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Concatenate string literals at compile time.

Browse source 
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}
This commit is contained in:
Jonas Olsson 2020-01-14 12:11:31 +01:00 committed by Commit Bot
parent 6153e15d31
commit b2b2031457
98 changed files with 544 additions and 353 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

17
api/video_codecs/video_encoder.cc
View file

@ -112,18 +112,21 @@ std::string VideoEncoder::EncoderInfo::ToString() const {
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 @@ std::string VideoEncoder::EncoderInfo::ToString() const {
}
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();
}

2
audio/channel_receive.cc
View file

@ -834,7 +834,7 @@ void ChannelReceive::UpdatePlayoutTimestamp(bool rtcp, int64_t now_ms) {
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;
}

2
audio/channel_send.cc
View file

@ -420,7 +420,7 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
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;
}
}

6
call/flexfec_receive_stream_impl.cc
View file

@ -87,7 +87,7 @@ std::unique_ptr<FlexfecReceiver> MaybeCreateFlexfecReceiver(
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 @@ std::unique_ptr<FlexfecReceiver> MaybeCreateFlexfecReceiver(
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;
}

3
call/rtp_stream_receiver_controller.cc
View file

@ -25,7 +25,8 @@ RtpStreamReceiverController::Receiver::Receiver(
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 << ".";
}
}

4
call/rtp_transport_controller_send.cc
View file

@ -390,7 +390,7 @@ void RtpTransportControllerSend::SetSdpBitrateParameters(
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetSdpBitrateParameters: "
<< "nothing to update";
"nothing to update";
}
}
@ -411,7 +411,7 @@ void RtpTransportControllerSend::SetClientBitratePreferences(
} else {
RTC_LOG(LS_VERBOSE)
<< "WebRTC.RtpTransportControllerSend.SetClientBitratePreferences: "
<< "nothing to update";
"nothing to update";
}
}

7
call/simulated_network.cc
View file

@ -100,9 +100,10 @@ void SimulatedNetwork::SetConfig(const Config& config) {
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 =

2
common_video/bitrate_adjuster.cc
View file

@ -140,7 +140,7 @@ void BitrateAdjuster::UpdateBitrate(uint32_t current_time_ms) {
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)

6
examples/peerconnection/client/conductor.cc
View file

@ -345,7 +345,8 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
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 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
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())) {

6
examples/unityplugin/simple_peer_connection.cc
View file

@ -342,7 +342,8 @@ bool SimplePeerConnection::SetRemoteDescription(const char* type,
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 @@ bool SimplePeerConnection::AddIceCandidate(const char* candidate,
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())) {

3
logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
View file

@ -674,7 +674,8 @@ std::string RtcEventLogEncoderLegacy::EncodeVideoSendStreamConfig(
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;
}
}

6
logging/rtc_event_log/rtc_event_log2rtp_dump.cc
View file

@ -249,7 +249,9 @@ int main(int argc, char* argv[]) {
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;
}

3
logging/rtc_event_log/rtc_event_log_impl.cc
View file

@ -93,7 +93,8 @@ bool RtcEventLogImpl::StartLogging(std::unique_ptr<RtcEventLogOutput> output,
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;

4
media/base/rtp_data_engine.cc
View file

@ -319,8 +319,8 @@ bool RtpDataMediaChannel::SendData(const SendDataParams& params,
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();

16
media/engine/webrtc_video_engine.cc
View file

@ -735,7 +735,7 @@ void WebRtcVideoChannel::RequestEncoderSwitch(
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 @@ webrtc::RtpParameters WebRtcVideoChannel::GetRtpSendParameters(
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 @@ webrtc::RTCError WebRtcVideoChannel::SetRtpSendParameters(
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::RTCError WebRtcVideoChannel::SetRtpSendParameters(
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 @@ webrtc::RtpParameters WebRtcVideoChannel::GetRtpReceiveParameters(
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 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetFrameDecryptor(
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);
}
}

