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/neteq/rtcp.h b/modules/audio_coding/neteq/rtcp.h
index ce2035b..45bb058 100644
--- a/modules/audio_coding/neteq/rtcp.h
+++ b/modules/audio_coding/neteq/rtcp.h
@@ -22,9 +22,7 @@
class Rtcp {
public:
- Rtcp() {
- Init(0);
- }
+ Rtcp() { Init(0); }
~Rtcp() {}
@@ -39,17 +37,17 @@
void GetStatistics(bool no_reset, RtcpStatistics* stats);
private:
- uint16_t cycles_; // The number of wrap-arounds for the sequence number.
- uint16_t max_seq_no_; // The maximum sequence number received. Starts over
- // from 0 after wrap-around.
+ uint16_t cycles_; // The number of wrap-arounds for the sequence number.
+ uint16_t max_seq_no_; // The maximum sequence number received. Starts over
+ // from 0 after wrap-around.
uint16_t base_seq_no_; // The sequence number of the first received packet.
uint32_t received_packets_; // The number of packets that have been received.
uint32_t received_packets_prior_; // Number of packets received when last
// report was generated.
uint32_t expected_prior_; // Expected number of packets, at the time of the
// last report.
- int64_t jitter_; // Current jitter value in Q4.
- int32_t transit_; // Clock difference for previous packet.
+ int64_t jitter_; // Current jitter value in Q4.
+ int32_t transit_; // Clock difference for previous packet.
RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);
};