5b3b6b1784
When WebRTC is built as a part of Chromium, some of the stuff in webrtc.gyp will not be found. This CL fixes this. TEST=trybots passing. I also did some manual builds for Android with the android_builder_webrtc target in build/all_android.gyp of a Chromium checkout. BUG=chromium:304143 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2353004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4949 4adac7df-926f-26a2-2b94-8c16560cd09d
7 lines
355 B
Plaintext
Executable File
7 lines
355 B
Plaintext
Executable File
# Place in ~/.pulse/ to add null sinks for the audio end-to-end quality test.
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.include /etc/pulse/default.pa
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load-module module-null-sink sink_name=render sink_properties=device.description=render format=s16 rate=48000 channels=1
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load-module module-null-sink sink_name=capture sink_properties=device.description=capture format=s16 rate=48000 channels=1
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