Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ half of the remaining folders git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -E "^(api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I8b2cac13f4587d3ce9b2fccc7362967283f57ea2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302062 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39977}
diff --git a/api/audio_codecs/audio_decoder.cc b/api/audio_codecs/audio_decoder.cc index 28f5b8a..0a131f1 100644 --- a/api/audio_codecs/audio_decoder.cc +++ b/api/audio_codecs/audio_decoder.cc
@@ -10,7 +10,6 @@ #include "api/audio_codecs/audio_decoder.h" - #include <memory> #include <utility>
diff --git a/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc b/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc index 0052c42..e159bd7 100644 --- a/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc +++ b/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc
@@ -32,8 +32,9 @@ const AudioEncoderMultiChannelOpusConfig&) = default; AudioEncoderMultiChannelOpusConfig::~AudioEncoderMultiChannelOpusConfig() = default; -AudioEncoderMultiChannelOpusConfig& AudioEncoderMultiChannelOpusConfig:: -operator=(const AudioEncoderMultiChannelOpusConfig&) = default; +AudioEncoderMultiChannelOpusConfig& +AudioEncoderMultiChannelOpusConfig::operator=( + const AudioEncoderMultiChannelOpusConfig&) = default; bool AudioEncoderMultiChannelOpusConfig::IsOk() const { if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
diff --git a/api/media_stream_interface.cc b/api/media_stream_interface.cc index e079079..6b0a6a9 100644 --- a/api/media_stream_interface.cc +++ b/api/media_stream_interface.cc
@@ -9,6 +9,7 @@ */ #include "api/media_stream_interface.h" + #include "api/media_types.h" namespace webrtc {
diff --git a/api/neteq/default_neteq_controller_factory.cc b/api/neteq/default_neteq_controller_factory.cc index 22274dc..4e0a0df 100644 --- a/api/neteq/default_neteq_controller_factory.cc +++ b/api/neteq/default_neteq_controller_factory.cc
@@ -9,6 +9,7 @@ */ #include "api/neteq/default_neteq_controller_factory.h" + #include "modules/audio_coding/neteq/decision_logic.h" namespace webrtc {
diff --git a/api/neteq/neteq_controller.h b/api/neteq/neteq_controller.h index a64a233..6f42e83 100644 --- a/api/neteq/neteq_controller.h +++ b/api/neteq/neteq_controller.h
@@ -13,7 +13,6 @@ #include <cstddef> #include <cstdint> - #include <functional> #include <memory>
diff --git a/api/stats/rtc_stats.h b/api/stats/rtc_stats.h index e38373a..5cb97cc 100644 --- a/api/stats/rtc_stats.h +++ b/api/stats/rtc_stats.h
@@ -163,7 +163,9 @@ return std::make_unique<this_class>(*this); \ } \ \ - const char* this_class::type() const { return this_class::kType; } \ + const char* this_class::type() const { \ + return this_class::kType; \ + } \ \ std::vector<const webrtc::RTCStatsMemberInterface*> \ this_class::MembersOfThisObjectAndAncestors( \ @@ -194,7 +196,9 @@ return std::make_unique<this_class>(*this); \ } \ \ - const char* this_class::type() const { return this_class::kType; } \ + const char* this_class::type() const { \ + return this_class::kType; \ + } \ \ std::vector<const webrtc::RTCStatsMemberInterface*> \ this_class::MembersOfThisObjectAndAncestors( \
diff --git a/api/units/timestamp_unittest.cc b/api/units/timestamp_unittest.cc index 43b2985..f49b8dd 100644 --- a/api/units/timestamp_unittest.cc +++ b/api/units/timestamp_unittest.cc
@@ -8,9 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "api/units/timestamp.h" + #include <limits> -#include "api/units/timestamp.h" #include "test/gtest.h" namespace webrtc {
diff --git a/api/video_codecs/video_encoder.cc b/api/video_codecs/video_encoder.cc index e4b44ae..b0fe078 100644 --- a/api/video_codecs/video_encoder.cc +++ b/api/video_codecs/video_encoder.cc
@@ -11,6 +11,7 @@ #include "api/video_codecs/video_encoder.h" #include <string.h> + #include <algorithm> #include "rtc_base/checks.h"
diff --git a/api/voip/test/voip_engine_factory_unittest.cc b/api/voip/test/voip_engine_factory_unittest.cc index f967a0b..7d717c1 100644 --- a/api/voip/test/voip_engine_factory_unittest.cc +++ b/api/voip/test/voip_engine_factory_unittest.cc
@@ -8,10 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "api/voip/voip_engine_factory.h" + #include <utility> #include "api/task_queue/default_task_queue_factory.h" -#include "api/voip/voip_engine_factory.h" #include "modules/audio_device/include/mock_audio_device.h" #include "modules/audio_processing/include/mock_audio_processing.h" #include "test/gtest.h"
diff --git a/call/adaptation/test/mock_resource_listener.h b/call/adaptation/test/mock_resource_listener.h index f0f998f..1c4df31 100644 --- a/call/adaptation/test/mock_resource_listener.h +++ b/call/adaptation/test/mock_resource_listener.h
@@ -12,7 +12,6 @@ #define CALL_ADAPTATION_TEST_MOCK_RESOURCE_LISTENER_H_ #include "api/adaptation/resource.h" - #include "test/gmock.h" namespace webrtc {
diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc index cd96138..c575a3f 100644 --- a/call/flexfec_receive_stream_unittest.cc +++ b/call/flexfec_receive_stream_unittest.cc
@@ -93,9 +93,7 @@ receive_stream_->RegisterWithTransport(&rtp_stream_receiver_controller_); } - ~FlexfecReceiveStreamTest() { - receive_stream_->UnregisterFromTransport(); - } + ~FlexfecReceiveStreamTest() { receive_stream_->UnregisterFromTransport(); } rtc::AutoThread main_thread_; MockTransport rtcp_send_transport_;
diff --git a/call/rtp_demuxer_unittest.cc b/call/rtp_demuxer_unittest.cc index 2b394d3..e850528 100644 --- a/call/rtp_demuxer_unittest.cc +++ b/call/rtp_demuxer_unittest.cc
@@ -711,7 +711,6 @@ EXPECT_TRUE(demuxer_.OnRtpPacket(*packet)); } - // If one sink is associated with SSRC x, and another sink with RSID y, then if // we receive a packet with both SSRC x and RSID y, route that to only the sink // for RSID y since we believe RSID tags to be more trustworthy than signaled
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc index b793b24..9108e83 100644 --- a/call/rtp_video_sender.cc +++ b/call/rtp_video_sender.cc
@@ -923,14 +923,14 @@ *sent_nack_rate_bps = 0; *sent_fec_rate_bps = 0; for (const RtpStreamSender& stream : rtp_streams_) { - stream.rtp_rtcp->SetFecProtectionParams(*delta_params, *key_params); + stream.rtp_rtcp->SetFecProtectionParams(*delta_params, *key_params); - auto send_bitrate = stream.rtp_rtcp->GetSendRates(); - *sent_video_rate_bps += send_bitrate[RtpPacketMediaType::kVideo].bps(); - *sent_fec_rate_bps += - send_bitrate[RtpPacketMediaType::kForwardErrorCorrection].bps(); - *sent_nack_rate_bps += - send_bitrate[RtpPacketMediaType::kRetransmission].bps(); + auto send_bitrate = stream.rtp_rtcp->GetSendRates(); + *sent_video_rate_bps += send_bitrate[RtpPacketMediaType::kVideo].bps(); + *sent_fec_rate_bps += + send_bitrate[RtpPacketMediaType::kForwardErrorCorrection].bps(); + *sent_nack_rate_bps += + send_bitrate[RtpPacketMediaType::kRetransmission].bps(); } return 0; }
diff --git a/common_audio/resampler/sinc_resampler.h b/common_audio/resampler/sinc_resampler.h index b89bba7..c6a43ab 100644 --- a/common_audio/resampler/sinc_resampler.h +++ b/common_audio/resampler/sinc_resampler.h
