First version of ALAC (Apple Lossless) decoder

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7547 a1c6a512-1295-4272-9138-f99709370657

3 files changed, 399 insertions, 3 deletions

diff --git a/apps/codecs/Makefile b/apps/codecs/Makefile
index abd108f..8f869b3 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
+   CODECLIBS = -lmad -la52 -lFLAC -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
+.PHONY: libmad liba52 libFLAC libTremor libwavpack dumb libmusepack libalac
 
 OUTPUT = $(SOFTWARECODECS)
 
@@ -60,6 +60,7 @@ $(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)
 $(OBJDIR)/wavpack.elf: $(OBJDIR)/wavpack.o $(CODECDEPS) $(BUILDDIR)/libwavpack.a
+$(OBJDIR)/alac.elf: $(OBJDIR)/alac.o $(CODECDEPS) $(BUILDDIR)/libalac.a
 
 $(OBJDIR)/%.elf: $(OBJDIR)/%.o $(CODECDEPS)
 	$(ELFIT)
@@ -152,14 +153,20 @@ libmusepack:
 	@mkdir -p $(OBJDIR)/libmusepack
 	@$(MAKE) -C libmusepack OBJDIR=$(OBJDIR)/libmusepack OUTPUT=$(BUILDDIR)/libmusepack.a
 
