Commit work started in FS#12153 to put timing/position information in PCM

buffer chunks.

* Samples and position indication is closely associated with audio data
  instead of compensating by a latency constant. Alleviates problems with
  using the elapsed as a track indicator where it could be off by several
  steps.

* Timing is accurate throughout track even if resampling for pitch shift,
  whereas before it updated during transition latency at the normal 1:1 rate.

* Simpler PCM buffer with a constant chunk size, no linked lists.

In converting crossfade, a minor change was made to not change the WPS until
the fade-in of the incoming track, whereas before it would change upon the
start of the fade-out of the outgoing track possibly having the WPS change
with far too much lead time.

Codec changes are to set elapsed times *before* writing next PCM frame because
 time and position data last set are saved in the next committed PCM chunk. 


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 3 insertions, 2 deletions

diff --git a/apps/codecs/aac.c b/apps/codecs/aac.c
index 52e08c7b56..f1e81bad81 100644
--- a/apps/codecs/aac.c
+++ b/apps/codecs/aac.c
@@ -134,8 +134,6 @@ enum codec_status codec_run(void)
         if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
                           &sound_samples_done, (int*) &i)) {
             sound_samples_done *= sbr_fac;
-            elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
-            ci->set_elapsed(elapsed_time);
         } else {
             sound_samples_done = 0;
         }
@@ -143,6 +141,9 @@ enum codec_status codec_run(void)
     } else {
         sound_samples_done = 0;
     }
+
+    elapsed_time = (sound_samples_done * 10) / (ci->id3->frequency / 100);
+    ci->set_elapsed(elapsed_time);
     
     if (i == 0) 
     {