Fix problem when 'ended' event was fired before stream really ended.

That caused impression that rewind does not work. With that
change small JS program

 var a = new Audio("file:///home/enal/temp/click2/click2.wav");
 var num_played = 0;
 a.addEventListener('canplaythrough', function() {
   a.play();
 });

 a.addEventListener('ended', function() {
   num_played ++;
   if (num_played < 10) {
     a.currentTime = 0;
     a.play();
   }
 });

works correctly, you hear 10 clicks one after another, and it takes
~1.5 seconds to play all 10 sounds (one click is 146ms). Current
Chrome plays only beginnings of the first 9 clicks and then entire
10th click -- 'ended' event fires too early, so rewind stops audio
playback for all clicks but last one.

With that fix you can easily create pool of audio objects -- on 'ended'
event just add audio object to the pool.

Fix consists of 3 parts:
1) For low-latency code path pass entire "audio state" object to the renderer
   process. That allows renderer take into account number of pending bytes
   in the buffer.
2) When using low-latency code path renderer not only fills the buffer with
   data, but also writes length of data into first word of the buffer. That
   allows host process to pass correct byte counts to renderer.
3) Renderer now keeps track of the earliest time playback can end based on the
   number of rendered bytes, and will not call 'ended' callback till that time.

BUG=http://code.google.com/p/chromium/issues/detail?id=78992

http://codereview.chromium.org/7328030

Review URL: http://codereview.chromium.org/7328030

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@92749 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 9 insertions, 7 deletions

diff --git a/media/audio/audio_output_controller.cc b/media/audio/audio_output_controller.cc
index ec75cde..30882f4 100644
--- a/media/audio/audio_output_controller.cc
+++ b/media/audio/audio_output_controller.cc
@@ -5,6 +5,7 @@
 #include "media/audio/audio_output_controller.h"
 
 #include "base/message_loop.h"
+#include "media/audio/audio_buffers_state.h"
 
 namespace media {
 
@@ -202,7 +203,7 @@ void AudioOutputController::DoPause() {
 
   if (LowLatencyMode()) {
     // Send a special pause mark to the low-latency audio thread.
-    sync_reader_->UpdatePendingBytes(kPauseMark);
+    sync_reader_->UpdateBufferState(AudioBuffersState(kPauseMark, 0));
   }
 
   handler_->OnPaused(this);
@@ -268,13 +269,13 @@ void AudioOutputController::DoReportError(int code) {
 uint32 AudioOutputController::OnMoreData(
     AudioOutputStream* stream, uint8* dest,
     uint32 max_size, AudioBuffersState buffers_state) {
-  // If regular latency mode is used.
-  if (!sync_reader_) {
-    base::AutoLock auto_lock(lock_);
+  base::AutoLock auto_lock(lock_);
 
-    // Save current buffers state.
-    buffers_state_ = buffers_state;
+  // Save current buffers state.
+  buffers_state_ = buffers_state;
 
+  // If regular latency mode is used.
+  if (!sync_reader_) {
     if (state_ != kPlaying) {
       // Don't read anything. Save the number of bytes in the hardware buffer.
       return 0;
@@ -288,7 +289,8 @@ uint32 AudioOutputController::OnMoreData(
 
   // Low latency mode.
   uint32 size =  sync_reader_->Read(dest, max_size);
-  sync_reader_->UpdatePendingBytes(buffers_state.total_bytes() + size);
+  buffers_state_.pending_bytes += size;
+  sync_reader_->UpdateBufferState(buffers_state_);
   return size;
 }