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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

33
modules/audio_coding/codecs/opus/opus_fec_test.cc
View File

@ -83,8 +83,8 @@ void OpusFecTest::SetUp() {
rewind(fp);
// Allocate memory to contain the whole file.
in_data_.reset(new int16_t[loop_length_samples_ +
block_length_sample_ * channels_]);
in_data_.reset(
new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);
// Copy the file into the buffer.
ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
@ -130,14 +130,12 @@ OpusFecTest::OpusFecTest()
max_bytes_(0),
encoded_bytes_(0),
opus_encoder_(NULL),
opus_decoder_(NULL) {
}
opus_decoder_(NULL) {}
void OpusFecTest::EncodeABlock() {
int value = WebRtcOpus_Encode(opus_encoder_,
&in_data_[data_pointer_],
block_length_sample_,
max_bytes_, &bit_stream_[0]);
int value =
WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
block_length_sample_, max_bytes_, &bit_stream_[0]);
EXPECT_GT(value, 0);
encoded_bytes_ = static_cast<size_t>(value);
@ -151,9 +149,9 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
// Decode previous frame.
if (!lost_current &&
WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
value_1 = WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0],
encoded_bytes_, &out_data_[0],
&audio_type);
value_1 =
WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
&out_data_[0], &audio_type);
} else {
value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1);
}
@ -173,16 +171,14 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
int time_now_ms, fec_frames;
int actual_packet_loss_rate;
bool lost_current, lost_previous;
mode mode_set[3] = {{true, 0},
{false, 0},
{true, 50}};
mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};
lost_current = false;
for (int i = 0; i < 3; i++) {
if (mode_set[i].fec) {
EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_,
mode_set[i].target_packet_loss_rate));
EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
opus_encoder_, mode_set[i].target_packet_loss_rate));
printf("FEC is ON, target at packet loss rate %d percent.\n",
mode_set[i].target_packet_loss_rate);
} else {
@ -218,7 +214,7 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
// |data_pointer_| is incremented and wrapped across
// |loop_length_samples_|.
data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
loop_length_samples_;
loop_length_samples_;
}
if (mode_set[i].fec) {
printf("%.2f percent frames has FEC.\n",
@ -242,7 +238,6 @@ const coding_param param_set[] = {
string("pcm"))};
// 64 kbps, stereo
INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest,
::testing::ValuesIn(param_set));
INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
} // namespace webrtc