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authorDave Chapman <dave@dchapman.com>2005-10-26 12:35:58 +0000
committerDave Chapman <dave@dchapman.com>2005-10-26 12:35:58 +0000
commit273d2e81f72c7721447ab9c539877f6712faaecc (patch)
tree79c66b0b699ca65ac2cc2369cd5bbed764c35a4b /apps
parent48be8e6a8b45fa16380af97829ba944430a78f17 (diff)
downloadrockbox-273d2e81f72c7721447ab9c539877f6712faaecc.tar.gz
rockbox-273d2e81f72c7721447ab9c539877f6712faaecc.tar.bz2
rockbox-273d2e81f72c7721447ab9c539877f6712faaecc.zip
New FLAC decoder from the ffmpeg project
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7656 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps')
-rw-r--r--apps/FILES6
-rw-r--r--apps/codecs/Makefile18
-rw-r--r--apps/codecs/flac.c405
-rw-r--r--apps/codecs/libffmpegFLAC/COPYING504
-rw-r--r--apps/codecs/libffmpegFLAC/Makefile47
-rw-r--r--apps/codecs/libffmpegFLAC/Makefile.test21
-rw-r--r--apps/codecs/libffmpegFLAC/README19
-rw-r--r--apps/codecs/libffmpegFLAC/README.rockbox46
-rw-r--r--apps/codecs/libffmpegFLAC/SOURCES2
-rw-r--r--apps/codecs/libffmpegFLAC/bitstream.c62
-rw-r--r--apps/codecs/libffmpegFLAC/bitstream.h333
-rw-r--r--apps/codecs/libffmpegFLAC/decoder.c584
-rw-r--r--apps/codecs/libffmpegFLAC/decoder.h46
-rw-r--r--apps/codecs/libffmpegFLAC/golomb.h106
-rw-r--r--apps/codecs/libffmpegFLAC/main.c292
-rw-r--r--apps/plugins/Makefile2
16 files changed, 2197 insertions, 296 deletions
diff --git a/apps/FILES b/apps/FILES
index 07719a4501..816b8c60d9 100644
--- a/apps/FILES
+++ b/apps/FILES
@@ -34,11 +34,7 @@ codecs/SOURCES
codecs/*.[ch]
codecs/Makefile
codecs/liba52/*
-codecs/libFLAC/*
-codecs/libFLAC/include/private/*.h
-codecs/libFLAC/include/protected/*.h
-codecs/libFLAC/include/FLAC/*.h
-codecs/libFLAC/include/*.h
+codecs/libffmpegFLAC/*
codecs/libmad/*
codecs/libwavpack/*
codecs/Tremor/*
diff --git a/apps/codecs/Makefile b/apps/codecs/Makefile
index 8f869b350c..b29ed738b0 100644
--- a/apps/codecs/Makefile
+++ b/apps/codecs/Makefile
@@ -17,7 +17,7 @@ ifdef APPEXTRA
endif
ifdef SOFTWARECODECS
- CODECLIBS = -lmad -la52 -lFLAC -lTremor -lwavpack -lmusepack -lalac
+ CODECLIBS = -lmad -la52 -lffmpegFLAC -lTremor -lwavpack -lmusepack -lalac
endif
# we "borrow" the plugin LDS file
@@ -39,7 +39,7 @@ DIRS = .
