Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
diff --git a/api/video_codecs/video_encoder.cc b/api/video_codecs/video_encoder.cc
index 0ee25c5..52e2866 100644
--- a/api/video_codecs/video_encoder.cc
+++ b/api/video_codecs/video_encoder.cc
@@ -112,18 +112,21 @@
rtc::SimpleStringBuilder oss(string_buf);
oss << "EncoderInfo { "
- << "ScalingSettings { ";
+ "ScalingSettings { ";
if (scaling_settings.thresholds) {
oss << "Thresholds { "
- << "low = " << scaling_settings.thresholds->low
+ "low = "
+ << scaling_settings.thresholds->low
<< ", high = " << scaling_settings.thresholds->high << "}, ";
}
oss << "min_pixels_per_frame = " << scaling_settings.min_pixels_per_frame
<< " }";
oss << ", requested_resolution_alignment = " << requested_resolution_alignment
<< ", supports_native_handle = " << supports_native_handle
- << ", implementation_name = '" << implementation_name << "'"
- << ", has_trusted_rate_controller = " << has_trusted_rate_controller
+ << ", implementation_name = '" << implementation_name
+ << "'"
+ ", has_trusted_rate_controller = "
+ << has_trusted_rate_controller
<< ", is_hardware_accelerated = " << is_hardware_accelerated
<< ", has_internal_source = " << has_internal_source
<< ", fps_allocation = [";
@@ -154,13 +157,15 @@
}
ResolutionBitrateLimits l = resolution_bitrate_limits[i];
oss << "Limits { "
- << "frame_size_pixels = " << l.frame_size_pixels
+ "frame_size_pixels = "
+ << l.frame_size_pixels
<< ", min_start_bitrate_bps = " << l.min_start_bitrate_bps
<< ", min_bitrate_bps = " << l.min_bitrate_bps
<< ", max_bitrate_bps = " << l.max_bitrate_bps << "} ";
}
oss << "] "
- << ", supports_simulcast = " << supports_simulcast << "}";
+ ", supports_simulcast = "
+ << supports_simulcast << "}";
return oss.str();
}
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 2ecc3cf..d0c17fb 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -834,7 +834,7 @@
if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
RTC_DLOG(LS_WARNING)
<< "ChannelReceive::UpdatePlayoutTimestamp() failed to read"
- << " playout delay from the ADM";
+ " playout delay from the ADM";
return;
}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 5541d75..de77158 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -420,7 +420,7 @@
payload = encrypted_audio_payload;
} else if (crypto_options_.sframe.require_frame_encryption) {
RTC_DLOG(LS_ERROR) << "Channel::SendData() failed sending audio payload: "
- << "A frame encryptor is required but one is not set.";
+ "A frame encryptor is required but one is not set.";
return -1;
}
}
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 9ffa515..40005ef 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -87,7 +87,7 @@
if (config.payload_type < 0) {
RTC_LOG(LS_WARNING)
<< "Invalid FlexFEC payload type given. "
- << "This FlexfecReceiveStream will therefore be useless.";
+ "This FlexfecReceiveStream will therefore be useless.";
return nullptr;
}
RTC_DCHECK_GE(config.payload_type, 0);
@@ -95,13 +95,13 @@
if (config.remote_ssrc == 0) {
RTC_LOG(LS_WARNING)
<< "Invalid FlexFEC SSRC given. "
- << "This FlexfecReceiveStream will therefore be useless.";
+ "This FlexfecReceiveStream will therefore be useless.";
return nullptr;
}
if (config.protected_media_ssrcs.empty()) {
RTC_LOG(LS_WARNING)
<< "No protected media SSRC supplied. "
- << "This FlexfecReceiveStream will therefore be useless.";
+ "This FlexfecReceiveStream will therefore be useless.";
return nullptr;
}
diff --git a/call/rtp_stream_receiver_controller.cc b/call/rtp_stream_receiver_controller.cc
index 0fc8b26..f440b42 100644
--- a/call/rtp_stream_receiver_controller.cc
+++ b/call/rtp_stream_receiver_controller.cc
@@ -25,7 +25,8 @@
if (!sink_added) {
RTC_LOG(LS_ERROR)
<< "RtpStreamReceiverController::Receiver::Receiver: Sink "
- << "could not be added for SSRC=" << ssrc << ".";
+ "could not be added for SSRC="
+ << ssrc << ".";
}
}
diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index 282a3ad..a5878ab 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -390,7 +390,7 @@
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetSdpBitrateParameters: "
- << "nothing to update";
+ "nothing to update";
}
}
@@ -411,7 +411,7 @@
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetClientBitratePreferences: "
- << "nothing to update";
+ "nothing to update";
}
}
diff --git a/call/simulated_network.cc b/call/simulated_network.cc
index d6a7369..f904464 100644
--- a/call/simulated_network.cc
+++ b/call/simulated_network.cc
@@ -100,9 +100,10 @@
int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));
RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
- << "For a total packet loss of " << config.loss_percent << "%% then"
- << " avg_burst_loss_length must be " << min_avg_burst_loss_length + 1
- << " or higher.";
+ << "For a total packet loss of " << config.loss_percent
+ << "%% then"
+ " avg_burst_loss_length must be "
+ << min_avg_burst_loss_length + 1 << " or higher.";
config_state_.prob_loss_bursting = (1.0 - 1.0 / avg_burst_loss_length);
config_state_.prob_start_bursting =
diff --git a/common_video/bitrate_adjuster.cc b/common_video/bitrate_adjuster.cc
index e2d3b3d..ca52ed9 100644
--- a/common_video/bitrate_adjuster.cc
+++ b/common_video/bitrate_adjuster.cc
@@ -140,7 +140,7 @@
float last_adjusted_bitrate_bps = adjusted_bitrate_bps_;
if (adjusted_bitrate_bps != last_adjusted_bitrate_bps) {
RTC_LOG(LS_VERBOSE) << "Adjusting encoder bitrate:"
- << "\n target_bitrate:"
+ "\n target_bitrate:"
<< static_cast<uint32_t>(target_bitrate_bps)
<< "\n estimated_bitrate:"
<< static_cast<uint32_t>(estimated_bitrate_bps)
diff --git a/examples/peerconnection/client/conductor.cc b/examples/peerconnection/client/conductor.cc
index 10fbc79..005a9d6 100644
--- a/examples/peerconnection/client/conductor.cc
+++ b/examples/peerconnection/client/conductor.cc
@@ -345,7 +345,8 @@
webrtc::CreateSessionDescription(type, sdp, &error);
if (!session_description) {
RTC_LOG(WARNING) << "Can't parse received session description message. "
- << "SdpParseError was: " << error.description;
+ "SdpParseError was: "
+ << error.description;
return;
}
RTC_LOG(INFO) << " Received session description :" << message;
@@ -373,7 +374,8 @@
webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, sdp, &error));
if (!candidate.get()) {
RTC_LOG(WARNING) << "Can't parse received candidate message. "
- << "SdpParseError was: " << error.description;
+ "SdpParseError was: "
+ << error.description;
return;
}
if (!peer_connection_->AddIceCandidate(candidate.get())) {
diff --git a/examples/unityplugin/simple_peer_connection.cc b/examples/unityplugin/simple_peer_connection.cc
index 8a719ba..05282fa 100644
--- a/examples/unityplugin/simple_peer_connection.cc
+++ b/examples/unityplugin/simple_peer_connection.cc
@@ -342,7 +342,8 @@
webrtc::CreateSessionDescription(sdp_type, remote_desc, &error));
if (!session_description) {
RTC_LOG(WARNING) << "Can't parse received session description message. "
- << "SdpParseError was: " << error.description;
+ "SdpParseError was: "
+ << error.description;
return false;
}
RTC_LOG(INFO) << " Received session description :" << remote_desc;
@@ -363,7 +364,8 @@
webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate, &error));
if (!ice_candidate.get()) {
RTC_LOG(WARNING) << "Can't parse received candidate message. "
- << "SdpParseError was: " << error.description;
+ "SdpParseError was: "
+ << error.description;
return false;
}
if (!peer_connection_->AddIceCandidate(ice_candidate.get())) {
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
index 7b35485..2f1c5a4 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -674,7 +674,8 @@
if (event.config().codecs.size() > 1) {
RTC_LOG(WARNING)
<< "LogVideoSendStreamConfig currently only supports one "
- << "codec. Logging codec :" << codec.payload_name;
+ "codec. Logging codec :"
+ << codec.payload_name;
break;
}
}
diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index 405f702..c9d4a6c 100644
--- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -249,7 +249,9 @@
event_processor.ProcessEventsInOrder();
std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
- << " RTP packets and " << rtcp_counter << " RTCP packets to the "
- << "output file." << std::endl;
+ << " RTP packets and " << rtcp_counter
+ << " RTCP packets to the "
+ "output file."
+ << std::endl;
return 0;
}
diff --git a/logging/rtc_event_log/rtc_event_log_impl.cc b/logging/rtc_event_log/rtc_event_log_impl.cc
index e1e1aab..4a272f0 100644
--- a/logging/rtc_event_log/rtc_event_log_impl.cc
+++ b/logging/rtc_event_log/rtc_event_log_impl.cc
@@ -93,7 +93,8 @@
const int64_t timestamp_us = rtc::TimeMicros();
const int64_t utc_time_us = rtc::TimeUTCMicros();
RTC_LOG(LS_INFO) << "Starting WebRTC event log. (Timestamp, UTC) = "
- << "(" << timestamp_us << ", " << utc_time_us << ").";
+ "("
+ << timestamp_us << ", " << utc_time_us << ").";
RTC_DCHECK_RUN_ON(&logging_state_checker_);
logging_state_started_ = true;
diff --git a/media/base/rtp_data_engine.cc b/media/base/rtp_data_engine.cc
index 6161085..0303cd3 100644
--- a/media/base/rtp_data_engine.cc
+++ b/media/base/rtp_data_engine.cc
@@ -319,8 +319,8 @@
packet.AppendData(payload);
RTC_LOG(LS_VERBOSE) << "Sent RTP data packet: "
- << " stream=" << found_stream->id
- << " ssrc=" << header.ssrc
+ " stream="
+ << found_stream->id << " ssrc=" << header.ssrc
<< ", seqnum=" << header.seq_num
<< ", timestamp=" << header.timestamp
<< ", len=" << payload.size();
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index c93494b..a2944d5 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -735,7 +735,7 @@
if (!allow_codec_switching_) {
RTC_LOG(LS_INFO) << "Encoder switch requested but codec switching has"
- << " not been enabled yet.";
+ " not been enabled yet.";
requested_encoder_switch_ = conf;
return;
}
@@ -857,7 +857,8 @@
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
- << "with ssrc " << ssrc << " which doesn't exist.";
+ "with ssrc "
+ << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
@@ -878,7 +879,8 @@
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream "
- << "with ssrc " << ssrc << " which doesn't exist.";
+ "with ssrc "
+ << ssrc << " which doesn't exist.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
@@ -887,7 +889,7 @@
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
- << "is not currently supported.";
+ "is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
@@ -922,7 +924,8 @@
if (it == receive_streams_.end()) {
RTC_LOG(LS_WARNING)
<< "Attempting to get RTP receive parameters for stream "
- << "with SSRC " << ssrc << " which doesn't exist.";
+ "with SSRC "
+ << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
rtp_params = it->second->GetRtpParameters();
@@ -2736,7 +2739,8 @@
if (stream_) {
RTC_LOG(LS_INFO)
<< "Setting FrameDecryptor (recv) because of SetFrameDecryptor, "
- << "remote_ssrc=" << config_.rtp.remote_ssrc;
+ "remote_ssrc="
+ << config_.rtp.remote_ssrc;
stream_->SetFrameDecryptor(frame_decryptor);
}
}
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index bff3172..a36fc6e 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -170,9 +170,10 @@
// fail. If codec is not multi-rate and |bps| exceeds or equal the fixed
// bitrate then ignore.