20
media/engine/webrtc_voice_engine.cc
View file

@ -170,9 +170,10 @@ absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
// 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 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
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 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters(
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 @@ webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
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::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
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 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
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();

135
media/sctp/sctp_transport.cc
View file

@ -284,7 +284,8 @@ class SctpTransport::UsrSctpWrapper {
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::Start(int local_sctp_port,
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 @@ bool SctpTransport::OpenStream(int sid) {
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 @@ bool SctpTransport::ResetStream(int sid) {
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 @@ bool SctpTransport::SendData(const SendDataParams& params,
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 @@ SendDataResult SctpTransport::SendMessageInternal(OutgoingMessage* message) {
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 @@ SendDataResult SctpTransport::SendMessageInternal(OutgoingMessage* message) {
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 @@ bool SctpTransport::Connect() {
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 @@ bool SctpTransport::Connect() {
params.spp_pathmtu = kSctpMtu - sizeof(struct sctp_common_header);
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, &params,
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::Connect() {
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 @@ bool SctpTransport::OpenSctpSocket() {
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 @@ bool SctpTransport::ConfigureSctpSocket() {
// 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 @@ bool SctpTransport::ConfigureSctpSocket() {
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 @@ bool SctpTransport::ConfigureSctpSocket() {
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 @@ bool SctpTransport::ConfigureSctpSocket() {
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 @@ bool SctpTransport::ConfigureSctpSocket() {
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 @@ bool SctpTransport::ConfigureSctpSocket() {
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 @@ void SctpTransport::OnPacketRead(rtc::PacketTransportInternal* transport,
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 @@ void SctpTransport::OnPacketFromSctpToNetwork(
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 @@ void SctpTransport::OnInboundPacketFromSctpToTransport(
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 @@ void SctpTransport::OnDataFromSctpToTransport(
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 @@ void SctpTransport::OnNotificationFromSctp(
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)

3
modules/audio_coding/neteq/decision_logic.cc
View file

@ -53,7 +53,8 @@ DecisionLogic::DecisionLogic(NetEqController::Config config)
{&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_;
}

3
modules/audio_coding/neteq/delay_manager.cc
View file

@ -71,7 +71,8 @@ DelayHistogramConfig GetDelayHistogramConfig() {
}
}
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);

5
modules/audio_coding/neteq/neteq_impl.cc
View file

@ -141,8 +141,9 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
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);

25
modules/audio_coding/neteq/tools/neteq_input.cc
View file

@ -21,13 +21,24 @@ NetEqInput::PacketData::~PacketData() = default;
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();
}

10
modules/audio_coding/neteq/tools/neteq_rtpplay.cc
View file

@ -242,8 +242,9 @@ bool ValidateOutputFilesOptions(bool textlog,
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 @@ bool ValidateOutputFilesOptions(bool textlog,
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;

3
modules/audio_device/android/aaudio_player.cc
View file

@ -158,7 +158,8 @@ aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
// 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;
}

3
modules/audio_device/android/aaudio_recorder.cc
View file

@ -161,7 +161,8 @@ aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
// 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;
}

3
modules/audio_device/android/audio_manager.cc
View file

@ -275,7 +275,8 @@ void AudioManager::OnCacheAudioParameters(JNIEnv* env,
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)

36
modules/audio_device/audio_device_buffer.cc
View file

@ -413,11 +413,19 @@ void AudioDeviceBuffer::LogStats(LogState state) {
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 @@ void AudioDeviceBuffer::LogStats(LogState state) {
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;

4
modules/audio_device/dummy/file_audio_device_factory.cc
View file

@ -29,8 +29,8 @@ FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice() {
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;
}

6
modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc
View file

@ -181,7 +181,8 @@ int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) {
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 @@ int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) {
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;

3
modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc
View file

@ -455,7 +455,8 @@ int32_t AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(
RTC_LOG(LS_VERBOSE)
<< "AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
<< " => available=" << available;
" => available="
<< available;
return 0;
}

8
modules/audio_device/mac/audio_device_mac.cc
View file

@ -1034,7 +1034,7 @@ int32_t AudioDeviceMac::InitPlayout() {
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 @@ int32_t AudioDeviceMac::StopRecording() {
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 @@ int32_t AudioDeviceMac::StopRecording() {
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 @@ int32_t AudioDeviceMac::StopPlayout() {
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.

