commit | e7d08df83764208982a61f6128be544e7b590442 | [log] [tgz] |
---|---|---|
author | Artem Titov <titovartem@webrtc.org> | Wed Jan 16 14:49:44 2019 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 16 15:19:57 2019 +0000 |
tree | 6149bc46c83c46154028542cd27f5d90f0d9fd80 | |
parent | 90d0a627c48f629732d9058296a0a557aa5d3d9d [diff] [blame] |
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}
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() {