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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
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13
modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
View File

@ -34,8 +34,7 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
encoded_bytes_(0),
encoding_time_ms_(0.0),
decoding_time_ms_(0.0),
out_file_(NULL) {
}
out_file_(NULL) {}
void AudioCodecSpeedTest::SetUp() {
channels_ = get<0>(GetParam());
@ -52,8 +51,8 @@ void AudioCodecSpeedTest::SetUp() {
rewind(fp);
// Allocate memory to contain the whole file.
in_data_.reset(new int16_t[loop_length_samples_ +
input_length_sample_ * channels_]);
in_data_.reset(
new int16_t[loop_length_samples_ + input_length_sample_ * channels_]);
data_pointer_ = 0;
@ -111,11 +110,11 @@ void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
decoding_time_ms_ += time_ms;
if (save_out_data_) {
fwrite(&out_data_[0], sizeof(int16_t),
output_length_sample_ * channels_, out_file_);
fwrite(&out_data_[0], sizeof(int16_t), output_length_sample_ * channels_,
out_file_);
}
data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
loop_length_samples_;
loop_length_samples_;
time_now_ms += block_duration_ms_;
}

9
modules/audio_coding/codecs/tools/audio_codec_speed_test.h
View File

@ -36,15 +36,18 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
// 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size,
// 3. assign |encoded_bytes| with the length of the bit stream (in bytes),
// 4. return the cost of time (in millisecond) spent on actual encoding.
virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t max_bytes, size_t* encoded_bytes) = 0;
virtual float EncodeABlock(int16_t* in_data,
uint8_t* bit_stream,
size_t max_bytes,
size_t* encoded_bytes) = 0;
// DecodeABlock(...) does the following:
// 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes|
// (in bytes),
// 2. save the decoded audio in |out_data|,
// 3. return the cost of time (in millisecond) spent on actual decoding.
virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
virtual float DecodeABlock(const uint8_t* bit_stream,
size_t encoded_bytes,
int16_t* out_data) = 0;
// Encoding and decode an audio of |audio_duration| (in seconds) and