summaryrefslogtreecommitdiffstats
path: root/apps
diff options
context:
space:
mode:
authorDave Chapman <dave@dchapman.com>2006-02-01 16:42:02 +0000
committerDave Chapman <dave@dchapman.com>2006-02-01 16:42:02 +0000
commitfbd8e5d29c34c3c389ee32b5fedb613716985545 (patch)
tree11cd6dcdbc940a712256d8741b17b192bdf0f830 /apps
parent6479e4c95eafcdc13fc49bebe6dc24a8ea3b6d15 (diff)
downloadrockbox-fbd8e5d29c34c3c389ee32b5fedb613716985545.tar.gz
rockbox-fbd8e5d29c34c3c389ee32b5fedb613716985545.zip
Patch #1421483 - AIFF codec by Jvo Studer
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8524 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps')
-rw-r--r--apps/codecs/Makefile1
-rw-r--r--apps/codecs/SOURCES1
-rw-r--r--apps/codecs/aiff.c314
-rw-r--r--apps/metadata.c81
-rw-r--r--apps/playback.c5
-rw-r--r--apps/tree.c4
6 files changed, 404 insertions, 2 deletions
diff --git a/apps/codecs/Makefile b/apps/codecs/Makefile
index d0cd97db9b..915e922278 100644
--- a/apps/codecs/Makefile
+++ b/apps/codecs/Makefile
@@ -63,6 +63,7 @@ $(OBJDIR)/wavpack.elf: $(OBJDIR)/wavpack.o $(CODECDEPS) $(BUILDDIR)/libwavpack.a
$(OBJDIR)/alac.elf: $(OBJDIR)/alac.o $(CODECDEPS) $(BUILDDIR)/libalac.a $(BUILDDIR)/libm4a.a
$(OBJDIR)/aac.elf: $(OBJDIR)/aac.o $(CODECDEPS) $(BUILDDIR)/libfaad.a $(BUILDDIR)/libm4a.a
$(OBJDIR)/shorten.elf: $(OBJDIR)/shorten.o $(CODECDEPS) $(BUILDDIR)/libffmpegFLAC.a
+$(OBJDIR)/aiff.elf: $(OBJDIR)/aiff.o $(CODECDEPS)
$(OBJDIR)/%.elf: $(OBJDIR)/%.o $(CODECDEPS)
$(ELFIT)
diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES
index 911ee3b705..05172d6da9 100644
--- a/apps/codecs/SOURCES
+++ b/apps/codecs/SOURCES
@@ -11,4 +11,5 @@ alac.c
aac.c
#endif
shorten.c
+aiff.c
#endif
diff --git a/apps/codecs/aiff.c b/apps/codecs/aiff.c
new file mode 100644
index 0000000000..091e621bb7
--- /dev/null
+++ b/apps/codecs/aiff.c
@@ -0,0 +1,314 @@
+/***************************************************************************
+ * __________ __ ___.
+ * Open \______ \ ____ ____ | | _\_ |__ _______ ___
+ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
+ * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
+ * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
+ * \/ \/ \/ \/ \/
+ * $Id$
+ *
+ * Copyright (c) 2005 Jvo Studer
+ *
+ * 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 "codeclib.h"
+#include "inttypes.h"
+
+CODEC_HEADER
+
+struct codec_api* rb;
+
+/* This codec supports AIFF files with the following formats:
+ * - PCM, 8 and 16 bits, mono or stereo
+ */
+
+enum
+{
+ AIFF_FORMAT_PCM = 0x0001, /* AIFF PCM Format (big endian) */
+ IEEE_FORMAT_FLOAT = 0x0003, /* IEEE Float */
+ AIFF_FORMAT_ALAW = 0x0004, /* AIFC ALaw compressed */
+ AIFF_FORMAT_ULAW = 0x0005 /* AIFC uLaw compressed */
+};
+
+/* Maximum number of bytes to process in one iteration */
+/* for 44.1kHz stereo 16bits, this represents 0.023s ~= 1/50s */
+#define AIF_CHUNK_SIZE (1024*2)
+
+#ifdef USE_IRAM
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+extern char iedata[];
+extern char iend[];
+#endif
+
+static int16_t int16_samples[AIF_CHUNK_SIZE] IBSS_ATTR;
+
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api)
+{
+ struct codec_api* ci;
+ uint32_t numbytes, bytesdone;
+ uint16_t numChannels = 0;
+ uint32_t numSampleFrames = 0;
+ uint16_t sampleSize = 0;
+ uint32_t sampleRate = 0;
+ uint32_t i;
+ size_t n, aifbufsize;
+ int endofstream;
+ unsigned char* buf;
+ uint16_t* aifbuf;
+ long chunksize;
+ uint32_t offset2snd = 0;
+ uint16_t blockSize = 0;
+ uint32_t avgbytespersec = 0;
+ off_t firstblockposn; /* position of the first block in file */
+ int shortorlong = 1; /* do we output shorts (1) or longs (2)? */
+ int32_t * const int32_samples = (int32_t*)int16_samples;
+
+ /* Generic codec initialisation */
