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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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20
modules/audio_coding/test/PacketLossTest.h
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@ -20,8 +20,11 @@ namespace webrtc {
class ReceiverWithPacketLoss : public Receiver {
public:
ReceiverWithPacketLoss();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string out_file_name, int channels, int loss_rate,
void Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string out_file_name,
int channels,
int loss_rate,
int burst_length);
bool IncomingPacket() override;
@ -37,20 +40,27 @@ class ReceiverWithPacketLoss : public Receiver {
class SenderWithFEC : public Sender {
public:
SenderWithFEC();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string in_file_name, int sample_rate, int channels,
void Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string in_file_name,
int sample_rate,
int channels,
int expected_loss_rate);
bool SetPacketLossRate(int expected_loss_rate);
bool SetFEC(bool enable_fec);
protected:
int expected_loss_rate_;
};
class PacketLossTest : public ACMTest {
public:
PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate,
PacketLossTest(int channels,
int expected_loss_rate_,
int actual_loss_rate,
int burst_length);
void Perform();
protected:
int channels_;
std::string in_file_name_;