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

76
modules/audio_coding/test/delay_test.cc
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@ -83,26 +83,25 @@ class DelayTest {
void Initialize() {
test_cntr_ = 0;
std::string file_name = webrtc::test::ResourcePath(
"audio_coding/testfile32kHz", "pcm");
std::string file_name =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
if (strlen(FLAG_input_file) > 0)
file_name = FLAG_input_file;
in_file_a_.Open(file_name, 32000, "rb");
ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
"Couldn't initialize receiver.\n";
ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
"Couldn't initialize receiver.\n";
ASSERT_EQ(0, acm_a_->InitializeReceiver())
<< "Couldn't initialize receiver.\n";
ASSERT_EQ(0, acm_b_->InitializeReceiver())
<< "Couldn't initialize receiver.\n";
if (FLAG_delay > 0) {
ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAG_delay)) <<
"Failed to set minimum delay.\n";
ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAG_delay))
<< "Failed to set minimum delay.\n";
}
int num_encoders = acm_a_->NumberOfCodecs();
CodecInst my_codec_param;
for (int n = 0; n < num_encoders; n++) {
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
"Failed to get codec.";
EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) << "Failed to get codec.";
if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
my_codec_param.channels = 1;
else if (my_codec_param.channels > 1)
@ -118,12 +117,14 @@ class DelayTest {
}
// Create and connect the channel
ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
"Couldn't register Transport callback.\n";
ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_))
<< "Couldn't register Transport callback.\n";
channel_a2b_->RegisterReceiverACM(acm_b_.get());
}
void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
void Perform(const TestSettings* config,
size_t num_tests,
int duration_sec,
const char* output_prefix) {
for (size_t n = 0; n < num_tests; ++n) {
ApplyConfig(config[n]);
@ -134,14 +135,15 @@ class DelayTest {
private:
void ApplyConfig(const TestSettings& config) {
printf("====================================\n");
printf("Test %d \n"
"Codec: %s, %d kHz, %d channel(s)\n"
"ACM: DTX %s, FEC %s\n"
"Channel: %s\n",
++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
config.codec.num_channels, config.acm.dtx ? "on" : "off",
config.acm.fec ? "on" : "off",
config.packet_loss ? "with packet-loss" : "no packet-loss");
printf(
"Test %d \n"
"Codec: %s, %d kHz, %d channel(s)\n"
"ACM: DTX %s, FEC %s\n"
"Channel: %s\n",
++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
config.codec.num_channels, config.acm.dtx ? "on" : "off",
config.acm.fec ? "on" : "off",
config.packet_loss ? "with packet-loss" : "no packet-loss");
SendCodec(config.codec);
ConfigAcm(config.acm);
ConfigChannel(config.packet_loss);
@ -149,20 +151,20 @@ class DelayTest {
void SendCodec(const CodecSettings& config) {
CodecInst my_codec_param;
ASSERT_EQ(0, AudioCodingModule::Codec(
config.name, &my_codec_param, config.sample_rate_hz,
config.num_channels)) << "Specified codec is not supported.\n";
ASSERT_EQ(
0, AudioCodingModule::Codec(config.name, &my_codec_param,
config.sample_rate_hz, config.num_channels))
<< "Specified codec is not supported.\n";
encoding_sample_rate_hz_ = my_codec_param.plfreq;
ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
"Failed to register send-codec.\n";
ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param))
<< "Failed to register send-codec.\n";
}
void ConfigAcm(const AcmSettings& config) {
ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
"Failed to set VAD.\n";
ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
"Failed to set RED.\n";
ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr))
<< "Failed to set VAD.\n";
ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) << "Failed to set RED.\n";
}
void ConfigChannel(bool packet_loss) {
@ -172,7 +174,8 @@ class DelayTest {
void OpenOutFile(const char* output_id) {
std::stringstream file_stream;
file_stream << "delay_test_" << FLAG_codec << "_" << FLAG_sample_rate_hz
<< "Hz" << "_" << FLAG_delay << "ms.pcm";
<< "Hz"
<< "_" << FLAG_delay << "ms.pcm";
std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
std::string file_name = webrtc::test::OutputPath() + file_stream.str();
out_file_b_.Open(file_name.c_str(), 32000, "wb");
@ -197,14 +200,15 @@ class DelayTest {
if ((num_frames & 0x3F) == 0x3F) {
NetworkStatistics statistics;
acm_b_->GetNetworkStatistics(&statistics);
fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
fprintf(stdout,
"delay: min=%3d max=%3d mean=%3d median=%3d"
" ts-based average = %6.3f, "
"curr buff-lev = %4u opt buff-lev = %4u \n",
statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
average_delay, statistics.currentBufferSize,
statistics.preferredBufferSize);
fflush (stdout);
fflush(stdout);
}
in_file_a_.Read10MsData(audio_frame);
@ -256,10 +260,8 @@ int main(int argc, char* argv[]) {
webrtc::TestSettings test_setting;
strcpy(test_setting.codec.name, FLAG_codec);
if (FLAG_sample_rate_hz != 8000 &&
FLAG_sample_rate_hz != 16000 &&
FLAG_sample_rate_hz != 32000 &&
FLAG_sample_rate_hz != 48000) {
if (FLAG_sample_rate_hz != 8000 && FLAG_sample_rate_hz != 16000 &&
FLAG_sample_rate_hz != 32000 && FLAG_sample_rate_hz != 48000) {
std::cout << "Invalid sampling rate.\n";
return 1;
}