More stable playback with reduced stuttering when skipping tracks.

Removed CODEC_SET_FILEBUF_LIMIT setting; now playback.c determines how
to buffer the files.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7970 a1c6a512-1295-4272-9138-f99709370657

12 files changed, 22 insertions, 26 deletions

diff --git a/apps/codecs/a52.c b/apps/codecs/a52.c
index 050bfff25d..ee80f66892 100644
--- a/apps/codecs/a52.c
+++ b/apps/codecs/a52.c
@@ -140,7 +140,6 @@ enum codec_status codec_start(struct codec_api *api)
     ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)30);
     ci->configure(DSP_SET_CLIP_MAX, (long *)((1 << 30) - 1));
     ci->configure(DSP_SET_CLIP_MIN, (long *)-(1 << 30));
-    ci->configure(CODEC_SET_FILEBUF_LIMIT, (long *)(1024*1024*2));
     ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (long *)(1024*128));
 
 next_track:
diff --git a/apps/codecs/aac.c b/apps/codecs/aac.c
index f6a6de53da..c62fb2d007 100644
--- a/apps/codecs/aac.c
+++ b/apps/codecs/aac.c
@@ -60,7 +60,6 @@ enum codec_status codec_start(struct codec_api* api)
     rb->memcpy(iramstart, iramcopy, iramend-iramstart);
 #endif
 
-    ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
     ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
     ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
 
diff --git a/apps/codecs/alac.c b/apps/codecs/alac.c
index 1ecaab6b85..deb1e289a7 100644
--- a/apps/codecs/alac.c
+++ b/apps/codecs/alac.c
@@ -57,7 +57,6 @@ enum codec_status codec_start(struct codec_api* api)
   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));
 
diff --git a/apps/codecs/flac.c b/apps/codecs/flac.c
index b94152dec5..7639b9a352 100644
--- a/apps/codecs/flac.c
+++ b/apps/codecs/flac.c
@@ -233,7 +233,6 @@ enum codec_status codec_start(struct codec_api* api)
     ci->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));
 
diff --git a/apps/codecs/mpa.c b/apps/codecs/mpa.c
index c6342df262..0f575924d0 100644
--- a/apps/codecs/mpa.c
+++ b/apps/codecs/mpa.c
@@ -95,7 +95,6 @@ enum codec_status codec_start(struct codec_api *api)
     ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(MAD_F_FRACBITS));
     ci->configure(DSP_SET_CLIP_MIN, (int *)-MAD_F_ONE);
     ci->configure(DSP_SET_CLIP_MAX, (int *)(MAD_F_ONE - 1));
-    ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
     ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
     
     ci->memset(&stream, 0, sizeof(struct mad_stream));
diff --git a/apps/codecs/mpc.c b/apps/codecs/mpc.c
index 0f113c7bdc..db2501525e 100644
--- a/apps/codecs/mpc.c
+++ b/apps/codecs/mpc.c
@@ -96,7 +96,6 @@ enum codec_status codec_start(struct codec_api *api)
     /* disable these until we can figure out what's going on.
     ci->configure(DSP_SET_CLIP_MAX, (long *)MPC_FIXED_POINT_SCALE);
     ci->configure(DSP_SET_CLIP_MIN, (long *)-MPC_FIXED_POINT_SCALE);*/
-    ci->configure(CODEC_SET_FILEBUF_LIMIT, (long *)(1024*1024*2));
     ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (long *)(1024*16));
     
     /* Create a decoder instance */
diff --git a/apps/codecs/shorten.c b/apps/codecs/shorten.c
index 50d72997ce..26b5e09d6c 100644
--- a/apps/codecs/shorten.c
+++ b/apps/codecs/shorten.c
@@ -56,7 +56,6 @@ enum codec_status codec_start(struct codec_api* api)
     ci->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));
 
diff --git a/apps/codecs/vorbis.c b/apps/codecs/vorbis.c
index 1fbe131e10..69eba24bb8 100644
--- a/apps/codecs/vorbis.c
+++ b/apps/codecs/vorbis.c
@@ -135,7 +135,6 @@ enum codec_status codec_start(struct codec_api *api)
     /* Note: These are sane defaults for these values.  Perhaps
      * they should be set differently based on quality setting
      */
-    rb->configure(CODEC_SET_FILEBUF_LIMIT, (long *)(1024*1024*2));
 
