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 <[email protected]>
Reviewed-by: Karl Wiberg <[email protected]>
Commit-Queue: Jonas Olsson <[email protected]>
Cr-Commit-Position: refs/heads/master@{#30251}

api/video_codecs/video_encoder.cc

@@ -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();
 }
 

audio/channel_receive.cc

@@ -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;
   }
 

audio/channel_send.cc

@@ -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;
     }
   }

call/flexfec_receive_stream_impl.cc

@@ -87,21 +87,21 @@ 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);
   RTC_DCHECK_LE(config.payload_type, 127);
   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;
   }
 

call/rtp_stream_receiver_controller.cc

@@ -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 << ".";
   }
 }
 

call/rtp_transport_controller_send.cc

@@ -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";
   }
 }
 

call/simulated_network.cc

@@ -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 =

common_video/bitrate_adjuster.cc

@@ -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)

examples/peerconnection/client/conductor.cc

@@ -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())) {

examples/unityplugin/simple_peer_connection.cc

@@ -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())) {

logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc

@@ -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;
     }
   }

logging/rtc_event_log/rtc_event_log2rtp_dump.cc

@@ -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;
 }

logging/rtc_event_log/rtc_event_log_impl.cc

@@ -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;

media/base/rtp_data_engine.cc

@@ -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();

media/engine/webrtc_video_engine.cc

@@ -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);
   }
 }

media/engine/webrtc_voice_engine.cc

@@ -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();