@@ -157,10 +157,10 @@ // Data from the source is copied into this buffer for each processing pass. std::unique_ptr<float[], AlignedFreeDeleter> input_buffer_; -// Stores the runtime selection of which Convolve function to use. -// TODO(ajm): Move to using a global static which must only be initialized -// once by the user. We're not doing this initially, because we don't have -// e.g. a LazyInstance helper in webrtc. + // Stores the runtime selection of which Convolve function to use. + // TODO(ajm): Move to using a global static which must only be initialized + // once by the user. We're not doing this initially, because we don't have + // e.g. a LazyInstance helper in webrtc. typedef float (*ConvolveProc)(const float*, const float*, const float*,
diff --git a/common_audio/third_party/ooura/fft_size_256/fft4g.cc b/common_audio/third_party/ooura/fft_size_256/fft4g.cc index d2f7c1c..2573f23 100644 --- a/common_audio/third_party/ooura/fft_size_256/fft4g.cc +++ b/common_audio/third_party/ooura/fft_size_256/fft4g.cc
@@ -286,11 +286,11 @@ w[] and ip[] are compatible with all routines. */ +#include "common_audio/third_party/ooura/fft_size_256/fft4g.h" + #include <math.h> #include <stddef.h> -#include "common_audio/third_party/ooura/fft_size_256/fft4g.h" - namespace webrtc { namespace {
diff --git a/common_audio/third_party/ooura/fft_size_256/fft4g.h b/common_audio/third_party/ooura/fft_size_256/fft4g.h index d41d2c6..5a465a3 100644 --- a/common_audio/third_party/ooura/fft_size_256/fft4g.h +++ b/common_audio/third_party/ooura/fft_size_256/fft4g.h
@@ -11,6 +11,8 @@ #ifndef COMMON_AUDIO_THIRD_PARTY_OOURA_FFT_SIZE_256_FFT4G_H_ #define COMMON_AUDIO_THIRD_PARTY_OOURA_FFT_SIZE_256_FFT4G_H_ +#include <stddef.h> + namespace webrtc { // Refer to fft4g.c for documentation.
diff --git a/common_audio/wav_header.h b/common_audio/wav_header.h index 2cccd7d..a1aa942 100644 --- a/common_audio/wav_header.h +++ b/common_audio/wav_header.h
@@ -13,6 +13,7 @@ #include <stddef.h> #include <stdint.h> + #include <algorithm> #include "rtc_base/checks.h"
diff --git a/examples/androidnativeapi/jni/android_call_client.cc b/examples/androidnativeapi/jni/android_call_client.cc index ae0a40b..2713a56 100644 --- a/examples/androidnativeapi/jni/android_call_client.cc +++ b/examples/androidnativeapi/jni/android_call_client.cc
@@ -10,9 +10,8 @@ #include "examples/androidnativeapi/jni/android_call_client.h" -#include <utility> - #include <memory> +#include <utility> #include "api/peer_connection_interface.h" #include "api/rtc_event_log/rtc_event_log_factory.h"
diff --git a/examples/androidvoip/jni/android_voip_client.cc b/examples/androidvoip/jni/android_voip_client.cc index cf07e87..92fad22 100644 --- a/examples/androidvoip/jni/android_voip_client.cc +++ b/examples/androidvoip/jni/android_voip_client.cc
@@ -12,6 +12,7 @@ #include <errno.h> #include <sys/socket.h> + #include <algorithm> #include <map> #include <memory>
diff --git a/media/base/stream_params.h b/media/base/stream_params.h index c9c8a09..60c67a1 100644 --- a/media/base/stream_params.h +++ b/media/base/stream_params.h
@@ -303,8 +303,7 @@ return RemoveStream( streams, [&ssrc](const StreamParams& sp) { return sp.has_ssrc(ssrc); }); } -inline bool RemoveStreamByIds(StreamParamsVec* streams, - const std::string& id) { +inline bool RemoveStreamByIds(StreamParamsVec* streams, const std::string& id) { return RemoveStream(streams, [&id](const StreamParams& sp) { return sp.id == id; }); }
diff --git a/media/base/video_adapter.cc b/media/base/video_adapter.cc index 01aaad1..daac8cf 100644 --- a/media/base/video_adapter.cc +++ b/media/base/video_adapter.cc
@@ -40,8 +40,8 @@ // Determines number of output pixels if both width and height of an input of // `input_pixels` pixels is scaled with the fraction numerator / denominator. int scale_pixel_count(int input_pixels) { - return (numerator * numerator * static_cast<int64_t>(input_pixels)) - / (denominator * denominator); + return (numerator * numerator * static_cast<int64_t>(input_pixels)) / + (denominator * denominator); } };
