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}diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 07a23e3..51635f6 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -59,18 +59,16 @@
const double kTotalOutputEnergy = 0.25;
const double kTotalOutputDuration = 0.5;
-const CallStatistics kCallStats = {
- 345, 678, 901, 234, -12, 3456, 7890, 567, 890, 123};
-const CodecInst kCodecInst = {
- 123, "codec_name_recv", 96000, -187, 0, -103};
+const CallStatistics kCallStats = {345, 678, 901, 234, -12,
+ 3456, 7890, 567, 890, 123};
+const CodecInst kCodecInst = {123, "codec_name_recv", 96000, -187, 0, -103};
const NetworkStatistics kNetworkStats = {
123, 456, false, 789012, 3456, 123, 456, 0, {}, 789, 12,
345, 678, 901, 0, -1, -1, -1, -1, -1, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
- ConfigHelper()
- : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {}
+ ConfigHelper() : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {}
explicit ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer)
: audio_mixer_(audio_mixer) {
@@ -88,23 +86,21 @@
EXPECT_CALL(*channel_proxy_, SetRemoteSSRC(kRemoteSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
EXPECT_CALL(*channel_proxy_,
- RegisterReceiverCongestionControlObjects(&packet_router_))
- .Times(1);
+ RegisterReceiverCongestionControlObjects(&packet_router_))
+ .Times(1);
EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects())
.Times(1);
EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2);
testing::Expectation expect_set =
- EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
- .Times(1);
+ EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
.Times(1)
.After(expect_set);
EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
- .WillRepeatedly(
- Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
- EXPECT_THAT(codecs, testing::IsEmpty());
- }));
+ .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
+ EXPECT_THAT(codecs, testing::IsEmpty());
+ }));
stream_config_.rtp.local_ssrc = kLocalSsrc;
stream_config_.rtp.remote_ssrc = kRemoteSsrc;
@@ -120,11 +116,8 @@
std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() {
return std::unique_ptr<internal::AudioReceiveStream>(
new internal::AudioReceiveStream(
- &rtp_stream_receiver_controller_,
- &packet_router_,
- stream_config_,
- audio_state_,
- &event_log_,
+ &rtp_stream_receiver_controller_, &packet_router_, stream_config_,
+ audio_state_, &event_log_,
std::unique_ptr<voe::ChannelProxy>(channel_proxy_)));
}
@@ -323,7 +316,7 @@
ConfigHelper helper;
auto recv_stream = helper.CreateAudioReceiveStream();
EXPECT_CALL(*helper.channel_proxy(),
- SetChannelOutputVolumeScaling(FloatEq(0.765f)));
+ SetChannelOutputVolumeScaling(FloatEq(0.765f)));
recv_stream->SetGain(0.765f);
}
@@ -371,10 +364,10 @@
new_config.rtp.nack.rtp_history_ms = 300 + 20;
new_config.rtp.extensions.clear();
new_config.rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
- new_config.rtp.extensions.push_back(RtpExtension(
- RtpExtension::kTransportSequenceNumberUri,
- kTransportSequenceNumberId + 1));
+ RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
+ new_config.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+ kTransportSequenceNumberId + 1));
new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
MockVoEChannelProxy& channel_proxy = *helper.channel_proxy();