zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Reformat the WebRTC code base

Browse Source 
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

38
modules/audio_coding/test/opus_test.cc
View File

@ -211,8 +211,11 @@ void OpusTest::Perform() {
#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 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
// 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 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
}
// 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 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
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::Run(TestPackStereo* channel, size_t channels, int bitrate,
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");
}