memcpy(iramstart, iramcopy, iramend-iramstart); ci->memcpy(iramstart, iramcopy, iramend-iramstart);
@ -254,19 +128,17 @@ enum codec_status codec_start(struct codec_api* api)
ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*10)); ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*10));
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512)); ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*1024)); ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
ci->configure(CODEC_DSP_ENABLE, (bool *)true); ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false); ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED); ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16)); ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(28));
next_track: next_track:
metadata_length = 0;
seek_table = NULL;
stream_info = NULL;
if (codec_init(api)) { if (!flac_init(&fc)) {
LOGF("FLAC: Error initialising codec\n");
return CODEC_ERROR; return CODEC_ERROR;
} }
@ -274,87 +146,58 @@ next_track:
ci->yield(); ci->yield();
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency)); ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
codec_set_replaygain(ci->id3);
/* Create a decoder instance */ /* The main decoding loop */
flacDecoder = FLAC__seekable_stream_decoder_new(); bytesleft=ci->read_filebuf(buf,sizeof(buf));
while (bytesleft) {
/* Set up the decoder and the callback functions - this must be done before init */
/* The following are required for stream_decoder and higher */
FLAC__seekable_stream_decoder_set_client_data(flacDecoder,ci);
FLAC__seekable_stream_decoder_set_write_callback(flacDecoder, flac_write_handler);
FLAC__seekable_stream_decoder_set_read_callback(flacDecoder, flac_read_handler);
FLAC__seekable_stream_decoder_set_metadata_callback(flacDecoder, flac_metadata_handler);
FLAC__seekable_stream_decoder_set_error_callback(flacDecoder, flac_error_handler);
FLAC__seekable_stream_decoder_set_metadata_respond_all(flacDecoder);
/* The following are only for the seekable_stream_decoder */
FLAC__seekable_stream_decoder_set_seek_callback(flacDecoder, flac_seek_handler);
FLAC__seekable_stream_decoder_set_tell_callback(flacDecoder, flac_tell_handler);
FLAC__seekable_stream_decoder_set_length_callback(flacDecoder, flac_length_handler);
FLAC__seekable_stream_decoder_set_eof_callback(flacDecoder, flac_eof_handler);
/* QUESTION: What do we do when the init fails? */
if (FLAC__seekable_stream_decoder_init(flacDecoder)) {
return CODEC_ERROR;
}
/* The first thing to do is to parse the metadata */
FLAC__seekable_stream_decoder_process_until_end_of_metadata(flacDecoder);
if (ci->id3->offset && stream_info) {
FLAC__uint64 sample;
sample = find_sample_number(ci, ci->id3->offset - metadata_length);
ci->advance_buffer(ci->id3->offset);
FLAC__seekable_stream_decoder_seek_absolute(flacDecoder, sample);
FLAC__seekable_stream_decoder_get_decode_position(flacDecoder, &offset);
ci->set_offset(offset);
samplesdone = (uint32_t)sample;
ci->set_elapsed(sample/(ci->id3->frequency/1000));
} else {
samplesdone = 0;
ci->set_elapsed(0);
}
/* The main decoder loop */
while (FLAC__seekable_stream_decoder_get_state(flacDecoder) != FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM) {
ci->yield(); ci->yield();
if (ci->stop_codec || ci->reload_codec) { if (ci->stop_codec || ci->reload_codec) {
break; break;
} }
/* Deal with any pending seek requests */
if (ci->seek_time) { if (ci->seek_time) {
int sample_loc; /* We only support seeking to start of track at the moment */
if (ci->seek_time==1) {
sample_loc = ci->seek_time/1000 * ci->id3->frequency; if (ci->seek_buffer(fc.metadatalength)) {
if (FLAC__seekable_stream_decoder_seek_absolute(flacDecoder, sample_loc)) { /* Refill the input buffer */
samplesdone = sample_loc; bytesleft=ci->read_filebuf(buf,sizeof(buf));
ci->set_elapsed(samplesdone/(ci->id3->frequency/1000)); ci->set_elapsed(0);
} }
ci->seek_time = 0; }
ci->seek_time = 0;
} }
FLAC__seekable_stream_decoder_process_single(flacDecoder); if(flac_decode_frame(&fc,decoded0,decoded1,buf,
FLAC__seekable_stream_decoder_get_decode_position(flacDecoder, &offset); bytesleft,ci->yield) < 0) {
ci->set_offset(offset); LOGF("FLAC: Decode error, aborting\n");
return CODEC_ERROR;
}
consumed=fc.gb.index/8;
ci->yield();
while(!ci->pcmbuf_insert_split((char*)decoded0,(char*)decoded1,
fc.blocksize*4)) {
ci->yield();
}
/* Update the elapsed-time indicator */
samplesdone=fc.samplenumber+fc.blocksize;
elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
ci->set_elapsed(elapsedtime);
memmove(buf,&buf[consumed],bytesleft-consumed);
bytesleft-=consumed;
n=ci->read_filebuf(&buf[bytesleft],sizeof(buf)-bytesleft);
if (n > 0) {
bytesleft+=n;
}
} }
LOGF("FLAC: Decoded %d samples\n",samplesdone);
/* Flush the libFLAC buffers */ if (ci->request_next_track())
FLAC__seekable_stream_decoder_finish(flacDecoder);
if (ci->request_next_track()) {
if (stream_info) {
FLAC__metadata_object_delete(stream_info);
}
if (seek_table) {
FLAC__metadata_object_delete(seek_table);
}
metadata_length = 0;
goto next_track; goto next_track;
}
return CODEC_OK; return CODEC_OK;
} }

504
apps/codecs/libffmpegFLAC/COPYING Normal file
View file

@ -0,0 +1,504 @@
GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
[This is the first released version of the Lesser GPL. It also counts
as the successor of the GNU Library Public License, version 2, hence
the version number 2.1.]
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
Licenses are intended to guarantee your freedom to share and change
free software--to make sure the software is free for all its users.
This license, the Lesser General Public License, applies to some
specially designated software packages--typically libraries--of the
Free Software Foundation and other authors who decide to use it. You
can use it too, but we suggest you first think carefully about whether
this license or the ordinary General Public License is the better
strategy to use in any particular case, based on the explanations below.
When we speak of free software, we are referring to freedom of use,
not price. Our General Public Licenses are designed to make sure that
you have the freedom to distribute copies of free software (and charge
for this service if you wish); that you receive source code or can get
it if you want it; that you can change the software and use pieces of
it in new free programs; and that you are informed that you can do
these things.
To protect your rights, we need to make restrictions that forbid
distributors to deny you these rights or to ask you to surrender these
rights. These restrictions translate to certain responsibilities for
you if you distribute copies of the library or if you modify it.
For example, if you distribute copies of the library, whether gratis
or for a fee, you must give the recipients all the rights that we gave
you. You must make sure that they, too, receive or can get the source
code. If you link other code with the library, you must provide
complete object files to the recipients, so that they can relink them
with the library after making changes to the library and recompiling
it. And you must show them these terms so they know their rights.
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That's all there is to it!

