Reformat the WebRTC code base
Running clang-format with chromium's style guide. The goal is n-fold: * providing consistency and readability (that's what code guidelines are for) * preventing noise with presubmit checks and git cl format * building on the previous point: making it easier to automatically fix format issues * you name it Please consider using git-hyper-blame to ignore this commit. Bug: webrtc:9340 Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87 Reviewed-on: https://webrtc-review.googlesource.com/81185 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23660}
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@ -95,10 +95,9 @@ std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
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RTC_CHECK(audio_source_->Read(input_block_size_samples_,
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input_frame_.mutable_data()));
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if (input_frame_.num_channels_ > 1) {
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InputAudioFile::DuplicateInterleaved(input_frame_.data(),
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input_block_size_samples_,
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input_frame_.num_channels_,
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input_frame_.mutable_data());
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InputAudioFile::DuplicateInterleaved(
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input_frame_.data(), input_block_size_samples_,
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input_frame_.num_channels_, input_frame_.mutable_data());
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}
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data_to_send_ = false;
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RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
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@ -138,7 +137,7 @@ std::unique_ptr<Packet> AcmSendTestOldApi::CreatePacket() {
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packet_memory[0] = 0x80;
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packet_memory[1] = static_cast<uint8_t>(payload_type_);
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packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
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packet_memory[3] = (sequence_number_) & 0xFF;
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packet_memory[3] = (sequence_number_)&0xFF;
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packet_memory[4] = (timestamp_ >> 24) & 0xFF;
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packet_memory[5] = (timestamp_ >> 16) & 0xFF;
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packet_memory[6] = (timestamp_ >> 8) & 0xFF;
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@ -152,8 +151,7 @@ std::unique_ptr<Packet> AcmSendTestOldApi::CreatePacket() {
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++sequence_number_;
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// Copy the payload data.
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memcpy(packet_memory + kRtpHeaderSize,
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&last_payload_vec_[0],
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memcpy(packet_memory + kRtpHeaderSize, &last_payload_vec_[0],
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last_payload_vec_.size());
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std::unique_ptr<Packet> packet(
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new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()));
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