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Unified Diff: webrtc/call/rampup_tests.cc

Issue 1604563002: Add send-side BWE to WebRtcVoiceEngine under a finch experiment. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Remove extension filter test which is no longer needed. Created 4 years, 11 months ago
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« talk/media/webrtc/webrtcvoiceengine.cc ('K') | « webrtc/call/bitrate_estimator_tests.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »
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Index: webrtc/call/rampup_tests.cc
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
index 81f1e81c685b45adf6515d4a56f3204b3734d5d1..847ffbd3a914a037fed3b63e596b5d531f560028 100644
--- a/webrtc/call/rampup_tests.cc
+++ b/webrtc/call/rampup_tests.cc
@@ -200,7 +200,6 @@ void RampUpTester::ModifyAudioConfigs(
}
for (AudioReceiveStream::Config& recv_config : *receive_configs) {
- recv_config.combined_audio_video_bwe = true;
recv_config.rtp.transport_cc = transport_cc;
recv_config.rtp.extensions = send_config->rtp.extensions;
recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
« talk/media/webrtc/webrtcvoiceengine.cc ('K') | « webrtc/call/bitrate_estimator_tests.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »

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