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, 1 insertions, 1 deletions

diff --git a/media/audio/audio_io.h b/media/audio/audio_io.h
index 74b5c47..2550873 100644
--- a/media/audio/audio_io.h
+++ b/media/audio/audio_io.h
@@ -112,7 +112,7 @@ class AudioInputStream {
     // available. This is called from a special audio thread and the
     // implementation should return as soon as possible.
     virtual void OnData(AudioInputStream* stream, const uint8* src,
-                        uint32 size) = 0;
+                        uint32 size, uint32 hardware_delay_bytes) = 0;
 
     // The stream is done with this callback, the last call received by this
     // audio sink.