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Unified Diff: webrtc/audio/test/low_bandwidth_audio_test.cc

Issue 2774463003: Don't hardcode MediaType::ANY in FakeNetworkPipe. (Closed)
Patch Set: Update FakeCall::DeliverPacket, for consistency with Call::DeliverRtp. Created 3 years, 9 months ago
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Index: webrtc/audio/test/low_bandwidth_audio_test.cc
diff --git a/webrtc/audio/test/low_bandwidth_audio_test.cc b/webrtc/audio/test/low_bandwidth_audio_test.cc
index c1b1a02235610b51cfa708aa934eb6183ccc3275..50d7e674e83d11833f43808e6572cc86c2df842b 100644
--- a/webrtc/audio/test/low_bandwidth_audio_test.cc
+++ b/webrtc/audio/test/low_bandwidth_audio_test.cc
@@ -80,12 +80,14 @@ test::PacketTransport* AudioQualityTest::CreateSendTransport(
Call* sender_call) {
return new test::PacketTransport(
sender_call, this, test::PacketTransport::kSender,
+ MediaType::AUDIO,
GetNetworkPipeConfig());
}
test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
return new test::PacketTransport(nullptr, this,
- test::PacketTransport::kReceiver, GetNetworkPipeConfig());
+ test::PacketTransport::kReceiver, MediaType::AUDIO,
+ GetNetworkPipeConfig());
}
void AudioQualityTest::ModifyAudioConfigs(
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