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

webrtc/modules/utility/source/coder.h

Index: webrtc/modules/utility/source/coder.h
diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h
index 9536a027d0e913aad4b50c3c3db77365eaa85cf8..cd17574784b644352c3c23f90f02718d1bc0f8fe 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;

[perkj_webrtc, 2016/05/04 10:43:55]

fix the naming while your are at it? acm_ here and below. 

[kwiberg-webrtc, 2016/05/04 10:49:04]

Sure, but can I do it in a follow-up CL? One of the main advantages of a CL with
nothing but clang-format changes is that you can be very sure that no mistakes
have crept in, since no human has touched anything.

+  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_