Concatenate string literals at compile time.

This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
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

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.";

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.";

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;
}

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@ -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;
}

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;
}

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)