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Reformat the WebRTC code base

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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}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
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8
modules/audio_coding/neteq/tools/neteq_quality_test.h
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@ -36,7 +36,7 @@ enum LossModes {
class LossModel {
public:
virtual ~LossModel() {};
virtual ~LossModel(){};
virtual bool Lost(int now_ms) = 0;
};
@ -110,8 +110,10 @@ class NetEqQualityTest : public ::testing::Test {
// |block_size_samples| (samples per channel),
// 2. save the bit stream to |payload| of |max_bytes| bytes in size,
// 3. returns the length of the payload (in bytes),
virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
rtc::Buffer* payload, size_t max_bytes) = 0;
virtual int EncodeBlock(int16_t* in_data,
size_t block_size_samples,
rtc::Buffer* payload,
size_t max_bytes) = 0;
// PacketLost(...) determines weather a packet sent at an indicated time gets
// lost or not.