47
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# __________ __ ___.
# Open \______ \ ____ ____ | | _\_ |__ _______ ___
# Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
# Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
# Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
# \/ \/ \/ \/ \/
# $Id$
#
INCLUDES=-I$(APPSDIR) -I.. -I. -I$(FIRMDIR)/include -I$(FIRMDIR)/export \
-I$(FIRMDIR)/common -I$(FIRMDIR)/drivers -I$(BUILDDIR)
ifdef APPEXTRA
INCLUDES += -I$(APPSDIR)/$(APPEXTRA)
endif
FLACOPTS = -O2
CFLAGS = $(GCCOPTS) $(FLACOPTS) $(INCLUDES) $(TARGET) $(EXTRA_DEFINES) -DMEM=${MEMORYSIZE}
# This sets up 'SRC' based on the files mentioned in SOURCES
include $(TOOLSDIR)/makesrc.inc
SOURCES = $(SRC)
OBJS2 := $(SRC:%.c=$(OBJDIR)/%.o)
OBJS = $(patsubst %.S, $(OBJDIR)/%.o, $(OBJS2))
DEPFILE = $(OBJDIR)/dep-libalac
DIRS =
OUTPUT = $(BUILDDIR)/libalac.a
all: $(OUTPUT)
$(OUTPUT): $(OBJS)
@echo "AR $@"
@$(AR) ruv $@ $+ >/dev/null 2>&1
$(OBJDIR)/libalac/%.o: $(APPSDIR)/codecs/libalac/%.c
@echo "(libalac) CC $<"
@$(CC) -c $(CFLAGS) -I$(APPSDIR)/codecs/libalac/ $< -o $@
include $(TOOLSDIR)/make.inc
clean:
@echo "cleaning libffmpegFLAC"
@rm -f $(OBJS) $(OUTPUT) $(DEPFILE)
-include $(DEPFILE)

21
apps/codecs/libffmpegFLAC/Makefile.test Normal file
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# Simple Makefile for standalone testing of decoder
OBJS=main.o decoder.o bitstream.o
CFLAGS=-O2 -Wall -DBUILD_STANDALONE
CC=gcc
# Uncomment the next line to build on a big-endian target:
#CFLAGS += -DBUILD_BIGENDIAN
all: test
main.o: main.c decoder.h
decoder.o: decoder.c bitstream.h golomb.h decoder.h
bitstream.o: bitstream.c
test: $(OBJS)
gcc -o test $(OBJS)
clean:
rm -f test $(OBJS) *~

19
apps/codecs/libffmpegFLAC/README Normal file
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FFmpeg README
-------------
1) Documentation
----------------
* Read the documentation in the doc/ directory.
2) Licensing
------------
* Read the file COPYING. ffmpeg and the associated libraries EXCEPT
liba52 and libpostproc are licensed under the Lesser GNU General
Public License.
* liba52 and libpostproc are distributed under the GNU General Public
License and their compilation and use is optional in ffmpeg.
Fabrice Bellard.

46
apps/codecs/libffmpegFLAC/README.rockbox Normal file
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Library: libffmpegFLAC
Imported: 2005-10-26 by Dave Chapman
This directory contains a FLAC decoder based on the flac.c (and a small
number of other files) from the ffmpeg project.
LICENSING INFORMATION
ffmpeg is licensed under the Lesser GNU General Public License and the
file flac.c is Copyright 2003 Alex Beregszaszi.
IMPORT DETAILS
The decoder is based on ffmpeg-cvs from 26 October 2005.
The file libavcodec/flac.c was renamed to decoder.c and all ffmpeg
related code was removed. It was also heavily modified to remove
debugging information and unused experiemental code.
It was also changed to provide output as 32-bit integers (with the
data left-shifted to 28 bit accuracy). The original flac.c provided
16-bit output.
In order to minimise memory usage, a small number of hard-coded limits
are present in decoder.h - mainly limiting the supported blocksize
(number of samples in a frame) to 4608. This is the default value
used by the reference FLAC encoder at all standard compression
settings.
TESTING
A test program is contained in the file main.c and this should compile
in any Unix-like environment using the command "make -f Makefile.test".
Running "test file.flac" will decode the audio data to a WAV file
called "test.wav" in the current directory. This has been tested with
16-bit and 24-bit mono and stereo FLAC files.
This can be used to test the accuracy of the decoder by comparing the
md5sum of the WAV file produced by this decoder with a WAV file
produced by the official libFLAC decoder.
This test program could be extended to perform an internal md5sum
calculation and comparing that against the md5sum stored in the FLAC
file's header.

2
apps/codecs/libffmpegFLAC/SOURCES Normal file
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bitstream.c
decoder.c

62
apps/codecs/libffmpegFLAC/bitstream.c Normal file
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/*
* Common bit i/o utils
* Copyright (c) 2000, 2001 Fabrice Bellard.
* Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* alternative bitstream reader & writer by Michael Niedermayer <michaelni@gmx.at>
*/
/**
* @file bitstream.c
* bitstream api.
*/
#include <stdio.h>
#include "bitstream.h"
/* bit input functions */
/**
* reads 0-32 bits.
*/
unsigned int get_bits_long(GetBitContext *s, int n){
if(n<=17) return get_bits(s, n);
else{
int ret= get_bits(s, 16) << (n-16);
return ret | get_bits(s, n-16);
}
}
/**
* shows 0-32 bits.
*/
unsigned int show_bits_long(GetBitContext *s, int n){
if(n<=17) return show_bits(s, n);
else{
GetBitContext gb= *s;
int ret= get_bits_long(s, n);
*s= gb;
return ret;
}
}
void align_get_bits(GetBitContext *s)
{
int n= (-get_bits_count(s)) & 7;
if(n) skip_bits(s, n);
}