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)
@@ -55,7 +55,7 @@ ELFIT = $(SILENT)(file=`basename $@`; \
$(OBJDIR)/mpa.elf: $(OBJDIR)/mpa.o $(CODECDEPS) $(BUILDDIR)/libmad.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)/mpc.elf: $(OBJDIR)/mpc.o $(CODECDEPS) $(BUILDDIR)/libmusepack.a
$(OBJDIR)/wav.elf: $(OBJDIR)/wav.o $(CODECDEPS)
@@ -128,10 +128,10 @@ liba52:
@mkdir -p $(OBJDIR)/liba52
@$(MAKE) -C liba52 OBJDIR=$(OBJDIR)/liba52 OUTPUT=$(BUILDDIR)/liba52.a
-libFLAC:
- @echo "MAKE in libFLAC"
- @mkdir -p $(OBJDIR)/libFLAC
- @$(MAKE) -C libFLAC OBJDIR=$(OBJDIR)/libFLAC OUTPUT=$(BUILDDIR)/libFLAC.a
+libffmpegFLAC:
+ @echo "MAKE in libffmpegFLAC"
+ @mkdir -p $(OBJDIR)/libffmpegFLAC
+ @$(MAKE) -C libffmpegFLAC OBJDIR=$(OBJDIR)/libffmpegFLAC OUTPUT=$(BUILDDIR)/libffmpegFLAC.a
libTremor:
@echo "MAKE in Tremor"
@@ -160,10 +160,10 @@ libalac:
clean:
@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 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 libwavpack clean OBJDIR=$(OBJDIR)/libwavpack
@$(MAKE) -C libmusepack clean OBJDIR=$(OBJDIR)/libmusepack
diff --git a/apps/codecs/flac.c b/apps/codecs/flac.c
index 45fc21466c..b9a3a8a249 100644
--- a/apps/codecs/flac.c
+++ b/apps/codecs/flac.c
@@ -7,7 +7,7 @@
* \/ \/ \/ \/ \/
* $Id$
*
- * Copyright (C) 2002 Björn Stenberg
+ * 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.
@@ -18,156 +18,7 @@
****************************************************************************/
#include "codeclib.h"
-#include <codecs/libFLAC/include/FLAC/seekable_stream_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;
- }
-}
+#include <codecs/libffmpegFLAC/decoder.h>
#ifndef SIMULATOR
extern char iramcopy[];
@@ -175,78 +26,101 @@ extern char iramstart[];
extern char iramend[];
#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;
- FLAC__uint64 prev_sample, next_sample;
- size_t prev_offset, next_offset;
- int percent;
+ unsigned char buf[255];
+ bool found_streaminfo=false;
+ int endofmetadata=0;
+ int blocklength;
- if (offset >= (ci->id3->filesize - metadata_length)) {
- return stream_info->data.stream_info.total_samples;
+ if (ci->read_filebuf(buf, 4) < 4)
+ {
+ return false;
}
- prev_offset = 0;
- prev_sample = 0;
- next_offset = ci->id3->filesize - metadata_length;
- next_sample = stream_info->data.stream_info.total_samples;
-
- if (seek_table) {
- int left, right, middle;
+ if (ci->memcmp(buf,"fLaC",4) != 0)
+ {
+ return false;
+ }
+ fc->metadatalength = 4;
- middle = 0; /* Silence compiler warnings */
- points = seek_table->data.seek_table.points;
- left = 0;
- right = seek_table->data.seek_table.num_points - 1;
+ while (!endofmetadata) {
+ if (ci->read_filebuf(buf, 4) < 4)
+ {
+ return false;
+ }
- /* Do a binary search to find the matching seek point */
- while (left <= right) {
- middle = (left + right) / 2;
+ endofmetadata=(buf[0]&0x80);
+ blocklength = (buf[1] << 16) | (buf[2] << 8) | buf[3];
+ fc->metadatalength+=blocklength+4;
- if ((FLAC__uint64)offset < points[middle].stream_offset) {
- right = middle - 1;
- } else if ((FLAC__uint64)offset > points[middle].stream_offset) {
- left = middle + 1;
- } else {
- return points[middle].sample_number;
+ if ((buf[0] & 0x7f) == 0) /* 0 is the STREAMINFO block */
+ {
+ /* FIXME: Don't trust the value of blocklength */
+ if (ci->read_filebuf(buf, blocklength) < 0)
+ {
+ return false;
}
- }
-
- /* Didn't find a matching seek point, so get the sample numbers of the
- * seek points to the left and right of offset to make our guess more
- * accurate. Accuracy depends on how close these sample numbers are to
- * each other.
