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/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index 0d082c8..08004ea 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -117,9 +117,9 @@
virtual void set_last_pack_cng_or_dtmf(int value);
private:
- static const int kLimitProbability = 53687091; // 1/20 in Q30.
+ static const int kLimitProbability = 53687091; // 1/20 in Q30.
static const int kLimitProbabilityStreaming = 536871; // 1/2000 in Q30.
- static const int kMaxStreamingPeakPeriodMs = 600000; // 10 minutes in ms.
+ static const int kMaxStreamingPeakPeriodMs = 600000; // 10 minutes in ms.
static const int kCumulativeSumDrift = 2; // Drift term for cumulative sum
// |iat_cumulative_sum_|.
// Steady-state forgetting factor for |iat_vector_|, 0.9993 in Q15.
@@ -146,28 +146,29 @@
bool first_packet_received_;
const size_t max_packets_in_buffer_; // Capacity of the packet buffer.
- IATVector iat_vector_; // Histogram of inter-arrival times.
+ IATVector iat_vector_; // Histogram of inter-arrival times.
int iat_factor_; // Forgetting factor for updating the IAT histogram (Q15).
const TickTimer* tick_timer_;
// Time elapsed since last packet.
std::unique_ptr<TickTimer::Stopwatch> packet_iat_stopwatch_;
- int base_target_level_; // Currently preferred buffer level before peak
- // detection and streaming mode (Q0).
+ int base_target_level_; // Currently preferred buffer level before peak
+ // detection and streaming mode (Q0).
// TODO(turajs) change the comment according to the implementation of
// minimum-delay.
- int target_level_; // Currently preferred buffer level in (fractions)
- // of packets (Q8), before adding any extra delay.
+ int target_level_; // Currently preferred buffer level in (fractions)
+ // of packets (Q8), before adding any extra delay.
int packet_len_ms_; // Length of audio in each incoming packet [ms].
bool streaming_mode_;
- uint16_t last_seq_no_; // Sequence number for last received packet.
- uint32_t last_timestamp_; // Timestamp for the last received packet.
- int minimum_delay_ms_; // Externally set minimum delay.
+ uint16_t last_seq_no_; // Sequence number for last received packet.
+ uint32_t last_timestamp_; // Timestamp for the last received packet.
+ int minimum_delay_ms_; // Externally set minimum delay.
int least_required_delay_ms_; // Smallest preferred buffer level (same unit
- // as |target_level_|), before applying
- // |minimum_delay_ms_| and/or |maximum_delay_ms_|.
- int maximum_delay_ms_; // Externally set maximum allowed delay.
- int iat_cumulative_sum_; // Cumulative sum of delta inter-arrival times.
- int max_iat_cumulative_sum_; // Max of |iat_cumulative_sum_|.
+ // as |target_level_|), before applying
+ // |minimum_delay_ms_| and/or
+ // |maximum_delay_ms_|.
+ int maximum_delay_ms_; // Externally set maximum allowed delay.
+ int iat_cumulative_sum_; // Cumulative sum of delta inter-arrival times.
+ int max_iat_cumulative_sum_; // Max of |iat_cumulative_sum_|.
// Time elapsed since maximum was observed.
std::unique_ptr<TickTimer::Stopwatch> max_iat_stopwatch_;
DelayPeakDetector& peak_detector_;