WebRtc_Word32 -> int32_t in utility/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1307005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3797 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/utility/interface/rtp_dump.h b/webrtc/modules/utility/interface/rtp_dump.h
index 9291a1c..9a852d0 100644
--- a/webrtc/modules/utility/interface/rtp_dump.h
+++ b/webrtc/modules/utility/interface/rtp_dump.h
@@ -31,10 +31,10 @@
// Open the file fileNameUTF8 for writing RTP/RTCP packets.
// Note: this API also adds the rtpplay header.
- virtual WebRtc_Word32 Start(const char* fileNameUTF8) = 0;
+ virtual int32_t Start(const char* fileNameUTF8) = 0;
// Close the existing file. No more packets will be recorded.
- virtual WebRtc_Word32 Stop() = 0;
+ virtual int32_t Stop() = 0;
// Return true if a file is open for recording RTP/RTCP packets.
virtual bool IsActive() const = 0;
@@ -42,8 +42,8 @@
// Writes the RTP/RTCP packet in packet with length packetLength in bytes.
// Note: packet should contain the RTP/RTCP part of the packet. I.e. the
// first bytes of packet should be the RTP/RTCP header.
- virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet,
- WebRtc_UWord16 packetLength) = 0;
+ virtual int32_t DumpPacket(const uint8_t* packet,
+ uint16_t packetLength) = 0;
protected:
virtual ~RtpDump();