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
Not only WAV but also Sun audio, SMAF, vox and WAV64 can resume.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25289 a1c6a512-1295-4272-9138-f99709370657
- does not get dwAvgBytesPerSec
wave/aiff/smaf/wave64 codec
- corrects the problem that codec_main() returns invalid value.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25051 a1c6a512-1295-4272-9138-f99709370657