333
apps/codecs/libffmpegFLAC/bitstream.h Normal file
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/**
* @file bitstream.h
* bitstream api header.
*/
#ifndef BITSTREAM_H
#define BITSTREAM_H
#include <inttypes.h>
#ifndef BUILD_STANDALONE
#include <config.h>
#include <system.h>
#else
#include <stdio.h>
#define IBSS_ATTR
#define ICONST_ATTR
#endif
/* Endian conversion routines for standalone compilation */
#ifdef BUILD_STANDALONE
#ifdef BUILD_BIGENDIAN
#define betoh32(x) (x)
#define letoh32(x) swap32(x)
#else
#define letoh32(x) (x)
#define betoh32(x) swap32(x)
#endif
/* Taken from rockbox/firmware/export/system.h */
static inline unsigned short swap16(unsigned short value)
/*
result[15..8] = value[ 7..0];
result[ 7..0] = value[15..8];
*/
{
return (value >> 8) | (value << 8);
}
static inline unsigned long swap32(unsigned long value)
/*
result[31..24] = value[ 7.. 0];
result[23..16] = value[15.. 8];
result[15.. 8] = value[23..16];
result[ 7.. 0] = value[31..24];
*/
{
unsigned long hi = swap16(value >> 16);
unsigned long lo = swap16(value & 0xffff);
return (lo << 16) | hi;
}
#endif
/* FLAC files are big-endian */
#define ALT_BITSTREAM_READER_BE
#define NEG_SSR32(a,s) (((int32_t)(a))>>(32-(s)))
#define NEG_USR32(a,s) (((uint32_t)(a))>>(32-(s)))
/* bit input */
/* buffer, buffer_end and size_in_bits must be present and used by every reader */
typedef struct GetBitContext {
const uint8_t *buffer, *buffer_end;
int index;
int size_in_bits;
} GetBitContext;
#define VLC_TYPE int16_t
typedef struct VLC {
int bits;
VLC_TYPE (*table)[2]; ///< code, bits
int table_size, table_allocated;
} VLC;
typedef struct RL_VLC_ELEM {
int16_t level;
int8_t len;
uint8_t run;
} RL_VLC_ELEM;
#if defined(ARCH_SPARC) || defined(ARCH_ARMV4L)
#define UNALIGNED_STORES_ARE_BAD
#endif
/* used to avoid missaligned exceptions on some archs (alpha, ...) */
#if defined(ARCH_X86) || defined(ARCH_X86_64)
# define unaligned32(a) (*(const uint32_t*)(a))
#else
# ifdef __GNUC__
static inline uint32_t unaligned32(const void *v) {
struct Unaligned {
uint32_t i;
} __attribute__((packed));
return ((const struct Unaligned *) v)->i;
}
# elif defined(__DECC)
static inline uint32_t unaligned32(const void *v) {
return *(const __unaligned uint32_t *) v;
}
# else
static inline uint32_t unaligned32(const void *v) {
return *(const uint32_t *) v;
}
# endif
#endif //!ARCH_X86
/* Bitstream reader API docs:
name
abritary name which is used as prefix for the internal variables
gb
getbitcontext
OPEN_READER(name, gb)
loads gb into local variables
CLOSE_READER(name, gb)
stores local vars in gb
UPDATE_CACHE(name, gb)
refills the internal cache from the bitstream
after this call at least MIN_CACHE_BITS will be available,
GET_CACHE(name, gb)
will output the contents of the internal cache, next bit is MSB of 32 or 64 bit (FIXME 64bit)
SHOW_UBITS(name, gb, num)
will return the next num bits
SHOW_SBITS(name, gb, num)
will return the next num bits and do sign extension
SKIP_BITS(name, gb, num)
will skip over the next num bits
note, this is equivalent to SKIP_CACHE; SKIP_COUNTER
SKIP_CACHE(name, gb, num)
will remove the next num bits from the cache (note SKIP_COUNTER MUST be called before UPDATE_CACHE / CLOSE_READER)
SKIP_COUNTER(name, gb, num)
will increment the internal bit counter (see SKIP_CACHE & SKIP_BITS)
LAST_SKIP_CACHE(name, gb, num)
will remove the next num bits from the cache if it is needed for UPDATE_CACHE otherwise it will do nothing
LAST_SKIP_BITS(name, gb, num)
is equivalent to SKIP_LAST_CACHE; SKIP_COUNTER
for examples see get_bits, show_bits, skip_bits, get_vlc
*/
static inline int unaligned32_be(const void *v)
{
#ifdef CONFIG_ALIGN
const uint8_t *p=v;
return (((p[0]<<8) | p[1])<<16) | (p[2]<<8) | (p[3]);
#else
return betoh32( unaligned32(v)); //original
#endif
}
static inline int unaligned32_le(const void *v)
{
#ifdef CONFIG_ALIGN
const uint8_t *p=v;
return (((p[3]<<8) | p[2])<<16) | (p[1]<<8) | (p[0]);
#else
return letoh32( unaligned32(v)); //original
#endif
}
# define MIN_CACHE_BITS 25
# define OPEN_READER(name, gb)\
int name##_index= (gb)->index;\
int name##_cache= 0;\
# define CLOSE_READER(name, gb)\
(gb)->index= name##_index;\
# ifdef ALT_BITSTREAM_READER_LE
# define UPDATE_CACHE(name, gb)\
name##_cache= unaligned32_le( ((const uint8_t *)(gb)->buffer)+(name##_index>>3) ) >> (name##_index&0x07);\
# define SKIP_CACHE(name, gb, num)\
name##_cache >>= (num);
# else
# define UPDATE_CACHE(name, gb)\
name##_cache= unaligned32_be( ((const uint8_t *)(gb)->buffer)+(name##_index>>3) ) << (name##_index&0x07);\
# define SKIP_CACHE(name, gb, num)\
name##_cache <<= (num);
# endif
// FIXME name?
# define SKIP_COUNTER(name, gb, num)\
name##_index += (num);\
# define SKIP_BITS(name, gb, num)\
{\
SKIP_CACHE(name, gb, num)\
SKIP_COUNTER(name, gb, num)\
}\
# define LAST_SKIP_BITS(name, gb, num) SKIP_COUNTER(name, gb, num)
# define LAST_SKIP_CACHE(name, gb, num) ;
# ifdef ALT_BITSTREAM_READER_LE
# define SHOW_UBITS(name, gb, num)\
((name##_cache) & (NEG_USR32(0xffffffff,num)))
# else
# define SHOW_UBITS(name, gb, num)\
NEG_USR32(name##_cache, num)
# endif
# define SHOW_SBITS(name, gb, num)\
NEG_SSR32(name##_cache, num)
# define GET_CACHE(name, gb)\
((uint32_t)name##_cache)
static inline int get_bits_count(GetBitContext *s){
return s->index;
}
static inline int get_sbits(GetBitContext *s, int n){
register int tmp;
OPEN_READER(re, s)
UPDATE_CACHE(re, s)
tmp= SHOW_SBITS(re, s, n);
LAST_SKIP_BITS(re, s, n)
CLOSE_READER(re, s)
return tmp;
}
/**
* reads 0-17 bits.
* Note, the alt bitstream reader can read up to 25 bits, but the libmpeg2 reader can't
*/
static inline unsigned int get_bits(GetBitContext *s, int n){
register int tmp;
OPEN_READER(re, s)
UPDATE_CACHE(re, s)
tmp= SHOW_UBITS(re, s, n);
LAST_SKIP_BITS(re, s, n)
CLOSE_READER(re, s)
return tmp;
}
unsigned int get_bits_long(GetBitContext *s, int n);
/**
* shows 0-17 bits.
* Note, the alt bitstream reader can read up to 25 bits, but the libmpeg2 reader can't
*/
static inline unsigned int show_bits(GetBitContext *s, int n){
register int tmp;
OPEN_READER(re, s)
UPDATE_CACHE(re, s)
tmp= SHOW_UBITS(re, s, n);
// CLOSE_READER(re, s)
return tmp;
}
unsigned int show_bits_long(GetBitContext *s, int n);
static inline void skip_bits(GetBitContext *s, int n){
//Note gcc seems to optimize this to s->index+=n for the ALT_READER :))
OPEN_READER(re, s)
UPDATE_CACHE(re, s)
LAST_SKIP_BITS(re, s, n)
CLOSE_READER(re, s)
}
static inline unsigned int get_bits1(GetBitContext *s){
int index= s->index;
uint8_t result= s->buffer[ index>>3 ];
#ifdef ALT_BITSTREAM_READER_LE
result>>= (index&0x07);
result&= 1;
#else
result<<= (index&0x07);
result>>= 8 - 1;
#endif
index++;
s->index= index;
return result;
}
static inline unsigned int show_bits1(GetBitContext *s){
return show_bits(s, 1);
}
static inline void skip_bits1(GetBitContext *s){
skip_bits(s, 1);
}
/**
* init GetBitContext.
* @param buffer bitstream buffer, must be FF_INPUT_BUFFER_PADDING_SIZE bytes larger then the actual read bits
* because some optimized bitstream readers read 32 or 64 bit at once and could read over the end
* @param bit_size the size of the buffer in bits
*/
static inline void init_get_bits(GetBitContext *s,
const uint8_t *buffer, int bit_size)
{
int buffer_size= (bit_size+7)>>3;
if(buffer_size < 0 || bit_size < 0) {
buffer_size = bit_size = 0;
buffer = 0;
}
s->buffer= buffer;
s->size_in_bits= bit_size;
s->buffer_end= buffer + buffer_size;
s->index=0;
{
OPEN_READER(re, s)
UPDATE_CACHE(re, s)
UPDATE_CACHE(re, s)
CLOSE_READER(re, s)
}
}
void align_get_bits(GetBitContext *s);
#endif /* BITSTREAM_H */