RTC_LOG(LS_ERROR) << "Failed to set codec " << spec.format.name
- << " to bitrate " << bps << " bps"
- << ", requires at least " << spec.info.min_bitrate_bps
- << " bps.";
+ << " to bitrate " << bps
+ << " bps"
+ ", requires at least "
+ << spec.info.min_bitrate_bps << " bps.";
return absl::nullopt;
}
@@ -1181,7 +1182,7 @@
return true;
} else {
RTC_LOG(LS_ERROR) << "Failed to SetBaseMinimumPlayoutDelayMs"
- << " on AudioReceiveStream on SSRC="
+ " on AudioReceiveStream on SSRC="
<< config_.rtp.remote_ssrc
<< " with delay_ms=" << delay_ms;
return false;
@@ -1351,7 +1352,8 @@
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
- << "with ssrc " << ssrc << " which doesn't exist.";
+ "with ssrc "
+ << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
@@ -1371,7 +1373,8 @@
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
RTC_LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
- << "with ssrc " << ssrc << " which doesn't exist.";
+ "with ssrc "
+ << ssrc << " which doesn't exist.";
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
@@ -1380,7 +1383,7 @@
webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
if (current_parameters.codecs != parameters.codecs) {
RTC_DLOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
- << "is not currently supported.";
+ "is not currently supported.";
return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER);
}
@@ -1426,7 +1429,8 @@
if (it == recv_streams_.end()) {
RTC_LOG(LS_WARNING)
<< "Attempting to get RTP receive parameters for stream "
- << "with ssrc " << ssrc << " which doesn't exist.";
+ "with ssrc "
+ << ssrc << " which doesn't exist.";
return webrtc::RtpParameters();
}
rtp_params = it->second->GetRtpParameters();
diff --git a/media/sctp/sctp_transport.cc b/media/sctp/sctp_transport.cc
index 31489eb..3a083b42 100644
--- a/media/sctp/sctp_transport.cc
+++ b/media/sctp/sctp_transport.cc
@@ -284,7 +284,8 @@
uint8_t set_df) {
SctpTransport* transport = static_cast<SctpTransport*>(addr);
RTC_LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
- << "addr: " << addr << "; length: " << length
+ "addr: "
+ << addr << "; length: " << length
<< "; tos: " << rtc::ToHex(tos)
<< "; set_df: " << rtc::ToHex(set_df);
@@ -511,9 +512,11 @@
bool SctpTransport::OpenStream(int sid) {
RTC_DCHECK_RUN_ON(network_thread_);
if (sid > kMaxSctpSid) {
- RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
- << "Not adding data stream "
- << "with sid=" << sid << " because sid is too high.";
+ RTC_LOG(LS_WARNING) << debug_name_
+ << "->OpenStream(...): "
+ "Not adding data stream "
+ "with sid="
+ << sid << " because sid is too high.";
return false;
}
auto it = stream_status_by_sid_.find(sid);
@@ -522,16 +525,18 @@
return true;
}
if (it->second.is_open()) {
- RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
- << "Not adding data stream "
- << "with sid=" << sid
- << " because stream is already open.";
+ RTC_LOG(LS_WARNING) << debug_name_
+ << "->OpenStream(...): "
+ "Not adding data stream "
+ "with sid="
+ << sid << " because stream is already open.";
return false;
} else {
- RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
- << "Not adding data stream "
- << " with sid=" << sid
- << " because stream is still closing.";
+ RTC_LOG(LS_WARNING) << debug_name_
+ << "->OpenStream(...): "
+ "Not adding data stream "
+ " with sid="
+ << sid << " because stream is still closing.";
return false;
}
}
@@ -546,8 +551,9 @@
return false;
}
- RTC_LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid << "): "
- << "Queuing RE-CONFIG chunk.";
+ RTC_LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid
+ << "): "
+ "Queuing RE-CONFIG chunk.";
it->second.closure_initiated = true;
// Signal our stream-reset logic that it should try to send now, if it can.
@@ -597,8 +603,9 @@
SendDataResult SctpTransport::SendMessageInternal(OutgoingMessage* message) {
RTC_DCHECK_RUN_ON(network_thread_);
if (!sock_) {
- RTC_LOG(LS_WARNING) << debug_name_ << "->SendMessageInternal(...): "
- << "Not sending packet with sid="
+ RTC_LOG(LS_WARNING) << debug_name_
+ << "->SendMessageInternal(...): "
+ "Not sending packet with sid="
<< message->send_params().sid
<< " len=" << message->size() << " before Start().";
return SDR_ERROR;
@@ -607,8 +614,9 @@
auto it = stream_status_by_sid_.find(message->send_params().sid);
if (it == stream_status_by_sid_.end() || !it->second.is_open()) {
RTC_LOG(LS_WARNING)
- << debug_name_ << "->SendMessageInternal(...): "
- << "Not sending data because sid is unknown or closing: "
+ << debug_name_
+ << "->SendMessageInternal(...): "
+ "Not sending data because sid is unknown or closing: "
<< message->send_params().sid;
return SDR_ERROR;
}
@@ -636,9 +644,9 @@
return SDR_BLOCK;
}
- RTC_LOG_ERRNO(LS_ERROR)
- << "ERROR:" << debug_name_ << "->SendMessageInternal(...): "
- << " usrsctp_sendv: ";
+ RTC_LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_
+ << "->SendMessageInternal(...): "
+ " usrsctp_sendv: ";
return SDR_ERROR;
}
@@ -711,9 +719,10 @@
int connect_result = usrsctp_connect(
sock_, reinterpret_cast<sockaddr*>(&remote_sconn), sizeof(remote_sconn));
if (connect_result < 0 && errno != SCTP_EINPROGRESS) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
- << "Failed usrsctp_connect. got errno=" << errno
- << ", but wanted " << SCTP_EINPROGRESS;
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->Connect(): "
+ "Failed usrsctp_connect. got errno="
+ << errno << ", but wanted " << SCTP_EINPROGRESS;
CloseSctpSocket();
return false;
}
@@ -727,8 +736,9 @@
params.spp_pathmtu = kSctpMtu - sizeof(struct sctp_common_header);
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, ¶ms,
sizeof(params))) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
- << "Failed to set SCTP_PEER_ADDR_PARAMS.";
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->Connect(): "
+ "Failed to set SCTP_PEER_ADDR_PARAMS.";
}
// Since this is a fresh SCTP association, we'll always start out with empty
// queues, so "ReadyToSendData" should be true.
@@ -739,8 +749,9 @@
bool SctpTransport::OpenSctpSocket() {
RTC_DCHECK_RUN_ON(network_thread_);
if (sock_) {
- RTC_LOG(LS_WARNING) << debug_name_ << "->OpenSctpSocket(): "
- << "Ignoring attempt to re-create existing socket.";
+ RTC_LOG(LS_WARNING) << debug_name_
+ << "->OpenSctpSocket(): "
+ "Ignoring attempt to re-create existing socket.";
return false;
}
@@ -755,8 +766,9 @@
AF_CONN, SOCK_STREAM, IPPROTO_SCTP, &UsrSctpWrapper::OnSctpInboundPacket,
&UsrSctpWrapper::SendThresholdCallback, kSendThreshold, this);
if (!sock_) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->OpenSctpSocket(): "
- << "Failed to create SCTP socket.";
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->OpenSctpSocket(): "
+ "Failed to create SCTP socket.";
UsrSctpWrapper::DecrementUsrSctpUsageCount();
return false;
}
@@ -779,8 +791,9 @@
// Make the socket non-blocking. Connect, close, shutdown etc will not block
// the thread waiting for the socket operation to complete.
if (usrsctp_set_non_blocking(sock_, 1) < 0) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
- << "Failed to set SCTP to non blocking.";
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->ConfigureSctpSocket(): "
+ "Failed to set SCTP to non blocking.";
return false;
}
@@ -792,8 +805,9 @@
linger_opt.l_linger = 0;
if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt,
sizeof(linger_opt))) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
- << "Failed to set SO_LINGER.";
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->ConfigureSctpSocket(): "
+ "Failed to set SO_LINGER.";
return false;
}
@@ -803,9 +817,9 @@
stream_rst.assoc_value = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET,
&stream_rst, sizeof(stream_rst))) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
-
- << "Failed to set SCTP_ENABLE_STREAM_RESET.";
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->ConfigureSctpSocket(): "
+ "Failed to set SCTP_ENABLE_STREAM_RESET.";
return false;
}
@@ -813,8 +827,9 @@
uint32_t nodelay = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay,
sizeof(nodelay))) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
- << "Failed to set SCTP_NODELAY.";
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->ConfigureSctpSocket(): "
+ "Failed to set SCTP_NODELAY.";
return false;
}
@@ -822,8 +837,9 @@
uint32_t eor = 1;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EXPLICIT_EOR, &eor,
sizeof(eor))) {
- RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
- << "Failed to set SCTP_EXPLICIT_EOR.";
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->ConfigureSctpSocket(): "
+ "Failed to set SCTP_EXPLICIT_EOR.";
return false;
}
@@ -838,10 +854,10 @@
event.se_type = event_types[i];
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event,
sizeof(event)) < 0) {
- RTC_LOG_ERRNO(LS_ERROR)
- << debug_name_ << "->ConfigureSctpSocket(): "
-
- << "Failed to set SCTP_EVENT type: " << event.se_type;
+ RTC_LOG_ERRNO(LS_ERROR) << debug_name_
+ << "->ConfigureSctpSocket(): "
+ "Failed to set SCTP_EVENT type: "
+ << event.se_type;
return false;
}
}
@@ -974,8 +990,10 @@
return;
}
- RTC_LOG(LS_VERBOSE) << debug_name_ << "->OnPacketRead(...): "
- << " length=" << len << ", started: " << started_;
+ RTC_LOG(LS_VERBOSE) << debug_name_
+ << "->OnPacketRead(...): "
+ " length="
+ << len << ", started: " << started_;
// Only give receiving packets to usrsctp after if connected. This enables two
// peers to each make a connect call, but for them not to receive an INIT
// packet before they have called connect; least the last receiver of the INIT
@@ -1023,10 +1041,11 @@
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread_);
if (buffer.size() > (kSctpMtu)) {
- RTC_LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): "
- << "SCTP seems to have made a packet that is bigger "
- << "than its official MTU: " << buffer.size()
- << " vs max of " << kSctpMtu;
+ RTC_LOG(LS_ERROR) << debug_name_
+ << "->OnPacketFromSctpToNetwork(...): "
+ "SCTP seems to have made a packet that is bigger "
+ "than its official MTU: "
+ << buffer.size() << " vs max of " << kSctpMtu;
}
TRACE_EVENT0("webrtc", "SctpTransport::OnPacketFromSctpToNetwork");
@@ -1048,8 +1067,9 @@
RTC_DCHECK_RUN_ON(network_thread_);
RTC_LOG(LS_VERBOSE) << debug_name_
<< "->OnInboundPacketFromSctpToTransport(...): "
- << "Received SCTP data:"
- << " sid=" << params.sid
+ "Received SCTP data:"
+ " sid="
+ << params.sid
<< " notification: " << (flags & MSG_NOTIFICATION)
<< " length=" << buffer.size();
// Sending a packet with data == NULL (no data) is SCTPs "close the
@@ -1071,9 +1091,10 @@
const ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread_);
- RTC_LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToTransport(...): "
- << "Posting with length: " << buffer.size()
- << " on stream " << params.sid;
+ RTC_LOG(LS_VERBOSE) << debug_name_
+ << "->OnDataFromSctpToTransport(...): "
+ "Posting with length: "
+ << buffer.size() << " on stream " << params.sid;
// Reports all received messages to upper layers, no matter whether the sid
// is known.
SignalDataReceived(params, buffer);
@@ -1119,7 +1140,7 @@
const struct sctp_send_failed_event& ssfe =
notification.sn_send_failed_event;
RTC_LOG(LS_WARNING) << "SCTP_SEND_FAILED_EVENT: message with"
- << " PPID = "
+ " PPID = "
<< rtc::NetworkToHost32(ssfe.ssfe_info.snd_ppid)
<< " SID = " << ssfe.ssfe_info.snd_sid
<< " flags = " << rtc::ToHex(ssfe.ssfe_info.snd_flags)
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index b223b2d..8e1ffaf 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -53,7 +53,8 @@
{&estimate_dtx_delay_, &time_stretch_cn_, &target_level_window_ms_},
field_trial_name);
RTC_LOG(LS_INFO) << "NetEq decision logic settings:"
- << " estimate_dtx_delay=" << estimate_dtx_delay_
+ " estimate_dtx_delay="
+ << estimate_dtx_delay_
<< " time_stretch_cn=" << time_stretch_cn_
<< " target_level_window_ms=" << target_level_window_ms_;
}
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index 0003d32..4ae6d10 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -71,7 +71,8 @@
}
}
RTC_LOG(LS_INFO) << "Delay histogram config:"
- << " quantile=" << config.quantile
+ " quantile="
+ << config.quantile
<< " forget_factor=" << config.forget_factor
<< " start_forget_weight="
<< config.start_forget_weight.value_or(0);
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 05e7b73..d0945d7 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -141,8 +141,9 @@
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
- RTC_LOG(LS_ERROR) << "Sample rate " << fs << " Hz not supported. "
- << "Changing to 8000 Hz.";
+ RTC_LOG(LS_ERROR) << "Sample rate " << fs
+ << " Hz not supported. "
+ "Changing to 8000 Hz.";
fs = 8000;
}
controller_->SetMaximumDelay(config.max_delay_ms);
diff --git a/modules/audio_coding/neteq/tools/neteq_input.cc b/modules/audio_coding/neteq/tools/neteq_input.cc
index 645894d..de41634 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_input.cc
@@ -21,13 +21,24 @@
std::string NetEqInput::PacketData::ToString() const {
rtc::StringBuilder ss;
ss << "{"
- << "time_ms: " << static_cast<int64_t>(time_ms) << ", "
- << "header: {"
- << "pt: " << static_cast<int>(header.payloadType) << ", "
- << "sn: " << header.sequenceNumber << ", "
- << "ts: " << header.timestamp << ", "
- << "ssrc: " << header.ssrc << "}, "
- << "payload bytes: " << payload.size() << "}";
+ "time_ms: "
+ << static_cast<int64_t>(time_ms)
+ << ", "
+ "header: {"
+ "pt: "
+ << static_cast<int>(header.payloadType)
+ << ", "
+ "sn: "
+ << header.sequenceNumber
+ << ", "
+ "ts: "
+ << header.timestamp
+ << ", "
+ "ssrc: "
+ << header.ssrc
+ << "}, "
+ "payload bytes: "
+ << payload.size() << "}";
return ss.Release();
}
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index aa73b85..b37bea1 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -242,8 +242,9 @@
bool output_files_base_name_specified = !output_files_base_name.empty();
if (!textlog && !plotting && output_files_base_name_specified) {
std::cout << "Error: --output_files_base_name cannot be used without at "
- << "least one of the following flags: --textlog, --matlabplot, "
- << "--pythonplot." << std::endl;
+ "least one of the following flags: --textlog, --matlabplot, "
+ "--pythonplot."
