zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Reformat the WebRTC code base

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

33
modules/audio_coding/test/PacketLossTest.cc
View File

@ -23,11 +23,10 @@ ReceiverWithPacketLoss::ReceiverWithPacketLoss()
burst_length_(1),
packet_counter_(0),
lost_packet_counter_(0),
burst_lost_counter_(burst_length_) {
}
burst_lost_counter_(burst_length_) {}
void ReceiverWithPacketLoss::Setup(AudioCodingModule *acm,
RTPStream *rtpStream,
void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string out_file_name,
int channels,
int loss_rate,
@ -84,13 +83,14 @@ bool ReceiverWithPacketLoss::PacketLost() {
return false;
}
SenderWithFEC::SenderWithFEC()
: expected_loss_rate_(0) {
}
SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string in_file_name, int sample_rate,
int channels, int expected_loss_rate) {
void SenderWithFEC::Setup(AudioCodingModule* acm,
RTPStream* rtpStream,
std::string in_file_name,
int sample_rate,
int channels,
int expected_loss_rate) {
Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
EXPECT_TRUE(SetFEC(true));
EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
@ -111,18 +111,19 @@ bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
return false;
}
PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
int actual_loss_rate, int burst_length)
PacketLossTest::PacketLossTest(int channels,
int expected_loss_rate,
int actual_loss_rate,
int burst_length)
: channels_(channels),
in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
"audio_coding/teststereo32kHz"),
in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
: "audio_coding/teststereo32kHz"),
sample_rate_hz_(32000),
sender_(new SenderWithFEC),
receiver_(new ReceiverWithPacketLoss),
expected_loss_rate_(expected_loss_rate),
actual_loss_rate_(actual_loss_rate),
burst_length_(burst_length) {
}
burst_length_(burst_length) {}
void PacketLossTest::Perform() {
#ifndef WEBRTC_CODEC_OPUS