Adding delay estimation to input audio.

On linux, snd_pcm_delay API will be used to get the delay value from ALSA;
On mac, the AudioQueue implementation will just use the buffer size as delay value since we may not use this impl for recording; 
the low-latency AudioUnits implementation will use some AudioUnitGetProperty() API to get the accurate delay value from the driver;
On windows, for the Wave implementation, based on the tests we do, it should be OK to use the buffer size as the delay value; 
for the to-be-implemented core audio implementation, we will see what is the appropriate way to get the delay there.



Review URL: http://codereview.chromium.org/8068005

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@104002 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 2 insertions, 1 deletions

diff --git a/media/audio/mac/audio_input_mac.cc b/media/audio/mac/audio_input_mac.cc
index 3efe86c..e96277f 100644
--- a/media/audio/mac/audio_input_mac.cc
+++ b/media/audio/mac/audio_input_mac.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
@@ -159,6 +159,7 @@ void PCMQueueInAudioInputStream::HandleInputBuffer(
   if (audio_buffer->mAudioDataByteSize)
     callback_->OnData(this,
                       reinterpret_cast<const uint8*>(audio_buffer->mAudioData),
+                      audio_buffer->mAudioDataByteSize,
                       audio_buffer->mAudioDataByteSize);
   // Recycle the buffer.
   OSStatus err = QueueNextBuffer(audio_buffer);