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}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions

View File

@ -34,9 +34,7 @@ class RtpDataEngine : public DataEngineInterface {
virtual DataMediaChannel* CreateChannel(const MediaConfig& config);
virtual const std::vector<DataCodec>& data_codecs() {
return data_codecs_;
}
virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
private:
std::vector<DataCodec> data_codecs_;
@ -88,10 +86,9 @@ class RtpDataMediaChannel : public DataMediaChannel {
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {}
virtual void OnReadyToSend(bool ready) {}
virtual bool SendData(
const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result);
virtual bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result);
virtual rtc::DiffServCodePoint PreferredDscp() const;
private: