tree dbd3b14b73e00c47d991848de39c799827423990
parent 8cfa49594feb82a0c1f91d1a661b22cc43ca44ce
author andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> 1367607706 +0000
committer andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> 1367607706 +0000

Remove 44.1 kHz workaround from AudioDevice on PulseAudio.

We currently inform VoE that 44.1 kHz audio is 44 kHz. We now have arbitrary
resampling in VoE, allowing us to pass in the native 44.1 kHz.

Our ALSA interface always requires 48 kHz, allowing ALSA to handle resampling.

This also removes WEBRTC_PA_GTALK which was not defined anywhere.

BUG=webrtc:1395
TESTED=Using 44.1 for capture and render in loopback, ran through all codec channel/rate combinations. Quality is good. Testing AEC was difficult as I can't find a way to change the sample rate of an individual device in PulseAudio. Using a webcam at 32 kHz, other problems were the overriding contribution to quality degradation (delay issues, possible clock drift from the camera). At least I verified that the quality got no worse with this patch.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3955 4adac7df-926f-26a2-2b94-8c16560cd09d