+libalac:
+	@echo "MAKE in libalac"
+	@mkdir -p $(OBJDIR)/libalac
+	@$(MAKE) -C libalac OBJDIR=$(OBJDIR)/libalac OUTPUT=$(BUILDDIR)/libalac.a
+
 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
+	$(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 $(BUILDDIR)/libalac.a
 	@$(MAKE) -C libmad clean OBJDIR=$(OBJDIR)/libmad
 	@$(MAKE) -C liba52 clean OBJDIR=$(OBJDIR)/liba52
 	@$(MAKE) -C libFLAC clean OBJDIR=$(OBJDIR)/libFLAC
 	@$(MAKE) -C Tremor clean OBJDIR=$(OBJDIR)/Tremor
 	@$(MAKE) -C libwavpack clean OBJDIR=$(OBJDIR)/libwavpack
 	@$(MAKE) -C libmusepack clean OBJDIR=$(OBJDIR)/libmusepack
+	@$(MAKE) -C libalac clean OBJDIR=$(OBJDIR)/libalac
 	@$(MAKE) -C dumb clean OBJDIR=$(OBJDIR)/dumb
 	@$(MAKE) -C lib clean OBJDIR=$(OBJDIR)/lib
diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES
index 14a053b..14cf847 100644
--- a/apps/codecs/SOURCES
+++ b/apps/codecs/SOURCES
@@ -6,4 +6,5 @@ wav.c
 a52.c
 mpc.c
 wavpack.c
+alac.c
 #endif
diff --git a/apps/codecs/alac.c b/apps/codecs/alac.c
new file mode 100644
index 0000000..f00ae97
--- /dev/null
+++ b/apps/codecs/alac.c
@@ -0,0 +1,388 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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.
+ *
+ ****************************************************************************/
+
+#include "codec.h"
+
+#include <codecs/libalac/demux.h>
+#include <codecs/libalac/decomp.h>
+#include <codecs/libalac/stream.h>
+
+#include "playback.h"
+#include "dsp.h"
+#include "lib/codeclib.h"
+
+#ifndef SIMULATOR
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+#define destBufferSize (1024*16)
+
+char inputBuffer[1024*32];    /* Input buffer */
+size_t mdat_offset;
+struct codec_api* rb;
+struct codec_api* ci;
+
+/* Implementation of the stream.h functions used by libalac */
+
+#define _Swap32(v) do { \
+                   v = (((v) & 0x000000FF) << 0x18) | \
+                       (((v) & 0x0000FF00) << 0x08) | \
+                       (((v) & 0x00FF0000) >> 0x08) | \
+                       (((v) & 0xFF000000) >> 0x18); } while(0)
+
+#define _Swap16(v) do { \
+                   v = (((v) & 0x00FF) << 0x08) | \
+                       (((v) & 0xFF00) >> 0x08); } while (0)
+
+/* A normal read without any byte-swapping */
+void stream_read(stream_t *stream, size_t size, void *buf)
+{
+    ci->read_filebuf(buf,size);
+    if (ci->curpos >= ci->filesize) { stream->eof=1; }
+}
+
+int32_t stream_read_int32(stream_t *stream)
+{
+    int32_t v;
+    stream_read(stream, 4, &v);
+#ifdef ROCKBOX_LITTLE_ENDIAN
+    _Swap32(v);
+#endif
+    return v;
+}
+
+uint32_t stream_read_uint32(stream_t *stream)
+{
+    uint32_t v;
+    stream_read(stream, 4, &v);
+#ifdef ROCKBOX_LITTLE_ENDIAN
+    _Swap32(v);
+#endif
+    return v;
+}
+
+int16_t stream_read_int16(stream_t *stream)
+{
+    int16_t v;
+    stream_read(stream, 2, &v);
+#ifdef ROCKBOX_LITTLE_ENDIAN
+    _Swap16(v);
+#endif
+    return v;
+}
+
+uint16_t stream_read_uint16(stream_t *stream)
+{
+    uint16_t v;
+    stream_read(stream, 2, &v);
+#ifdef ROCKBOX_LITTLE_ENDIAN
+    _Swap16(v);
+#endif
+    return v;
+}
+
+int8_t stream_read_int8(stream_t *stream)
+{
+    int8_t v;
+    stream_read(stream, 1, &v);
+    return v;
+}
+
+uint8_t stream_read_uint8(stream_t *stream)
+{
+    uint8_t v;
+    stream_read(stream, 1, &v);
+    return v;
+}
+
+void stream_skip(stream_t *stream, size_t skip)
+{
+  (void)stream;
+  ci->advance_buffer(skip);
+}
+
+int stream_eof(stream_t *stream)
+{
+    return stream->eof;
+}
+
+void stream_create(stream_t *stream)
+{
+    stream->eof=0;
+}
+
+/* This function was part of the original alac decoder implementation */
+
+static int get_sample_info(demux_res_t *demux_res, uint32_t samplenum,
+                           uint32_t *sample_duration,
+                           uint32_t *sample_byte_size)
+{
+    unsigned int duration_index_accum = 0;
+    unsigned int duration_cur_index = 0;
+
+    if (samplenum >= demux_res->num_sample_byte_sizes) { 
+        return 0;
+    }
+
+    if (!demux_res->num_time_to_samples) {
+        return 0;
+    }
+
+    while ((demux_res->time_to_sample[duration_cur_index].sample_count
+            + duration_index_accum) <= samplenum) {
+        duration_index_accum +=
+        demux_res->time_to_sample[duration_cur_index].sample_count;
+
+        duration_cur_index++;
+        if (duration_cur_index >= demux_res->num_time_to_samples) {
+            return 0;
+        }
+    }
+
+    *sample_duration =
+     demux_res->time_to_sample[duration_cur_index].sample_duration;
+    *sample_byte_size = demux_res->sample_byte_size[samplenum];
+
+    return 1;
+}
+
+/* Seek to sample_loc (or close to it).  Return 1 on success (and
+   modify samplesdone and currentblock), 0 if failed 
+
+   Seeking uses the following two arrays:
+
+     1) the sample_byte_size array contains the length in bytes of
+        each block ("sample" in Applespeak).
+
+     2) the time_to_sample array contains the duration (in samples) of
+        each block of data.
+
+     So we just find the block number we are going to seek to (using
+     time_to_sample) and then find the offset in the file (using
+     sample_byte_size).
+
+     Each ALAC block seems to be independent of all the others.
+ */
+
+static unsigned int alac_seek (demux_res_t* demux_res, 
+                               unsigned int sample_loc, 
+                               size_t* samplesdone, int* currentblock)
+{
+  int flag;
+  unsigned int i,j;