diff --git a/media/engine/fake_webrtc_video_engine.cc b/media/engine/fake_webrtc_video_engine.cc index 3cd2855..d10e618 100644 --- a/media/engine/fake_webrtc_video_engine.cc +++ b/media/engine/fake_webrtc_video_engine.cc
@@ -202,8 +202,7 @@ // Video encoder factory. FakeWebRtcVideoEncoderFactory::FakeWebRtcVideoEncoderFactory() - : num_created_encoders_(0), - vp8_factory_mode_(false) {} + : num_created_encoders_(0), vp8_factory_mode_(false) {} std::vector<webrtc::SdpVideoFormat> FakeWebRtcVideoEncoderFactory::GetSupportedFormats() const {
diff --git a/net/dcsctp/packet/chunk/data_chunk.cc b/net/dcsctp/packet/chunk/data_chunk.cc index 769be2d..cf866b7 100644 --- a/net/dcsctp/packet/chunk/data_chunk.cc +++ b/net/dcsctp/packet/chunk/data_chunk.cc
@@ -89,10 +89,10 @@ rtc::StringBuilder sb; sb << "DATA, type=" << (options().is_unordered ? "unordered" : "ordered") << "::" - << (*options().is_beginning && *options().is_end - ? "complete" - : *options().is_beginning ? "first" - : *options().is_end ? "last" : "middle") + << (*options().is_beginning && *options().is_end ? "complete" + : *options().is_beginning ? "first" + : *options().is_end ? "last" + : "middle") << ", tsn=" << *tsn() << ", sid=" << *stream_id() << ", ssn=" << *ssn() << ", ppid=" << *ppid() << ", length=" << payload().size(); return sb.Release();
diff --git a/net/dcsctp/packet/chunk/idata_chunk.cc b/net/dcsctp/packet/chunk/idata_chunk.cc index 378c527..9f19c7f 100644 --- a/net/dcsctp/packet/chunk/idata_chunk.cc +++ b/net/dcsctp/packet/chunk/idata_chunk.cc
@@ -92,10 +92,10 @@ rtc::StringBuilder sb; sb << "I-DATA, type=" << (options().is_unordered ? "unordered" : "ordered") << "::" - << (*options().is_beginning && *options().is_end - ? "complete" - : *options().is_beginning ? "first" - : *options().is_end ? "last" : "middle") + << (*options().is_beginning && *options().is_end ? "complete" + : *options().is_beginning ? "first" + : *options().is_end ? "last" + : "middle") << ", tsn=" << *tsn() << ", stream_id=" << *stream_id() << ", message_id=" << *message_id();
diff --git a/p2p/base/basic_async_resolver_factory.cc b/p2p/base/basic_async_resolver_factory.cc index 3fdf75b..2769f82 100644 --- a/p2p/base/basic_async_resolver_factory.cc +++ b/p2p/base/basic_async_resolver_factory.cc
@@ -25,7 +25,6 @@ return new rtc::AsyncResolver(); } - std::unique_ptr<webrtc::AsyncDnsResolverInterface> WrappingAsyncDnsResolverFactory::Create() { return std::make_unique<WrappingAsyncDnsResolver>(wrapped_factory_->Create());
diff --git a/p2p/base/connection_info.h b/p2p/base/connection_info.h index cd2a913..e7ed1b4 100644 --- a/p2p/base/connection_info.h +++ b/p2p/base/connection_info.h
@@ -36,16 +36,16 @@ ConnectionInfo(const ConnectionInfo&); ~ConnectionInfo(); - bool best_connection; // Is this the best connection we have? - bool writable; // Has this connection received a STUN response? - bool receiving; // Has this connection received anything? - bool timeout; // Has this connection timed out? - size_t rtt; // The STUN RTT for this connection. + bool best_connection; // Is this the best connection we have? + bool writable; // Has this connection received a STUN response? + bool receiving; // Has this connection received anything? + bool timeout; // Has this connection timed out? + size_t rtt; // The STUN RTT for this connection. size_t sent_discarded_bytes; // Number of outgoing bytes discarded due to // socket errors. size_t sent_total_bytes; // Total bytes sent on this connection. Does not // include discarded bytes. - size_t sent_bytes_second; // Bps over the last measurement interval. + size_t sent_bytes_second; // Bps over the last measurement interval. size_t sent_discarded_packets; // Number of outgoing packets discarded due to // socket errors. size_t sent_total_packets; // Number of total outgoing packets attempted for