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/*
* FLAC (Free Lossless Audio Codec) decoder
* Copyright (c) 2003 Alex Beregszaszi
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/**
* @file flac.c
* FLAC (Free Lossless Audio Codec) decoder
* @author Alex Beregszaszi
*
* For more information on the FLAC format, visit:
* http://flac.sourceforge.net/
*
* This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
* through, starting from the initial 'fLaC' signature; or by passing the
* 34-byte streaminfo structure through avctx->extradata[_size] followed
* by data starting with the 0xFFF8 marker.
*/
#include <inttypes.h>
#include <stdbool.h>
#ifndef BUILD_STANDALONE
#include "../codec.h"
#endif
#include "bitstream.h"
#include "golomb.h"
#include "decoder.h"
#define INT_MAX 2147483647
#define FFMAX(a,b) ((a) > (b) ? (a) : (b))
#define FFMIN(a,b) ((a) > (b) ? (b) : (a))
/* Needed by golomb.h */
const uint8_t ff_log2_tab[256] ICONST_ATTR = {
0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7
};
static const int sample_rate_table[] ICONST_ATTR =
{ 0, 0, 0, 0,
8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
0, 0, 0, 0 };
static const int sample_size_table[] ICONST_ATTR =
{ 0, 8, 12, 0, 16, 20, 24, 0 };
static const int blocksize_table[] ICONST_ATTR = {
0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0,
256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
};
static const uint8_t table_crc8[256] ICONST_ATTR = {
0x00, 0x07, 0x0e, 0x09, 0x1c, 0x1b, 0x12, 0x15,
0x38, 0x3f, 0x36, 0x31, 0x24, 0x23, 0x2a, 0x2d,
0x70, 0x77, 0x7e, 0x79, 0x6c, 0x6b, 0x62, 0x65,
0x48, 0x4f, 0x46, 0x41, 0x54, 0x53, 0x5a, 0x5d,
0xe0, 0xe7, 0xee, 0xe9, 0xfc, 0xfb, 0xf2, 0xf5,
0xd8, 0xdf, 0xd6, 0xd1, 0xc4, 0xc3, 0xca, 0xcd,
0x90, 0x97, 0x9e, 0x99, 0x8c, 0x8b, 0x82, 0x85,
0xa8, 0xaf, 0xa6, 0xa1, 0xb4, 0xb3, 0xba, 0xbd,
0xc7, 0xc0, 0xc9, 0xce, 0xdb, 0xdc, 0xd5, 0xd2,
0xff, 0xf8, 0xf1, 0xf6, 0xe3, 0xe4, 0xed, 0xea,
0xb7, 0xb0, 0xb9, 0xbe, 0xab, 0xac, 0xa5, 0xa2,
0x8f, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9d, 0x9a,
0x27, 0x20, 0x29, 0x2e, 0x3b, 0x3c, 0x35, 0x32,
0x1f, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0d, 0x0a,
0x57, 0x50, 0x59, 0x5e, 0x4b, 0x4c, 0x45, 0x42,
0x6f, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7d, 0x7a,
0x89, 0x8e, 0x87, 0x80, 0x95, 0x92, 0x9b, 0x9c,
0xb1, 0xb6, 0xbf, 0xb8, 0xad, 0xaa, 0xa3, 0xa4,
0xf9, 0xfe, 0xf7, 0xf0, 0xe5, 0xe2, 0xeb, 0xec,
0xc1, 0xc6, 0xcf, 0xc8, 0xdd, 0xda, 0xd3, 0xd4,
0x69, 0x6e, 0x67, 0x60, 0x75, 0x72, 0x7b, 0x7c,
0x51, 0x56, 0x5f, 0x58, 0x4d, 0x4a, 0x43, 0x44,
0x19, 0x1e, 0x17, 0x10, 0x05, 0x02, 0x0b, 0x0c,
0x21, 0x26, 0x2f, 0x28, 0x3d, 0x3a, 0x33, 0x34,
0x4e, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5c, 0x5b,
0x76, 0x71, 0x78, 0x7f, 0x6a, 0x6d, 0x64, 0x63,
0x3e, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2c, 0x2b,
0x06, 0x01, 0x08, 0x0f, 0x1a, 0x1d, 0x14, 0x13,
0xae, 0xa9, 0xa0, 0xa7, 0xb2, 0xb5, 0xbc, 0xbb,
0x96, 0x91, 0x98, 0x9f, 0x8a, 0x8d, 0x84, 0x83,
0xde, 0xd9, 0xd0, 0xd7, 0xc2, 0xc5, 0xcc, 0xcb,
0xe6, 0xe1, 0xe8, 0xef, 0xfa, 0xfd, 0xf4, 0xf3
};
static int64_t get_utf8(GetBitContext *gb)
{
uint64_t val;
int ones=0, bytes;
while(get_bits1(gb))
ones++;
if (ones==0) bytes=0;
else if(ones==1) return -1;
else bytes= ones - 1;
val= get_bits(gb, 7-ones);
while(bytes--){
const int tmp = get_bits(gb, 8);
if((tmp>>6) != 2)
return -1;
val<<=6;
val|= tmp&0x3F;
}
return val;
}
static int get_crc8(const uint8_t *buf, int count){
int crc=0;
int i;
for(i=0; i<count; i++){
crc = table_crc8[crc ^ buf[i]];
}
return crc;
}
static int decode_residuals(FLACContext *s, int32_t* decoded, int pred_order)
{
int i, tmp, partition, method_type, rice_order;
int sample = 0, samples;
method_type = get_bits(&s->gb, 2);
if (method_type != 0){
//fprintf(stderr,"illegal residual coding method %d\n", method_type);
return -1;
}
rice_order = get_bits(&s->gb, 4);
samples= s->blocksize >> rice_order;
sample=
i= pred_order;
for (partition = 0; partition < (1 << rice_order); partition++)
{
tmp = get_bits(&s->gb, 4);
if (tmp == 15)
{
//fprintf(stderr,"fixed len partition\n");
tmp = get_bits(&s->gb, 5);
for (; i < samples; i++, sample++)
decoded[sample] = get_sbits(&s->gb, tmp);
}
else
{
for (; i < samples; i++, sample++){
decoded[sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
}
}
i= 0;
}
return 0;
}
static int decode_subframe_fixed(FLACContext *s, int32_t* decoded, int pred_order)
{
int i;
/* warm up samples */
for (i = 0; i < pred_order; i++)
{
decoded[i] = get_sbits(&s->gb, s->curr_bps);
}
if (decode_residuals(s, decoded, pred_order) < 0)
return -1;
switch(pred_order)
{
case 0:
break;
case 1:
for (i = pred_order; i < s->blocksize; i++)
decoded[i] += decoded[i-1];
break;
case 2:
for (i = pred_order; i < s->blocksize; i++)
decoded[i] += 2*decoded[i-1]
- decoded[i-2];
break;
case 3:
for (i = pred_order; i < s->blocksize; i++)
decoded[i] += 3*decoded[i-1]
- 3*decoded[i-2]
+ decoded[i-3];
break;
case 4:
for (i = pred_order; i < s->blocksize; i++)
decoded[i] += 4*decoded[i-1]
- 6*decoded[i-2]
+ 4*decoded[i-3]
- decoded[i-4];
break;
default:
return -1;
}
return 0;
}
static int decode_subframe_lpc(FLACContext *s, int32_t* decoded, int pred_order)
{
int sum, i, j;
int64_t wsum;
int coeff_prec, qlevel;
int coeffs[pred_order];
/* warm up samples */
for (i = 0; i < pred_order; i++)
{
decoded[i] = get_sbits(&s->gb, s->curr_bps);
}
coeff_prec = get_bits(&s->gb, 4) + 1;
if (coeff_prec == 16)
{
//fprintf(stderr,"invalid coeff precision\n");
return -1;
}
qlevel = get_sbits(&s->gb, 5);
if (qlevel < 0)
{
//fprintf(stderr,"qlevel %d not supported, maybe buggy stream\n", qlevel);
return -1;
}
for (i = 0; i < pred_order; i++)
{
coeffs[i] = get_sbits(&s->gb, coeff_prec);
}
if (decode_residuals(s, decoded, pred_order) < 0)
return -1;
if ((s->bps + coeff_prec + av_log2(pred_order)) <= 32) {
for (i = pred_order; i < s->blocksize; i++)
{
sum = 0;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * decoded[i-j-1];
decoded[i] += sum >> qlevel;
}
} else {
for (i = pred_order; i < s->blocksize; i++)
{
wsum = 0;
for (j = 0; j < pred_order; j++)
