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/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index d3afd6b..40b5147 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -211,8 +211,11 @@
#endif
}
-void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
- size_t frame_length, int percent_loss) {
+void OpusTest::Run(TestPackStereo* channel,
+ size_t channels,
+ int bitrate,
+ size_t frame_length,
+ int percent_loss) {
AudioFrame audio_frame;
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
const size_t kBufferSizeSamples = 480 * 12 * 2; // 120 ms stereo audio.
@@ -237,8 +240,8 @@
// default.
const int kOpusComplexity5 = 5;
EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
- EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_,
- kOpusComplexity5));
+ EXPECT_EQ(0,
+ WebRtcOpus_SetComplexity(opus_stereo_encoder_, kOpusComplexity5));
#endif
// Fast-forward 1 second (100 blocks) since the files start with silence.
@@ -263,19 +266,16 @@
}
// If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
- EXPECT_EQ(480,
- resampler_.Resample10Msec(audio_frame.data(),
- audio_frame.sample_rate_hz_,
- 48000,
- channels,
- kBufferSizeSamples - written_samples,
- &audio[written_samples]));
+ EXPECT_EQ(480, resampler_.Resample10Msec(
+ audio_frame.data(), audio_frame.sample_rate_hz_, 48000,
+ channels, kBufferSizeSamples - written_samples,
+ &audio[written_samples]));
written_samples += 480 * channels;
// Sometimes we need to loop over the audio vector to produce the right
// number of packets.
- size_t loop_encode = (written_samples - read_samples) /
- (channels * frame_length);
+ size_t loop_encode =
+ (written_samples - read_samples) / (channels * frame_length);
if (loop_encode > 0) {
const size_t kMaxBytes = 1000; // Maximum number of bytes for one packet.
@@ -319,9 +319,9 @@
opus_stereo_decoder_, bitstream, bitstream_len_byte,
&out_audio[decoded_samples * channels], &audio_type);
} else {
- decoded_samples += WebRtcOpus_DecodePlc(
- opus_stereo_decoder_, &out_audio[decoded_samples * channels],
- 1);
+ decoded_samples +=
+ WebRtcOpus_DecodePlc(opus_stereo_decoder_,
+ &out_audio[decoded_samples * channels], 1);
}
}
@@ -377,14 +377,14 @@
void OpusTest::OpenOutFile(int test_number) {
std::string file_name;
std::stringstream file_stream;
- file_stream << webrtc::test::OutputPath() << "opustest_out_"
- << test_number << ".pcm";
+ file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number
+ << ".pcm";
file_name = file_stream.str();
out_file_.Open(file_name, 48000, "wb");
file_stream.str("");
file_name = file_stream.str();
file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
- << test_number << ".pcm";
+ << test_number << ".pcm";
file_name = file_stream.str();
out_file_standalone_.Open(file_name, 48000, "wb");
}