commit 665174fdbb4e0540eccb27cf7412348f1b65534c
author Yves Gerey <yvesg@webrtc.org> Tue Jun 19 15:03:05 2018 +0200
committer Yves Gerey <yvesg@webrtc.org> Tue Jun 19 14:00:39 2018 +0000
tree 34a9eb770e6de54c282f62ad0b65d2cd69c917f6
parent b602123a5aae1a9f0b6b581e611acf44eb525c08

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}

modules/audio_coding/codecs/opus/opus_interface.h

diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h
index 4b8e892..0b1c64d 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -318,8 +318,10 @@
  * 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 @@
  * 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 @@
  *                              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);
 
 /****************************************************************************