Prevent sending 0 as volume to AGC at the beginning of the audio stream.

We are using a timer and query the volume every 1 second, but since we don't have the correct volume at the beginning of the audio stream, we send 0 up to AGC, which will starting raising the volume from 0. This makes WebRtc audio barely usable since the volume is always set to around 0 at the beginning of the call.

BUG=266048
TEST=Open http://webrtc.googlecode.com/svn/trunk/samples/js/demos/html/pc1.html, make a call, verity that the volume is not set to 0 at the beginning.

Review URL: https://chromiumcodereview.appspot.com/21114004

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@215012 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 4 insertions, 2 deletions

diff --git a/media/audio/pulse/pulse_input.cc b/media/audio/pulse/pulse_input.cc
index cc62085..54dfc1e 100644
--- a/media/audio/pulse/pulse_input.cc
+++ b/media/audio/pulse/pulse_input.cc
@@ -61,13 +61,15 @@ void PulseAudioInputStream::Start(AudioInputCallback* callback) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(callback);
   DCHECK(handle_);
+
+  // AGC needs to be started out of the lock.
+  StartAgc();
+
   AutoPulseLock auto_lock(pa_mainloop_);
 
   if (stream_started_)
     return;
 
-  StartAgc();
-
   // Clean up the old buffer.
   pa_stream_drop(handle_);
   buffer_->Clear();