wsum += (int64_t)coeffs[j] * (int64_t)decoded[i-j-1];
decoded[i] += wsum >> qlevel;
}
}
return 0;
}
static inline int decode_subframe(FLACContext *s, int channel, int32_t* decoded)
{
int type, wasted = 0;
int i, tmp;
s->curr_bps = s->bps;
if(channel == 0){
if(s->decorrelation == RIGHT_SIDE)
s->curr_bps++;
}else{
if(s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
s->curr_bps++;
}
if (get_bits1(&s->gb))
{
//av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
return -1;
}
type = get_bits(&s->gb, 6);
// wasted = get_bits1(&s->gb);
// if (wasted)
// {
// while (!get_bits1(&s->gb))
// wasted++;
// if (wasted)
// wasted++;
// s->curr_bps -= wasted;
// }
#if 0
wasted= 16 - av_log2(show_bits(&s->gb, 17));
skip_bits(&s->gb, wasted+1);
s->curr_bps -= wasted;
#else
if (get_bits1(&s->gb))
{
wasted = 1;
while (!get_bits1(&s->gb))
wasted++;
s->curr_bps -= wasted;
//fprintf(stderr,"%d wasted bits\n", wasted);
}
#endif
//FIXME use av_log2 for types
if (type == 0)
{
//fprintf(stderr,"coding type: constant\n");
tmp = get_sbits(&s->gb, s->curr_bps);
for (i = 0; i < s->blocksize; i++)
decoded[i] = tmp;
}
else if (type == 1)
{
//fprintf(stderr,"coding type: verbatim\n");
for (i = 0; i < s->blocksize; i++)
decoded[i] = get_sbits(&s->gb, s->curr_bps);
}
else if ((type >= 8) && (type <= 12))
{
//fprintf(stderr,"coding type: fixed\n");
if (decode_subframe_fixed(s, decoded, type & ~0x8) < 0)
return -1;
}
else if (type >= 32)
{
//fprintf(stderr,"coding type: lpc\n");
if (decode_subframe_lpc(s, decoded, (type & ~0x20)+1) < 0)
return -1;
}
else
{
//fprintf(stderr,"Unknown coding type: %d\n",type);
return -1;
}
if (wasted)
{
int i;
for (i = 0; i < s->blocksize; i++)
decoded[i] <<= wasted;
}
return 0;
}
static int decode_frame(FLACContext *s,
int32_t* decoded0,
int32_t* decoded1,
void (*yield)(void))
{
int blocksize_code, sample_rate_code, sample_size_code, assignment, crc8;
int decorrelation, bps, blocksize, samplerate;
blocksize_code = get_bits(&s->gb, 4);
sample_rate_code = get_bits(&s->gb, 4);
assignment = get_bits(&s->gb, 4); /* channel assignment */
if (assignment < 8 && s->channels == assignment+1)
decorrelation = INDEPENDENT;
else if (assignment >=8 && assignment < 11 && s->channels == 2)
decorrelation = LEFT_SIDE + assignment - 8;
else
{
return -1;
}
sample_size_code = get_bits(&s->gb, 3);
if(sample_size_code == 0)
bps= s->bps;
else if((sample_size_code != 3) && (sample_size_code != 7))
bps = sample_size_table[sample_size_code];
else
{
return -1;
}
if (get_bits1(&s->gb))
{
return -1;
}
/* Get the samplenumber of the first sample in this block */
s->samplenumber=get_utf8(&s->gb);
/* samplenumber actually contains the frame number for streams
with a constant block size - so we multiply by blocksize to
get the actual sample number */
if (s->min_blocksize == s->max_blocksize) {
s->samplenumber*=s->min_blocksize;
}
#if 0
if (/*((blocksize_code == 6) || (blocksize_code == 7)) &&*/
(s->min_blocksize != s->max_blocksize)){
}else{
}
#endif
if (blocksize_code == 0)
blocksize = s->min_blocksize;
else if (blocksize_code == 6)
blocksize = get_bits(&s->gb, 8)+1;
else if (blocksize_code == 7)
blocksize = get_bits(&s->gb, 16)+1;
else
blocksize = blocksize_table[blocksize_code];
if(blocksize > s->max_blocksize){
return -1;
}
if (sample_rate_code == 0){
samplerate= s->samplerate;
}else if ((sample_rate_code > 3) && (sample_rate_code < 12))
samplerate = sample_rate_table[sample_rate_code];
else if (sample_rate_code == 12)
samplerate = get_bits(&s->gb, 8) * 1000;
else if (sample_rate_code == 13)
samplerate = get_bits(&s->gb, 16);
else if (sample_rate_code == 14)
samplerate = get_bits(&s->gb, 16) * 10;
else{
return -1;
}
skip_bits(&s->gb, 8);
crc8= get_crc8(s->gb.buffer, get_bits_count(&s->gb)/8);
if(crc8){
return -1;
}
s->blocksize = blocksize;
s->samplerate = samplerate;
s->bps = bps;
s->decorrelation= decorrelation;
yield();
/* subframes */
if (decode_subframe(s, 0, decoded0) < 0)
return -1;
yield();
if (s->channels==2) {
if (decode_subframe(s, 1, decoded1) < 0)
return -1;
}
yield();
align_get_bits(&s->gb);
/* frame footer */
skip_bits(&s->gb, 16); /* data crc */
return 0;
}
int flac_decode_frame(FLACContext *s,
int32_t* decoded0,
int32_t* decoded1,
uint8_t *buf, int buf_size,
void (*yield)(void))
{
int tmp = 0, i, input_buf_size = 0;
int framesize;
int scale;
init_get_bits(&s->gb, buf, buf_size*8);
tmp = show_bits(&s->gb, 16);
if(tmp != 0xFFF8){
//fprintf(stderr,"FRAME HEADER not here\n");
while(get_bits_count(&s->gb)/8+2 < buf_size && show_bits(&s->gb, 16) != 0xFFF8)
skip_bits(&s->gb, 8);
goto end; // we may not have enough bits left to decode a frame, so try next time
}
skip_bits(&s->gb, 16);
if ((framesize=decode_frame(s,decoded0,decoded1,yield)) < 0){
s->bitstream_size=0;
s->bitstream_index=0;
return -1;
}
yield();
scale=FLAC_OUTPUT_DEPTH-s->bps;
switch(s->decorrelation)
{
case INDEPENDENT:
if (s->channels==1) {;
for (i = 0; i < s->blocksize; i++)
{
decoded0[i] = decoded0[i] << scale;
}
} else {
for (i = 0; i < s->blocksize; i++)
{
decoded0[i] = decoded0[i] << scale;
decoded1[i] = decoded1[i] << scale;
}
}
break;
case LEFT_SIDE:
//assert(s->channels == 2);
for (i = 0; i < s->blocksize; i++)
{
decoded1[i] = (decoded0[i] - decoded1[i]) << scale;
decoded0[i] = decoded0[i] << scale;
}
break;
case RIGHT_SIDE:
//assert(s->channels == 2);
for (i = 0; i < s->blocksize; i++)
{
decoded0[i] = (decoded0[i] + decoded1[i]) << scale;
decoded1[i] = decoded1[i] << scale;
}
break;
case MID_SIDE:
//assert(s->channels == 2);
for (i = 0; i < s->blocksize; i++)
{
int mid, side;
mid = decoded0[i];
side = decoded1[i];
#if 1 //needs to be checked but IMHO it should be binary identical
mid -= side>>1;
decoded0[i] = (mid + side) << scale;
decoded1[i] = mid << scale;
#else
mid <<= 1;
if (side & 1)
mid++;
decoded0[i] = ((mid + side) >> 1) << scale;
decoded1[i] = ((mid - side) >> 1) << scale;
#endif
}
break;
}
end:
i= (get_bits_count(&s->gb)+7)/8;;
if(i > buf_size){
s->bitstream_size=0;
s->bitstream_index=0;
return -1;
}
if(s->bitstream_size){
s->bitstream_index += i;
s->bitstream_size -= i;
return input_buf_size;
}else
return i;
}