+ rb = api;
+ ci = api;
+
+#ifdef USE_IRAM
+ ci->memcpy(iramstart, iramcopy, iramend-iramstart);
+ ci->memset(iedata, 0, iend - iedata);
+#endif
+
+ ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
+ ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
+
+ ci->configure(DSP_DITHER, (bool *)false);
+
+ next_track:
+
+ if (codec_init(api)) {
+ i = CODEC_ERROR;
+ goto exit;
+ }
+
+ while (!*ci->taginfo_ready)
+ ci->yield();
+
+ /* assume the AIFF header is less than 1024 bytes */
+ buf=ci->request_buffer((long *)&n,1024);
+ if (n<44) {
+ i = CODEC_ERROR;
+ goto exit;
+ }
+ if ((memcmp(buf,"FORM",4)!=0) || (memcmp(&buf[8],"AIFF",4)!=0)) {
+ i = CODEC_ERROR;
+ goto exit;
+ }
+
+ buf += 12;
+ n -= 12;
+ numbytes = 0;
+
+ /* read until 'SSND' chunk, which typically is last */
+ while(numbytes == 0 && n >= 8) {
+ /* chunkSize */
+ i = ((buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7]);
+ if (memcmp(buf,"COMM",4)==0) {
+ if (i != 18) {
+ DEBUGF("CODEC_ERROR: 'COMM' chunk size=%lu != 18\n",i);
+ i = CODEC_ERROR;
+ goto exit;
+ }
+ /* numChannels */
+ numChannels = ((buf[8]<<8)|buf[9]);
+ /* numSampleFrames */
+ numSampleFrames = ((buf[10]<<24)|(buf[11]<<16)|(buf[12]<<8)|buf[13]);
+ /* sampleSize */
+ sampleSize = ((buf[14]<<8)|buf[15]);
+ /* sampleRate (don't use last 4 bytes, only integer fs) */
+ if (buf[16] != 0x40) {
+ DEBUGF("CODEC_ERROR: wierd sampling rate (no @)\n",i);
+ i = CODEC_ERROR;
+ goto exit;
+ }
+ sampleRate = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21])+1;
+ sampleRate = sampleRate >> (16+14-buf[17]);
+ /* calc average bytes per second */
+ avgbytespersec = sampleRate*numChannels*sampleSize/8;
+ }
+ else if (memcmp(buf,"SSND",4)==0) {
+ if (sampleSize == 0) {
+ DEBUGF("CODEC_ERROR: unsupported chunk order\n");
+ i = CODEC_ERROR;
+ goto exit;
+ }
+ /* offset2snd */
+ offset2snd = ((buf[8]<<8)|buf[9]);
+ /* blockSize */
+ blockSize = ((buf[10]<<8)|buf[11]);
+ if (blockSize == 0)
+ blockSize = numChannels*sampleSize;
+ numbytes = i-8-offset2snd;
+ i = 8+offset2snd; /* advance to the beginning of data */
+ }
+ else {
+ DEBUGF("unsupported AIFF chunk: '%c%c%c%c', size=%lu\n",
+ buf[0], buf[1], buf[2], buf[3], i);
+ }
+
+ if (i & 0x01) /* odd chunk sizes must be padded */
+ i++;
+ buf += i+8;
+ if (n < (i+8)) {
+ DEBUGF("CODEC_ERROR: AIFF header size > 1024\n");
+ i = CODEC_ERROR;
+ goto exit;
+ }
+ n -= i+8;
+ } /* while 'SSND' */
+
+ if (numChannels == 0) {
+ DEBUGF("CODEC_ERROR: 'COMM' chunk not found or 0-channels file\n");
+ i = CODEC_ERROR;
+ goto exit;
+ }
+ if (numbytes == 0) {
+ DEBUGF("CODEC_ERROR: 'SSND' chunk not found or has zero length\n");
+ i = CODEC_ERROR;
+ goto exit;
+ }
+ if (sampleSize > 24) {
+ DEBUGF("CODEC_ERROR: PCM with more than 24 bits per sample "
+ "is unsupported\n");
+ i = CODEC_ERROR;
+ goto exit;
+ }
+
+ ci->configure(CODEC_DSP_ENABLE, (bool *)true);
+ ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
+
+ if (sampleSize <= 16) {
+ ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
+ } else {
+ shortorlong = 2;
+ ci->configure(DSP_DITHER, (bool *)false);
+ ci->configure(DSP_SET_SAMPLE_DEPTH, (long *) (32));
+ ci->configure(DSP_SET_CLIP_MAX, (long *) (2147483647));
+ ci->configure(DSP_SET_CLIP_MIN, (long *) (-2147483647-1));
+ }
+
+ if (numChannels == 2) {
+ ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
+ } else if (numChannels == 1) {
+ ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
+ } else {
+ DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
+ i = CODEC_ERROR;
+ goto exit;
+ }
+
+ firstblockposn = (1024-n);
+ ci->advance_buffer(firstblockposn);
+
+ /* The main decoder loop */
+
+ bytesdone=0;
+ ci->set_elapsed(0);
+ endofstream=0;
+ /* chunksize is computed so that one chunk is about 1/50s.
