commit c951d1b0f6f450ab668ecea6635f2e5249beff2a
author Alfred E. Heggestad <aheggestad@nvidia.com> Mon Sep 18 11:58:24 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Thu Sep 21 05:42:29 2023
tree 3d94f63c94a73f9d779a5d11bcf343087f12a5c2
parent 6fc4d9750c5210b5a7dce0acf97bffb4136a4d62

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}

audio/voip/test/audio_channel_unittest.cc

diff --git a/audio/voip/test/audio_channel_unittest.cc b/audio/voip/test/audio_channel_unittest.cc
index 7097e7f..0c8312b 100644
--- a/audio/voip/test/audio_channel_unittest.cc
+++ b/audio/voip/test/audio_channel_unittest.cc
@@ -232,7 +232,7 @@
   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 @@
   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);
 

audio/voip/test/audio_egress_unittest.cc

diff --git a/audio/voip/test/audio_egress_unittest.cc b/audio/voip/test/audio_egress_unittest.cc
index 8501b2d..83df26e 100644
--- a/audio/voip/test/audio_egress_unittest.cc
+++ b/audio/voip/test/audio_egress_unittest.cc
@@ -218,7 +218,7 @@
 
   // 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));