webrtc/api/rtpsenderreceiver_unittest.cc - Issue 1816143002: Removed MediaStreamTrackInterface::set_state

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Unified Diff: webrtc/api/rtpsenderreceiver_unittest.cc

Issue 1816143002: Removed MediaStreamTrackInterface::set_state (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@track_state_listen_on_source2

Patch Set: Rebased Created 4 years, 9 months ago

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Index: webrtc/api/rtpsenderreceiver_unittest.cc

diff --git a/webrtc/api/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc

index fd1c1f33b1556ec9c2f72c5d2ae65cc22298d29f..7f32f29bc4bf40e48a0a3114afed296b57576b47 100644

--- a/webrtc/api/rtpsenderreceiver_unittest.cc

+++ b/webrtc/api/rtpsenderreceiver_unittest.cc

@@ -137,8 +137,9 @@ class RtpSenderReceiverTest : public testing::Test {

kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));

EXPECT_TRUE(stream_->AddTrack(audio_track_));

EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));

- audio_rtp_receiver_ = new AudioRtpReceiver(stream_->GetAudioTracks()[0],

+ audio_rtp_receiver_ = new AudioRtpReceiver(stream_, kAudioTrackId,

kAudioSsrc, &audio_provider_);

+ audio_track_ = audio_rtp_receiver_->audio_track();

}

void CreateVideoRtpReceiver() {

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