commit 17b050f8f833580c99178b374e44d8688925f59f
author Benjamin Wright <benwright@webrtc.org> Wed Mar 13 17:35:46 2019 -0700
committer Commit Bot <commit-bot@chromium.org> Fri Mar 15 01:55:52 2019 +0000
tree c81c2463060c3e066b2cb2ce7af0fde5f96b82e1
parent 8965fbc542246c5a234e525c65aea8247395798b

Fixes ClangTidy errors in audio/

These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.

Bug: webrtc:10410
Change-Id: I1b46653b91bce012afabfa0f2d249718e6de2df8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127626
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27139}

audio/channel_send.cc

diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 9f95ebb..5ca6f6a 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -76,11 +76,10 @@
 class TransportSequenceNumberProxy;
 class VoERtcpObserver;
 
-class ChannelSend
-    : public ChannelSendInterface,
-      public AudioPacketizationCallback,  // receive encoded packets from the
-                                          // ACM
-      public TargetTransferRateObserver {
+class ChannelSend : public ChannelSendInterface,
+                    public AudioPacketizationCallback,  // receive encoded
+                                                        // packets from the ACM
+                    public TargetTransferRateObserver {
  public:
   // TODO(nisse): Make OnUplinkPacketLossRate public, and delete friend
   // declaration.
@@ -603,7 +602,7 @@
       // Sample count isn't conveniently available from the AudioCodingModule,
       // and needs some refactoring to wire up in a good way. For now, left as
       // zero.
-      /*sample_count=*/0,
+      /*samples_per_channel=*/0,
 
       /*sequence_number=*/media_transport_sequence_number_,
       MediaTransportFrameTypeForWebrtcFrameType(frameType), payloadType,