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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

15
modules/audio_coding/codecs/opus/opus_interface.h
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@ -318,8 +318,10 @@ void WebRtcOpus_DecoderInit(OpusDecInst* inst);
* Return value : >0 - Samples per channel in decoded vector
* -1 - Error
*/
int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
size_t encoded_bytes, int16_t* decoded,
int WebRtcOpus_Decode(OpusDecInst* inst,
const uint8_t* encoded,
size_t encoded_bytes,
int16_t* decoded,
int16_t* audio_type);
/****************************************************************************
@ -336,7 +338,8 @@ int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
* Return value : >0 - number of samples in decoded PLC vector
* -1 - Error
*/
int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
int WebRtcOpus_DecodePlc(OpusDecInst* inst,
int16_t* decoded,
int number_of_lost_frames);
/****************************************************************************
@ -357,8 +360,10 @@ int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
* 0 - No FEC data in the packet
* -1 - Error
*/
int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
size_t encoded_bytes, int16_t* decoded,
int WebRtcOpus_DecodeFec(OpusDecInst* inst,
const uint8_t* encoded,
size_t encoded_bytes,
int16_t* decoded,
int16_t* audio_type);
/****************************************************************************