New FLAC decoder from the ffmpeg project

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7656 a1c6a512-1295-4272-9138-f99709370657

17 files changed, 2201 insertions, 300 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
+
+ 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
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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
diff --git a/tools/configure b/tools/configure
index 9c6f2df319..4a0a4ee47a 100755
--- a/tools/configure
+++ b/tools/configure
@@ -523,7 +523,7 @@ appsdir='\$(ROOTDIR)/apps'
     archosrom=""
     flash=""
     plugins="yes"
-    codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac"
+    codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
     ;;
 
    10)
@@ -537,7 +537,7 @@ appsdir='\$(ROOTDIR)/apps'
     archosrom=""
     flash=""
     plugins="yes"
-    codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac"
+    codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
     ;;
 
    11)
@@ -551,7 +551,7 @@ appsdir='\$(ROOTDIR)/apps'
     archosrom=""
     flash=""
     plugins="yes"
-    codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac"
+    codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
     ;;
 
    12)
@@ -565,7 +565,7 @@ appsdir='\$(ROOTDIR)/apps'
     archosrom=""
     flash=""
     plugins="yes"
-    codecs="libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac"
+    codecs="libmad liba52 libffmpegFLAC libTremor libwavpack dumb libmusepack libalac"
     ;;
 
    *)