Add first version of WAV playback to iRiver - only 16-bit Stereo 44.1KHz files supported.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6654 a1c6a512-1295-4272-9138-f99709370657

2 files changed, 183 insertions, 1 deletions

diff --git a/apps/playback.c b/apps/playback.c
index 71cb479059..4cefed20c2 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -598,7 +598,7 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
     int rc, i;
     int copy_n;
     /* Used by the FLAC metadata parser */
-    unsigned long totalsamples;
+    unsigned long totalsamples,bytespersample,channels,bitspersample,numbytes;
     unsigned char* buf;
     
     if (track_count >= MAX_TRACK)
@@ -677,6 +677,57 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
         tracks[track_widx].taginfo_ready = true;
         break ;
 
+    case AFMT_PCM_WAV:
+        /* Use the trackname part of the id3 structure as a temporary buffer */
+        buf=tracks[track_widx].id3.path;
+
+        lseek(fd, 0, SEEK_SET);
+
+        rc = read(fd, buf, 44);
+        if (rc < 44) {
+          close(fd);
+          return false;
+        }
+
+        if ((memcmp(buf,"RIFF",4)!=0) || 
+            (memcmp(&buf[8],"WAVEfmt",7)!=0)) {
+          logf("%s is not a WAV file\n",trackname);
+          close(fd);
+          return(false);
+        }
+
+        /* FIX: Correctly parse WAV header - we assume canonical 
+           44-byte header */
+
+        bitspersample=buf[34];
+        channels=buf[22];
+
+        if ((bitspersample!=16) || (channels != 2)) {
+          logf("Unsupported WAV file - %d bitspersample, %d channels\n",
+               bitspersample,channels);
+          close(fd);
+          return(false);
+        }
+
+        bytespersample=((bitspersample/8)*channels);
+        numbytes=(buf[40]|(buf[41]<<8)|(buf[42]<<16)|(buf[43]<<24));
+        totalsamples=numbytes/bytespersample;
+
+        tracks[track_widx].id3.vbr=false;   /* All WAV files are CBR */
+        tracks[track_widx].id3.filesize=filesize(fd);
+        tracks[track_widx].id3.frequency=buf[24]|(buf[25]<<8)|(buf[26]<<16)|(buf[27]<<24);
+
+        /* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
+        tracks[track_widx].id3.length=(totalsamples/tracks[track_widx].id3.frequency)*1000;
+        tracks[track_widx].id3.bitrate=(tracks[track_widx].id3.frequency*bytespersample)/1024;
+
+        lseek(fd, 0, SEEK_SET);
+        strncpy(tracks[track_widx].id3.path,trackname,sizeof(tracks[track_widx].id3.path));
+        tracks[track_widx].taginfo_ready = true;
+
+        break;
+    
+
     case AFMT_FLAC:
         /* A simple parser to read vital metadata from a FLAC file - length, frequency, bitrate etc. */
         /* This code should either be moved to a seperate file, or discarded in favour of the libFLAC code */
diff --git a/apps/plugins/codecwav.c b/apps/plugins/codecwav.c
new file mode 100644
index 0000000000..a02f2cdbff
--- /dev/null
+++ b/apps/plugins/codecwav.c
@@ -0,0 +1,131 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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 "plugin.h"
+#include "playback.h"
+#include "lib/codeclib.h"
+
+#ifndef SIMULATOR
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+/* This is probably a waste of IRAM, but why not? */
+static unsigned char wavbuf[16384] IDATA_ATTR;
+
+/* this is the plugin entry point */
+enum plugin_status plugin_start(struct plugin_api* api, void* parm)
+{
+  struct plugin_api* rb = (struct plugin_api*)api;
+  struct codec_api* ci = (struct codec_api*)parm;
+  unsigned long samplerate,numbytes,totalsamples,samplesdone,nsamples;
+  int channels,bytespersample,bitspersample;
+  unsigned int i,j,n;
+  int endofstream;
+
+  /* Generic plugin initialisation */
+  TEST_PLUGIN_API(api);
+
+  /* if you are using a global api pointer, don't forget to copy it!
+     otherwise you will get lovely "I04: IllInstr" errors... :-) */
+  rb = 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));
+
+  next_track:
+
+  if (codec_init(api, ci)) {
+    return PLUGIN_ERROR;
+  }
+
+  /* FIX: Correctly parse WAV header - we assume canonical 44-byte header */
+
+  n=(unsigned)(ci->read_filebuf(wavbuf,44));
+  if (n!=44) {
+    return PLUGIN_ERROR;
+  }
+  if ((memcmp(wavbuf,"RIFF",4)!=0) || (memcmp(&wavbuf[8],"WAVEfmt",7)!=0)) {
+    return PLUGIN_ERROR;
+  }
+
+  samplerate=wavbuf[24]|(wavbuf[25]<<8)|(wavbuf[26]<<16)|(wavbuf[27]<<24);
+  bitspersample=wavbuf[34];
+  channels=wavbuf[22];
+  bytespersample=((bitspersample/8)*channels);
+  numbytes=(wavbuf[40]|(wavbuf[41]<<8)|(wavbuf[42]<<16)|(wavbuf[43]<<24));
+  totalsamples=numbytes/bytespersample;
+
+  if ((bitspersample!=16) || (channels != 2)) {
+    return PLUGIN_ERROR;
+  }
+    
+  /* The main decoder loop */
+
+  samplesdone=0;
+  ci->set_elapsed(0);
+  endofstream=0;
+  while (!endofstream) {
+    rb->yield();
+    if (ci->stop_codec || ci->reload_codec) {
+      break;
+    }
+
+    n=(unsigned)(ci->read_filebuf(wavbuf,sizeof(wavbuf)));
+
+    rb->yield();
+
+    if (n==0) break; /* End of stream */
+
+    nsamples=(n/bytespersample);
+
+    /* WAV files can contain extra data at the end - so we can't just
+       process until the end of the file */
+
+    if (samplesdone+nsamples > totalsamples) {
+      nsamples=(totalsamples-samplesdone);
+      n=nsamples*bytespersample;
+      endofstream=1;
+    }
+
+    /* Byte-swap data */
+    for (i=0;i<n;i+=2) {
+        j=wavbuf[i];
+        wavbuf[i]=wavbuf[i+1];
+        wavbuf[i+1]=j;
+    }
+
+    samplesdone+=nsamples;
+    ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
+   
+    rb->yield();
+    while (!ci->audiobuffer_insert(wavbuf, n))
+      rb->yield();
+  }
+
+  if (ci->request_next_track())
+    goto next_track;
+
+  return PLUGIN_OK;
+}