+ * this make 4096 for 44.1kHz 16bits stereo.
+ * It also has to be a multiple of blockalign */
+ chunksize = (1 + avgbytespersec / (50*blockSize)) * blockSize;
+ /* check that the output buffer is big enough (convert to samplespersec,
+ then round to the blockSize multiple below) */
+ if (((uint64_t)chunksize*ci->id3->frequency*numChannels*shortorlong)
+ / (uint64_t)avgbytespersec >= AIF_CHUNK_SIZE) {
+ chunksize = ((uint64_t)AIF_CHUNK_SIZE * avgbytespersec
+ / ((uint64_t)ci->id3->frequency * numChannels * shortorlong
+ * blockSize)) * blockSize;
+ }
+
+ while (!endofstream) {
+ uint8_t *aifbuf8;
+
+ ci->yield();
+ if (ci->stop_codec || ci->reload_codec) {
+ break;
+ }
+
+ if (ci->seek_time) {
+ uint32_t newpos;
+
+ /* use avgbytespersec to round to the closest blockalign multiple,
+ add firstblockposn. 64-bit casts to avoid overflows. */
+ newpos = (((uint64_t)avgbytespersec * (ci->seek_time - 1))
+ / (1000LL*blockSize)) * blockSize;
+ if (newpos > numbytes)
+ break;
+ if (ci->seek_buffer(firstblockposn + newpos)) {
+ bytesdone = newpos;
+ }
+ ci->seek_complete();
+ }
+ aifbuf=ci->request_buffer((long *)&n,chunksize);
+ aifbuf8 = (uint8_t*)aifbuf;
+
+ if (n==0)
+ break; /* End of stream */
+
+ if (bytesdone + n > numbytes) {
+ n = numbytes - bytesdone;
+ endofstream = 1;
+ }
+
+ aifbufsize = sizeof(int16_samples);
+
+ if (sampleSize > 24) {
+ for (i=0;i<n;i+=4) {
+ int32_samples[i/4]=(int32_t)((aifbuf8[i]<<24)|
+ (aifbuf8[i+1]<<16)|(aifbuf8[i+2]<<8)|aifbuf8[i+3]);
+ }
+ aifbufsize = n;
+ } else if (sampleSize > 16) {
+ for (i=0;i<n;i+=3) {
+ int32_samples[i/3]=(int32_t)((aifbuf8[i]<<24)|
+ (aifbuf8[i+1]<<16)|(aifbuf8[i+2]<<8));
+ }
+ aifbufsize = n*4/3;
+ } else if (sampleSize > 8) {
+ /* copy data. */
+ for (i=0;i<n;i+=2) {
+ int16_samples[i/2]=(int16_t)((aifbuf8[i]<<8)|aifbuf8[i+1]);
+ }
+ aifbufsize = n;
+ } else {
+ for (i=0;i<n;i++) {
+ int16_samples[i] = (aifbuf8[i]<<8) - 0x8000;
+ }
+ aifbufsize = n*2;
+ }
+
+ while (!ci->pcmbuf_insert((char*)int16_samples, aifbufsize)) {
+ ci->yield();
+ }
+
+ ci->advance_buffer(n);
+ bytesdone += n;
+ if (bytesdone >= numbytes) {
+ endofstream=1;
+ }
+
+ ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
+ }
+
+ if (ci->request_next_track())
+ goto next_track;
+
+ i = CODEC_OK;
+exit:
+ return i;
+}
+
diff --git a/apps/metadata.c b/apps/metadata.c
index 41ea0196c0..531969b8aa 100644
--- a/apps/metadata.c
+++ b/apps/metadata.c
@@ -78,6 +78,8 @@ static const struct format_list formats[] =
{ AFMT_ALAC, "m4a" },
{ AFMT_AAC, "mp4" },
{ AFMT_SHN, "shn" },
+ { AFMT_AIFF, "aif" },
+ { AFMT_AIFF, "aiff" },
};
static const unsigned short a52_bitrates[] =
@@ -894,7 +896,6 @@ static bool get_wave_metadata(int fd, struct mp3entry* id3)
}
-
static bool get_m4a_metadata(int fd, struct mp3entry* id3)
{
unsigned char* buf;
@@ -1245,6 +1246,76 @@ static bool get_musepack_metadata(int fd, struct mp3entry *id3)
return true;
}
+static bool get_aiff_metadata(int fd, struct mp3entry* id3)
+{
+ /* Use the trackname part of the id3 structure as a temporary buffer */
+ unsigned char* buf = id3->path;
+ unsigned long numChannels = 0;
+ unsigned long numSampleFrames = 0;
+ unsigned long sampleSize = 0;
+ unsigned long sampleRate = 0;
+ unsigned long numbytes = 0;
+ int read_bytes;
+ int i;
+
+ if ((lseek(fd, 0, SEEK_SET) < 0)
+ || ((read_bytes = read(fd, buf, sizeof(id3->path))) < 44))
+ {
+ return false;
+ }
+
+ if ((memcmp(buf, "FORM",4) != 0)
+ || (memcmp(&buf[8], "AIFF", 4) !=0 ))
