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

@ -31,8 +31,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
return SetEncoderForMock(payload_type, &encoder);
}
MOCK_METHOD2(SetEncoderForMock,
bool(int payload_type,
std::unique_ptr<AudioEncoder>* encoder));
bool(int payload_type, std::unique_ptr<AudioEncoder>* encoder));
MOCK_METHOD1(
ModifyEncoder,
void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
@ -59,8 +58,8 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());
MOCK_CONST_METHOD0(GetTotalOutputDuration, double());
MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
MOCK_METHOD2(SetSendTelephoneEventPayloadType, bool(int payload_type,
int payload_frequency));
MOCK_METHOD2(SetSendTelephoneEventPayloadType,
bool(int payload_type, int payload_frequency));
MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink));
@ -71,8 +70,8 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
MOCK_METHOD2(GetAudioFrameWithInfo,
AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
AudioFrame* audio_frame));
AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
AudioFrame* audio_frame));
MOCK_CONST_METHOD0(PreferredSampleRate, int());
// GMock doesn't like move-only types, like std::unique_ptr.
virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) {
@ -84,8 +83,8 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_METHOD1(AssociateSendChannel,
void(const ChannelProxy& send_channel_proxy));
MOCK_METHOD0(DisassociateSendChannel, void());
MOCK_CONST_METHOD2(GetRtpRtcp, void(RtpRtcp** rtp_rtcp,
RtpReceiver** rtp_receiver));
MOCK_CONST_METHOD2(GetRtpRtcp,
void(RtpRtcp** rtp_rtcp, RtpReceiver** rtp_receiver));
MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));