audio: fix some typos

Bug: None
Change-Id: I255a23a893d008dc58c3c9cb3facf61419c88c72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320620
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40779}
This commit is contained in:
Alfred E. Heggestad 2023-09-18 13:58:24 +02:00 committed by WebRTC LUCI CQ
parent 6fc4d9750c
commit c951d1b0f6
2 changed files with 3 additions and 3 deletions

View file

@ -232,7 +232,7 @@ TEST_F(AudioChannelTest, TestChannelStatistics) {
EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
EXPECT_CALL(transport_, SendRtcp).WillRepeatedly(Invoke(loop_rtcp));
// Simulate microphone giving audio frame (10 ms). This will trigger tranport
// Simulate microphone giving audio frame (10 ms). This will trigger transport
// to send RTP as handled in loop_rtp above.
auto audio_sender = audio_channel_->GetAudioSender();
audio_sender->SendAudioData(GetAudioFrame(0));
@ -245,7 +245,7 @@ TEST_F(AudioChannelTest, TestChannelStatistics) {
audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
// Force sending RTCP SR report in order to have remote_rtcp field available
// in channel statistics. This will trigger tranport to send RTCP as handled
// in channel statistics. This will trigger transport to send RTCP as handled
// in loop_rtcp above.
audio_channel_->SendRTCPReportForTesting(kRtcpSr);

View file

@ -218,7 +218,7 @@ TEST_F(AudioEgressTest, SkipAudioEncodingAfterStopSend) {
// It should be safe to exit the test case while encoder_queue_ has
// outstanding data to process. We are making sure that this doesn't
// result in crahses or sanitizer errors due to remaining data.
// result in crashes or sanitizer errors due to remaining data.
for (size_t i = 0; i < kExpected * 2; i++) {
egress_->SendAudioData(GetAudioFrame(i));
time_controller_.AdvanceTime(TimeDelta::Millis(10));