Patch #1421483 - AIFF codec by Jvo Studer

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@8524 a1c6a512-1295-4272-9138-f99709370657

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)

apps/codecs/SOURCES

@@ -11,4 +11,5 @@ alac.c
 aac.c
 #endif
 shorten.c
+aiff.c
 #endif

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;
+}
+

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 */

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;

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 },

docs/CREDITS

@@ -161,3 +161,4 @@ Tomasz Malesinski
 Andrew Pilley
 Matt v.d. Westhuizen
 Tim Crist
+Jvo Studer

firmware/export/id3.h

@@ -41,6 +41,7 @@ enum {
     AFMT_ALAC,         /* Apple Lossless Audio Codec */
     AFMT_AAC,          /* Advanced Audio Coding (AAC) in M4A container */
     AFMT_SHN,          /* Shorten */
+    AFMT_AIFF,         /* Audio Interchange File Format */
 
     /* New formats must be added to the end of this list */
 

firmware/id3.c

@@ -102,6 +102,7 @@ static const char* const codec_labels[] = {
   "ALAC", /* Apple Lossless Audio Codec */
   "AAC",  /* Advanced Audio Coding in M4A container */
   "SHN",  /* Shorten */
+  "AIFF", /* Audio Interchange File Format */
 #endif
 };