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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

8
modules/audio_coding/neteq/tools/rtp_generator.cc
View File

@ -32,8 +32,8 @@ uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
uint32_t this_send_time = next_send_time_ms_;
assert(samples_per_ms_ > 0);
next_send_time_ms_ += ((1.0 + drift_factor_) * payload_length_samples) /
samples_per_ms_;
next_send_time_ms_ +=
((1.0 + drift_factor_) * payload_length_samples) / samples_per_ms_;
return this_send_time;
}
@ -46,8 +46,8 @@ void RtpGenerator::set_drift_factor(double factor) {
uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
size_t payload_length_samples,
RTPHeader* rtp_header) {
uint32_t ret = RtpGenerator::GetRtpHeader(
payload_type, payload_length_samples, rtp_header);
uint32_t ret = RtpGenerator::GetRtpHeader(payload_type,
payload_length_samples, rtp_header);
if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
jump_from_timestamp_ &&
timestamp_ > jump_from_timestamp_) {