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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}
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2 changed files with 3 additions and 3 deletions
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@ -232,7 +232,7 @@ TEST_F(AudioChannelTest, TestChannelStatistics) {
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EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
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EXPECT_CALL(transport_, SendRtcp).WillRepeatedly(Invoke(loop_rtcp));
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// Simulate microphone giving audio frame (10 ms). This will trigger tranport
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// Simulate microphone giving audio frame (10 ms). This will trigger transport
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// to send RTP as handled in loop_rtp above.
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auto audio_sender = audio_channel_->GetAudioSender();
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audio_sender->SendAudioData(GetAudioFrame(0));
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@ -245,7 +245,7 @@ TEST_F(AudioChannelTest, TestChannelStatistics) {
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audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
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// Force sending RTCP SR report in order to have remote_rtcp field available
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// in channel statistics. This will trigger tranport to send RTCP as handled
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// in channel statistics. This will trigger transport to send RTCP as handled
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// in loop_rtcp above.
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audio_channel_->SendRTCPReportForTesting(kRtcpSr);
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@ -218,7 +218,7 @@ TEST_F(AudioEgressTest, SkipAudioEncodingAfterStopSend) {
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// It should be safe to exit the test case while encoder_queue_ has
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// outstanding data to process. We are making sure that this doesn't
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// result in crahses or sanitizer errors due to remaining data.
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// result in crashes or sanitizer errors due to remaining data.
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for (size_t i = 0; i < kExpected * 2; i++) {
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egress_->SendAudioData(GetAudioFrame(i));
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time_controller_.AdvanceTime(TimeDelta::Millis(10));
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