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_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index ddd8137..1d4d5b7 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -240,8 +240,7 @@
EXPECT_TRUE(channel_send_);
EXPECT_CALL(*channel_send_, SetSendTelephoneEventPayloadType(
kTelephoneEventPayloadType,
- kTelephoneEventPayloadFrequency))
- .WillOnce(Return(true));
+ kTelephoneEventPayloadFrequency));
EXPECT_CALL(
*channel_send_,
SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
@@ -500,7 +499,7 @@
// to ConfigHelper (say to WillRepeatedly) would silently make this test
// useless.
EXPECT_CALL(*helper.channel_send(), SetEncoderForMock(_, _))
- .WillOnce(Return(true));
+ .WillOnce(Return());
helper.config().send_codec_spec =
AudioSendStream::Config::SendCodecSpec(9, kG722Format);