- */
- if ((unsigned)left >= seek_table->data.seek_table.num_points) {
- prev_offset = points[middle].stream_offset;
- prev_sample = points[middle].sample_number;
- } else if (right < 0) {
- next_offset = points[middle].stream_offset;
- next_sample = points[middle].sample_number;
+
+ fc->filesize = ci->filesize;
+ 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 */
+ ci->advance_buffer(blocklength);
} else {
- middle--;
- prev_offset = points[middle].stream_offset;
- prev_sample = points[middle].sample_number;
- next_offset = points[middle+1].stream_offset;
- next_sample = points[middle+1].sample_number;
+ /* Skip to next metadata block */
+ ci->advance_buffer(blocklength);
}
}
- /* Either there's no seek table or we didn't find our seek point, so now we
- * have to guess.
- */
- percent = ((offset - prev_offset) * 100) / (next_offset - prev_offset);
- return (FLAC__uint64)(percent * (next_sample - prev_sample) / 100 + prev_sample);
+ if (found_streaminfo) {
+ fc->bitrate = ((fc->filesize-fc->metadatalength) * 8) / fc->length;
+ return true;
+ } else {
+ return false;
+ }
}
/* this is the codec entry point */
enum codec_status codec_start(struct codec_api* api)
{
- struct codec_api* ci = api;
- FLAC__SeekableStreamDecoder* flacDecoder;
- FLAC__uint64 offset;
+ size_t n;
+ static int8_t buf[MAX_FRAMESIZE];
+ 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
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_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(DSP_DITHER, (bool *)false);
- ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
- ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
-
-next_track:
- metadata_length = 0;
- seek_table = NULL;
- stream_info = NULL;
+ ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
+ ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(28));
+
+ next_track:
- if (codec_init(api)) {
+ if (!flac_init(&fc)) {
+ LOGF("FLAC: Error initialising codec\n");
return CODEC_ERROR;
}
@@ -274,87 +146,58 @@ next_track:
ci->yield();
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
- codec_set_replaygain(ci->id3);
-
- /* Create a decoder instance */
- flacDecoder = FLAC__seekable_stream_decoder_new();
- /* 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) {
+ /* The main decoding loop */
+ bytesleft=ci->read_filebuf(buf,sizeof(buf));
+ while (bytesleft) {
ci->yield();
if (ci->stop_codec || ci->reload_codec) {
break;
}
+ /* Deal with any pending seek requests */
if (ci->seek_time) {
- int sample_loc;
+ /* We only support seeking to start of track at the moment */
+ if (ci->seek_time==1) {
+ if (ci->seek_buffer(fc.metadatalength)) {
+ /* Refill the input buffer */
+ bytesleft=ci->read_filebuf(buf,sizeof(buf));
+ ci->set_elapsed(0);
+ }
+ }
+ ci->seek_time = 0;
+ }
- sample_loc = ci->seek_time/1000 * ci->id3->frequency;
- if (FLAC__seekable_stream_decoder_seek_absolute(flacDecoder, sample_loc)) {
- samplesdone = sample_loc;
- ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
- }
- ci->seek_time = 0;
+ if(flac_decode_frame(&fc,decoded0,decoded1,buf,
+ bytesleft,ci->yield) < 0) {
+ LOGF("FLAC: Decode error, aborting\n");
+ return CODEC_ERROR;
}
+ consumed=fc.gb.index/8;
- FLAC__seekable_stream_decoder_process_single(flacDecoder);
- FLAC__seekable_stream_decoder_get_decode_position(flacDecoder, &offset);
- ci->set_offset(offset);
- }
+ ci->yield();
+ while(!ci->pcmbuf_insert_split((char*)decoded0,(char*)decoded1,
+ fc.blocksize*4)) {
+ ci->yield();
+ }
- /* Flush the libFLAC buffers */
- FLAC__seekable_stream_decoder_finish(flacDecoder);
+ /* Update the elapsed-time indicator */
+ samplesdone=fc.samplenumber+fc.blocksize;
+ elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
+ ci->set_elapsed(elapsedtime);
- if (ci->request_next_track()) {
- if (stream_info) {
- FLAC__metadata_object_delete(stream_info);
- }
- if (seek_table) {
- FLAC__metadata_object_delete(seek_table);
+ memmove(buf,&buf[consumed],bytesleft-consumed);
+ bytesleft-=consumed;
+
+ n=ci->read_filebuf(&buf[bytesleft],sizeof(buf)-bytesleft);
+ if (n > 0) {
+ bytesleft+=n;
}
- metadata_length = 0;
- goto next_track;
}
+ LOGF("FLAC: Decoded %d samples\n",samplesdone);
+
+ if (ci->request_next_track())
+ goto next_track;
return CODEC_OK;
}
diff --git a/apps/codecs/libffmpegFLAC/COPYING b/apps/codecs/libffmpegFLAC/COPYING
new file mode 100644
index 0000000000..223ede7de3
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/COPYING
@@ -0,0 +1,504 @@
+ GNU LESSER GENERAL PUBLIC LICENSE
+ Version 2.1, February 1999
+
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diff --git a/apps/codecs/libffmpegFLAC/Makefile b/apps/codecs/libffmpegFLAC/Makefile
new file mode 100644
index 0000000000..240ce28a00
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/Makefile
@@ -0,0 +1,47 @@
+# __________ __ ___.