49
modules/audio_device/win/audio_device_core_win.cc
View file

@ -265,10 +265,10 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported() {
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 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported() {
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 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
// 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 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
_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 @@ AudioDeviceWindowsCore::~AudioDeviceWindowsCore() {
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 @@ int32_t AudioDeviceWindowsCore::InitSpeaker() {
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 @@ int32_t AudioDeviceWindowsCore::InitMicrophone() {
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 @@ int32_t AudioDeviceWindowsCore::InitPlayout() {
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 @@ int32_t AudioDeviceWindowsCore::InitRecording() {
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 @@ int32_t AudioDeviceWindowsCore::StartRecording() {
// 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 @@ int32_t AudioDeviceWindowsCore::StopPlayout() {
// 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 @@ DWORD AudioDeviceWindowsCore::DoRenderThread() {
_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 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread() {
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 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread() {
// 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 @@ int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
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)) {

5
modules/audio_device/win/core_audio_utility_win.cc
View file

@ -284,7 +284,8 @@ ComPtr<IMMDevice> CreateDeviceInternal(const std::string& device_id,
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 @@ HRESULT GetBufferSizeLimits(IAudioClient2* client,
// 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);

11
modules/audio_mixer/audio_mixer_test.cc
View file

@ -144,9 +144,14 @@ int main(int argc, char* argv[]) {
// 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";
}

10
modules/audio_processing/agc/agc_manager_direct.cc
View file

@ -244,9 +244,13 @@ void MonoAgc::SetLevel(int new_level) {
}
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;
}

2
modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
View file

@ -362,7 +362,7 @@ PitchInfo CheckLowerPitchPeriodsAndComputePitchGain(
}
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|.

4
modules/audio_processing/agc2/rnn_vad/rnn.cc
View file

@ -331,7 +331,7 @@ GatedRecurrentLayer::GatedRecurrentLayer(
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 @@ GatedRecurrentLayer::GatedRecurrentLayer(
RTC_DCHECK_EQ(kNumGruGates * output_size_ * output_size_,
recurrent_weights_.size())
<< "Mismatching input-output size and recurrent weight coefficients array"
<< " size.";
" size.";
Reset();
}

3
modules/audio_processing/audio_processing_impl.cc
View file

@ -352,7 +352,8 @@ AudioProcessingImpl::AudioProcessingImpl(
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: "

39
modules/audio_processing/audio_processing_unittest.cc
View file

@ -2174,21 +2174,36 @@ std::string ProduceDebugText(int render_input_sample_rate_hz,
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();
}

18
modules/audio_processing/gain_controller2.cc
View file

@ -107,15 +107,15 @@ std::string GainController2::ToString(
// 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();
}

13
modules/audio_processing/include/audio_processing.cc
View file

@ -72,13 +72,16 @@ std::string AudioProcessing::Config::ToString() const {
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

11
modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
View file

@ -76,9 +76,14 @@ int main(int argc, char* argv[]) {
// 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.

3
modules/audio_processing/transient/wpd_tree_unittest.cc
View file

@ -145,7 +145,8 @@ TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
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.