+ {
+ return false;
+ }
+
+ buf += 12;
+ read_bytes -= 12;
+
+ while ((numbytes == 0) && (read_bytes >= 8))
+ {
+ /* chunkSize */
+ i = ((buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7]);
+
+ if (memcmp(buf, "COMM", 4) == 0)
+ {
+ /* numChannels */
+ numChannels = ((buf[8]<<8)|buf[9]);
+ /* numSampleFrames */
+ numSampleFrames =((buf[10]<<24)|(buf[11]<<16)|(buf[12]<<8)|buf[13]);
+ /* sampleSize */
+ sampleSize = ((buf[14]<<8)|buf[15]);
+ /* sampleRate */
+ sampleRate = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21]);
+ sampleRate = sampleRate >> (16+14-buf[17]);
+ /* save format infos */
+ id3->bitrate = (sampleSize * numChannels * sampleRate) / 1000;
+ id3->frequency = sampleRate;
+ id3->length = (numSampleFrames / id3->frequency) * 1000;
+ id3->vbr = false; /* AIFF files are CBR */
+ id3->filesize = filesize(fd);
+ }
+ else if (memcmp(buf, "SSND", 4) == 0)
+ {
+ numbytes = i - 8;
+ }
+
+ if (i & 0x01)
+ {
+ i++; /* odd chunk sizes must be padded */
+ }
+ buf += i + 8;
+ read_bytes -= i + 8;
+ }
+
+ if ((numbytes == 0) || (numChannels == 0))
+ {
+ return false;
+ }
+ return true;
+}
+
/* Simple file type probing by looking at the filename extension. */
static unsigned int probe_file_format(const char *filename)
{
@@ -1448,6 +1519,14 @@ bool get_metadata(struct track_info* track, int fd, const char* trackname,
/* TODO: read the id3v2 header if it exists */
break;
+ case AFMT_AIFF:
+ if (!get_aiff_metadata(fd, &(track->id3)))
+ {
+ return false;
+ }
+
+ break;
+
default:
/* If we don't know how to read the metadata, assume we can't play
the file */
diff --git a/apps/playback.c b/apps/playback.c
index 22ee3362c4..5ed6c5e00c 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -81,6 +81,7 @@ static volatile bool paused;
#define CODEC_ALAC "/.rockbox/codecs/alac.codec"
#define CODEC_AAC "/.rockbox/codecs/aac.codec"
#define CODEC_SHN "/.rockbox/codecs/shorten.codec"
+#define CODEC_AIFF "/.rockbox/codecs/aiff.codec"
#define AUDIO_DEFAULT_FIRST_LIMIT (1024*1024*10)
#define AUDIO_FILL_CYCLE (1024*256)
@@ -950,6 +951,10 @@ static bool loadcodec(bool start_play)
logf("Codec: SHN");
codec_path = CODEC_SHN;
break;
+ case AFMT_AIFF:
+ logf("Codec: PCM AIFF");
+ codec_path = CODEC_AIFF;
+ break;
default:
logf("Codec: Unsupported");
codec_path = NULL;
diff --git a/apps/tree.c b/apps/tree.c
index 15624d7618..74c1059a60 100644
--- a/apps/tree.c
+++ b/apps/tree.c
@@ -85,7 +85,7 @@ const struct filetype filetypes[] = {
{ "ogg", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
{ "wma", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
{ "wav", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
- { "flac", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
+ { "flac",TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
{ "ac3", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
{ "a52", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
{ "mpc", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
@@ -93,6 +93,8 @@ const struct filetype filetypes[] = {
{ "m4a", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
{ "mp4", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
{ "shn", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
+ { "aif", TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
+ { "aiff",TREE_ATTR_MPA, Icon_Audio, VOICE_EXT_MPA },
#endif
{ "m3u", TREE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },
{ "cfg", TREE_ATTR_CFG, Icon_Config, VOICE_EXT_CFG },