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Unified Diff: webrtc/build/webrtc.gni

Issue 2250483002: Add a .gyp option to use dummy audio file devices. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add gn version Created 4 years, 4 months ago
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Index: webrtc/build/webrtc.gni
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index 30e505d1d796a7852f3d9d9115d7c6018afa0b95..b345d735f1f412e580abc34a43eae39c0e13499c 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -112,6 +112,10 @@ declare_args() {
# Determines whether QUIC code will be built.
rtc_use_quic = false
+ # By default, use normal platform audio support or dummy audio, but don't
+ # use file-based audio playout and record.
+ rtc_use_dummy_audio_file_devices = false
+
# FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
# by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
# only be initialized once. Projects that initialize FFmpeg externally, such
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