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/media/base/mediachannel.h b/media/base/mediachannel.h
index a9b586a..ea8b186 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -50,7 +50,7 @@
namespace webrtc {
class AudioSinkInterface;
class VideoFrame;
-}
+} // namespace webrtc
namespace cricket {
@@ -76,16 +76,16 @@
template <class T>
static std::string VectorToString(const std::vector<T>& vals) {
- std::ostringstream ost;
- ost << "[";
- for (size_t i = 0; i < vals.size(); ++i) {
- if (i > 0) {
- ost << ", ";
- }
- ost << vals[i].ToString();
+ std::ostringstream ost; // no-presubmit-check TODO(webrtc:8982)
+ ost << "[";
+ for (size_t i = 0; i < vals.size(); ++i) {
+ if (i > 0) {
+ ost << ", ";
}
- ost << "]";
- return ost.str();
+ ost << vals[i].ToString();
+ }
+ ost << "]";
+ return ost.str();
}
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
@@ -170,7 +170,8 @@
const rtc::PacketOptions& options) = 0;
virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) = 0;
- virtual int SetOption(SocketType type, rtc::Socket::Option opt,
+ virtual int SetOption(SocketType type,
+ rtc::Socket::Option opt,
int option) = 0;
virtual ~NetworkInterface() {}
};
@@ -240,13 +241,9 @@
// This method sets DSCP |value| on both RTP and RTCP channels.
int SetDscp(rtc::DiffServCodePoint value) {
int ret;
- ret = SetOption(NetworkInterface::ST_RTP,
- rtc::Socket::OPT_DSCP,
- value);
+ ret = SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_DSCP, value);
if (ret == 0) {
- ret = SetOption(NetworkInterface::ST_RTCP,
- rtc::Socket::OPT_DSCP,
- value);
+ ret = SetOption(NetworkInterface::ST_RTCP, rtc::Socket::OPT_DSCP, value);
}
return ret;
}
@@ -290,9 +287,7 @@
struct MediaSenderInfo {
MediaSenderInfo();
~MediaSenderInfo();
- void add_ssrc(const SsrcSenderInfo& stat) {
- local_stats.push_back(stat);
- }
+ void add_ssrc(const SsrcSenderInfo& stat) { local_stats.push_back(stat); }
// Temporary utility function for call sites that only provide SSRC.
// As more info is added into SsrcSenderInfo, this function should go away.
void add_ssrc(uint32_t ssrc) {
@@ -337,9 +332,7 @@
struct MediaReceiverInfo {
MediaReceiverInfo();
~MediaReceiverInfo();
- void add_ssrc(const SsrcReceiverInfo& stat) {
- local_stats.push_back(stat);
- }
+ void add_ssrc(const SsrcReceiverInfo& stat) { local_stats.push_back(stat); }
// Temporary utility function for call sites that only provide SSRC.
// As more info is added into SsrcSenderInfo, this function should go away.
void add_ssrc(uint32_t ssrc) {
@@ -655,8 +648,7 @@
std::map<std::string, std::string> ToStringMap() const override;
};
-struct AudioRecvParameters : RtpParameters<AudioCodec> {
-};
+struct AudioRecvParameters : RtpParameters<AudioCodec> {};
class VoiceMediaChannel : public MediaChannel {
public:
@@ -727,8 +719,7 @@
// TODO(deadbeef): Rename to VideoReceiverParameters, since they're intended to
// encapsulate all the parameters needed for a video RtpReceiver.
-struct VideoRecvParameters : RtpParameters<VideoCodec> {
-};
+struct VideoRecvParameters : RtpParameters<VideoCodec> {};
class VideoMediaChannel : public MediaChannel {
public:
@@ -836,11 +827,9 @@
enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
-struct DataSendParameters : RtpSendParameters<DataCodec> {
-};
+struct DataSendParameters : RtpSendParameters<DataCodec> {};
-struct DataRecvParameters : RtpParameters<DataCodec> {
-};
+struct DataRecvParameters : RtpParameters<DataCodec> {};
class DataMediaChannel : public MediaChannel {
public:
@@ -860,14 +849,12 @@
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override {}
- virtual bool SendData(
- const SendDataParams& params,
- const rtc::CopyOnWriteBuffer& payload,
- SendDataResult* result = NULL) = 0;
+ virtual bool SendData(const SendDataParams& params,
+ const rtc::CopyOnWriteBuffer& payload,
+ SendDataResult* result = NULL) = 0;
// Signals when data is received (params, data, len)
- sigslot::signal3<const ReceiveDataParams&,
- const char*,
- size_t> SignalDataReceived;
+ sigslot::signal3<const ReceiveDataParams&, const char*, size_t>
+ SignalDataReceived;
// Signal when the media channel is ready to send the stream. Arguments are:
// writable(bool)
sigslot::signal1<bool> SignalReadyToSend;