+ << std::endl;
return false;
}
// Without |output_audio_filename|, |output_files_base_name| is required when
@@ -252,8 +253,9 @@
if (output_audio_filename.empty() && plotting &&
!output_files_base_name_specified) {
std::cout << "Error: when no output audio file is specified and "
- << "--matlabplot and/or --pythonplot are used, "
- << "--output_files_base_name must be also used." << std::endl;
+ "--matlabplot and/or --pythonplot are used, "
+ "--output_files_base_name must be also used."
+ << std::endl;
return false;
}
return true;
diff --git a/modules/audio_device/android/aaudio_player.cc b/modules/audio_device/android/aaudio_player.cc
index def0322..6d310ed 100644
--- a/modules/audio_device/android/aaudio_player.cc
+++ b/modules/audio_device/android/aaudio_player.cc
@@ -158,7 +158,8 @@
// utilized.
if (first_data_callback_) {
RTC_LOG(INFO) << "--- First output data callback: "
- << "device id=" << aaudio_.device_id();
+ "device id="
+ << aaudio_.device_id();
first_data_callback_ = false;
}
diff --git a/modules/audio_device/android/aaudio_recorder.cc b/modules/audio_device/android/aaudio_recorder.cc
index 3c5dae9..3a29bb8 100644
--- a/modules/audio_device/android/aaudio_recorder.cc
+++ b/modules/audio_device/android/aaudio_recorder.cc
@@ -161,7 +161,8 @@
// is obtained.
if (first_data_callback_) {
RTC_LOG(INFO) << "--- First input data callback: "
- << "device id=" << aaudio_.device_id();
+ "device id="
+ << aaudio_.device_id();
aaudio_.ClearInputStream(audio_data, num_frames);
first_data_callback_ = false;
}
diff --git a/modules/audio_device/android/audio_manager.cc b/modules/audio_device/android/audio_manager.cc
index 9c2bdd4..9c8137b 100644
--- a/modules/audio_device/android/audio_manager.cc
+++ b/modules/audio_device/android/audio_manager.cc
@@ -275,7 +275,8 @@
jint input_buffer_size) {
RTC_LOG(INFO)
<< "OnCacheAudioParameters: "
- << "hardware_aec: " << static_cast<bool>(hardware_aec)
+ "hardware_aec: "
+ << static_cast<bool>(hardware_aec)
<< ", hardware_agc: " << static_cast<bool>(hardware_agc)
<< ", hardware_ns: " << static_cast<bool>(hardware_ns)
<< ", low_latency_output: " << static_cast<bool>(low_latency_output)
diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 8b56330..336846e 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -413,11 +413,19 @@
abs_diff_rate_in_percent);
RTC_LOG(INFO) << "[REC : " << time_since_last << "msec, "
<< rec_sample_rate / 1000 << "kHz] callbacks: "
- << stats.rec_callbacks - last_stats_.rec_callbacks << ", "
- << "samples: " << diff_samples << ", "
- << "rate: " << static_cast<int>(rate + 0.5) << ", "
- << "rate diff: " << abs_diff_rate_in_percent << "%, "
- << "level: " << stats.max_rec_level;
+ << stats.rec_callbacks - last_stats_.rec_callbacks
+ << ", "
+ "samples: "
+ << diff_samples
+ << ", "
+ "rate: "
+ << static_cast<int>(rate + 0.5)
+ << ", "
+ "rate diff: "
+ << abs_diff_rate_in_percent
+ << "%, "
+ "level: "
+ << stats.max_rec_level;
}
diff_samples = stats.play_samples - last_stats_.play_samples;
@@ -431,11 +439,19 @@
abs_diff_rate_in_percent);
RTC_LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
<< play_sample_rate / 1000 << "kHz] callbacks: "
- << stats.play_callbacks - last_stats_.play_callbacks << ", "
- << "samples: " << diff_samples << ", "
- << "rate: " << static_cast<int>(rate + 0.5) << ", "
- << "rate diff: " << abs_diff_rate_in_percent << "%, "
- << "level: " << stats.max_play_level;
+ << stats.play_callbacks - last_stats_.play_callbacks
+ << ", "
+ "samples: "
+ << diff_samples
+ << ", "
+ "rate: "
+ << static_cast<int>(rate + 0.5)
+ << ", "
+ "rate diff: "
+ << abs_diff_rate_in_percent
+ << "%, "
+ "level: "
+ << stats.max_play_level;
}
}
last_stats_ = stats;
diff --git a/modules/audio_device/dummy/file_audio_device_factory.cc b/modules/audio_device/dummy/file_audio_device_factory.cc
index 60ef92b..0f56e06 100644
--- a/modules/audio_device/dummy/file_audio_device_factory.cc
+++ b/modules/audio_device/dummy/file_audio_device_factory.cc
@@ -29,8 +29,8 @@
if (!_isConfigured) {
RTC_LOG(LS_WARNING)
<< "WebRTC configured with WEBRTC_DUMMY_FILE_DEVICES but "
- << "no device files supplied. Will fall back to dummy "
- << "audio.";
+ "no device files supplied. Will fall back to dummy "
+ "audio.";
return nullptr;
}
diff --git a/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc b/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
index 8a755f6..4368ec9 100644
--- a/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
+++ b/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
@@ -181,7 +181,8 @@
if (errVal < 0) {
RTC_LOG(LS_ERROR)
<< "snd_mixer_selem_register(_outputMixerHandle, NULL, NULL), "
- << "error: " << LATE(snd_strerror)(errVal);
+ "error: "
+ << LATE(snd_strerror)(errVal);
_outputMixerHandle = NULL;
return -1;
}
@@ -262,7 +263,8 @@
if (errVal < 0) {
RTC_LOG(LS_ERROR)
<< "snd_mixer_selem_register(_inputMixerHandle, NULL, NULL), "
- << "error: " << LATE(snd_strerror)(errVal);
+ "error: "
+ << LATE(snd_strerror)(errVal);
_inputMixerHandle = NULL;
return -1;
diff --git a/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc b/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
index 3728a3d..c507e62 100644
--- a/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
+++ b/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
@@ -455,7 +455,8 @@
RTC_LOG(LS_VERBOSE)
<< "AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
- << " => available=" << available;
+ " => available="
+ << available;
return 0;
}
diff --git a/modules/audio_device/mac/audio_device_mac.cc b/modules/audio_device/mac/audio_device_mac.cc
index 345935f..e894cf3 100644
--- a/modules/audio_device/mac/audio_device_mac.cc
+++ b/modules/audio_device/mac/audio_device_mac.cc
@@ -1034,7 +1034,7 @@
if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) {
RTC_LOG(LS_ERROR) << "Non-interleaved audio data is not supported."
- << "AudioHardware streams should not have this format.";
+ "AudioHardware streams should not have this format.";
return -1;
}
@@ -1333,7 +1333,7 @@
if (!_stopEventRec.Wait(2000)) {
rtc::CritScope critScoped(&_critSect);
RTC_LOG(LS_WARNING) << "Timed out stopping the capture IOProc."
- << "We may have failed to detect a device removal.";
+ "We may have failed to detect a device removal.";
WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
WEBRTC_CA_LOG_WARN(
AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
@@ -1361,7 +1361,7 @@
if (!_stopEvent.Wait(2000)) {
rtc::CritScope critScoped(&_critSect);
RTC_LOG(LS_WARNING) << "Timed out stopping the shared IOProc."
- << "We may have failed to detect a device removal.";
+ "We may have failed to detect a device removal.";
// We assume rendering on a shared device has stopped as well if
// the IOProc times out.
WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
@@ -1468,7 +1468,7 @@
if (!_stopEvent.Wait(2000)) {
rtc::CritScope critScoped(&_critSect);
RTC_LOG(LS_WARNING) << "Timed out stopping the render IOProc."
- << "We may have failed to detect a device removal.";
+ "We may have failed to detect a device removal.";
// We assume capturing on a shared device has stopped as well if the
// IOProc times out.
diff --git a/modules/audio_device/win/audio_device_core_win.cc b/modules/audio_device/win/audio_device_core_win.cc
index fbcd7fc..a7aecb0 100644
--- a/modules/audio_device/win/audio_device_core_win.cc
+++ b/modules/audio_device/win/audio_device_core_win.cc
@@ -265,10 +265,10 @@
if (FAILED(hr)) {
RTC_LOG(LS_ERROR) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " Failed to create the required COM object (hr=" << hr
- << ")";
+ " Failed to create the required COM object (hr="
+ << hr << ")";
RTC_LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " CoCreateInstance(MMDeviceEnumerator) failed (hr="
+ " CoCreateInstance(MMDeviceEnumerator) failed (hr="
<< hr << ")";
const DWORD dwFlags =
@@ -295,7 +295,8 @@
MMDeviceIsAvailable = true;
RTC_LOG(LS_VERBOSE)
<< "AudioDeviceWindowsCore::CoreAudioIsSupported()"
- << " CoCreateInstance(MMDeviceEnumerator) succeeded (hr=" << hr << ")";
+ " CoCreateInstance(MMDeviceEnumerator) succeeded (hr="
+ << hr << ")";
SAFE_RELEASE(pIMMD);
}
@@ -404,7 +405,7 @@
// Handle is valid (should only happen if OS larger than vista & win7).
// Try to get the function addresses.
RTC_LOG(LS_VERBOSE) << "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
- << " The Avrt DLL module is now loaded";
+ " The Avrt DLL module is now loaded";
_PAvRevertMmThreadCharacteristics =
(PAvRevertMmThreadCharacteristics)GetProcAddress(
@@ -419,13 +420,13 @@
_PAvSetMmThreadCharacteristicsA && _PAvSetMmThreadPriority) {
RTC_LOG(LS_VERBOSE)
<< "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
- << " AvRevertMmThreadCharacteristics() is OK";
+ " AvRevertMmThreadCharacteristics() is OK";
RTC_LOG(LS_VERBOSE)
<< "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
- << " AvSetMmThreadCharacteristicsA() is OK";
+ " AvSetMmThreadCharacteristicsA() is OK";
RTC_LOG(LS_VERBOSE)
<< "AudioDeviceWindowsCore::AudioDeviceWindowsCore()"
- << " AvSetMmThreadPriority() is OK";
+ " AvSetMmThreadPriority() is OK";
_winSupportAvrt = true;
}
}
@@ -535,10 +536,10 @@
if (!freeOK) {
RTC_LOG(LS_WARNING)
<< "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
- << " failed to free the loaded Avrt DLL module correctly";
+ " failed to free the loaded Avrt DLL module correctly";
} else {
RTC_LOG(LS_WARNING) << "AudioDeviceWindowsCore::~AudioDeviceWindowsCore()"
- << " the Avrt DLL module is now unloaded";
+ " the Avrt DLL module is now unloaded";
}
}
}
@@ -653,7 +654,7 @@
int16_t nDevices = PlayoutDevices();
if (_outputDeviceIndex > (nDevices - 1)) {
RTC_LOG(LS_ERROR) << "current device selection is invalid => unable to"
- << " initialize";
+ " initialize";
return -1;
}
}
@@ -722,7 +723,7 @@
int16_t nDevices = RecordingDevices();
if (_inputDeviceIndex > (nDevices - 1)) {
RTC_LOG(LS_ERROR) << "current device selection is invalid => unable to"
- << " initialize";
+ " initialize";
return -1;
}
}
@@ -1878,8 +1879,8 @@
RTC_LOG(INFO) << "nChannels=" << Wfx.nChannels
<< ", nSamplesPerSec=" << Wfx.nSamplesPerSec
<< " is not supported. Closest match: "
- << "nChannels=" << pWfxClosestMatch->nChannels
- << ", nSamplesPerSec="
+ "nChannels="
+ << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
<< pWfxClosestMatch->nSamplesPerSec;
CoTaskMemFree(pWfxClosestMatch);
pWfxClosestMatch = NULL;
@@ -2199,8 +2200,8 @@
RTC_LOG(INFO) << "nChannels=" << Wfx.Format.nChannels
<< ", nSamplesPerSec=" << Wfx.Format.nSamplesPerSec
<< " is not supported. Closest match: "
- << "nChannels=" << pWfxClosestMatch->nChannels
- << ", nSamplesPerSec="
+ "nChannels="
+ << pWfxClosestMatch->nChannels << ", nSamplesPerSec="
<< pWfxClosestMatch->nSamplesPerSec;
CoTaskMemFree(pWfxClosestMatch);
pWfxClosestMatch = NULL;
@@ -2338,7 +2339,7 @@
// give it render data to process.