46
apps/codecs/libffmpegFLAC/decoder.h Normal file
View file

@ -0,0 +1,46 @@
#ifndef _FLAC_DECODER_H
#define _FLAC_DECODER_H
#include "bitstream.h"
#define MAX_CHANNELS 2 /* Maximum supported channels */
#define MAX_BLOCKSIZE 4608 /* Maxsize in samples of one uncompressed frame */
#define MAX_FRAMESIZE 32768 /* Maxsize in bytes of one compressed frame */
#define FLAC_OUTPUT_DEPTH 28 /* Provide samples left-shifted to 28-bits */
enum decorrelation_type {
INDEPENDENT,
LEFT_SIDE,
RIGHT_SIDE,
MID_SIDE,
};
typedef struct FLACContext {
GetBitContext gb;
int min_blocksize, max_blocksize;
int min_framesize, max_framesize;
int samplerate, channels;
int blocksize/*, last_blocksize*/;
int bps, curr_bps;
unsigned long samplenumber;
unsigned long totalsamples;
enum decorrelation_type decorrelation;
int filesize;
int length;
int bitrate;
int metadatalength;
int bitstream_size;
int bitstream_index;
} FLACContext;
int flac_decode_frame(FLACContext *s,
int32_t* decoded0,
int32_t* decoded1,
uint8_t *buf, int buf_size,
void (*yield)(void));
#endif

