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:
@ -712,8 +712,10 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(JNIEnv* jni,
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(current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms;
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ALOGD << "Frames decoded: " << frames_decoded_
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<< ". Received: " << frames_received_
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<< ". Bitrate: " << current_bitrate << " kbps"
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<< ". Fps: " << current_fps
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<< ". Bitrate: " << current_bitrate
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<< " kbps"
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". Fps: "
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<< current_fps
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<< ". DecTime: " << (current_decoding_time_ms_ / current_frames_)
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<< ". DelayTime: " << (current_delay_time_ms_ / current_frames_)
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<< " for last " << statistic_time_ms << " ms.";
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@ -649,7 +649,8 @@ int32_t MediaCodecVideoEncoder::Encode(
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if (input_frame_infos_.size() > MAX_ENCODER_Q_SIZE) {
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ALOGD << "Already " << input_frame_infos_.size()
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<< " frames in the queue, dropping"
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<< ". TS: " << static_cast<int>(current_timestamp_us_ / 1000)
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". TS: "
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<< static_cast<int>(current_timestamp_us_ / 1000)
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<< ". Fps: " << last_set_fps_
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<< ". Consecutive drops: " << consecutive_full_queue_frame_drops_;
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current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
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@ -1134,8 +1135,10 @@ void MediaCodecVideoEncoder::LogStatistics(bool force_log) {
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(current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms;
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ALOGD << "Encoded frames: " << frames_encoded_
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<< ". Bitrate: " << current_bitrate
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<< ", target: " << last_set_bitrate_kbps_ << " kbps"
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<< ", fps: " << current_fps << ", encTime: "
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<< ", target: " << last_set_bitrate_kbps_
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<< " kbps"
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", fps: "
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<< current_fps << ", encTime: "
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<< (current_encoding_time_ms_ / current_frames_divider)
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<< ". QP: " << (current_acc_qp_ / current_frames_divider)
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<< " for last " << statistic_time_ms << " ms.";
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@ -248,7 +248,8 @@ rtc::NetworkBindingResult AndroidNetworkMonitor::BindSocketToNetwork(
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if (!network_binding_supported) {
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RTC_LOG(LS_WARNING)
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<< "BindSocketToNetwork is not supported on this platform "
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<< "(Android SDK: " << android_sdk_int_ << ")";
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"(Android SDK: "
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<< android_sdk_int_ << ")";
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return rtc::NetworkBindingResult::NOT_IMPLEMENTED;
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}
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@ -174,7 +174,8 @@ aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
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// utilized.
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if (first_data_callback_) {
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RTC_LOG(INFO) << "--- First output data callback: "
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<< "device id=" << aaudio_.device_id();
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"device id="
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<< aaudio_.device_id();
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first_data_callback_ = false;
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}
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@ -173,7 +173,8 @@ aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
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// is obtained.
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if (first_data_callback_) {
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RTC_LOG(INFO) << "--- First input data callback: "
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<< "device id=" << aaudio_.device_id();
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"device id="
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<< aaudio_.device_id();
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aaudio_.ClearInputStream(audio_data, num_frames);
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first_data_callback_ = false;
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}
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@ -330,7 +330,7 @@ ScopedJavaLocalRef<jobject> CreatePeerConnectionFactoryForJava(
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CreateModularPeerConnectionFactory(std::move(dependencies));
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RTC_CHECK(factory) << "Failed to create the peer connection factory; "
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<< "WebRTC/libjingle init likely failed on this device";
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"WebRTC/libjingle init likely failed on this device";
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// TODO(honghaiz): Maybe put the options as the argument of
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// CreatePeerConnectionFactory.
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if (options)
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