commit 665174fdbb4e0540eccb27cf7412348f1b65534c
author Yves Gerey <yvesg@webrtc.org> Tue Jun 19 15:03:05 2018 +0200
committer Yves Gerey <yvesg@webrtc.org> Tue Jun 19 14:00:39 2018 +0000
tree 34a9eb770e6de54c282f62ad0b65d2cd69c917f6
parent b602123a5aae1a9f0b6b581e611acf44eb525c08

Reformat the WebRTC code base

Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}

audio/audio_state.cc

diff --git a/audio/audio_state.cc b/audio/audio_state.cc
index d738884..bff818e 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -27,8 +27,7 @@
 
 AudioState::AudioState(const AudioState::Config& config)
     : config_(config),
-      audio_transport_(config_.audio_mixer,
-                       config_.audio_processing.get()) {
+      audio_transport_(config_.audio_mixer, config_.audio_processing.get()) {
   process_thread_checker_.DetachFromThread();
   RTC_DCHECK(config_.audio_mixer);
   RTC_DCHECK(config_.audio_device_module);
@@ -50,7 +49,7 @@
   RTC_DCHECK_EQ(0, receiving_streams_.count(stream));
   receiving_streams_.insert(stream);
   if (!config_.audio_mixer->AddSource(
-      static_cast<internal::AudioReceiveStream*>(stream))) {
+          static_cast<internal::AudioReceiveStream*>(stream))) {
     RTC_DLOG(LS_ERROR) << "Failed to add source to mixer.";
   }
 
@@ -79,7 +78,8 @@
 }
 
 void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
-                                  int sample_rate_hz, size_t num_channels) {
+                                  int sample_rate_hz,
+                                  size_t num_channels) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   auto& properties = sending_streams_[stream];
   properties.sample_rate_hz = sample_rate_hz;
@@ -121,8 +121,7 @@
       }
     } else {
       config_.audio_device_module->StopPlayout();
-      null_audio_poller_ =
-          rtc::MakeUnique<NullAudioPoller>(&audio_transport_);
+      null_audio_poller_ = rtc::MakeUnique<NullAudioPoller>(&audio_transport_);
     }
   }
 }