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/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index 532a8eb..3c20a54 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -83,26 +83,25 @@
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 @@
}
// 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 @@
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 @@
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 @@
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 @@
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 @@
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;
}