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}
diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index 7eab442..f6f92db 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -20,8 +20,11 @@
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 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_;