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Reformat the WebRTC code base

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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
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55
media/engine/webrtcvideoengine.cc
View File

@ -187,7 +187,6 @@ void AddDefaultFeedbackParams(VideoCodec* codec) {
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
}
// This function will assign dynamic payload types (in the range [96, 127]) to
// the input codecs, and also add ULPFEC, RED, FlexFEC, and associated RTX
// codecs for recognized codecs (VP8, VP9, H264, and RED). It will also add
@ -590,8 +589,8 @@ RtpCapabilities WebRtcVideoEngine::GetCapabilities() const {
webrtc::RtpExtension(webrtc::RtpExtension::kVideoContentTypeUri,
webrtc::RtpExtension::kVideoContentTypeDefaultId));
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri,
webrtc::RtpExtension::kVideoTimingDefaultId));
webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri,
webrtc::RtpExtension::kVideoTimingDefaultId));
// TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
// demuxing is completed.
// capabilities.header_extensions.push_back(webrtc::RtpExtension(
@ -663,10 +662,10 @@ bool WebRtcVideoChannel::NonFlexfecReceiveCodecsHaveChanged(
// to support munging the SDP in this way without recreating receive
// streams, we ignore the order of the received codecs so that
// changing the order doesn't cause this "blink".
auto comparison =
[](const VideoCodecSettings& codec1, const VideoCodecSettings& codec2) {
return codec1.codec.id > codec2.codec.id;
};
auto comparison = [](const VideoCodecSettings& codec1,
const VideoCodecSettings& codec2) {
return codec1.codec.id > codec2.codec.id;
};
std::sort(before.begin(), before.end(), comparison);
std::sort(after.begin(), after.end(), comparison);
@ -1064,8 +1063,7 @@ bool WebRtcVideoChannel::SetVideoSend(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
TRACE_EVENT0("webrtc", "SetVideoSend");
RTC_DCHECK(ssrc != 0);
RTC_LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc
<< ", options: "
RTC_LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", options: "
<< (options ? options->ToString() : "nullptr")
<< ", source = " << (source ? "(source)" : "nullptr") << ")";
@ -1128,9 +1126,8 @@ bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) {
WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
call_, sp, std::move(config), default_send_options_,
video_config_.enable_cpu_adaptation,
bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
send_params_);
video_config_.enable_cpu_adaptation, bitrate_config_.max_bitrate_bps,
send_codec_, send_rtp_extensions_, send_params_);
uint32_t ssrc = sp.first_ssrc();
RTC_DCHECK(ssrc != 0);
@ -1385,7 +1382,7 @@ bool WebRtcVideoChannel::GetStats(VideoMediaInfo* info) {
}
void WebRtcVideoChannel::FillSenderStats(VideoMediaInfo* video_media_info,
bool log_stats) {
bool log_stats) {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
send_streams_.begin();
@ -1396,7 +1393,7 @@ void WebRtcVideoChannel::FillSenderStats(VideoMediaInfo* video_media_info,
}
void WebRtcVideoChannel::FillReceiverStats(VideoMediaInfo* video_media_info,
bool log_stats) {
bool log_stats) {
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
receive_streams_.begin();
@ -1429,9 +1426,8 @@ void WebRtcVideoChannel::FillSendAndReceiveCodecStats(
}
}
void WebRtcVideoChannel::OnPacketReceived(
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
const webrtc::PacketReceiver::DeliveryStatus delivery_result =
@ -1487,9 +1483,8 @@ void WebRtcVideoChannel::OnPacketReceived(
}
}
void WebRtcVideoChannel::OnRtcpReceived(
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
void WebRtcVideoChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
// TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
@ -1519,16 +1514,14 @@ void WebRtcVideoChannel::OnNetworkRouteChanged(
void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
MediaChannel::SetInterface(iface);
// Set the RTP recv/send buffer to a bigger size
MediaChannel::SetOption(NetworkInterface::ST_RTP,
rtc::Socket::OPT_RCVBUF,
MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_RCVBUF,
kVideoRtpBufferSize);
// Speculative change to increase the outbound socket buffer size.
// In b/15152257, we are seeing a significant number of packets discarded
// due to lack of socket buffer space, although it's not yet clear what the
// ideal value should be.
MediaChannel::SetOption(NetworkInterface::ST_RTP,
rtc::Socket::OPT_SNDBUF,
MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_SNDBUF,
kVideoRtpBufferSize);
}
@ -1981,8 +1974,8 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::ReconfigureEncoder() {
webrtc::VideoEncoderConfig encoder_config =
CreateVideoEncoderConfig(codec_settings.codec);
encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
codec_settings.codec);
encoder_config.encoder_specific_settings =
ConfigureVideoEncoderSettings(codec_settings.codec);
stream_->ReconfigureVideoEncoder(encoder_config.Copy());
@ -2383,8 +2376,7 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetRecvParameters(
}
}
void WebRtcVideoChannel::WebRtcVideoReceiveStream::
RecreateWebRtcVideoStream() {
void WebRtcVideoChannel::WebRtcVideoReceiveStream::RecreateWebRtcVideoStream() {
if (stream_) {
MaybeDissociateFlexfecFromVideo();
call_->DestroyVideoReceiveStream(stream_);
@ -2504,8 +2496,8 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
info.jitter_buffer_ms = stats.jitter_buffer_ms;
info.min_playout_delay_ms = stats.min_playout_delay_ms;
info.render_delay_ms = stats.render_delay_ms;
info.frames_received = stats.frame_counts.key_frames +
stats.frame_counts.delta_frames;
info.frames_received =
stats.frame_counts.key_frames + stats.frame_counts.delta_frames;
info.frames_decoded = stats.frames_decoded;
info.frames_rendered = stats.frames_rendered;
info.qp_sum = stats.qp_sum;
@ -2624,8 +2616,7 @@ WebRtcVideoChannel::MapCodecs(const std::vector<VideoCodec>& codecs) {
RTC_DCHECK(!video_codecs.empty());
for (std::map<int, int>::const_iterator it = rtx_mapping.begin();
it != rtx_mapping.end();
++it) {
it != rtx_mapping.end(); ++it) {
if (!payload_used[it->first]) {
RTC_LOG(LS_ERROR) << "RTX mapped to payload not in codec list.";
return std::vector<VideoCodecSettings>();