RTC_LOG(LS_ERROR)
<< "Playout must be started before recording when using"
- << " the built-in AEC";
+ " the built-in AEC";
return -1;
}
}
@@ -2571,7 +2572,7 @@
// playout to stop properly.
RTC_LOG(LS_WARNING)
<< "Recording should be stopped before playout when using the"
- << " built-in AEC";
+ " built-in AEC";
}
// Reset the playout delay value.
@@ -2822,7 +2823,7 @@
_UnLock();
RTC_LOG(LS_ERROR)
<< "output state has been modified during unlocked"
- << " period";
+ " period";
goto Exit;
}
if (nSamples != static_cast<int32_t>(_playBlockSize)) {
@@ -3261,7 +3262,7 @@
if (_ptrCaptureClient == NULL || _ptrClientIn == NULL) {
_UnLock();
RTC_LOG(LS_ERROR) << "input state has been modified during"
- << " unlocked period";
+ " unlocked period";
goto Exit;
}
}
@@ -3282,7 +3283,7 @@
// IAudioClient::Stop, IAudioClient::Reset, and releasing the audio
// client.
RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer returned"
- << " AUDCLNT_E_BUFFER_ERROR, hr = 0x"
+ " AUDCLNT_E_BUFFER_ERROR, hr = 0x"
<< rtc::ToHex(hr);
goto Exit;
}
@@ -3815,14 +3816,16 @@
if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt)) {
hr = E_FAIL;
RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue returned no value,"
- << " hr = 0x" << rtc::ToHex(hr);
+ " hr = 0x"
+ << rtc::ToHex(hr);
}
if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt)) {
// The returned value is not a wide null terminated string.
hr = E_UNEXPECTED;
RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue returned unexpected"
- << " type, hr = 0x" << rtc::ToHex(hr);
+ " type, hr = 0x"
+ << rtc::ToHex(hr);
}
if (SUCCEEDED(hr) && (varName.pwszVal != NULL)) {
diff --git a/modules/audio_device/win/core_audio_utility_win.cc b/modules/audio_device/win/core_audio_utility_win.cc
index 29f73c2..a570bfe 100644
--- a/modules/audio_device/win/core_audio_utility_win.cc
+++ b/modules/audio_device/win/core_audio_utility_win.cc
@@ -284,7 +284,8 @@
EDataFlow data_flow,
ERole role) {
RTC_DLOG(INFO) << "CreateDeviceInternal: "
- << "id=" << device_id << ", flow=" << FlowToString(data_flow)
+ "id="
+ << device_id << ", flow=" << FlowToString(data_flow)
<< ", role=" << RoleToString(role);
ComPtr<IMMDevice> audio_endpoint_device;
@@ -967,7 +968,7 @@
// This API seems to be supported in off-load mode only but it is not
// documented as a valid error code. Making a special note about it here.
RTC_LOG(LS_ERROR) << "IAudioClient2::GetBufferSizeLimits failed: "
- << "AUDCLNT_E_OFFLOAD_MODE_ONLY";
+ "AUDCLNT_E_OFFLOAD_MODE_ONLY";
} else if (FAILED(error.Error())) {
RTC_LOG(LS_ERROR) << "IAudioClient2::GetBufferSizeLimits failed: "
<< ErrorToString(error);
diff --git a/modules/audio_mixer/audio_mixer_test.cc b/modules/audio_mixer/audio_mixer_test.cc
index 816d229..5bdc485 100644
--- a/modules/audio_mixer/audio_mixer_test.cc
+++ b/modules/audio_mixer/audio_mixer_test.cc
@@ -144,9 +144,14 @@
// Print stats.
std::cout << "Limiting is: " << (absl::GetFlag(FLAGS_limiter) ? "on" : "off")
<< "\n"
- << "Channels: " << num_channels << "\n"
- << "Rate: " << sample_rate << "\n"
- << "Number of input streams: " << input_files.size() << "\n";
+ "Channels: "
+ << num_channels
+ << "\n"
+ "Rate: "
+ << sample_rate
+ << "\n"
+ "Number of input streams: "
+ << input_files.size() << "\n";
for (const auto& source : sources) {
std::cout << "\t" << source.ToString() << "\n";
}
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index 8af7c2d..999e19b 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -244,9 +244,13 @@
}
stream_analog_level_ = new_level;
- RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
- << "level_=" << level_ << ", "
- << "new_level=" << new_level;
+ RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level
+ << ", "
+ "level_="
+ << level_
+ << ", "
+ "new_level="
+ << new_level;
level_ = new_level;
}
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
index af3619b..f24a76f 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -362,7 +362,7 @@
}
RTC_DCHECK_NE(candidate_pitch_period, candidate_pitch_secondary_period)
<< "The lower pitch period and the additional sub-harmonic must not "
- << "coincide.";
+ "coincide.";
// Compute an auto-correlation score for the primary pitch candidate
// |candidate_pitch_period| by also looking at its possible sub-harmonic
// |candidate_pitch_secondary_period|.
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn.cc b/modules/audio_processing/agc2/rnn_vad/rnn.cc
index 1cd8ae7..55a51ff 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn.cc
@@ -331,7 +331,7 @@
optimization_(optimization) {
RTC_DCHECK_LE(output_size_, kRecurrentLayersMaxUnits)
<< "Static over-allocation of recurrent layers state vectors is not "
- << "sufficient.";
+ "sufficient.";
RTC_DCHECK_EQ(kNumGruGates * output_size_, bias_.size())
<< "Mismatching output size and bias terms array size.";
RTC_DCHECK_EQ(kNumGruGates * input_size_ * output_size_, weights_.size())
@@ -339,7 +339,7 @@
RTC_DCHECK_EQ(kNumGruGates * output_size_ * output_size_,
recurrent_weights_.size())
<< "Mismatching input-output size and recurrent weight coefficients array"
- << " size.";
+ " size.";
Reset();
}
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 1c88581..a0d9dd1 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -352,7 +352,8 @@
EnforceSplitBandHpf()),
capture_nonlocked_() {
RTC_LOG(LS_INFO) << "Injected APM submodules:"
- << "\nEcho control factory: " << !!echo_control_factory_
+ "\nEcho control factory: "
+ << !!echo_control_factory_
<< "\nEcho detector: " << !!submodules_.echo_detector
<< "\nCapture analyzer: " << !!submodules_.capture_analyzer
<< "\nCapture post processor: "
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 8f9e535..3911f31 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2174,21 +2174,36 @@
size_t capture_output_num_channels) {
rtc::StringBuilder ss;
ss << "Sample rates:"
+ "\n"
+ " Render input: "
+ << render_input_sample_rate_hz
+ << " Hz"
+ "\n"
+ " Render output: "
+ << render_output_sample_rate_hz
+ << " Hz"
+ "\n"
+ " Capture input: "
+ << capture_input_sample_rate_hz
+ << " Hz"
+ "\n"
+ " Capture output: "
+ << capture_output_sample_rate_hz
+ << " Hz"
+ "\n"
+ "Number of channels:"
+ "\n"
+ " Render input: "
+ << render_input_num_channels
<< "\n"
- << " Render input: " << render_input_sample_rate_hz << " Hz"
+ " Render output: "
+ << render_output_num_channels
<< "\n"
- << " Render output: " << render_output_sample_rate_hz << " Hz"
+ " Capture input: "
+ << capture_input_num_channels
<< "\n"
- << " Capture input: " << capture_input_sample_rate_hz << " Hz"
- << "\n"
- << " Capture output: " << capture_output_sample_rate_hz << " Hz"
- << "\n"
- << "Number of channels:"
- << "\n"
- << " Render input: " << render_input_num_channels << "\n"
- << " Render output: " << render_output_num_channels << "\n"
- << " Capture input: " << capture_input_num_channels << "\n"
- << " Capture output: " << capture_output_num_channels;
+ " Capture output: "
+ << capture_output_num_channels;
return ss.Release();
}
diff --git a/modules/audio_processing/gain_controller2.cc b/modules/audio_processing/gain_controller2.cc
index 8c764f8..b15a266 100644
--- a/modules/audio_processing/gain_controller2.cc
+++ b/modules/audio_processing/gain_controller2.cc
@@ -107,15 +107,15 @@
// clang-format off
// clang formatting doesn't respect custom nested style.
ss << "{"
- << "enabled: " << (config.enabled ? "true" : "false") << ", "
- << "fixed_digital: {gain_db: " << config.fixed_digital.gain_db << "}, "
- << "adaptive_digital: {"
- << "enabled: "
- << (config.adaptive_digital.enabled ? "true" : "false") << ", "
- << "level_estimator: " << adaptive_digital_level_estimator << ", "
- << "extra_saturation_margin_db:"
- << config.adaptive_digital.extra_saturation_margin_db << "}"
- << "}";
+ "enabled: " << (config.enabled ? "true" : "false") << ", "
+ "fixed_digital: {gain_db: " << config.fixed_digital.gain_db << "}, "
+ "adaptive_digital: {"
+ "enabled: "
+ << (config.adaptive_digital.enabled ? "true" : "false") << ", "
+ "level_estimator: " << adaptive_digital_level_estimator << ", "
+ "extra_saturation_margin_db:"
+ << config.adaptive_digital.extra_saturation_margin_db << "}"
+ "}";
// clang-format on
return ss.Release();
}
diff --git a/modules/audio_processing/include/audio_processing.cc b/modules/audio_processing/include/audio_processing.cc
index 98ec590..30d025d 100644
--- a/modules/audio_processing/include/audio_processing.cc
+++ b/modules/audio_processing/include/audio_processing.cc
@@ -72,13 +72,16 @@
char buf[1024];
rtc::SimpleStringBuilder builder(buf);
builder << "AudioProcessing::Config{ "
- << "pipeline: {"
- << "maximum_internal_processing_rate: "
+ "pipeline: {"
+ "maximum_internal_processing_rate: "
<< pipeline.maximum_internal_processing_rate
- << ", multi_channel_render: " << pipeline.multi_channel_render << ", "
- << ", multi_channel_capture: " << pipeline.multi_channel_capture
+ << ", multi_channel_render: " << pipeline.multi_channel_render
+ << ", "
+ ", multi_channel_capture: "
+ << pipeline.multi_channel_capture
<< "}, "
- << "pre_amplifier: { enabled: " << pre_amplifier.enabled
+ "pre_amplifier: { enabled: "
+ << pre_amplifier.enabled
<< ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
<< " }, high_pass_filter: { enabled: " << high_pass_filter.enabled
<< " }, echo_canceller: { enabled: " << echo_canceller.enabled
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc b/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
index de084d3..d58b57e 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
@@ -76,9 +76,14 @@
// Write config to file.
std::ofstream out_config(config_output_file);
out_config << "{"
- << "'frame_len_ms': " << absl::GetFlag(FLAGS_f) << ", "
- << "'attack_ms': " << absl::GetFlag(FLAGS_a) << ", "
- << "'decay_ms': " << absl::GetFlag(FLAGS_d) << "}\n";
+ "'frame_len_ms': "
+ << absl::GetFlag(FLAGS_f)
+ << ", "
+ "'attack_ms': "
+ << absl::GetFlag(FLAGS_a)
+ << ", "
+ "'decay_ms': "
+ << absl::GetFlag(FLAGS_d) << "}\n";
out_config.close();
// Measure level frame-by-frame.
