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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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12
modules/audio_coding/neteq/decision_logic.cc
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@ -95,7 +95,7 @@ void DecisionLogic::SoftReset() {
void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
// TODO(hlundin): Change to an enumerator and skip assert.
assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
fs_mult_ = fs_hz / 8000;
output_size_samples_ = output_size_samples;
}
@ -122,11 +122,11 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
const size_t cur_size_samples =
samples_left + packet_buffer_.NumSamplesInBuffer(decoder_frame_length);
prev_time_scale_ = prev_time_scale_ &&
(prev_mode == kModeAccelerateSuccess ||
prev_mode == kModeAccelerateLowEnergy ||
prev_mode == kModePreemptiveExpandSuccess ||
prev_mode == kModePreemptiveExpandLowEnergy);
prev_time_scale_ =
prev_time_scale_ && (prev_mode == kModeAccelerateSuccess ||
prev_mode == kModeAccelerateLowEnergy ||
prev_mode == kModePreemptiveExpandSuccess ||
prev_mode == kModePreemptiveExpandLowEnergy);
FilterBufferLevel(cur_size_samples, prev_mode);