+  unsigned int newblock;
+  unsigned int newsample;
+  unsigned int newpos;
+
+  /* First check we have the appropriate metadata - we should always
+     have it. */
+  if ((demux_res->num_time_to_samples==0) ||
+    (demux_res->num_sample_byte_sizes==0)) { return 0; }
+
+  /* Find the destination block from time_to_sample array */
+  i=0;
+  newblock=0;
+  newsample=0;
+  flag=0;
+
+  while ((i<demux_res->num_time_to_samples) && (flag==0) && 
+         (newsample < sample_loc)) {
+    j=(sample_loc-newsample) /
+      demux_res->time_to_sample[i].sample_duration;
+    
+    if (j <= demux_res->time_to_sample[i].sample_count) {
+      newblock+=j;
+      newsample+=j*demux_res->time_to_sample[i].sample_duration;
+      flag=1;
+    } else {
+      newsample+=(demux_res->time_to_sample[i].sample_duration
+                 * demux_res->time_to_sample[i].sample_count);
+      newblock+=demux_res->time_to_sample[i].sample_count;
+      i++;
+    }
+  }
+
+  /* We know the new block, now calculate the file position */
+  newpos=mdat_offset;
+  for (i=0;i<newblock;i++) {
+    newpos+=demux_res->sample_byte_size[i];
+  }
+
+  /* We know the new file position, so let's try to seek to it */
+  if (ci->seek_buffer(newpos)) {
+    *samplesdone=newsample;
+    *currentblock=newblock;
+    return 1;
+  } else {
+    return 0;
+  }
+}
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api)
+{
+  size_t n;
+  demux_res_t demux_res;
+  static stream_t input_stream;
+  uint32_t samplesdone;
+  uint32_t elapsedtime;
+  uint32_t sample_duration;
+  uint32_t sample_byte_size;
+  int outputBytes;
+  unsigned int i;
+  unsigned char* buffer;
+  alac_file alac;
+  int16_t* pDestBuffer;
+
+  /* Generic codec initialisation */
+  TEST_CODEC_API(api);
+
+  rb = api;
+  ci = (struct codec_api*)api;
+
+#ifndef SIMULATOR
+  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+
+  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*128));
+
+  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:
+
+  if (codec_init(api)) {
+    LOGF("ALAC: Error initialising codec\n");
+    return CODEC_ERROR;
+  }
+
+  while (!rb->taginfo_ready)
+    rb->yield();
+  
+  if (rb->id3->frequency != NATIVE_FREQUENCY) {
+    rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
+    rb->configure(CODEC_DSP_ENABLE, (bool *)true);
+  } else {
+    rb->configure(CODEC_DSP_ENABLE, (bool *)false);
+  }
+
+  stream_create(&input_stream);
+
+  /* if qtmovie_read returns successfully, the stream is up to
+   * the movie data, which can be used directly by the decoder */
+  if (!qtmovie_read(&input_stream, &demux_res)) {
+    LOGF("ALAC: Error initialising file\n");
+    return CODEC_ERROR;
+  }
+
+  /* Keep track of start of stream in file - used for seeking */
+  mdat_offset=ci->curpos;
+
+  /* initialise the sound converter */
+  create_alac(demux_res.sample_size, demux_res.num_channels,&alac);
+  alac_set_info(&alac, demux_res.codecdata);
+
+  i=0;
+  samplesdone=0;
+  /* The main decoding loop */
+  while (i < demux_res.num_sample_byte_sizes) {
+    rb->yield();
+    if (ci->stop_codec || ci->reload_codec) {
+      break;
+    }
+
+    /* Deal with any pending seek requests */
+    if (ci->seek_time) {
+      if (alac_seek(&demux_res,
+                    (ci->seek_time/10) * (ci->id3->frequency/100),
+                    &samplesdone, &i)) {
+        elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
+        ci->set_elapsed(elapsedtime);
+      }
+      ci->seek_time = 0;
+    }
+
+    /* Lookup the length (in samples and bytes) of block i */
+    if (!get_sample_info(&demux_res, i, &sample_duration, 
+                         &sample_byte_size)) {
+      LOGF("ALAC: Error in get_sample_info\n");
+      return CODEC_ERROR;
+    }
+
+    /* Request the required number of bytes from the input buffer */
+
+    buffer=ci->request_buffer(&n,sample_byte_size);
+    if (n!=sample_byte_size) {
+      /* The decode_frame function requires the whole frame, so if we
+         can't get it contiguously from the buffer, then we need to
+         copy it via a read - i.e. we are at the buffer wraparound
+         point */
+
+      /* Check we estimated the maximum buffer size correctly */
+      if (sample_byte_size > sizeof(inputBuffer)) {
+        LOGF("ALAC: Input buffer < %d bytes\n",sample_byte_size);
+        return CODEC_ERROR;
+      }
+
+      n=ci->read_filebuf(inputBuffer,sample_byte_size);
+      if (n!=sample_byte_size) {
+        LOGF("ALAC: Error reading data\n");
+        return CODEC_ERROR;
+      }
+      buffer=inputBuffer;
+    }
+
+    /* Decode one block - returned samples will be host-endian */
+    outputBytes = destBufferSize;
+    rb->yield();
+    pDestBuffer=decode_frame(&alac, buffer, &outputBytes);
+
+    /* Advance codec buffer - unless we did a read */
+    if ((char*)buffer!=(char*)inputBuffer) {
+      ci->advance_buffer(n);
+    }
+
+    /* Output the audio */
+    rb->yield();
+    while(!ci->pcmbuf_insert((char*)pDestBuffer,outputBytes))
+      rb->yield();
+
+    /* Update the elapsed-time indicator */
+    samplesdone+=sample_duration;
+    elapsedtime=(samplesdone*10)/(ci->id3->frequency/100);
+    ci->set_elapsed(elapsedtime);
+
+    /* Keep track of current position - for resuming */
+    ci->set_offset(elapsedtime);
+
+    i++;
+  }
+
+  LOGF("ALAC: Decoded %d samples\n",samplesdone);
+
+  if (ci->request_next_track())
+   goto next_track;
+
+  return CODEC_OK;
+}