2
modules/congestion_controller/bbr/rtt_stats.cc
View file

@ -49,7 +49,7 @@ void RttStats::UpdateRtt(TimeDelta send_delta,
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;
}

12
modules/congestion_controller/bbr/windowed_filter_unittest.cc
View file

@ -41,7 +41,8 @@ class WindowedFilterTest : public ::testing::Test {
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 @@ class WindowedFilterTest : public ::testing::Test {
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 @@ TEST_F(WindowedFilterTest, MonotonicallyIncreasingMin) {
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 @@ TEST_F(WindowedFilterTest, MonotonicallyDecreasingMax) {
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) {

37
modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
View file

@ -107,10 +107,12 @@ absl::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
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 @@ absl::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
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 @@ absl::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
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) << "]";

2
modules/congestion_controller/receive_side_congestion_controller.cc
View file

@ -99,7 +99,7 @@ void ReceiveSideCongestionController::WrappingBitrateEstimator::
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();
}

2
modules/desktop_capture/linux/mouse_cursor_monitor_x11.cc
View file

@ -46,7 +46,7 @@ Window GetTopLevelWindow(Display* display, Window window) {
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)

2
modules/desktop_capture/linux/window_list_utils.cc
View file

@ -61,7 +61,7 @@ DeferXFree::~DeferXFree() {
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;

4
modules/desktop_capture/mac/desktop_configuration_monitor.cc
View file

@ -54,8 +54,8 @@ void DesktopConfigurationMonitor::DisplaysReconfigured(
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);

35
modules/desktop_capture/win/screen_capturer_win_magnifier.cc
View file

@ -230,14 +230,15 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
!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 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
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 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
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 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
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 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
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 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
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 @@ void ScreenCapturerWinMagnifier::OnCaptured(void* data,
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;
}

5
modules/pacing/bitrate_prober.cc
View file

@ -132,8 +132,9 @@ Timestamp BitrateProber::NextProbeTime(Timestamp now) const {
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();
}

3
modules/remote_bitrate_estimator/aimd_rate_control.cc
View file

@ -111,7 +111,8 @@ AimdRateControl::AimdRateControl(const WebRtcKeyValueConfig* key_value_config,
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_;
}

4
modules/rtp_rtcp/source/flexfec_sender.cc
View file

@ -58,8 +58,8 @@ RtpHeaderExtensionMap RegisterSupportedExtensions(
} 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;

10
modules/rtp_rtcp/source/forward_error_correction.cc
View file

@ -131,7 +131,7 @@ int ForwardErrorCorrection::EncodeFec(const PacketList& media_packets,
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 @@ int ForwardErrorCorrection::EncodeFec(const PacketList& media_packets,
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 @@ bool ForwardErrorCorrection::StartPacketRecovery(
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 @@ bool ForwardErrorCorrection::FinishPacketRecovery(
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);

2
modules/rtp_rtcp/source/rtp_sender_video.cc
View file

@ -593,7 +593,7 @@ bool RTPSenderVideo::SendVideo(
} 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(

12
modules/rtp_rtcp/test/testFec/test_fec.cc
View file

@ -99,7 +99,7 @@ void RunTest(bool use_flexfec) {
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 @@ void RunTest(bool use_flexfec) {
<< "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 @@ void RunTest(bool use_flexfec) {
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;

6
modules/utility/source/process_thread_impl.cc
View file

@ -124,8 +124,10 @@ void ProcessThreadImpl::RegisterModule(Module* module,
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

21
modules/video_capture/windows/device_info_ds.cc
View file

@ -74,7 +74,8 @@ DeviceInfoDS::DeviceInfoDS()
//
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 @@ int32_t DeviceInfoDS::GetDeviceInfo(uint32_t deviceNumber,
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 @@ int32_t DeviceInfoDS::GetDeviceInfo(uint32_t deviceNumber,
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 @@ IBaseFilter* DeviceInfoDS::GetDeviceFilter(const char* deviceUniqueIdUTF8,
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 @@ int32_t DeviceInfoDS::CreateCapabilityMap(const char* deviceUniqueIdUTF8)
(void**)&streamConfig);
if (FAILED(hr)) {
RTC_LOG(LS_INFO) << "Failed to get IID_IAMStreamConfig interface "
<< "from capture device";
"from capture device";
return -1;
}

2
modules/video_coding/codecs/h264/h264_encoder_impl.cc
View file

@ -378,7 +378,7 @@ int32_t H264EncoderImpl::Encode(
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;
}

8
modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
View file

@ -72,9 +72,11 @@ Vp9FrameBufferPool::GetFrameBuffer(size_t min_size) {
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.

