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/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 6239985..4ed7a6b 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -61,17 +61,17 @@
const std::string& checksum_win_32,
const std::string& checksum_win_64) {
#if defined(WEBRTC_ANDROID)
- #ifdef WEBRTC_ARCH_64_BITS
- return checksum_android_64;
- #else
- return checksum_android_32;
- #endif // WEBRTC_ARCH_64_BITS
+#ifdef WEBRTC_ARCH_64_BITS
+ return checksum_android_64;
+#else
+ return checksum_android_32;
+#endif // WEBRTC_ARCH_64_BITS
#elif defined(WEBRTC_WIN)
- #ifdef WEBRTC_ARCH_64_BITS
- return checksum_win_64;
- #else
- return checksum_win_32;
- #endif // WEBRTC_ARCH_64_BITS
+#ifdef WEBRTC_ARCH_64_BITS
+ return checksum_win_64;
+#else
+ return checksum_win_32;
+#endif // WEBRTC_ARCH_64_BITS
#else
return checksum_general;
#endif // WEBRTC_WIN
@@ -107,7 +107,8 @@
stats->set_jitter(stats_raw.jitter);
}
-void AddMessage(FILE* file, rtc::MessageDigest* digest,
+void AddMessage(FILE* file,
+ rtc::MessageDigest* digest,
const std::string& message) {
int32_t size = message.length();
if (file)
@@ -164,7 +165,8 @@
explicit ResultSink(const std::string& output_file);
~ResultSink();
- template<typename T> void AddResult(const T* test_results, size_t length);
+ template <typename T>
+ void AddResult(const T* test_results, size_t length);
void AddResult(const NetEqNetworkStatistics& stats);
void AddResult(const RtcpStatistics& stats);
@@ -190,7 +192,7 @@
fclose(output_fp_);
}
-template<typename T>
+template <typename T>
void ResultSink::AddResult(const T* test_results, size_t length) {
if (output_fp_) {
ASSERT_EQ(length, fwrite(test_results, sizeof(T), length, output_fp_));
@@ -247,7 +249,7 @@
virtual void SetUp();
virtual void TearDown();
void SelectDecoders(NetEqDecoder* used_codec);
- void OpenInputFile(const std::string &rtp_file);
+ void OpenInputFile(const std::string& rtp_file);
void Process();
void DecodeAndCompare(const std::string& rtp_file,
@@ -265,9 +267,11 @@
uint8_t* payload,
size_t* payload_len);
- void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
+ void WrapTest(uint16_t start_seq_no,
+ uint32_t start_timestamp,
const std::set<uint16_t>& drop_seq_numbers,
- bool expect_seq_no_wrap, bool expect_timestamp_wrap);
+ bool expect_seq_no_wrap,
+ bool expect_timestamp_wrap);
void LongCngWithClockDrift(double drift_factor,
double network_freeze_ms,
@@ -316,7 +320,7 @@
delete neteq_;
}
-void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
+void NetEqDecodingTest::OpenInputFile(const std::string& rtp_file) {
rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
}
@@ -384,8 +388,8 @@
ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
ASSERT_NO_FATAL_FAILURE(Process());
- ASSERT_NO_FATAL_FAILURE(output.AddResult(
- out_frame_.data(), out_frame_.samples_per_channel_));
+ ASSERT_NO_FATAL_FAILURE(
+ output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
// Query the network statistics API once per second
if (sim_clock_ % 1000 == 0) {
@@ -447,7 +451,7 @@
rtp_info->ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info->payloadType = 98; // WB CNG.
rtp_info->markerBit = 0;
- payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
+ payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
*payload_len = 1; // Only noise level, no spectral parameters.
