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/iSACTest.cc
View File

@ -41,11 +41,11 @@ void SetISACConfigDefault(ACMTestISACConfig& isacConfig) {
return;
}
int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
int16_t SetISAConfig(ACMTestISACConfig& isacConfig,
AudioCodingModule* acm,
int testMode) {
if ((isacConfig.currentRateBitPerSec != 0)
|| (isacConfig.currentFrameSizeMsec != 0)) {
if ((isacConfig.currentRateBitPerSec != 0) ||
(isacConfig.currentFrameSizeMsec != 0)) {
auto sendCodec = acm->SendCodec();
EXPECT_TRUE(sendCodec);
if (isacConfig.currentRateBitPerSec < 0) {
@ -57,8 +57,8 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
sendCodec->rate = isacConfig.currentRateBitPerSec;
}
if (isacConfig.currentFrameSizeMsec != 0) {
sendCodec->pacsize = isacConfig.currentFrameSizeMsec
* (sendCodec->plfreq / 1000);
sendCodec->pacsize =
isacConfig.currentFrameSizeMsec * (sendCodec->plfreq / 1000);
}
EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
}
@ -81,15 +81,15 @@ void ISACTest::Setup() {
CodecInst codecParam;
for (codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs();
codecCntr++) {
codecCntr++) {
EXPECT_EQ(0, AudioCodingModule::Codec(codecCntr, &codecParam));
if (!STR_CASE_CMP(codecParam.plname, "ISAC")
&& codecParam.plfreq == 16000) {
if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
codecParam.plfreq == 16000) {
memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
_idISAC16kHz = codecCntr;
}
if (!STR_CASE_CMP(codecParam.plname, "ISAC")
&& codecParam.plfreq == 32000) {
if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
codecParam.plfreq == 32000) {
memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
_idISAC32kHz = codecCntr;
}
@ -115,8 +115,8 @@ void ISACTest::Setup() {
EXPECT_EQ(0, _acmB->RegisterTransportCallback(_channel_B2A.get()));
_channel_B2A->RegisterReceiverACM(_acmA.get());
file_name_swb_ = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
"pcm");
file_name_swb_ =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
@ -213,7 +213,8 @@ void ISACTest::Run10ms() {
_outFileB.Write10MsData(audioFrame);
}
void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
void ISACTest::EncodeDecode(int testNr,
ACMTestISACConfig& wbISACConfig,
ACMTestISACConfig& swbISACConfig) {
// Files in Side A and B
_inFileA.Open(file_name_swb_, 32000, "rb", true);
@ -241,8 +242,8 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
SetISAConfig(wbISACConfig, _acmB.get(), _testMode);
bool adaptiveMode = false;
if ((swbISACConfig.currentRateBitPerSec == -1)
|| (wbISACConfig.currentRateBitPerSec == -1)) {
if ((swbISACConfig.currentRateBitPerSec == -1) ||
(wbISACConfig.currentRateBitPerSec == -1)) {
adaptiveMode = true;
}
_myTimer.Reset();