8
modules/video_coding/codecs/vp9/vp9_impl.cc
View file

@ -1032,7 +1032,8 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
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 @@ VP9DecoderImpl::~VP9DecoderImpl() {
// 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.";
}
}

2
modules/video_coding/decoding_state.cc
View file

@ -297,7 +297,7 @@ bool VCMDecodingState::UsingFlexibleMode(const VCMFrameBuffer* frame) const {
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;

9
modules/video_coding/frame_buffer2.cc
View file

@ -350,7 +350,8 @@ bool FrameBuffer::HasBadRenderTiming(const EncodedFrame& frame,
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 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
<< 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 @@ bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame,
<< "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;

4
modules/video_coding/packet_buffer.cc
View file

@ -357,8 +357,8 @@ std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
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");

3
modules/video_coding/receiver.cc
View file

@ -114,7 +114,8 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
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()) >

19
modules/video_coding/rtp_frame_reference_finder.cc
View file

@ -384,7 +384,7 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
<< " and packet range [" << frame->first_seq_num()
<< ", " << frame->last_seq_num()
<< "] already received, "
<< " dropping frame.";
" dropping frame.";
return kDrop;
}
@ -585,8 +585,9 @@ bool RtpFrameReferenceFinder::MissingRequiredFrameVp9(uint16_t picture_id,
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 @@ void RtpFrameReferenceFinder::FrameReceivedVp9(uint16_t picture_id,
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 @@ void RtpFrameReferenceFinder::FrameReceivedVp9(uint16_t picture_id,
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 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameH264(
<< " and packet range [" << frame->first_seq_num()
<< ", " << frame->last_seq_num()
<< "] already received, "
<< " dropping frame.";
" dropping frame.";
return kDrop;
}

8
p2p/base/connection.cc
View file

@ -525,15 +525,15 @@ void Connection::HandleStunBindingOrGoogPingRequest(IceMessage* msg) {
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_) << ")";
}
}

2
p2p/base/p2p_transport_channel.cc
View file

@ -1105,7 +1105,7 @@ void P2PTransportChannel::ResolveHostnameCandidate(const Candidate& candidate) {
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;
}

3
p2p/base/port_unittest.cc
View file

@ -2677,7 +2677,8 @@ TEST_P(GoogPingTest, TestGoogPingAnnounceEnable) {
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 =

5
p2p/base/turn_port.cc
View file

@ -1224,8 +1224,9 @@ bool TurnPort::CreateOrRefreshEntry(const rtc::SocketAddress& addr,
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);
}

2
pc/channel.cc
View file

@ -423,7 +423,7 @@ bool BaseChannel::SendPacket(bool rtcp,
// (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;
}

2
pc/peer_connection.cc
View file

@ -1262,7 +1262,7 @@ bool PeerConnection::Initialize(
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;
}

3
pc/rtc_stats_integrationtest.cc
View file

@ -318,7 +318,8 @@ class RTCStatsVerifier {
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);
}

6
pc/session_description.cc
View file

@ -266,7 +266,7 @@ ContentInfo::~ContentInfo() {
// 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 @@ const MediaContentDescription* ContentInfo::media_description() const {
// 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 @@ MediaContentDescription* ContentInfo::media_description() {
// 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();

5
pc/webrtc_sdp.cc
View file

@ -1061,8 +1061,9 @@ bool ParseCandidate(const std::string& message,
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,

23
pc/webrtc_sdp_unittest.cc
View file

@ -1963,18 +1963,22 @@ class WebRtcSdpTest : public ::testing::Test {
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 @@ void TestMismatch(const std::string& string1, const std::string& string2) {
}
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) {

6
rtc_base/logging_unittest.cc
View file

@ -339,8 +339,10 @@ TEST(LogTest, Perf) {
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) {

2
rtc_base/network.cc
View file

@ -773,7 +773,7 @@ bool IsDefaultRoute(const std::string& network_name) {
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;

