commit 73987c9932b2b6a49bda467a448d428119e933f5
author kwiberg <kwiberg@webrtc.org> Wed May 04 05:12:19 2016 -0700
committer Commit bot <commit-bot@chromium.org> Wed May 04 12:12:26 2016 +0000
tree 39ac1be9da03d813553244c59f65d602d57db389
parent 053f91774149a5367ddd531999d4ca69a57dbaa3

Run "git cl format --full" on a pair of files with ancient formatting

Review-Url: https://codereview.webrtc.org/1946873003
Cr-Commit-Position: refs/heads/master@{#12625}

webrtc/modules/utility/source/coder.h

diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h
index 9536a02..cd17574 100644
--- a/webrtc/modules/utility/source/coder.h
+++ b/webrtc/modules/utility/source/coder.h
@@ -22,45 +22,47 @@
 namespace webrtc {
 class AudioFrame;
 
-class AudioCoder : public AudioPacketizationCallback
-{
-public:
-    AudioCoder(uint32_t instanceID);
-    ~AudioCoder();
+class AudioCoder : public AudioPacketizationCallback {
+ public:
+  AudioCoder(uint32_t instanceID);
+  ~AudioCoder();
 
-    int32_t SetEncodeCodec(const CodecInst& codecInst);
+  int32_t SetEncodeCodec(const CodecInst& codecInst);
 
-    int32_t SetDecodeCodec(const CodecInst& codecInst);
+  int32_t SetDecodeCodec(const CodecInst& codecInst);
 
-    int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
-                   const int8_t* incomingPayload, size_t payloadLength);
+  int32_t Decode(AudioFrame& decodedAudio,
+                 uint32_t sampFreqHz,
+                 const int8_t* incomingPayload,
+                 size_t payloadLength);
 
-    int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
+  int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
 
-    int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
-                   size_t& encodedLengthInBytes);
+  int32_t Encode(const AudioFrame& audio,
+                 int8_t* encodedData,
+                 size_t& encodedLengthInBytes);
 
-protected:
- int32_t SendData(FrameType frameType,
-                  uint8_t payloadType,
-                  uint32_t timeStamp,
-                  const uint8_t* payloadData,
-                  size_t payloadSize,
-                  const RTPFragmentationHeader* fragmentation) override;
+ protected:
+  int32_t SendData(FrameType frameType,
+                   uint8_t payloadType,
+                   uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   size_t payloadSize,
+                   const RTPFragmentationHeader* fragmentation) override;
 
-private:
- std::unique_ptr<AudioCodingModule> _acm;
- acm2::CodecManager codec_manager_;
- acm2::RentACodec rent_a_codec_;
+ private:
+  std::unique_ptr<AudioCodingModule> _acm;
+  acm2::CodecManager codec_manager_;
+  acm2::RentACodec rent_a_codec_;
 
-    CodecInst _receiveCodec;
+  CodecInst _receiveCodec;
 
-    uint32_t _encodeTimestamp;
-    int8_t*  _encodedData;
-    size_t _encodedLengthInBytes;
+  uint32_t _encodeTimestamp;
+  int8_t* _encodedData;
+  size_t _encodedLengthInBytes;
 
-    uint32_t _decodeTimestamp;
+  uint32_t _decodeTimestamp;
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
+#endif  // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_