Delete a few return values from audio streams and video send streams.
Bug: webrtc:10198
Change-Id: I583dbb717aea26c9d282a3786062d285121fbf66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125723
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26986}diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 4594e56..1ba9b69 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -255,8 +255,8 @@
std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
EXPECT_CALL(*helper.channel_receive(),
ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
- .WillOnce(Return(true));
- EXPECT_TRUE(recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size()));
+ .WillOnce(Return());
+ recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size());
}
TEST(AudioReceiveStreamTest, GetStats) {