106
apps/codecs/libffmpegFLAC/golomb.h Normal file
View file

@ -0,0 +1,106 @@
/*
* exp golomb vlc stuff
* Copyright (c) 2003 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2004 Alex Beregszaszi
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
/* From libavutil/common.h */
extern const uint8_t ff_log2_tab[256];
static inline int av_log2(unsigned int v)
{
int n;
n = 0;
if (v & 0xffff0000) {
v >>= 16;
n += 16;
}
if (v & 0xff00) {
v >>= 8;
n += 8;
}
n += ff_log2_tab[v];
return n;
}
/**
* @file golomb.h
* @brief
* exp golomb vlc stuff
* @author Michael Niedermayer <michaelni@gmx.at> and Alex Beregszaszi
*/
/**
* read unsigned golomb rice code (jpegls).
*/
static inline int get_ur_golomb_jpegls(GetBitContext *gb, int k, int limit, int esc_len){
unsigned int buf;
int log;
OPEN_READER(re, gb);
UPDATE_CACHE(re, gb);
buf=GET_CACHE(re, gb);
log= av_log2(buf);
if(log > 31-11){
buf >>= log - k;
buf += (30-log)<<k;
LAST_SKIP_BITS(re, gb, 32 + k - log);
CLOSE_READER(re, gb);
return buf;
}else{
int i;
for(i=0; SHOW_UBITS(re, gb, 1) == 0; i++){
LAST_SKIP_BITS(re, gb, 1);
UPDATE_CACHE(re, gb);
}
SKIP_BITS(re, gb, 1);
if(i < limit - 1){
if(k){
buf = SHOW_UBITS(re, gb, k);
LAST_SKIP_BITS(re, gb, k);
}else{
buf=0;
}
CLOSE_READER(re, gb);
return buf + (i<<k);
}else if(i == limit - 1){
buf = SHOW_UBITS(re, gb, esc_len);
LAST_SKIP_BITS(re, gb, esc_len);
CLOSE_READER(re, gb);
return buf + 1;
}else
return -1;
}
}
/**
* read signed golomb rice code (flac).
*/
static inline int get_sr_golomb_flac(GetBitContext *gb, int k, int limit, int esc_len){
int v= get_ur_golomb_jpegls(gb, k, limit, esc_len);
return (v>>1) ^ -(v&1);
}

