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}
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

View File

@ -23,9 +23,12 @@ class OpusSpeedTest : public AudioCodecSpeedTest {
OpusSpeedTest();
void SetUp() override;
void TearDown() override;
float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t max_bytes, size_t* encoded_bytes) override;
float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
float EncodeABlock(int16_t* in_data,
uint8_t* bit_stream,
size_t max_bytes,
size_t* encoded_bytes) override;
float DecodeABlock(const uint8_t* bit_stream,
size_t encoded_bytes,
int16_t* out_data) override;
WebRtcOpusEncInst* opus_encoder_;
WebRtcOpusDecInst* opus_decoder_;
@ -36,8 +39,7 @@ OpusSpeedTest::OpusSpeedTest()
kOpusSamplingKhz,
kOpusSamplingKhz),
opus_encoder_(NULL),
opus_decoder_(NULL) {
}
opus_decoder_(NULL) {}
void OpusSpeedTest::SetUp() {
AudioCodecSpeedTest::SetUp();
@ -57,12 +59,13 @@ void OpusSpeedTest::TearDown() {
EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
}
float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t max_bytes, size_t* encoded_bytes) {
float OpusSpeedTest::EncodeABlock(int16_t* in_data,
uint8_t* bit_stream,
size_t max_bytes,
size_t* encoded_bytes) {
clock_t clocks = clock();
int value = WebRtcOpus_Encode(opus_encoder_, in_data,
input_length_sample_, max_bytes,
bit_stream);
int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_,
max_bytes, bit_stream);
clocks = clock() - clocks;
EXPECT_GT(value, 0);
*encoded_bytes = static_cast<size_t>(value);
@ -70,7 +73,8 @@ float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
}
float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
size_t encoded_bytes, int16_t* out_data) {
size_t encoded_bytes,
int16_t* out_data) {
int value;
int16_t audio_type;
clock_t clocks = clock();
@ -84,13 +88,13 @@ float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
/* Test audio length in second. */
constexpr size_t kDurationSec = 400;
#define ADD_TEST(complexity) \
TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
/* Set complexity. */ \
printf("Setting complexity to %d ...\n", complexity); \
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
EncodeDecode(kDurationSec); \
}
#define ADD_TEST(complexity) \
TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
/* Set complexity. */ \
printf("Setting complexity to %d ...\n", complexity); \
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
EncodeDecode(kDurationSec); \
}
ADD_TEST(10);
ADD_TEST(9);
@ -136,7 +140,6 @@ const coding_param param_set[] = {
string("pcm"),
true)};
INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
::testing::ValuesIn(param_set));
INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest, ::testing::ValuesIn(param_set));
} // namespace webrtc