commit 0946a56023d821e0deca04029bb016ae1f23aa82
author pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Apr 09 00:28:06 2013 +0000
committer pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Tue Apr 09 00:28:06 2013 +0000
tree 504a7ac85144a61e871e5858ffdae54fddd99815
parent 6faf71d27b89ce4fc29ebd8148fc2ffd90983a61

WebRtc_Word32 => int32_t etc. in audio_coding/

BUG=314

Review URL: https://webrtc-codereview.appspot.com/1271006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h

diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
index f407a6b..9b58d4d 100644
--- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ b/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
@@ -26,22 +26,22 @@
 // TestPacketization callback which writes the encoded payloads to file
 class TestPacketization: public AudioPacketizationCallback {
  public:
-  TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
+  TestPacketization(RTPStream *rtpStream, uint16_t frequency);
   ~TestPacketization();
-  virtual WebRtc_Word32 SendData(const FrameType frameType,
-                                 const WebRtc_UWord8 payloadType,
-                                 const WebRtc_UWord32 timeStamp,
-                                 const WebRtc_UWord8* payloadData,
-                                 const WebRtc_UWord16 payloadSize,
-                                 const RTPFragmentationHeader* fragmentation);
+  virtual int32_t SendData(const FrameType frameType,
+                           const uint8_t payloadType,
+                           const uint32_t timeStamp,
+                           const uint8_t* payloadData,
+                           const uint16_t payloadSize,
+                           const RTPFragmentationHeader* fragmentation);
 
  private:
-  static void MakeRTPheader(WebRtc_UWord8* rtpHeader, WebRtc_UWord8 payloadType,
-                            WebRtc_Word16 seqNo, WebRtc_UWord32 timeStamp,
-                            WebRtc_UWord32 ssrc);
+  static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
+                            int16_t seqNo, uint32_t timeStamp,
+                            uint32_t ssrc);
   RTPStream* _rtpStream;
-  WebRtc_Word32 _frequency;
-  WebRtc_Word16 _seqNo;
+  int32_t _frequency;
+  int16_t _seqNo;
 };
 
 class Sender {
@@ -54,8 +54,8 @@
   bool Process();
 
   //for auto_test and logging
-  WebRtc_UWord8 testMode;
-  WebRtc_UWord8 codeId;
+  uint8_t testMode;
+  uint8_t codeId;
 
  private:
   AudioCodingModule* _acm;
@@ -74,22 +74,22 @@
   bool PlayoutData();
 
   //for auto_test and logging
-  WebRtc_UWord8 codeId;
-  WebRtc_UWord8 testMode;
+  uint8_t codeId;
+  uint8_t testMode;
 
  private:
   AudioCodingModule* _acm;
   RTPStream* _rtpStream;
   PCMFile _pcmFile;
-  WebRtc_Word16* _playoutBuffer;
-  WebRtc_UWord16 _playoutLengthSmpls;
-  WebRtc_UWord8 _incomingPayload[MAX_INCOMING_PAYLOAD];
-  WebRtc_UWord16 _payloadSizeBytes;
-  WebRtc_UWord16 _realPayloadSizeBytes;
-  WebRtc_Word32 _frequency;
+  int16_t* _playoutBuffer;
+  uint16_t _playoutLengthSmpls;
+  uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
+  uint16_t _payloadSizeBytes;
+  uint16_t _realPayloadSizeBytes;
+  int32_t _frequency;
   bool _firstTime;
   WebRtcRTPHeader _rtpInfo;
-  WebRtc_UWord32 _nextTime;
+  uint32_t _nextTime;
 };
 
 class EncodeDecodeTest: public ACMTest {
@@ -98,8 +98,8 @@
   EncodeDecodeTest(int testMode);
   virtual void Perform();
 
-  WebRtc_UWord16 _playoutFreq;
-  WebRtc_UWord8 _testMode;
+  uint16_t _playoutFreq;
+  uint8_t _testMode;
 
  private:
   void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);