     /* The chunk size below is magic.  If set any lower, resume
      * doesn't work properly (ov_raw_seek() does the wrong thing).
diff --git a/apps/codecs/wav.c b/apps/codecs/wav.c
index 6bf7bcd408..85e24124ab 100644
--- a/apps/codecs/wav.c
+++ b/apps/codecs/wav.c
@@ -237,7 +237,6 @@ enum codec_status codec_start(struct codec_api* api)
   ci->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*256));
 
diff --git a/apps/codecs/wavpack.c b/apps/codecs/wavpack.c
index 93f5002538..21c194773b 100644
--- a/apps/codecs/wavpack.c
+++ b/apps/codecs/wavpack.c
@@ -58,7 +58,6 @@ enum codec_status codec_start(struct codec_api* api)
     ci->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));
   
diff --git a/apps/dsp.h b/apps/dsp.h
index 19ba6db980..f250d89708 100644
--- a/apps/dsp.h
+++ b/apps/dsp.h
@@ -31,7 +31,6 @@
 enum {
     CODEC_SET_FILEBUF_WATERMARK = 1,
     CODEC_SET_FILEBUF_CHUNKSIZE,
-    CODEC_SET_FILEBUF_LIMIT,
     CODEC_DSP_ENABLE,
     DSP_SET_FREQUENCY,
     DSP_SWITCH_FREQUENCY,
diff --git a/apps/playback.c b/apps/playback.c
index 6a944c9dc8..2418eba367 100644
--- a/apps/playback.c
+++ b/apps/playback.c
@@ -81,6 +81,7 @@ static volatile bool paused;
 #define CODEC_AAC      "/.rockbox/codecs/aac.codec"
 #define CODEC_SHN      "/.rockbox/codecs/shorten.codec"
 
+#define AUDIO_DEFAULT_FIRST_LIMIT    (1024*1024*10)
 #define AUDIO_FILL_CYCLE             (1024*256)
 #define AUDIO_DEFAULT_WATERMARK      (1024*512)
 #define AUDIO_DEFAULT_FILECHUNK      (1024*32)
@@ -717,10 +718,6 @@ void codec_configure_callback(int setting, void *value)
         conf_filechunk = (unsigned int)value;
         break;
         
-    case CODEC_SET_FILEBUF_LIMIT:
-        conf_bufferlimit = (unsigned int)value;
-        break;
-        
     case CODEC_DSP_ENABLE:
         if ((bool)value)
             ci.pcmbuf_insert = codec_pcmbuf_insert_callback;
@@ -1009,9 +1006,11 @@ bool read_next_metadata(void)
     fd = open(trackname, O_RDONLY);
     if (fd < 0)
         return false;
-        
-    /* Start buffer refilling also because we need to spin-up the disk. */
-    filling = true;
+    
+    /** Start buffer refilling also because we need to spin-up the disk.
+     * In fact, it might be better not to start filling here, because if user
+     * is manipulating the playlist a lot, we will just lose battery. */
+    // filling = true;
     tracks[next_track].id3.codectype = probe_file_format(trackname);
     status = get_metadata(&tracks[next_track],fd,trackname,v1first);
     tracks[next_track].id3.codectype = 0;
@@ -1081,7 +1080,7 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
     if (start_play) {
         int last_codec = current_codec;
         current_codec = CODEC_IDX_AUDIO;
-        conf_bufferlimit = 0;
+        conf_bufferlimit = AUDIO_DEFAULT_FIRST_LIMIT;
         conf_watermark = AUDIO_DEFAULT_WATERMARK;
         conf_filechunk = AUDIO_DEFAULT_FILECHUNK;
         dsp_configure(DSP_RESET, 0);
@@ -1329,13 +1328,13 @@ void initialize_buffer_fill(void)
 {
     int cur_idx, i;
     
-    
-    fill_bytesleft = filebuflen - filebufused;
-    cur_ti->start_pos = ci.curpos;
-
+    /* Initialize only once; do not truncate the tracks. */
     if (filling)
         return ;
 
+    fill_bytesleft = filebuflen - filebufused;
+    cur_ti->start_pos = ci.curpos;
+
     pcmbuf_set_boost_mode(true);
     
     filling = true;
@@ -1392,8 +1391,14 @@ void audio_check_buffer(void)
     
     /* Load new files to fill the entire buffer. */
     if (audio_load_track(0, false, last_peek_offset + 1)) {
-    
-    } else if (tracks[track_widx].filerem == 0 || fill_bytesleft == 0) {
+        if (conf_bufferlimit)
+            fill_bytesleft = 0;
+    }
+    else if (tracks[track_widx].filerem == 0)
+        fill_bytesleft = 0;
+
+    if (fill_bytesleft <= 0)
+    {
         /* Read next unbuffered track's metadata as necessary. */
         read_next_metadata();
         
@@ -1659,7 +1664,7 @@ void audio_thread(void)
     bool play_pending = false;
     
     while (1) {
-        if (!play_pending)
+        if (!play_pending && queue_empty(&audio_queue))
         {
             yield_codecs();
             audio_check_buffer();
@@ -1683,12 +1688,14 @@ void audio_thread(void)
                  * fast to prevent UI lag. */
                 track_count = 0;
                 last_peek_offset = 0;
+                track_changed = true;
                 if (current_tick - last_tick < HZ/2)
                 {
                     play_pending = true;
                     break ;
                 }
                 play_pending = false;
+                last_tick = current_tick;
             
                 /* Do not start crossfading if audio is paused. */
                 if (paused)