ReplayGain support for Ogg Vorbis files (also called VorbisGain) added.

Note that there is a small delay from leaving a setting until the change
can be heard (due to audio data buffering).


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7234 a1c6a512-1295-4272-9138-f99709370657

apps/codecs/Tremor/ivorbisfile.h

@@ -121,6 +121,8 @@ extern vorbis_comment *ov_comment(OggVorbis_File *vf,int link);
 
 extern long ov_read(OggVorbis_File *vf,char *buffer,int length,
 		    int *bitstream);
+extern long ov_read_fixed(OggVorbis_File *vf,ogg_int32_t ***pcm_channels,
+                          int length,int *bitstream);
 
 #ifdef __cplusplus
 }

apps/codecs/Tremor/vorbisfile.c

@@ -1604,3 +1604,47 @@ long ov_read(OggVorbis_File *vf,char *buffer,int bytes_req,int *bitstream){
     return(samples);
   }
 }
+
+/* input values: pcm_channels) a float vector per channel of output
+                length) the sample length being read by the app
+
+   return values: <0) error/hole in data (OV_HOLE), partial open (OV_EINVAL)
+                   0) EOF
+                  n) number of samples of PCM actually returned.  The
+                  below works on a packet-by-packet basis, so the
+                  return length is not related to the 'length' passed
+                  in, just guaranteed to fit.
+
+           *section) set to the logical bitstream number */
+
+long ov_read_fixed(OggVorbis_File *vf,ogg_int32_t ***pcm_channels,int length,
+                  int *bitstream){
+  if(vf->ready_state<OPENED)return(OV_EINVAL);
+
+#if CONFIG_CPU == MCF5249
+  mcf5249_init_mac();
+#endif
+
+  while(1){
+    if(vf->ready_state==INITSET){
+      ogg_int32_t **pcm;
+      long samples=vorbis_synthesis_pcmout(&vf->vd,&pcm);
+      if(samples){
+       if(pcm_channels)*pcm_channels=pcm;
+       if(samples>length)samples=length;
+       vorbis_synthesis_read(&vf->vd,samples);
+       vf->pcm_offset+=samples;
+       if(bitstream)*bitstream=vf->current_link;
+       return samples;
+
+      }
+    }
+
+    /* suck in another packet */
+    {
+      int ret=_fetch_and_process_packet(vf,1,1);
+      if(ret==OV_EOF)return(0);
+      if(ret<=0)return(ret);
+    }
+  }
+}

apps/codecs/lib/codeclib.c

@@ -23,6 +23,7 @@
 #include "playback.h"
 #include "codeclib.h"
 #include "xxx2wav.h"
+#include "id3.h"
 
 struct codec_api *local_rb;
 
@@ -34,3 +35,11 @@ int codec_init(struct codec_api* rb)
   
     return 0;
 }
+
+void codec_set_replaygain(struct mp3entry* id3)
+{
+    local_rb->configure(DSP_SET_TRACK_GAIN, (long *) id3->track_gain);
+    local_rb->configure(DSP_SET_ALBUM_GAIN, (long *) id3->album_gain);
+    local_rb->configure(DSP_SET_TRACK_PEAK, (long *) id3->track_peak);
+    local_rb->configure(DSP_SET_ALBUM_PEAK, (long *) id3->album_peak);
+}

apps/codecs/lib/codeclib.h

@@ -37,4 +37,4 @@ int memcmp(const void *s1, const void *s2, size_t n);
 void* memmove(const void *s1, const void *s2, size_t n);
 
 int codec_init(struct codec_api* rb);
-
+void codec_set_replaygain(struct mp3entry* id3);

apps/codecs/vorbis.c

@@ -105,16 +105,11 @@ bool vorbis_set_codec_parameters(OggVorbis_File *vf)
         return false;
     }
 
-    if (rb->id3->frequency != NATIVE_FREQUENCY) {
-        rb->configure(CODEC_DSP_ENABLE, (bool *)true);
-    } else {
-        rb->configure(CODEC_DSP_ENABLE, (bool *)false);
-    }
-
     rb->configure(DSP_SET_FREQUENCY, (int *)rb->id3->frequency);
+    codec_set_replaygain(rb->id3);
 
     if (vi->channels == 2) {
-          rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
+          rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
     } else if (vi->channels == 1) {
           rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
     }
@@ -129,14 +124,12 @@ extern char iramend[];
 #endif
 
 
-/* reserve the PCM buffer in the IRAM area */
-static char pcmbuf[4096] IDATA_ATTR;
-
 /* this is the codec entry point */
 enum codec_status codec_start(struct codec_api* api)
 {
     ov_callbacks callbacks;
     OggVorbis_File vf;
+    ogg_int32_t** pcm;
 
     int error;
     long n;
@@ -157,10 +150,12 @@ enum codec_status codec_start(struct codec_api* api)
     #ifdef USE_IRAM
     rb->memcpy(iramstart, iramcopy, iramend-iramstart);
     #endif
-        
-    rb->configure(DSP_DITHER, (bool *)false);
-    rb->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
 
+    rb->configure(CODEC_DSP_ENABLE, (bool *)true);
+    rb->configure(DSP_DITHER, (bool *)false);
+    rb->configure(DSP_SET_SAMPLE_DEPTH, (long *) (24));
+    rb->configure(DSP_SET_CLIP_MAX, (long *)  ((1 << 24) - 1));
+    rb->configure(DSP_SET_CLIP_MIN, (long *) -((1 << 24) - 1));
     /* Note: These are sane defaults for these values.  Perhaps
      * they should be set differently based on quality setting
      */
@@ -244,9 +239,9 @@ enum codec_status codec_start(struct codec_api* api)
             }
             rb->seek_time = 0;
         }
-                
-        /* Read host-endian signed 16 bit PCM samples */
-        n=ov_read(&vf,pcmbuf,sizeof(pcmbuf),&current_section);
+
+        /* Read host-endian signed 24-bit PCM samples */
+        n=ov_read_fixed(&vf,&pcm,1024,&current_section);
 
         /* Change DSP and buffer settings for this bitstream */
         if ( current_section != previous_section ) {
@@ -262,9 +257,10 @@ enum codec_status codec_start(struct codec_api* api)
         } else if (n < 0) {
             DEBUGF("Error decoding frame\n");
         } else {
-            while (!rb->pcmbuf_insert(pcmbuf, n)) {
+            while (!rb->pcmbuf_insert_split(pcm[0], pcm[1], 
+                n * sizeof(ogg_int32_t))) {
                 rb->sleep(1);
-        }
+            }
 
         rb->set_offset(ov_raw_tell(&vf));
         rb->set_elapsed(ov_time_tell(&vf));