diff --git a/modules/audio_processing/transient/wpd_tree_unittest.cc b/modules/audio_processing/transient/wpd_tree_unittest.cc
index 11f75e6..97d69ae 100644
--- a/modules/audio_processing/transient/wpd_tree_unittest.cc
+++ b/modules/audio_processing/transient/wpd_tree_unittest.cc
@@ -145,7 +145,8 @@
ASSERT_EQ(kLeavesSamples, matlab_samples_read)
<< "Matlab test files are malformed.\n"
- << "File: 3_" << i;
+ "File: 3_"
+ << i;
// Get output data from the corresponding node
const float* node_data = tree.NodeAt(kLevels, i)->data();
// Compare with matlab files.
diff --git a/modules/congestion_controller/bbr/rtt_stats.cc b/modules/congestion_controller/bbr/rtt_stats.cc
index bbe5e42..2973463 100644
--- a/modules/congestion_controller/bbr/rtt_stats.cc
+++ b/modules/congestion_controller/bbr/rtt_stats.cc
@@ -49,7 +49,7 @@
Timestamp now) {
if (send_delta.IsInfinite() || send_delta <= TimeDelta::Zero()) {
RTC_LOG(LS_WARNING) << "Ignoring measured send_delta, because it's is "
- << "either infinite, zero, or negative. send_delta = "
+ "either infinite, zero, or negative. send_delta = "
<< ToString(send_delta);
return;
}
diff --git a/modules/congestion_controller/bbr/windowed_filter_unittest.cc b/modules/congestion_controller/bbr/windowed_filter_unittest.cc
index 61510d0..7ab4588 100644
--- a/modules/congestion_controller/bbr/windowed_filter_unittest.cc
+++ b/modules/congestion_controller/bbr/windowed_filter_unittest.cc
@@ -41,7 +41,8 @@
windowed_min_rtt_.Update(rtt_sample, now_ms);
RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << ToString(rtt_sample)
<< " mins: "
- << " " << ToString(windowed_min_rtt_.GetBest()) << " "
+ " "
+ << ToString(windowed_min_rtt_.GetBest()) << " "
<< ToString(windowed_min_rtt_.GetSecondBest()) << " "
<< ToString(windowed_min_rtt_.GetThirdBest());
now_ms += 25;
@@ -63,7 +64,8 @@
windowed_max_bw_.Update(bw_sample, now_ms);
RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << ToString(bw_sample)
<< " maxs: "
- << " " << ToString(windowed_max_bw_.GetBest()) << " "
+ " "
+ << ToString(windowed_max_bw_.GetBest()) << " "
<< ToString(windowed_max_bw_.GetSecondBest()) << " "
<< ToString(windowed_max_bw_.GetThirdBest());
now_ms += 25;
@@ -117,7 +119,8 @@
windowed_min_rtt_.Update(rtt_sample, now_ms);
RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << rtt_sample.ms()
<< " mins: "
- << " " << windowed_min_rtt_.GetBest().ms() << " "
+ " "
+ << windowed_min_rtt_.GetBest().ms() << " "
<< windowed_min_rtt_.GetSecondBest().ms() << " "
<< windowed_min_rtt_.GetThirdBest().ms();
if (i < 3) {
@@ -144,7 +147,8 @@
windowed_max_bw_.Update(bw_sample, now_ms);
RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << bw_sample.bps()
<< " maxs: "
- << " " << windowed_max_bw_.GetBest().bps() << " "
+ " "
+ << windowed_max_bw_.GetBest().bps() << " "
<< windowed_max_bw_.GetSecondBest().bps() << " "
<< windowed_max_bw_.GetThirdBest().bps();
if (i < 3) {
diff --git a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
index 0a636fc..b4a33eb 100644
--- a/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
+++ b/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
@@ -107,10 +107,12 @@
receive_interval <= TimeDelta::Zero() ||
receive_interval > kMaxProbeInterval) {
RTC_LOG(LS_INFO) << "Probing unsuccessful, invalid send/receive interval"
- << " [cluster id: " << cluster_id
- << "] [send interval: " << ToString(send_interval) << "]"
- << " [receive interval: " << ToString(receive_interval)
- << "]";
+ " [cluster id: "
+ << cluster_id
+ << "] [send interval: " << ToString(send_interval)
+ << "]"
+ " [receive interval: "
+ << ToString(receive_interval) << "]";
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
@@ -134,16 +136,20 @@
double ratio = receive_rate / send_rate;
if (ratio > kMaxValidRatio) {
RTC_LOG(LS_INFO) << "Probing unsuccessful, receive/send ratio too high"
- << " [cluster id: " << cluster_id
- << "] [send: " << ToString(send_size) << " / "
- << ToString(send_interval) << " = " << ToString(send_rate)
+ " [cluster id: "
+ << cluster_id << "] [send: " << ToString(send_size)
+ << " / " << ToString(send_interval) << " = "
+ << ToString(send_rate)
<< "]"
- << " [receive: " << ToString(receive_size) << " / "
+ " [receive: "
+ << ToString(receive_size) << " / "
<< ToString(receive_interval) << " = "
- << ToString(receive_rate) << " ]"
- << " [ratio: " << ToString(receive_rate) << " / "
- << ToString(send_rate) << " = " << ratio
- << " > kMaxValidRatio (" << kMaxValidRatio << ")]";
+ << ToString(receive_rate)
+ << " ]"
+ " [ratio: "
+ << ToString(receive_rate) << " / " << ToString(send_rate)
+ << " = " << ratio << " > kMaxValidRatio ("
+ << kMaxValidRatio << ")]";
if (event_log_) {
event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
@@ -151,11 +157,12 @@
return absl::nullopt;
}
RTC_LOG(LS_INFO) << "Probing successful"
- << " [cluster id: " << cluster_id
- << "] [send: " << ToString(send_size) << " / "
+ " [cluster id: "
+ << cluster_id << "] [send: " << ToString(send_size) << " / "
<< ToString(send_interval) << " = " << ToString(send_rate)
<< " ]"
- << " [receive: " << ToString(receive_size) << " / "
+ " [receive: "
+ << ToString(receive_size) << " / "
<< ToString(receive_interval) << " = "
<< ToString(receive_rate) << "]";
diff --git a/modules/congestion_controller/receive_side_congestion_controller.cc b/modules/congestion_controller/receive_side_congestion_controller.cc
index 628981f..7448ec2 100644
--- a/modules/congestion_controller/receive_side_congestion_controller.cc
+++ b/modules/congestion_controller/receive_side_congestion_controller.cc
@@ -99,7 +99,7 @@
if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
RTC_LOG(LS_INFO)
<< "WrappingBitrateEstimator: Switching to transmission "
- << "time offset RBE.";
+ "time offset RBE.";
using_absolute_send_time_ = false;
PickEstimator();
}
diff --git a/modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc b/modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc
index 9a2f5ff..e3668a5 100644
--- a/modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc
+++ b/modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc
@@ -46,7 +46,7 @@
if (!XQueryTree(display, window, &root, &parent, &children,
&num_children)) {
RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
- << "does not have a valid WM_STATE.";
+ "does not have a valid WM_STATE.";
return None;
}
if (children)
diff --git a/modules/desktop_capture/linux/window_list_utils.cc b/modules/desktop_capture/linux/window_list_utils.cc
index 4f05fc6..06660dd 100644
--- a/modules/desktop_capture/linux/window_list_utils.cc
+++ b/modules/desktop_capture/linux/window_list_utils.cc
@@ -61,7 +61,7 @@
if (!XQueryTree(cache->display(), window, &root, &parent, &children,
&num_children)) {
RTC_LOG(LS_ERROR) << "Failed to query for child windows although window"
- << "does not have a valid WM_STATE.";
+ "does not have a valid WM_STATE.";
return 0;
}
::Window app_window = 0;
diff --git a/modules/desktop_capture/mac/desktop_configuration_monitor.cc b/modules/desktop_capture/mac/desktop_configuration_monitor.cc
index cee8e70..e2225cd 100644
--- a/modules/desktop_capture/mac/desktop_configuration_monitor.cc
+++ b/modules/desktop_capture/mac/desktop_configuration_monitor.cc
@@ -54,8 +54,8 @@
CGDisplayChangeSummaryFlags flags) {
TRACE_EVENT0("webrtc", "DesktopConfigurationMonitor::DisplaysReconfigured");
RTC_LOG(LS_INFO) << "DisplaysReconfigured: "
- << "DisplayID " << display << "; ChangeSummaryFlags "
- << flags;
+ "DisplayID "
+ << display << "; ChangeSummaryFlags " << flags;
if (flags & kCGDisplayBeginConfigurationFlag) {
reconfiguring_displays_.insert(display);
diff --git a/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
index 8293ae5..1a7bbc1 100644
--- a/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
+++ b/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
@@ -230,14 +230,15 @@
!set_window_source_func_ || !set_window_filter_list_func_ ||
!set_image_scaling_callback_func_) {
RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
- << "library functions missing.";
+ "library functions missing.";
return false;
}
BOOL result = mag_initialize_func_();
if (!result) {
RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
- << "error from MagInitialize " << GetLastError();
+ "error from MagInitialize "
+ << GetLastError();
return false;
}
@@ -249,7 +250,8 @@
if (!result) {
mag_uninitialize_func_();
RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
- << "error from GetModulehandleExA " << GetLastError();
+ "error from GetModulehandleExA "
+ << GetLastError();
return false;
}
@@ -272,7 +274,7 @@
if (!host_window_) {
mag_uninitialize_func_();
RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
- << "error from creating host window "
+ "error from creating host window "
<< GetLastError();
return false;
}
@@ -284,7 +286,7 @@
if (!magnifier_window_) {
mag_uninitialize_func_();
RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
- << "error from creating magnifier window "
+ "error from creating magnifier window "
<< GetLastError();
return false;
}
@@ -299,7 +301,7 @@
if (!result) {
mag_uninitialize_func_();
RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
- << "error from MagSetImageScalingCallback "
+ "error from MagSetImageScalingCallback "
<< GetLastError();
return false;
}
@@ -311,7 +313,8 @@
mag_uninitialize_func_();
RTC_LOG_F(LS_WARNING)
<< "Failed to initialize ScreenCapturerWinMagnifier: "
- << "error from MagSetWindowFilterList " << GetLastError();
+ "error from MagSetWindowFilterList "
+ << GetLastError();
return false;
}
}
@@ -334,11 +337,19 @@
captured_bytes_per_pixel != DesktopFrame::kBytesPerPixel) {
RTC_LOG_F(LS_WARNING)
<< "Output format does not match the captured format: "
- << "width = " << header.width << ", "
- << "height = " << header.height << ", "
- << "stride = " << header.stride << ", "
- << "bpp = " << captured_bytes_per_pixel << ", "
- << "pixel format RGBA ? "
+ "width = "
+ << header.width
+ << ", "
+ "height = "
+ << header.height
+ << ", "
+ "stride = "
+ << header.stride
+ << ", "
+ "bpp = "
+ << captured_bytes_per_pixel
+ << ", "
+ "pixel format RGBA ? "
<< (header.format == GUID_WICPixelFormat32bppRGBA) << ".";
return;
}
diff --git a/modules/pacing/bitrate_prober.cc b/modules/pacing/bitrate_prober.cc
index 8dc89e4..719a602 100644
--- a/modules/pacing/bitrate_prober.cc
+++ b/modules/pacing/bitrate_prober.cc
@@ -132,8 +132,9 @@
if (next_probe_time_.IsFinite() &&
now - next_probe_time_ > config_.max_probe_delay.Get()) {
RTC_DLOG(LS_WARNING) << "Probe delay too high"
- << " (next_ms:" << next_probe_time_.ms()
- << ", now_ms: " << now.ms() << ")";
+ " (next_ms:"
+ << next_probe_time_.ms() << ", now_ms: " << now.ms()
+ << ")";
return Timestamp::PlusInfinity();
}
diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.cc b/modules/remote_bitrate_estimator/aimd_rate_control.cc
index 6c8e6eb..4d2e585 100644
--- a/modules/remote_bitrate_estimator/aimd_rate_control.cc
+++ b/modules/remote_bitrate_estimator/aimd_rate_control.cc
@@ -111,7 +111,8 @@
key_value_config->Lookup("WebRTC-BweAimdRateControlConfig"));
if (initial_backoff_interval_) {
RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval"
- << " " << ToString(*initial_backoff_interval_) << ".";
+ " "
+ << ToString(*initial_backoff_interval_) << ".";
}
RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
}
diff --git a/modules/rtp_rtcp/source/flexfec_sender.cc b/modules/rtp_rtcp/source/flexfec_sender.cc
index d35f4d6..70f1666 100644
--- a/modules/rtp_rtcp/source/flexfec_sender.cc
+++ b/modules/rtp_rtcp/source/flexfec_sender.cc
@@ -58,8 +58,8 @@
} else {
RTC_LOG(LS_INFO)
<< "FlexfecSender only supports RTP header extensions for "
- << "BWE and MID, so the extension " << extension.ToString()
- << " will not be used.";
+ "BWE and MID, so the extension "
+ << extension.ToString() << " will not be used.";
}
}
return map;
diff --git a/modules/rtp_rtcp/source/forward_error_correction.cc b/modules/rtp_rtcp/source/forward_error_correction.cc
index 120e11f..1812fbf 100644
--- a/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -131,7 +131,7 @@
if (media_packet->data.size() < kRtpHeaderSize) {
RTC_LOG(LS_WARNING) << "Media packet " << media_packet->data.size()
<< " bytes "
- << "is smaller than RTP header.";
+ "is smaller than RTP header.";
return -1;
}
// Ensure the FEC packets will fit in a typical MTU.