diff --git a/p2p/base/stun_request.cc b/p2p/base/stun_request.cc index d15a3e6..25d387c 100644 --- a/p2p/base/stun_request.cc +++ b/p2p/base/stun_request.cc
@@ -35,7 +35,7 @@ // The timeout doubles each retransmission, up to this many times // RFC 5389 says SHOULD retransmit 7 times. // This has been 8 for years (not sure why). -const int STUN_MAX_RETRANSMISSIONS = 8; // Total sends: 9 +const int STUN_MAX_RETRANSMISSIONS = 8; // Total sends: 9 // We also cap the doubling, even though the standard doesn't say to. // This has been 1.6 seconds for years, but for networks that
diff --git a/p2p/base/transport_description_unittest.cc b/p2p/base/transport_description_unittest.cc index 41d7336..c3746ba 100644 --- a/p2p/base/transport_description_unittest.cc +++ b/p2p/base/transport_description_unittest.cc
@@ -9,6 +9,7 @@ */ #include "p2p/base/transport_description.h" + #include "test/gtest.h" using webrtc::RTCErrorType;
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc index c6ad9e8..cd71726 100644 --- a/pc/channel_unittest.cc +++ b/pc/channel_unittest.cc
@@ -2380,5 +2380,4 @@ Base::SocketOptionsMergedOnSetTransport(); } - // TODO(pthatcher): TestSetReceiver?
diff --git a/pc/data_channel_utils.h b/pc/data_channel_utils.h index 85cacdb..8681ba4 100644 --- a/pc/data_channel_utils.h +++ b/pc/data_channel_utils.h
@@ -13,6 +13,7 @@ #include <stddef.h> #include <stdint.h> + #include <deque> #include <memory> #include <string>
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc index 4839fcf..0956d75 100644 --- a/pc/peer_connection.cc +++ b/pc/peer_connection.cc
@@ -634,20 +634,18 @@ } // Network thread initialization. - transport_controller_copy_ = - network_thread()->BlockingCall([&] { - RTC_DCHECK_RUN_ON(network_thread()); - network_thread_safety_ = PendingTaskSafetyFlag::Create(); - InitializePortAllocatorResult pa_result = InitializePortAllocator_n( - stun_servers, turn_servers, configuration); - // Send information about IPv4/IPv6 status. - PeerConnectionAddressFamilyCounter address_family = - pa_result.enable_ipv6 ? kPeerConnection_IPv6 : kPeerConnection_IPv4; - RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", - address_family, - kPeerConnectionAddressFamilyCounter_Max); - return InitializeTransportController_n(configuration, dependencies); - }); + transport_controller_copy_ = network_thread()->BlockingCall([&] { + RTC_DCHECK_RUN_ON(network_thread()); + network_thread_safety_ = PendingTaskSafetyFlag::Create(); + InitializePortAllocatorResult pa_result = + InitializePortAllocator_n(stun_servers, turn_servers, configuration); + // Send information about IPv4/IPv6 status. + PeerConnectionAddressFamilyCounter address_family = + pa_result.enable_ipv6 ? kPeerConnection_IPv6 : kPeerConnection_IPv4; + RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family, + kPeerConnectionAddressFamilyCounter_Max); + return InitializeTransportController_n(configuration, dependencies); + }); configuration_ = configuration;
diff --git a/pc/peer_connection.h b/pc/peer_connection.h index 32d304e..26d2531 100644 --- a/pc/peer_connection.h +++ b/pc/peer_connection.h
@@ -265,9 +265,7 @@ } rtc::Thread* worker_thread() const final { return context_->worker_thread(); } - std::string session_id() const override { - return session_id_; - } + std::string session_id() const override { return session_id_; } bool initial_offerer() const override { RTC_DCHECK_RUN_ON(signaling_thread());
