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)
