commit e7d08df83764208982a61f6128be544e7b590442
author Artem Titov <titovartem@webrtc.org> Wed Jan 16 13:49:44 2019
committer Commit Bot <commit-bot@chromium.org> Wed Jan 16 15:19:57 2019
tree 6149bc46c83c46154028542cd27f5d90f0d9fd80
parent 90d0a627c48f629732d9058296a0a557aa5d3d9d

Fix chromium roll into WebRTC.

Original error detected here:
https://webrtc-review.googlesource.com/c/src/+/117840/

BUG=None

Change-Id: I30c7dc6b1ddbf32a7081e07a261554cbe72db9ba
Reviewed-on: https://webrtc-review.googlesource.com/c/117880
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26281}

audio/audio_send_stream_unittest.cc

diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 1a173fc..1bf8bca 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -227,12 +227,12 @@
   void SetupMockForModifyEncoder() {
     // Let ModifyEncoder to invoke mock audio encoder.
     EXPECT_CALL(*channel_send_, ModifyEncoder(_))
-        .WillRepeatedly(Invoke(
+        .WillRepeatedly(
             [this](rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
                        modifier) {
               if (this->audio_encoder_)
                 modifier(&this->audio_encoder_);
-            }));
+            });
   }
 
   void SetupMockForSendTelephoneEvent() {