@@ -139,8 +139,8 @@
IP_PACKET_SIZE) {
RTC_LOG(LS_WARNING) << "Media packet " << media_packet->data.size()
<< " bytes "
- << "with overhead is larger than " << IP_PACKET_SIZE
- << " bytes.";
+ "with overhead is larger than "
+ << IP_PACKET_SIZE << " bytes.";
}
}
@@ -549,7 +549,7 @@
fec_packet.fec_header_size + fec_packet.protection_length) {
RTC_LOG(LS_WARNING)
<< "The FEC packet is truncated: it does not contain enough room "
- << "for its own header.";
+ "for its own header.";
return false;
}
if (fec_packet.protection_length >
@@ -590,7 +590,7 @@
ByteReader<uint16_t>::ReadBigEndian(&data[2]) + kRtpHeaderSize;
if (new_size > size_t{IP_PACKET_SIZE - kRtpHeaderSize}) {
RTC_LOG(LS_WARNING) << "The recovered packet had a length larger than a "
- << "typical IP packet, and is thus dropped.";
+ "typical IP packet, and is thus dropped.";
return false;
}
recovered_packet->pkt->data.SetSize(new_size);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 5f7735e..d1822f2 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -593,7 +593,7 @@
} else if (require_frame_encryption_) {
RTC_LOG(LS_WARNING)
<< "No FrameEncryptor is attached to this video sending stream but "
- << "one is required since require_frame_encryptor is set";
+ "one is required since require_frame_encryptor is set";
}
std::unique_ptr<RtpPacketizer> packetizer = RtpPacketizer::Create(
diff --git a/modules/rtp_rtcp/test/testFec/test_fec.cc b/modules/rtp_rtcp/test/testFec/test_fec.cc
index 505084f..db5ff15 100644
--- a/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ b/modules/rtp_rtcp/test/testFec/test_fec.cc
@@ -99,7 +99,7 @@
sizeof(kPacketMaskBurstyTbl) / sizeof(*kPacketMaskBurstyTbl)};
ASSERT_EQ(12, kMaxMediaPackets[1]) << "Max media packets for bursty mode not "
- << "equal to 12.";
+ "equal to 12.";
ForwardErrorCorrection::PacketList media_packet_list;
std::list<ForwardErrorCorrection::Packet*> fec_packet_list;
@@ -293,8 +293,10 @@
<< "EncodeFec() failed";
ASSERT_EQ(num_fec_packets, fec_packet_list.size())
- << "We requested " << num_fec_packets << " FEC packets, but "
- << "EncodeFec() produced " << fec_packet_list.size();
+ << "We requested " << num_fec_packets
+ << " FEC packets, but "
+ "EncodeFec() produced "
+ << fec_packet_list.size();
memset(media_loss_mask, 0, sizeof(media_loss_mask));
uint32_t media_packet_idx = 0;
@@ -419,12 +421,12 @@
ASSERT_EQ(recovered_packet->pkt->data.size(),
media_packet->data.size())
<< "Recovered packet length not identical to original "
- << "media packet";
+ "media packet";
ASSERT_EQ(0, memcmp(recovered_packet->pkt->data.cdata(),
media_packet->data.cdata(),
media_packet->data.size()))
<< "Recovered packet payload not identical to original "
- << "media packet";
+ "media packet";
recovered_packet_list.pop_front();
}
++media_packet_idx;
diff --git a/modules/utility/source/process_thread_impl.cc b/modules/utility/source/process_thread_impl.cc
index 506e8b6..67399371 100644
--- a/modules/utility/source/process_thread_impl.cc
+++ b/modules/utility/source/process_thread_impl.cc
@@ -124,8 +124,10 @@
rtc::CritScope lock(&lock_);
for (const ModuleCallback& mc : modules_) {
RTC_DCHECK(mc.module != module)
- << "Already registered here: " << mc.location.ToString() << "\n"
- << "Now attempting from here: " << from.ToString();
+ << "Already registered here: " << mc.location.ToString()
+ << "\n"
+ "Now attempting from here: "
+ << from.ToString();
}
}
#endif
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index 5018f52..a163579 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -74,7 +74,8 @@
//
RTC_LOG(LS_INFO) << __FUNCTION__
<< ": CoInitializeEx(NULL, COINIT_APARTMENTTHREADED)"
- << " => RPC_E_CHANGED_MODE, error 0x" << rtc::ToHex(hr);
+ " => RPC_E_CHANGED_MODE, error 0x"
+ << rtc::ToHex(hr);
}
}
}
@@ -163,7 +164,8 @@
deviceNameLength, NULL, NULL);
if (convResult == 0) {
RTC_LOG(LS_INFO) << "Failed to convert device name to UTF8, "
- << "error = " << GetLastError();
+ "error = "
+ << GetLastError();
return -1;
}
}
@@ -173,16 +175,16 @@
strncpy_s((char*)deviceUniqueIdUTF8, deviceUniqueIdUTF8Length,
(char*)deviceNameUTF8, convResult);
RTC_LOG(LS_INFO) << "Failed to get "
- << "deviceUniqueIdUTF8 using "
- << "deviceNameUTF8";
+ "deviceUniqueIdUTF8 using "
+ "deviceNameUTF8";
} else {
convResult = WideCharToMultiByte(
CP_UTF8, 0, varName.bstrVal, -1, (char*)deviceUniqueIdUTF8,
deviceUniqueIdUTF8Length, NULL, NULL);
if (convResult == 0) {
- RTC_LOG(LS_INFO)
- << "Failed to convert device "
- << "name to UTF8, error = " << GetLastError();
+ RTC_LOG(LS_INFO) << "Failed to convert device "
+ "name to UTF8, error = "
+ << GetLastError();
return -1;
}
if (productUniqueIdUTF8 && productUniqueIdUTF8Length > 0) {
@@ -261,7 +263,8 @@
if
FAILED(hr) {
RTC_LOG(LS_ERROR) << "Failed to bind to the selected "
- << "capture device " << hr;
+ "capture device "
+ << hr;
}
if (productUniqueIdUTF8 &&
@@ -334,7 +337,7 @@
(void**)&streamConfig);
if (FAILED(hr)) {
RTC_LOG(LS_INFO) << "Failed to get IID_IAMStreamConfig interface "
- << "from capture device";
+ "from capture device";
return -1;
}
diff --git a/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
index 24fd7a8..53fac77 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl.cc
@@ -378,7 +378,7 @@
if (!encoded_image_callback_) {
RTC_LOG(LS_WARNING)
<< "InitEncode() has been called, but a callback function "
- << "has not been set with RegisterEncodeCompleteCallback()";
+ "has not been set with RegisterEncodeCompleteCallback()";
ReportError();
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
diff --git a/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc b/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
index 4e47507..551ace2 100644
--- a/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
+++ b/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
@@ -72,9 +72,11 @@
allocated_buffers_.push_back(available_buffer);
if (allocated_buffers_.size() > max_num_buffers_) {
RTC_LOG(LS_WARNING)
- << allocated_buffers_.size() << " Vp9FrameBuffers have been "
- << "allocated by a Vp9FrameBufferPool (exceeding what is "
- << "considered reasonable, " << max_num_buffers_ << ").";
+ << allocated_buffers_.size()
+ << " Vp9FrameBuffers have been "
+ "allocated by a Vp9FrameBufferPool (exceeding what is "
+ "considered reasonable, "
+ << max_num_buffers_ << ").";
// TODO(phoglund): this limit is being hit in tests since Oct 5 2016.
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=6484.
diff --git a/modules/video_coding/codecs/vp9/vp9_impl.cc b/modules/video_coding/codecs/vp9/vp9_impl.cc
index 06b7fe1..df53cda 100644
--- a/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -1032,7 +1032,8 @@
if (rv != VPX_CODEC_OK) {
RTC_LOG(LS_ERROR) << "Encoding error: " << vpx_codec_err_to_string(rv)
<< "\n"
- << "Details: " << vpx_codec_error(encoder_) << "\n"
+ "Details: "
+ << vpx_codec_error(encoder_) << "\n"
<< vpx_codec_error_detail(encoder_);
return WEBRTC_VIDEO_CODEC_ERROR;
}
@@ -1608,8 +1609,9 @@
// The frame buffers are reference counted and frames are exposed after
// decoding. There may be valid usage cases where previous frames are still
// referenced after ~VP9DecoderImpl that is not a leak.