}
@@ -462,36 +466,29 @@
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
- const std::string output_checksum = PlatformChecksum(
- "0c6dc227f781c81a229970f8fceda1a012498cba",
- "15c4a2202877a414515e218bdb7992f0ad53e5af",
- "not used",
- "0c6dc227f781c81a229970f8fceda1a012498cba",
- "25fc4c863caa499aa447a5b8d059f5452cbcc500");
+ const std::string output_checksum =
+ PlatformChecksum("0c6dc227f781c81a229970f8fceda1a012498cba",
+ "15c4a2202877a414515e218bdb7992f0ad53e5af", "not used",
+ "0c6dc227f781c81a229970f8fceda1a012498cba",
+ "25fc4c863caa499aa447a5b8d059f5452cbcc500");
const std::string network_stats_checksum =
PlatformChecksum("4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
- "e339cb2adf5ab3dfc21cb7205d670a34751e8336",
- "not used",
+ "e339cb2adf5ab3dfc21cb7205d670a34751e8336", "not used",
"4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
"4b2370f5c794741d2a46be5c7935c66ef3fb53e9");
- const std::string rtcp_stats_checksum = PlatformChecksum(
- "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
- "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
- "not used",
- "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
- "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
+ const std::string rtcp_stats_checksum =
+ PlatformChecksum("b8880bf9fed2487efbddcb8d94b9937a29ae521d",
+ "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4", "not used",
+ "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
+ "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
- DecodeAndCompare(input_rtp_file,
- output_checksum,
- network_stats_checksum,
- rtcp_stats_checksum,
- FLAG_gen_ref);
+ DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+ rtcp_stats_checksum, FLAG_gen_ref);
}
-#if !defined(WEBRTC_IOS) && \
- defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
+#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS)
#define MAYBE_TestOpusBitExactness TestOpusBitExactness
#else
@@ -501,12 +498,12 @@
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
- const std::string output_checksum = PlatformChecksum(
- "14a63b3c7b925c82296be4bafc71bec85f2915c2",
- "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
- "5876e52dda90d5ca433c3726555b907b97c86374",
- "14a63b3c7b925c82296be4bafc71bec85f2915c2",
- "14a63b3c7b925c82296be4bafc71bec85f2915c2");
+ const std::string output_checksum =
+ PlatformChecksum("14a63b3c7b925c82296be4bafc71bec85f2915c2",
+ "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
+ "5876e52dda90d5ca433c3726555b907b97c86374",
+ "14a63b3c7b925c82296be4bafc71bec85f2915c2",
+ "14a63b3c7b925c82296be4bafc71bec85f2915c2");
const std::string network_stats_checksum =
PlatformChecksum("adb3272498e436d1c019cbfd71610e9510c54497",
@@ -515,22 +512,18 @@
"adb3272498e436d1c019cbfd71610e9510c54497",
"adb3272498e436d1c019cbfd71610e9510c54497");
- const std::string rtcp_stats_checksum = PlatformChecksum(
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
- "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
+ const std::string rtcp_stats_checksum =
+ PlatformChecksum("e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
+ "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
+ "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
+ "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
+ "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
- DecodeAndCompare(input_rtp_file,
- output_checksum,
- network_stats_checksum,
- rtcp_stats_checksum,
- FLAG_gen_ref);
+ DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+ rtcp_stats_checksum, FLAG_gen_ref);
}
-#if !defined(WEBRTC_IOS) && \
- defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
+#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS)
#define MAYBE_TestOpusDtxBitExactness TestOpusDtxBitExactness
#else
@@ -805,10 +798,8 @@
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
- LongCngWithClockDrift(kDriftFactor,
- kNetworkFreezeTimeMs,
- kGetAudioDuringFreezeRecovery,
- kDelayToleranceMs,
+ LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
@@ -819,10 +810,8 @@
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
- LongCngWithClockDrift(kDriftFactor,
- kNetworkFreezeTimeMs,
- kGetAudioDuringFreezeRecovery,
- kDelayToleranceMs,
+ LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
@@ -833,10 +822,8 @@
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 50;
const int kMaxTimeToSpeechMs = 200;
- LongCngWithClockDrift(kDriftFactor,
- kNetworkFreezeTimeMs,
- kGetAudioDuringFreezeRecovery,
- kDelayToleranceMs,
+ LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
@@ -847,10 +834,8 @@
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
- LongCngWithClockDrift(kDriftFactor,
- kNetworkFreezeTimeMs,
- kGetAudioDuringFreezeRecovery,
- kDelayToleranceMs,
+ LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
@@ -861,10 +846,8 @@
const bool kGetAudioDuringFreezeRecovery = true;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
- LongCngWithClockDrift(kDriftFactor,
- kNetworkFreezeTimeMs,
- kGetAudioDuringFreezeRecovery,
- kDelayToleranceMs,
+ LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
@@ -874,10 +857,8 @@
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 10;
const int kMaxTimeToSpeechMs = 50;
- LongCngWithClockDrift(kDriftFactor,
- kNetworkFreezeTimeMs,
- kGetAudioDuringFreezeRecovery,
- kDelayToleranceMs,
+ LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+ kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
@@ -1002,11 +983,11 @@
ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Next packet.
