| Index: webrtc/video/bitrate_estimator_tests.cc
|
| diff --git a/webrtc/video/bitrate_estimator_tests.cc b/webrtc/video/bitrate_estimator_tests.cc
|
| index 6e71ca24efe7d8b9269b96235af8df03ce6fcfb9..d7b0b374d35c12eb0733485d79c2a08b3cc00eb2 100644
|
| --- a/webrtc/video/bitrate_estimator_tests.cc
|
| +++ b/webrtc/video/bitrate_estimator_tests.cc
|
| @@ -204,6 +204,9 @@ class BitrateEstimatorTest : public test::CallTest {
|
| if (receive_audio) {
|
| AudioReceiveStream::Config receive_config;
|
| receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
|
| + // Bogus non-default id, every receive stream should correspond to an
|
| + // underlying channel.
|
| + receive_config.voe_channel_id = 0;
|
| receive_config.rtp.extensions.push_back(
|
| RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
|
| audio_receive_stream_ = test_->receiver_call_->CreateAudioReceiveStream(
|
|
|