- RTC_LOG(LS_INFO) << num_buffers_in_use << " Vp9FrameBuffers are still "
- << "referenced during ~VP9DecoderImpl.";
+ RTC_LOG(LS_INFO) << num_buffers_in_use
+ << " Vp9FrameBuffers are still "
+ "referenced during ~VP9DecoderImpl.";
}
}
diff --git a/modules/video_coding/decoding_state.cc b/modules/video_coding/decoding_state.cc
index f769ed0..a951358 100644
--- a/modules/video_coding/decoding_state.cc
+++ b/modules/video_coding/decoding_state.cc
@@ -297,7 +297,7 @@
frame->CodecSpecific()->codecSpecific.VP9.flexible_mode;
if (is_flexible_mode && frame->PictureId() == kNoPictureId) {
RTC_LOG(LS_WARNING) << "Frame is marked as using flexible mode but no"
- << "picture id is set.";
+ "picture id is set.";
return false;
}
return is_flexible_mode;
diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc
index f76b957..1074215 100644
--- a/modules/video_coding/frame_buffer2.cc
+++ b/modules/video_coding/frame_buffer2.cc
@@ -350,7 +350,8 @@
int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
RTC_LOG(LS_WARNING)
<< "A frame about to be decoded is out of the configured "
- << "delay bounds (" << frame_delay << " > " << kMaxVideoDelayMs
+ "delay bounds ("
+ << frame_delay << " > " << kMaxVideoDelayMs
<< "). Resetting the video jitter buffer.";
return true;
}
@@ -482,14 +483,14 @@
<< id.picture_id << ":"
<< static_cast<int>(id.spatial_layer)
<< ") but buffer is full, clearing"
- << " buffer and inserting the frame.";
+ " buffer and inserting the frame.";
ClearFramesAndHistory();
} else {
RTC_LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) ("
<< id.picture_id << ":"
<< static_cast<int>(id.spatial_layer)
<< ") could not be inserted due to the frame "
- << "buffer being full, dropping frame.";
+ "buffer being full, dropping frame.";
return last_continuous_picture_id;
}
}
@@ -662,7 +663,7 @@
<< "Frame with (picture_id:spatial_id) (" << id.picture_id << ":"
<< static_cast<int>(id.spatial_layer)
<< ") depends on a non-decoded frame more previous than"
- << " the last decoded frame, dropping frame.";
+ " the last decoded frame, dropping frame.";
last_log_non_decoded_ms_ = now_ms;
}
return false;
diff --git a/modules/video_coding/packet_buffer.cc b/modules/video_coding/packet_buffer.cc
index fb25c0ad..0fbd042 100644
--- a/modules/video_coding/packet_buffer.cc
+++ b/modules/video_coding/packet_buffer.cc
@@ -357,8 +357,8 @@
if (has_h264_idr && (!has_h264_sps || !has_h264_pps)) {
RTC_LOG(LS_WARNING)
<< "Received H.264-IDR frame "
- << "(SPS: " << has_h264_sps << ", PPS: " << has_h264_pps
- << "). Treating as "
+ "(SPS: "
+ << has_h264_sps << ", PPS: " << has_h264_pps << "). Treating as "
<< (sps_pps_idr_is_h264_keyframe_ ? "delta" : "key")
<< " frame since WebRTC-SpsPpsIdrIsH264Keyframe is "
<< (sps_pps_idr_is_h264_keyframe_ ? "enabled." : "disabled");
diff --git a/modules/video_coding/receiver.cc b/modules/video_coding/receiver.cc
index 855ece8..2db4e21 100644
--- a/modules/video_coding/receiver.cc
+++ b/modules/video_coding/receiver.cc
@@ -114,7 +114,8 @@
int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
RTC_LOG(LS_WARNING)
<< "A frame about to be decoded is out of the configured "
- << "delay bounds (" << frame_delay << " > " << max_video_delay_ms_
+ "delay bounds ("
+ << frame_delay << " > " << max_video_delay_ms_
<< "). Resetting the video jitter buffer.";
timing_error = true;
} else if (static_cast<int>(timing_->TargetVideoDelay()) >
diff --git a/modules/video_coding/rtp_frame_reference_finder.cc b/modules/video_coding/rtp_frame_reference_finder.cc
index 4932c70..e09c95f 100644
--- a/modules/video_coding/rtp_frame_reference_finder.cc
+++ b/modules/video_coding/rtp_frame_reference_finder.cc
@@ -384,7 +384,7 @@
<< " and packet range [" << frame->first_seq_num()
<< ", " << frame->last_seq_num()
<< "] already received, "
- << " dropping frame.";
+ " dropping frame.";
return kDrop;
}
@@ -585,8 +585,9 @@
size_t temporal_idx = info.gof->temporal_idx[gof_idx];
if (temporal_idx >= kMaxTemporalLayers) {
- RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers << " temporal "
- << "layers are supported.";
+ RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers
+ << " temporal "
+ "layers are supported.";
return true;
}
@@ -628,8 +629,9 @@
size_t temporal_idx = info->gof->temporal_idx[gof_idx];
if (temporal_idx >= kMaxTemporalLayers) {
- RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers << " temporal "
- << "layers are supported.";
+ RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers
+ << " temporal "
+ "layers are supported.";
return;
}
@@ -646,8 +648,9 @@
size_t temporal_idx = info->gof->temporal_idx[gof_idx];
if (temporal_idx >= kMaxTemporalLayers) {
- RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers << " temporal "
- << "layers are supported.";
+ RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers
+ << " temporal "
+ "layers are supported.";
return;
}
@@ -783,7 +786,7 @@
<< " and packet range [" << frame->first_seq_num()
<< ", " << frame->last_seq_num()
<< "] already received, "
- << " dropping frame.";
+ " dropping frame.";
return kDrop;
}
diff --git a/p2p/base/connection.cc b/p2p/base/connection.cc
index e11b4bc..cd5d290 100644
--- a/p2p/base/connection.cc
+++ b/p2p/base/connection.cc
@@ -525,15 +525,15 @@
if (last_ping_sent_ + kMinExtraPingDelayMs <= now) {
RTC_LOG(LS_INFO) << ToString()
<< "WebRTC-ExtraICEPing/Sending extra ping"
- << " last_ping_sent_: " << last_ping_sent_
- << " now: " << now
+ " last_ping_sent_: "
+ << last_ping_sent_ << " now: " << now
<< " (diff: " << (now - last_ping_sent_) << ")";
Ping(now);
} else {
RTC_LOG(LS_INFO) << ToString()
<< "WebRTC-ExtraICEPing/Not sending extra ping"
- << " last_ping_sent_: " << last_ping_sent_
- << " now: " << now
+ " last_ping_sent_: "
+ << last_ping_sent_ << " now: " << now
<< " (diff: " << (now - last_ping_sent_) << ")";
}
}
diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc
index c7cfe5a..75490ee 100644
--- a/p2p/base/p2p_transport_channel.cc
+++ b/p2p/base/p2p_transport_channel.cc
@@ -1105,7 +1105,7 @@
RTC_DCHECK_RUN_ON(network_thread_);
if (!async_resolver_factory_) {
RTC_LOG(LS_WARNING) << "Dropping ICE candidate with hostname address "
- << "(no AsyncResolverFactory)";
+ "(no AsyncResolverFactory)";
return;
}
diff --git a/p2p/base/port_unittest.cc b/p2p/base/port_unittest.cc
index 9abeb3a..f203d48 100644
--- a/p2p/base/port_unittest.cc
+++ b/p2p/base/port_unittest.cc
@@ -2677,7 +2677,8 @@
trials.announce_goog_ping = GetParam().first;
trials.enable_goog_ping = GetParam().second;
RTC_LOG(LS_INFO) << "Testing combination: "
- << " announce: " << trials.announce_goog_ping
+ " announce: "
+ << trials.announce_goog_ping
<< " enable:" << trials.enable_goog_ping;
auto port1_unique =
diff --git a/p2p/base/turn_port.cc b/p2p/base/turn_port.cc
index ed82e35..2e8024d 100644
--- a/p2p/base/turn_port.cc
+++ b/p2p/base/turn_port.cc
@@ -1224,8 +1224,9 @@
if (webrtc::field_trial::IsEnabled("WebRTC-TurnAddMultiMapping")) {
if (entry->get_remote_ufrag() != remote_ufrag) {
- RTC_LOG(LS_INFO) << ToString() << ": remote ufrag updated."
- << " Sending new permission request";
+ RTC_LOG(LS_INFO) << ToString()
+ << ": remote ufrag updated."
+ " Sending new permission request";
entry->set_remote_ufrag(remote_ufrag);
entry->SendCreatePermissionRequest(0);
}
diff --git a/pc/channel.cc b/pc/channel.cc
index 285291f..d6f884c 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -423,7 +423,7 @@
// (and SetSend(true) is called).
RTC_LOG(LS_ERROR)
<< "Can't send outgoing RTP packet when SRTP is inactive"
- << " and crypto is required";
+ " and crypto is required";
RTC_NOTREACHED();
return false;
}
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 96b2ce8..664a830 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -1262,7 +1262,7 @@
RTC_DCHECK(false)
<< "PeerConnecton is initialized with use_datagram_transport = true "
"or use_datagram_transport_for_data_channels = true "
- << "but media transport factory is not set in PeerConnectionFactory";
+ "but media transport factory is not set in PeerConnectionFactory";
return false;
}
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index ed4ac5b..381d5bd 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -318,7 +318,8 @@
EXPECT_TRUE(valid_reference)
<< stats_->type() << "." << member.name()
<< " is not a reference to an "
- << "existing dictionary of type " << expected_type << " (value: "
+ "existing dictionary of type "
+ << expected_type << " (value: "
<< (member.is_defined() ? member.ValueToString() : "null") << ").";
MarkMemberTested(member, valid_reference);
}
diff --git a/pc/session_description.cc b/pc/session_description.cc
index 07ab7db..4881f4d 100644
--- a/pc/session_description.cc
+++ b/pc/session_description.cc
@@ -266,7 +266,7 @@
// If description_ is null, we assume that a move operator
// has been applied.
RTC_LOG(LS_ERROR) << "ContentInfo::description has been updated by "
- << "assignment. This usage is deprecated.";
+ "assignment. This usage is deprecated.";
description_.reset(description); // ensure that it is destroyed.
}
}
@@ -295,7 +295,7 @@
// Someone's updated |description|, or used a move operator
// on the record.
RTC_LOG(LS_ERROR) << "ContentInfo::description has been updated by "
- << "assignment. This usage is deprecated.";
+ "assignment. This usage is deprecated.";
const_cast<ContentInfo*>(this)->description_.reset(description);
}
return description_.get();
@@ -306,7 +306,7 @@
// Someone's updated |description|, or used a move operator
// on the record.
RTC_LOG(LS_ERROR) << "ContentInfo::description has been updated by "
- << "assignment. This usage is deprecated.";
+ "assignment. This usage is deprecated.";
description_.reset(description);
}
return description_.get();
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index c0e959a..575f339 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -1061,8 +1061,9 @@
attribute_candidate != kAttributeCandidate) {
if (is_raw) {
rtc::StringBuilder description;
- description << "Expect line: " << kAttributeCandidate << ":"
- << "<candidate-str>";
+ description << "Expect line: " << kAttributeCandidate
+ << ":"
+ "<candidate-str>";
return ParseFailed(first_line, 0, description.str(), error);
} else {
return ParseFailedExpectLine(first_line, 0, kLineTypeAttributes,
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index e8e937a..5bb4ffc 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -1963,18 +1963,22 @@
os << "minptime=" << params.min_ptime << "; stereo=" << params.stereo
<< "; sprop-stereo=" << params.sprop_stereo
<< "; useinbandfec=" << params.useinband
- << "; maxaveragebitrate=" << params.maxaveragebitrate << "\r\n"
- << "a=ptime:" << params.ptime << "\r\n"
- << "a=maxptime:" << params.max_ptime << "\r\n";
+ << "; maxaveragebitrate=" << params.maxaveragebitrate
+ << "\r\n"
+ "a=ptime:"
+ << params.ptime
+ << "\r\n"
+ "a=maxptime:"
+ << params.max_ptime << "\r\n";
sdp += os.str();
os.clear();
os.str("");
// Pl type 100 preferred.
os << "m=video 9 RTP/SAVPF 99 95\r\n"
- << "a=rtpmap:99 VP8/90000\r\n"
- << "a=rtpmap:95 RTX/90000\r\n"
- << "a=fmtp:95 apt=99;\r\n";
+ "a=rtpmap:99 VP8/90000\r\n"
+ "a=rtpmap:95 RTX/90000\r\n"
+ "a=fmtp:95 apt=99;\r\n";
sdp += os.str();
// Deserialize
@@ -2118,8 +2122,11 @@
}
EXPECT_EQ(0, position) << "Strings mismatch at the " << position
<< " character\n"
- << " 1: " << string1.substr(position, 20) << "\n"
- << " 2: " << string2.substr(position, 20) << "\n";
+ " 1: "
+ << string1.substr(position, 20)
+ << "\n"
+ " 2: "
+ << string2.substr(position, 20) << "\n";
}
TEST_F(WebRtcSdpTest, SerializeSessionDescription) {
diff --git a/rtc_base/logging_unittest.cc b/rtc_base/logging_unittest.cc
index 969ffeb..a66f8b5 100644
--- a/rtc_base/logging_unittest.cc
+++ b/rtc_base/logging_unittest.cc
@@ -339,8 +339,10 @@
stream.Close();
EXPECT_EQ(str.size(), (message.size() + logging_overhead) * kRepetitions);
- RTC_LOG(LS_INFO) << "Total log time: " << TimeDiff(finish, start) << " ms "
- << " total bytes logged: " << str.size();
+ RTC_LOG(LS_INFO) << "Total log time: " << TimeDiff(finish, start)
+ << " ms "
+ " total bytes logged: "
+ << str.size();
}
TEST(LogTest, EnumsAreSupported) {
diff --git a/rtc_base/network.cc b/rtc_base/network.cc
index 369c582..df3487f 100644
--- a/rtc_base/network.cc
+++ b/rtc_base/network.cc
@@ -773,7 +773,7 @@
if (!f) {
RTC_LOG(LS_WARNING)
<< "Couldn't read /proc/net/route, skipping default "
- << "route check (assuming everything is a default route).";
+ "route check (assuming everything is a default route).";
return true;
}
bool is_default_route = false;
diff --git a/rtc_base/network_unittest.cc b/rtc_base/network_unittest.cc
index 024115a..db97d07 100644
--- a/rtc_base/network_unittest.cc
+++ b/rtc_base/network_unittest.cc
@@ -930,7 +930,7 @@
return;
}
RTC_LOG(LS_INFO) << "Found dummy, running again while ignoring non-default "
- << "routes.";
+ "routes.";
manager.set_ignore_non_default_routes(true);
list = GetNetworks(manager, false);
for (NetworkManager::NetworkList::iterator it = list.begin();
diff --git a/rtc_base/openssl_adapter.cc b/rtc_base/openssl_adapter.cc
index 0036aae..07c2b81 100644
--- a/rtc_base/openssl_adapter.cc
+++ b/rtc_base/openssl_adapter.cc
@@ -857,8 +857,10 @@
if (ctx == nullptr) {
unsigned long error = ERR_get_error(); // NOLINT: type used by OpenSSL.
