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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@ -58,10 +58,8 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
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uint32_t rtp_timestamp,
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rtc::ArrayView<const int16_t> audio,
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rtc::Buffer* encoded) {
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const size_t primary_offset = encoded->size();
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EncodedInfo info =
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speech_encoder_->Encode(rtp_timestamp, audio, encoded);
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EncodedInfo info = speech_encoder_->Encode(rtp_timestamp, audio, encoded);
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RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
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RTC_DCHECK_EQ(encoded->size() - primary_offset, info.encoded_bytes);
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@ -48,9 +48,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
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.WillRepeatedly(Return(sample_rate_hz_));
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}
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void TearDown() override {
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red_.reset();
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}
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void TearDown() override { red_.reset(); }
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void Encode() {
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ASSERT_TRUE(red_.get() != NULL);
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@ -73,8 +71,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
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const int red_payload_type_;
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};
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TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {
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}
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TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {}
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TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
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EXPECT_CALL(*mock_encoder_, SampleRateHz()).WillOnce(Return(17));
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