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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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11
modules/audio_coding/test/Channel.h
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@ -20,8 +20,8 @@
namespace webrtc {
#define MAX_NUM_PAYLOADS 50
#define MAX_NUM_FRAMESIZES 6
#define MAX_NUM_PAYLOADS 50
#define MAX_NUM_FRAMESIZES 6
// TODO(turajs): Write constructor for this structure.
struct ACMTestFrameSizeStats {
@ -45,7 +45,6 @@ struct ACMTestPayloadStats {
class Channel : public AudioPacketizationCallback {
public:
Channel(int16_t chID = -1);
~Channel() override;
@ -56,7 +55,7 @@ class Channel : public AudioPacketizationCallback {
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
void RegisterReceiverACM(AudioCodingModule *acm);
void RegisterReceiverACM(AudioCodingModule* acm);
void ResetStats();
@ -68,9 +67,7 @@ class Channel : public AudioPacketizationCallback {
void PrintStats(CodecInst& codecInst);
void SetIsStereo(bool isStereo) {
_isStereo = isStereo;
}
void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
uint32_t LastInTimestamp();