RTC_LOG(LS_WARNING) << "SSL_CTX creation failed: " << '"'
- << ERR_reason_error_string(error) << "\" "
- << "(error=" << error << ')';
+ << ERR_reason_error_string(error)
+ << "\" "
+ "(error="
+ << error << ')';
return nullptr;
}
@@ -906,7 +908,7 @@
for (const std::string& proto : alpn_protocols) {
if (proto.size() == 0 || proto.size() > 0xFF) {
RTC_LOG(LS_ERROR) << "OpenSSLAdapter::Error("
- << "TransformAlpnProtocols received proto with size "
+ "TransformAlpnProtocols received proto with size "
<< proto.size() << ")";
return "";
}
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index ead4e57..bd6a324 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -1265,8 +1265,8 @@
if (!pending_add_dispatchers_.erase(pdispatcher) &&
dispatchers_.find(pdispatcher) == dispatchers_.end()) {
RTC_LOG(LS_WARNING) << "PhysicalSocketServer asked to remove a unknown "
- << "dispatcher, potentially from a duplicate call to "
- << "Add.";
+ "dispatcher, potentially from a duplicate call to "
+ "Add.";
return;
}
@@ -1274,7 +1274,7 @@
} else if (!dispatchers_.erase(pdispatcher)) {
RTC_LOG(LS_WARNING)
<< "PhysicalSocketServer asked to remove a unknown "
- << "dispatcher, potentially from a duplicate call to Add.";
+ "dispatcher, potentially from a duplicate call to Add.";
return;
}
#if defined(WEBRTC_USE_EPOLL)
diff --git a/rtc_base/socket_unittest.cc b/rtc_base/socket_unittest.cc
index 80b28bb..2af3a8e 100644
--- a/rtc_base/socket_unittest.cc
+++ b/rtc_base/socket_unittest.cc
@@ -391,7 +391,7 @@
dns_lookup_finished);
if (!dns_lookup_finished) {
RTC_LOG(LS_WARNING) << "Skipping test; DNS resolution took longer than 5 "
- << "seconds.";
+ "seconds.";
return;
}
diff --git a/rtc_base/thread.cc b/rtc_base/thread.cc
index ba5b617..6c5830f 100644
--- a/rtc_base/thread.cc
+++ b/rtc_base/thread.cc
@@ -766,7 +766,7 @@
RTC_DCHECK(!IsCurrent());
if (Current() && !Current()->blocking_calls_allowed_) {
RTC_LOG(LS_WARNING) << "Waiting for the thread to join, "
- << "but blocking calls have been disallowed";
+ "but blocking calls have been disallowed";
}
#if defined(WEBRTC_WIN)
diff --git a/rtc_base/timestamp_aligner.cc b/rtc_base/timestamp_aligner.cc
index 2896f9c..b797420 100644
--- a/rtc_base/timestamp_aligner.cc
+++ b/rtc_base/timestamp_aligner.cc
@@ -122,8 +122,8 @@
// duplicate timestamps in case this function is called several times with
// exactly the same |system_time_us|.
RTC_LOG(LS_WARNING) << "too short translated timestamp interval: "
- << "system time (us) = " << system_time_us
- << ", interval (us) = "
+ "system time (us) = "
+ << system_time_us << ", interval (us) = "
<< system_time_us - prev_translated_time_us_;
time_us = system_time_us;
}
diff --git a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
index b2672e9..54613f9 100644
--- a/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
+++ b/sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
@@ -67,7 +67,7 @@
auto factory = CreateModularPeerConnectionFactory(std::move(pcf_deps));
RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << factory;
RTC_CHECK(factory) << "Failed to create the peer connection factory; "
- << "WebRTC/libjingle init likely failed on this device";
+ "WebRTC/libjingle init likely failed on this device";
return factory;
}
diff --git a/sdk/android/src/jni/android_media_decoder.cc b/sdk/android/src/jni/android_media_decoder.cc
index f61db2a..94ce42d 100644
--- a/sdk/android/src/jni/android_media_decoder.cc
+++ b/sdk/android/src/jni/android_media_decoder.cc
@@ -712,8 +712,10 @@
(current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms;
ALOGD << "Frames decoded: " << frames_decoded_
<< ". Received: " << frames_received_
- << ". Bitrate: " << current_bitrate << " kbps"
- << ". Fps: " << current_fps
+ << ". Bitrate: " << current_bitrate
+ << " kbps"
+ ". Fps: "
+ << current_fps
<< ". DecTime: " << (current_decoding_time_ms_ / current_frames_)
<< ". DelayTime: " << (current_delay_time_ms_ / current_frames_)
<< " for last " << statistic_time_ms << " ms.";
diff --git a/sdk/android/src/jni/android_media_encoder.cc b/sdk/android/src/jni/android_media_encoder.cc
index 78f313a..4b4ad10 100644
--- a/sdk/android/src/jni/android_media_encoder.cc
+++ b/sdk/android/src/jni/android_media_encoder.cc
@@ -649,7 +649,8 @@
if (input_frame_infos_.size() > MAX_ENCODER_Q_SIZE) {
ALOGD << "Already " << input_frame_infos_.size()
<< " frames in the queue, dropping"
- << ". TS: " << static_cast<int>(current_timestamp_us_ / 1000)
+ ". TS: "
+ << static_cast<int>(current_timestamp_us_ / 1000)
<< ". Fps: " << last_set_fps_
<< ". Consecutive drops: " << consecutive_full_queue_frame_drops_;
current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
@@ -1134,8 +1135,10 @@
(current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms;
ALOGD << "Encoded frames: " << frames_encoded_
<< ". Bitrate: " << current_bitrate
- << ", target: " << last_set_bitrate_kbps_ << " kbps"
- << ", fps: " << current_fps << ", encTime: "
+ << ", target: " << last_set_bitrate_kbps_
+ << " kbps"
+ ", fps: "
+ << current_fps << ", encTime: "
<< (current_encoding_time_ms_ / current_frames_divider)
<< ". QP: " << (current_acc_qp_ / current_frames_divider)
<< " for last " << statistic_time_ms << " ms.";
diff --git a/sdk/android/src/jni/android_network_monitor.cc b/sdk/android/src/jni/android_network_monitor.cc
index 993bbb5..d29be44 100644
--- a/sdk/android/src/jni/android_network_monitor.cc
+++ b/sdk/android/src/jni/android_network_monitor.cc
@@ -248,7 +248,8 @@
if (!network_binding_supported) {
RTC_LOG(LS_WARNING)
<< "BindSocketToNetwork is not supported on this platform "
- << "(Android SDK: " << android_sdk_int_ << ")";
+ "(Android SDK: "
+ << android_sdk_int_ << ")";
return rtc::NetworkBindingResult::NOT_IMPLEMENTED;
}
diff --git a/sdk/android/src/jni/audio_device/aaudio_player.cc b/sdk/android/src/jni/audio_device/aaudio_player.cc
index 8e1122d..4e1c7e33 100644
--- a/sdk/android/src/jni/audio_device/aaudio_player.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_player.cc
@@ -174,7 +174,8 @@
// utilized.
if (first_data_callback_) {
RTC_LOG(INFO) << "--- First output data callback: "
- << "device id=" << aaudio_.device_id();
+ "device id="
+ << aaudio_.device_id();
first_data_callback_ = false;
}
diff --git a/sdk/android/src/jni/audio_device/aaudio_recorder.cc b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
index b543645..65bef4b 100644
--- a/sdk/android/src/jni/audio_device/aaudio_recorder.cc
+++ b/sdk/android/src/jni/audio_device/aaudio_recorder.cc
@@ -173,7 +173,8 @@
// is obtained.
if (first_data_callback_) {
RTC_LOG(INFO) << "--- First input data callback: "
- << "device id=" << aaudio_.device_id();
+ "device id="
+ << aaudio_.device_id();
aaudio_.ClearInputStream(audio_data, num_frames);
first_data_callback_ = false;
}
diff --git a/sdk/android/src/jni/pc/peer_connection_factory.cc b/sdk/android/src/jni/pc/peer_connection_factory.cc
index 5b6efe8..48dd6e4 100644
--- a/sdk/android/src/jni/pc/peer_connection_factory.cc
+++ b/sdk/android/src/jni/pc/peer_connection_factory.cc
@@ -330,7 +330,7 @@
CreateModularPeerConnectionFactory(std::move(dependencies));
RTC_CHECK(factory) << "Failed to create the peer connection factory; "
- << "WebRTC/libjingle init likely failed on this device";
+ "WebRTC/libjingle init likely failed on this device";
// TODO(honghaiz): Maybe put the options as the argument of
// CreatePeerConnectionFactory.
if (options)
diff --git a/stats/rtc_stats.cc b/stats/rtc_stats.cc
index d0a8653..b8e9633 100644
--- a/stats/rtc_stats.cc
+++ b/stats/rtc_stats.cc
@@ -99,9 +99,13 @@
std::string RTCStats::ToJson() const {
rtc::StringBuilder sb;
- sb << "{\"type\":\"" << type() << "\","
- << "\"id\":\"" << id_ << "\","
- << "\"timestamp\":" << timestamp_us_;
+ sb << "{\"type\":\"" << type()
+ << "\","
+ "\"id\":\""
+ << id_
+ << "\","
+ "\"timestamp\":"
+ << timestamp_us_;
for (const RTCStatsMemberInterface* member : Members()) {
if (member->is_defined()) {
sb << ",\"" << member->name() << "\":";
diff --git a/test/call_test.cc b/test/call_test.cc
index 38c5d5b..a230e025 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -79,8 +79,8 @@
<< "URI " << extension.uri
<< (extension.encrypt ? " with " : " without ")
<< "encryption already registered with a different "
- << "ID (" << extension.id << " vs. " << registered_extension.id
- << ").";
+ "ID ("
+ << extension.id << " vs. " << registered_extension.id << ").";
}
}
rtp_extensions_.push_back(extension);
diff --git a/test/network/fake_network_socket_server.cc b/test/network/fake_network_socket_server.cc
index 3e9c0ef..60dfbe3 100644
--- a/test/network/fake_network_socket_server.cc
+++ b/test/network/fake_network_socket_server.cc
@@ -202,8 +202,8 @@
// but we won't to skip such error, so we will assert here.
RTC_CHECK(data_read == pending_->size())
<< "Too small buffer is provided for socket read. "
- << "Received data size: " << pending_->size()
- << "; Provided buffer size: " << cb;
+ "Received data size: "
+ << pending_->size() << "; Provided buffer size: " << cb;
pending_.reset();
diff --git a/test/testsupport/file_utils_unittest.cc b/test/testsupport/file_utils_unittest.cc
index c62bb7a..7b23cbe 100644
--- a/test/testsupport/file_utils_unittest.cc
+++ b/test/testsupport/file_utils_unittest.cc
@@ -173,8 +173,9 @@
#endif
ASSERT_THAT(result, EndsWith(expected_end));
- ASSERT_TRUE(FileExists(result)) << "Expected " << result << " to exist; did "
- << "ResourcePath return an incorrect path?";
+ ASSERT_TRUE(FileExists(result)) << "Expected " << result
+ << " to exist; did "
+ "ResourcePath return an incorrect path?";
}
TEST_F(FileUtilsTest, ResourcePathFromRootWorkingDir) {
diff --git a/video/overuse_frame_detector.cc b/video/overuse_frame_detector.cc
index 429dbc4..ade9303 100644
--- a/video/overuse_frame_detector.cc
+++ b/video/overuse_frame_detector.cc
@@ -677,9 +677,10 @@
in_quick_rampup_ ? kQuickRampUpDelayMs : current_rampup_delay_ms_;
RTC_LOG(LS_VERBOSE) << " Frame stats: "
- << " encode usage " << *encode_usage_percent_
- << " overuse detections " << num_overuse_detections_
- << " rampup delay " << rampup_delay;
+ " encode usage "
+ << *encode_usage_percent_ << " overuse detections "
+ << num_overuse_detections_ << " rampup delay "
+ << rampup_delay;
}
void OveruseFrameDetector::SetOptions(const CpuOveruseOptions& options) {
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index ae3475b..495d2dc 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -927,7 +927,8 @@
return;
RTC_LOG(LS_INFO) << "Found out of band supplied codec parameters for"
- << " payload type: " << static_cast<int>(payload_type);
+ " payload type: "
+ << static_cast<int>(payload_type);
H264SpropParameterSets sprop_decoder;
auto sprop_base64_it =
diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
index d1a01d0..3ddbf58 100644
--- a/video/video_stream_encoder.cc
+++ b/video/video_stream_encoder.cc
@@ -1172,7 +1172,7 @@
if (frame_dropping_enabled && frame_dropper_.DropFrame()) {
RTC_LOG(LS_VERBOSE)
<< "Drop Frame: "
- << "target bitrate "
+ "target bitrate "
<< (last_encoder_rate_settings_
? last_encoder_rate_settings_->encoder_target.bps()
: 0)
@@ -1901,8 +1901,8 @@
rtc::StringBuilder ss;
ss << "Successfully parsed WebRTC-NetworkCondition-EncoderSwitch field "
"trial."
- << " to_codec:" << result.to_codec
- << " to_param:" << result.to_param.value_or("<none>")
+ " to_codec:"
+ << result.to_codec << " to_param:" << result.to_param.value_or("<none>")
<< " to_value:" << result.to_value.value_or("<none>")
<< " codec_thresholds:";
@@ -1935,7 +1935,8 @@
}
RTC_LOG(LS_INFO) << "Automatic animation detection experiment settings:"
- << " min_duration_ms=" << result.min_duration_ms
+ " min_duration_ms="
+ << result.min_duration_ms
<< " min_area_ration=" << result.min_area_ratio
<< " min_fps=" << result.min_fps;