+# 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)
diff --git a/apps/codecs/libffmpegFLAC/Makefile.test b/apps/codecs/libffmpegFLAC/Makefile.test
new file mode 100644
index 0000000000..a6850e8246
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/Makefile.test
@@ -0,0 +1,21 @@
+# 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) *~
diff --git a/apps/codecs/libffmpegFLAC/README b/apps/codecs/libffmpegFLAC/README
new file mode 100644
index 0000000000..9bc0f6ed8a
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/README
@@ -0,0 +1,19 @@
+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. \ No newline at end of file
diff --git a/apps/codecs/libffmpegFLAC/README.rockbox b/apps/codecs/libffmpegFLAC/README.rockbox
new file mode 100644
index 0000000000..850cdfcb7d
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/README.rockbox
@@ -0,0 +1,46 @@
+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.
diff --git a/apps/codecs/libffmpegFLAC/SOURCES b/apps/codecs/libffmpegFLAC/SOURCES
new file mode 100644
index 0000000000..6b23e7ca3e
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/SOURCES
@@ -0,0 +1,2 @@
+bitstream.c
+decoder.c
diff --git a/apps/codecs/libffmpegFLAC/bitstream.c b/apps/codecs/libffmpegFLAC/bitstream.c
new file mode 100644
index 0000000000..e53ec0fd46
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/bitstream.c
@@ -0,0 +1,62 @@
+/*
+ * 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);
+}
+
diff --git a/apps/codecs/libffmpegFLAC/bitstream.h b/apps/codecs/libffmpegFLAC/bitstream.h
new file mode 100644
index 0000000000..62e98c07ea
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/bitstream.h
@@ -0,0 +1,333 @@
+/**
+ * @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 */
diff --git a/apps/codecs/libffmpegFLAC/decoder.c b/apps/codecs/libffmpegFLAC/decoder.c
new file mode 100644
index 0000000000..0033feb4f9
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/decoder.c
@@ -0,0 +1,584 @@
+/*
+ * 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;
+}
diff --git a/apps/codecs/libffmpegFLAC/decoder.h b/apps/codecs/libffmpegFLAC/decoder.h
new file mode 100644
index 0000000000..5d503adc78
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/decoder.h
@@ -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
diff --git a/apps/codecs/libffmpegFLAC/golomb.h b/apps/codecs/libffmpegFLAC/golomb.h
new file mode 100644
index 0000000000..9dcbcb2d54
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/golomb.h
@@ -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);
+}
diff --git a/apps/codecs/libffmpegFLAC/main.c b/apps/codecs/libffmpegFLAC/main.c
new file mode 100644
index 0000000000..eea97f4f18
--- /dev/null
+++ b/apps/codecs/libffmpegFLAC/main.c
@@ -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);
+}
diff --git a/apps/plugins/Makefile b/apps/plugins/Makefile
index 172b757acc..c7aa438b60 100644
--- a/apps/plugins/Makefile
+++ b/apps/plugins/Makefile
@@ -17,7 +17,7 @@ ifdef APPEXTRA
endif
ifdef SOFTWARECODECS
- CODECLIBS = -lmad -la52 -lFLAC -lTremor -lwavpack -lmusepack
+ CODECLIBS = -lmad -la52 -lffmpegFLAC -lTremor -lwavpack -lmusepack
endif
LDS := plugin.lds