Index: webrtc/audio/audio_receive_stream_unittest.cc |
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc |
index 8703d6ed324819a0a041e823a88bf90db6f9e7ca..4da503fb539f22221a0360ae99a6d8d3e25d6947 100644 |
--- a/webrtc/audio/audio_receive_stream_unittest.cc |
+++ b/webrtc/audio/audio_receive_stream_unittest.cc |
@@ -120,6 +120,7 @@ struct ConfigHelper { |
AudioReceiveStream::Config& config() { return stream_config_; } |
rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; } |
MockVoiceEngine& voice_engine() { return voice_engine_; } |
+ MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } |
void SetupMockForBweFeedback(bool send_side_bwe) { |
EXPECT_CALL(congestion_controller_, |
@@ -254,6 +255,11 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) { |
rtp_packet.size() - kExpectedHeaderLength, |
VerifyHeaderExtension(expected_extension), false)) |
.Times(1); |
+ EXPECT_CALL(*helper.channel_proxy(), |
+ ReceivedRTPPacket(&rtp_packet[0], |
+ rtp_packet.size(), |
+ _)) |
+ .WillOnce(Return(true)); |
EXPECT_TRUE( |
recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); |
} |