diff --git a/pc/proxy.h b/pc/proxy.h index b0782bb..f39b4a5 100644 --- a/pc/proxy.h +++ b/pc/proxy.h
@@ -200,8 +200,12 @@ typedef class_name##Interface C; \ \ public: \ - const INTERNAL_CLASS* internal() const { return c(); } \ - INTERNAL_CLASS* internal() { return c(); } + const INTERNAL_CLASS* internal() const { \ + return c(); \ + } \ + INTERNAL_CLASS* internal() { \ + return c(); \ + } // clang-format off // clang-format would put the semicolon alone, @@ -245,9 +249,15 @@ } \ \ private: \ - const INTERNAL_CLASS* c() const { return c_.get(); } \ - INTERNAL_CLASS* c() { return c_.get(); } \ - void DestroyInternal() { c_ = nullptr; } \ + const INTERNAL_CLASS* c() const { \ + return c_.get(); \ + } \ + INTERNAL_CLASS* c() { \ + return c_.get(); \ + } \ + void DestroyInternal() { \ + c_ = nullptr; \ + } \ rtc::scoped_refptr<INTERNAL_CLASS> c_; // Note: This doesn't use a unique_ptr, because it intends to handle a corner @@ -264,9 +274,15 @@ } \ \ private: \ - const INTERNAL_CLASS* c() const { return c_; } \ - INTERNAL_CLASS* c() { return c_; } \ - void DestroyInternal() { delete c_; } \ + const INTERNAL_CLASS* c() const { \ + return c_; \ + } \ + INTERNAL_CLASS* c() { \ + return c_; \ + } \ + void DestroyInternal() { \ + delete c_; \ + } \ INTERNAL_CLASS* c_; #define BEGIN_PRIMARY_PROXY_MAP(class_name) \ @@ -292,16 +308,20 @@ primary_thread, secondary_thread, std::move(c)); \ } -#define PROXY_PRIMARY_THREAD_DESTRUCTOR() \ - private: \ - rtc::Thread* destructor_thread() const { return primary_thread_; } \ - \ +#define PROXY_PRIMARY_THREAD_DESTRUCTOR() \ + private: \ + rtc::Thread* destructor_thread() const { \ + return primary_thread_; \ + } \ + \ public: // NOLINTNEXTLINE -#define PROXY_SECONDARY_THREAD_DESTRUCTOR() \ - private: \ - rtc::Thread* destructor_thread() const { return secondary_thread_; } \ - \ +#define PROXY_SECONDARY_THREAD_DESTRUCTOR() \ + private: \ + rtc::Thread* destructor_thread() const { \ + return secondary_thread_; \ + } \ + \ public: // NOLINTNEXTLINE #if defined(RTC_DISABLE_PROXY_TRACE_EVENTS)
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc index 4535597..8a1ded7 100644 --- a/pc/rtc_stats_collector.cc +++ b/pc/rtc_stats_collector.cc
@@ -873,8 +873,7 @@ } remote_inbound->total_round_trip_time = report_block_data.sum_rtts().seconds<double>(); - remote_inbound->round_trip_time_measurements = - report_block_data.num_rtts(); + remote_inbound->round_trip_time_measurements = report_block_data.num_rtts(); std::string local_id = RTCOutboundRtpStreamStatsIDFromSSRC( transport_id, media_type, report_block.source_ssrc);
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc index 59f0b01..6c9e647 100644 --- a/pc/sdp_offer_answer.cc +++ b/pc/sdp_offer_answer.cc
@@ -4764,13 +4764,11 @@ // - crbug.com/1187289 for (const auto& entry : channels) { std::string error; - bool success = - context_->worker_thread()->BlockingCall([&]() { - return (source == cricket::CS_LOCAL) - ? entry.first->SetLocalContent(entry.second, type, error) - : entry.first->SetRemoteContent(entry.second, type, - error); - }); + bool success = context_->worker_thread()->BlockingCall([&]() { + return (source == cricket::CS_LOCAL) + ? entry.first->SetLocalContent(entry.second, type, error) + : entry.first->SetRemoteContent(entry.second, type, error); + }); if (!success) { return RTCError(RTCErrorType::INVALID_PARAMETER, error); }
diff --git a/pc/session_description.cc b/pc/session_description.cc index 0346f8c..e1152eb 100644 --- a/pc/session_description.cc +++ b/pc/session_description.cc
@@ -281,8 +281,7 @@ return content_groups; } -ContentInfo::~ContentInfo() { -} +ContentInfo::~ContentInfo() {} // Copy operator. ContentInfo::ContentInfo(const ContentInfo& o)