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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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34
modules/audio_coding/neteq/include/neteq.h
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@ -33,25 +33,25 @@ class AudioFrame;
class AudioDecoderFactory;
struct NetEqNetworkStatistics {
uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
uint16_t preferred_buffer_size_ms; // Target buffer size in ms.
uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
// jitter; 0 otherwise.
uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
uint16_t expand_rate; // Fraction (of original stream) of synthesized
// audio inserted through expansion (in Q14).
uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
// jitter; 0 otherwise.
uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
uint16_t expand_rate; // Fraction (of original stream) of synthesized
// audio inserted through expansion (in Q14).
uint16_t speech_expand_rate; // Fraction (of original stream) of synthesized
// speech inserted through expansion (in Q14).
uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration
// (in Q14).
uint16_t secondary_decoded_rate; // Fraction of data coming from FEC/RED
// decoding (in Q14).
uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration
// (in Q14).
uint16_t secondary_decoded_rate; // Fraction of data coming from FEC/RED
// decoding (in Q14).
uint16_t secondary_discarded_rate; // Fraction of discarded FEC/RED data (in
// Q14).
int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
// (positive or negative).
int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
// (positive or negative).
size_t added_zero_samples; // Number of zero samples added in "off" mode.
// Statistics for packet waiting times, i.e., the time between a packet
// arrives until it is decoded.
@ -104,11 +104,7 @@ class NetEq {
absl::optional<AudioCodecPairId> codec_pair_id;
};
enum ReturnCodes {
kOK = 0,
kFail = -1,
kNotImplemented = -2
};
enum ReturnCodes { kOK = 0, kFail = -1, kNotImplemented = -2 };
// Creates a new NetEq object, with parameters set in |config|. The |config|
// object will only have to be valid for the duration of the call to this