292
apps/codecs/libffmpegFLAC/main.c Normal file
View file

@ -0,0 +1,292 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
/* A test program for the Rockbox version of the ffmpeg FLAC decoder.
Compile using Makefile.test - run it as "./test file.flac" to decode the
FLAC file to the file "test.wav" in the current directory
This test program should support 16-bit and 24-bit mono and stereo files.
The resulting "test.wav" should have the same md5sum as a WAV file created
by the official FLAC decoder (it produces the same 44-byte canonical WAV
header.
*/
#include <stdio.h>
#include <string.h>
#include <inttypes.h>
#include <stdbool.h>
#include <fcntl.h>
#include <unistd.h>
#include <sys/types.h>
#include <sys/stat.h>
#include "decoder.h"
static unsigned char wav_header[44]={
'R','I','F','F',// 0 - ChunkID
0,0,0,0, // 4 - ChunkSize (filesize-8)
'W','A','V','E',// 8 - Format
'f','m','t',' ',// 12 - SubChunkID
16,0,0,0, // 16 - SubChunk1ID // 16 for PCM
1,0, // 20 - AudioFormat (1=Uncompressed)
2,0, // 22 - NumChannels
0,0,0,0, // 24 - SampleRate in Hz
0,0,0,0, // 28 - Byte Rate (SampleRate*NumChannels*(BitsPerSample/8)
4,0, // 32 - BlockAlign (== NumChannels * BitsPerSample/8)
16,0, // 34 - BitsPerSample
'd','a','t','a',// 36 - Subchunk2ID
0,0,0,0 // 40 - Subchunk2Size
};
int open_wav(char* filename) {
int fd;
fd=open(filename,O_CREAT|O_WRONLY|O_TRUNC,S_IRUSR|S_IWUSR);
if (fd >= 0) {
write(fd,wav_header,sizeof(wav_header));
}
return(fd);
}
void close_wav(int fd, FLACContext* fc) {
int x;
int filesize;
int bytespersample;
bytespersample=fc->bps/8;
filesize=fc->totalsamples*bytespersample*fc->channels+44;
// ChunkSize
x=filesize-8;
wav_header[4]=(x&0xff);
wav_header[5]=(x&0xff00)>>8;
wav_header[6]=(x&0xff0000)>>16;
wav_header[7]=(x&0xff000000)>>24;
// Number of channels
wav_header[22]=fc->channels;
// Samplerate
wav_header[24]=fc->samplerate&0xff;
wav_header[25]=(fc->samplerate&0xff00)>>8;
wav_header[26]=(fc->samplerate&0xff0000)>>16;
wav_header[27]=(fc->samplerate&0xff000000)>>24;
// ByteRate
x=fc->samplerate*(fc->bps/8)*fc->channels;
wav_header[28]=(x&0xff);
wav_header[29]=(x&0xff00)>>8;
wav_header[30]=(x&0xff0000)>>16;
wav_header[31]=(x&0xff000000)>>24;
// BlockAlign
wav_header[32]=(fc->bps/8)*fc->channels;
// Bits per sample
wav_header[34]=fc->bps;
// Subchunk2Size
x=filesize-44;
wav_header[40]=(x&0xff);
wav_header[41]=(x&0xff00)>>8;
wav_header[42]=(x&0xff0000)>>16;
wav_header[43]=(x&0xff000000)>>24;
lseek(fd,0,SEEK_SET);
write(fd,wav_header,sizeof(wav_header));
close(fd);
}
static void dump_headers(FLACContext *s)
{
fprintf(stderr," Blocksize: %d .. %d\n", s->min_blocksize,
s->max_blocksize);
fprintf(stderr," Framesize: %d .. %d\n", s->min_framesize,
s->max_framesize);
fprintf(stderr," Samplerate: %d\n", s->samplerate);
fprintf(stderr," Channels: %d\n", s->channels);
fprintf(stderr," Bits per sample: %d\n", s->bps);
fprintf(stderr," Metadata length: %d\n", s->metadatalength);
fprintf(stderr," Total Samples: %lu\n",s->totalsamples);
fprintf(stderr," Duration: %d ms\n",s->length);
fprintf(stderr," Bitrate: %d kbps\n",s->bitrate);
}
static bool flac_init(int fd, FLACContext* fc)
{
unsigned char buf[255];
struct stat statbuf;
bool found_streaminfo=false;
int endofmetadata=0;
int blocklength;
if (lseek(fd, 0, SEEK_SET) < 0)
{
return false;
}
if (read(fd, buf, 4) < 4)
{
return false;
}
if (memcmp(buf,"fLaC",4) != 0)
{
return false;
}
fc->metadatalength = 4;
while (!endofmetadata) {
if (read(fd, buf, 4) < 4)
{
return false;
}
endofmetadata=(buf[0]&0x80);
blocklength = (buf[1] << 16) | (buf[2] << 8) | buf[3];
fc->metadatalength+=blocklength+4;
if ((buf[0] & 0x7f) == 0) /* 0 is the STREAMINFO block */
{
/* FIXME: Don't trust the value of blocklength */
if (read(fd, buf, blocklength) < 0)
{
return false;
}
fstat(fd,&statbuf);
fc->filesize = statbuf.st_size;
fc->min_blocksize = (buf[0] << 8) | buf[1];
fc->max_blocksize = (buf[2] << 8) | buf[3];
fc->min_framesize = (buf[4] << 16) | (buf[5] << 8) | buf[6];
fc->max_framesize = (buf[7] << 16) | (buf[8] << 8) | buf[9];
fc->samplerate = (buf[10] << 12) | (buf[11] << 4)
| ((buf[12] & 0xf0) >> 4);
fc->channels = ((buf[12]&0x0e)>>1) + 1;
fc->bps = (((buf[12]&0x01) << 4) | ((buf[13]&0xf0)>>4) ) + 1;
/* totalsamples is a 36-bit field, but we assume <= 32 bits are
used */
fc->totalsamples = (buf[14] << 24) | (buf[15] << 16)
| (buf[16] << 8) | buf[17];
/* Calculate track length (in ms) and estimate the bitrate
(in kbit/s) */
fc->length = (fc->totalsamples / fc->samplerate) * 1000;
found_streaminfo=true;
} else if ((buf[0] & 0x7f) == 3) { /* 3 is the SEEKTABLE block */
fprintf(stderr,"Seektable length = %d bytes\n",blocklength);
if (lseek(fd, blocklength, SEEK_CUR) < 0) {
return false;
}
} else {
/* Skip to next metadata block */
if (lseek(fd, blocklength, SEEK_CUR) < 0)
{
return false;
}
}
}
if (found_streaminfo) {
fc->bitrate = ((fc->filesize-fc->metadatalength) * 8) / fc->length;
return true;
} else {
return false;
}
}
/* Dummy function needed to pass to flac_decode_frame() */
void yield() {
}
int main(int argc, char* argv[]) {
FLACContext fc;
int fd,fdout;
int n;
int i;
int bytesleft;
int consumed;
char buf[MAX_FRAMESIZE]; /* The input buffer */
/* The output buffers containing the decoded samples (channels 0 and 1) */
int32_t decoded0[MAX_BLOCKSIZE];
int32_t decoded1[MAX_BLOCKSIZE];
/* For testing */
int8_t wavbuf[MAX_CHANNELS*MAX_BLOCKSIZE*3];
int8_t* p;
int scale;
fd=open(argv[1],O_RDONLY);
if (fd < 0) {
fprintf(stderr,"Can not parse %s\n",argv[1]);
return(1);
}
/* Read the metadata and position the file pointer at the start of the
first audio frame */
flac_init(fd,&fc);
dump_headers(&fc);
fdout=open_wav("test.wav");
bytesleft=read(fd,buf,sizeof(buf));
while (bytesleft) {
if(flac_decode_frame(&fc,decoded0,decoded1,buf,bytesleft,yield) < 0) {
fprintf(stderr,"DECODE ERROR, ABORTING\n");
break;
}
consumed=fc.gb.index/8;
scale=FLAC_OUTPUT_DEPTH-fc.bps;
p=wavbuf;
for (i=0;i<fc.blocksize;i++) {
/* Left sample */
decoded0[i]=decoded0[i]>>scale;
*(p++)=decoded0[i]&0xff;
*(p++)=(decoded0[i]&0xff00)>>8;
if (fc.bps==24) *(p++)=(decoded0[i]&0xff0000)>>16;
if (fc.channels==2) {
/* Right sample */
decoded1[i]=decoded1[i]>>scale;
*(p++)=decoded1[i]&0xff;
*(p++)=(decoded1[i]&0xff00)>>8;
if (fc.bps==24) *(p++)=(decoded1[i]&0xff0000)>>16;
}
}
write(fdout,wavbuf,fc.blocksize*fc.channels*(fc.bps/8));
memmove(buf,&buf[consumed],bytesleft-consumed);
bytesleft-=consumed;
n=read(fd,&buf[bytesleft],sizeof(buf)-bytesleft);
if (n > 0) {
bytesleft+=n;
}
}
close_wav(fdout,&fc);
close(fd);
return(0);
}

2
apps/plugins/Makefile
View file

@ -17,7 +17,7 @@ ifdef APPEXTRA
endif endif
ifdef SOFTWARECODECS ifdef SOFTWARECODECS
CODECLIBS = -lmad -la52 -lFLAC -lTremor -lwavpack -lmusepack CODECLIBS = -lmad -la52 -lffmpegFLAC -lTremor -lwavpack -lmusepack
endif endif
LDS := plugin.lds LDS := plugin.lds

8
tools/configure vendored
View file

@ -523,7 +523,7 @@ appsdir='\$(ROOTDIR)/apps'
archosrom="" archosrom=""
flash="" flash=""
plugins="yes" plugins="yes"
codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac" codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
;; ;;
10) 10)
@ -537,7 +537,7 @@ appsdir='\$(ROOTDIR)/apps'
archosrom="" archosrom=""
flash="" flash=""
plugins="yes" plugins="yes"
codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac" codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
;; ;;
11) 11)
@ -551,7 +551,7 @@ appsdir='\$(ROOTDIR)/apps'
archosrom="" archosrom=""
flash="" flash=""
plugins="yes" plugins="yes"
codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac" codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
;; ;;
12) 12)
@ -565,7 +565,7 @@ appsdir='\$(ROOTDIR)/apps'
archosrom="" archosrom=""
flash="" flash=""
plugins="yes" plugins="yes"
codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac" codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
;; ;;
*) *)