- rtp_info.timestamp += rtc::checked_cast<uint32_t>(
- expected_samples_per_channel);
+ rtp_info.timestamp +=
+ rtc::checked_cast<uint32_t>(expected_samples_per_channel);
rtp_info.sequenceNumber++;
- receive_timestamp += rtc::checked_cast<uint32_t>(
- expected_samples_per_channel);
+ receive_timestamp +=
+ rtc::checked_cast<uint32_t>(expected_samples_per_channel);
}
output.Reset();
@@ -1099,8 +1080,8 @@
if (packets_inserted > 4) {
// Expect preferred and actual buffer size to be no more than 2 frames.
EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2);
- EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2 +
- algorithmic_delay_ms_);
+ EXPECT_LE(network_stats.current_buffer_size_ms,
+ kFrameSizeMs * 2 + algorithmic_delay_ms_);
}
last_seq_no = seq_no;
last_timestamp = timestamp;
@@ -1166,8 +1147,8 @@
const int kSamples = kFrameSizeMs * kSampleRateKhz;
const size_t kPayloadBytes = kSamples * 2;
- const int algorithmic_delay_samples = std::max(
- algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
+ const int algorithmic_delay_samples =
+ std::max(algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
// Insert three speech packets. Three are needed to get the frame length
// correct.
uint8_t payload[kPayloadBytes] = {0};
@@ -1239,7 +1220,9 @@
*playout_timestamp);
}
-TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
+TEST_F(NetEqDecodingTest, DiscardDuplicateCng) {
+ DuplicateCng();
+}
TEST_F(NetEqDecodingTest, CngFirst) {
uint16_t seq_no = 0;
@@ -1493,25 +1476,25 @@
return ::testing::AssertionFailure() << "timestamp_ diff (" << a.timestamp_
<< " != " << b.timestamp_ << ")";
if (a.sample_rate_hz_ != b.sample_rate_hz_)
- return ::testing::AssertionFailure() << "sample_rate_hz_ diff ("
- << a.sample_rate_hz_
- << " != " << b.sample_rate_hz_ << ")";
+ return ::testing::AssertionFailure()
+ << "sample_rate_hz_ diff (" << a.sample_rate_hz_
+ << " != " << b.sample_rate_hz_ << ")";
if (a.samples_per_channel_ != b.samples_per_channel_)
return ::testing::AssertionFailure()
<< "samples_per_channel_ diff (" << a.samples_per_channel_
<< " != " << b.samples_per_channel_ << ")";
if (a.num_channels_ != b.num_channels_)
- return ::testing::AssertionFailure() << "num_channels_ diff ("
- << a.num_channels_
- << " != " << b.num_channels_ << ")";
+ return ::testing::AssertionFailure()
+ << "num_channels_ diff (" << a.num_channels_
+ << " != " << b.num_channels_ << ")";
if (a.speech_type_ != b.speech_type_)
- return ::testing::AssertionFailure() << "speech_type_ diff ("
- << a.speech_type_
- << " != " << b.speech_type_ << ")";
+ return ::testing::AssertionFailure()
+ << "speech_type_ diff (" << a.speech_type_
+ << " != " << b.speech_type_ << ")";
if (a.vad_activity_ != b.vad_activity_)
- return ::testing::AssertionFailure() << "vad_activity_ diff ("
- << a.vad_activity_
- << " != " << b.vad_activity_ << ")";
+ return ::testing::AssertionFailure()
+ << "vad_activity_ diff (" << a.vad_activity_
+ << " != " << b.vad_activity_ << ")";
return ::testing::AssertionSuccess();
}
@@ -1520,9 +1503,9 @@
::testing::AssertionResult res = AudioFramesEqualExceptData(a, b);
if (!res)
return res;
- if (memcmp(
- a.data(), b.data(),
- a.samples_per_channel_ * a.num_channels_ * sizeof(*a.data())) != 0) {
+ if (memcmp(a.data(), b.data(),
+ a.samples_per_channel_ * a.num_channels_ * sizeof(*a.data())) !=
+ 0) {
return ::testing::AssertionFailure() << "data_ diff";
}
return ::testing::AssertionSuccess();