2
rtc_base/network_unittest.cc
View file

@ -930,7 +930,7 @@ TEST_F(NetworkTest, TestIgnoreNonDefaultRoutes) {
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();

8
rtc_base/openssl_adapter.cc
View file

@ -857,8 +857,10 @@ SSL_CTX* OpenSSLAdapter::CreateContext(SSLMode mode, bool enable_cache) {
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 @@ std::string TransformAlpnProtocols(
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 "";
}

6
rtc_base/physical_socket_server.cc
View file

@ -1265,8 +1265,8 @@ void PhysicalSocketServer::Remove(Dispatcher* pdispatcher) {
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 @@ void PhysicalSocketServer::Remove(Dispatcher* pdispatcher) {
} 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)

2
rtc_base/socket_unittest.cc
View file

@ -391,7 +391,7 @@ void SocketTest::ConnectWithDnsLookupFailInternal(const IPAddress& loopback) {
dns_lookup_finished);
if (!dns_lookup_finished) {
RTC_LOG(LS_WARNING) << "Skipping test; DNS resolution took longer than 5 "
<< "seconds.";
"seconds.";
return;
}

2
rtc_base/thread.cc
View file

@ -766,7 +766,7 @@ void Thread::Join() {
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)

4
rtc_base/timestamp_aligner.cc
View file

@ -122,8 +122,8 @@ int64_t TimestampAligner::ClipTimestamp(int64_t filtered_time_us,
// 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;
}

2
sdk/android/native_unittests/peerconnection/peer_connection_factory_unittest.cc
View file

@ -67,7 +67,7 @@ rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> CreateTestPCF(
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;
}

6
sdk/android/src/jni/android_media_decoder.cc
View file

@ -712,8 +712,10 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(JNIEnv* jni,
(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.";

9
sdk/android/src/jni/android_media_encoder.cc
View file

@ -649,7 +649,8 @@ int32_t MediaCodecVideoEncoder::Encode(
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 @@ void MediaCodecVideoEncoder::LogStatistics(bool force_log) {
(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.";

3
sdk/android/src/jni/android_network_monitor.cc
View file

@ -248,7 +248,8 @@ rtc::NetworkBindingResult AndroidNetworkMonitor::BindSocketToNetwork(
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;
}

3
sdk/android/src/jni/audio_device/aaudio_player.cc
View file

@ -174,7 +174,8 @@ aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
// 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;
}

3
sdk/android/src/jni/audio_device/aaudio_recorder.cc
View file

@ -173,7 +173,8 @@ aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
// 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;
}

2
sdk/android/src/jni/pc/peer_connection_factory.cc
View file

@ -330,7 +330,7 @@ ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
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)

10
stats/rtc_stats.cc
View file

@ -99,9 +99,13 @@ bool RTCStats::operator!=(const RTCStats& other) const {
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() << "\":";

4
test/call_test.cc
View file

@ -79,8 +79,8 @@ void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
<< "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);

4
test/network/fake_network_socket_server.cc
View file

@ -202,8 +202,8 @@ int FakeNetworkSocket::RecvFrom(void* pv,
// 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();

5
test/testsupport/file_utils_unittest.cc
View file

@ -173,8 +173,9 @@ TEST_F(FileUtilsTest, ResourcePathReturnsCorrectPath) {
#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) {

7
video/overuse_frame_detector.cc
View file

@ -677,9 +677,10 @@ void OveruseFrameDetector::CheckForOveruse(
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) {

3
video/rtp_video_stream_receiver.cc
View file

@ -927,7 +927,8 @@ void RtpVideoStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) {
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 =

9
video/video_stream_encoder.cc
View file

@ -1172,7 +1172,7 @@ void VideoStreamEncoder::MaybeEncodeVideoFrame(const VideoFrame& video_frame,
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 @@ VideoStreamEncoder::ParseEncoderSwitchFieldTrial() const {
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 @@ VideoStreamEncoder::ParseAutomatincAnimationDetectionFieldTrial() const {
}
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;