Remove traces of deprecated WebRtc_Word types.

BUG=314
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1385004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3933 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/common_types.h

@@ -266,23 +266,23 @@ enum VadModes                 // degree of bandwidth reduction
 struct NetworkStatistics           // NETEQ statistics
 {
     // current jitter buffer size in ms
-    WebRtc_UWord16 currentBufferSize;
+    uint16_t currentBufferSize;
     // preferred (optimal) buffer size in ms
-    WebRtc_UWord16 preferredBufferSize;
+    uint16_t preferredBufferSize;
     // adding extra delay due to "peaky jitter"
     bool jitterPeaksFound;
     // loss rate (network + late) in percent (in Q14)
-    WebRtc_UWord16 currentPacketLossRate;
+    uint16_t currentPacketLossRate;
     // late loss rate in percent (in Q14)
-    WebRtc_UWord16 currentDiscardRate;
+    uint16_t currentDiscardRate;
     // fraction (of original stream) of synthesized speech inserted through
     // expansion (in Q14)
-    WebRtc_UWord16 currentExpandRate;
+    uint16_t currentExpandRate;
     // fraction of synthesized speech inserted through pre-emptive expansion
     // (in Q14)
-    WebRtc_UWord16 currentPreemptiveRate;
+    uint16_t currentPreemptiveRate;
     // fraction of data removed through acceleration (in Q14)
-    WebRtc_UWord16 currentAccelerateRate;
+    uint16_t currentAccelerateRate;
     // clock-drift in parts-per-million (negative or positive)
     int32_t clockDriftPPM;
     // average packet waiting time in the jitter buffer (ms)

webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -82,8 +82,7 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_METHOD1(SetRtxReceivePayloadType,
       void(int));
   MOCK_METHOD2(IncomingPacket,
-      int32_t(const WebRtc_UWord8* incomingPacket,
+      int32_t(const uint8_t* incomingPacket, const uint16_t packetLength));
-              const WebRtc_UWord16 packetLength));
   MOCK_METHOD4(IncomingAudioNTP,
       int32_t(const uint32_t audioReceivedNTPsecs,
               const uint32_t audioReceivedNTPfrac,

webrtc/modules/video_coding/main/test/receiver_tests.h

@@ -26,9 +26,9 @@ class RtpDataCallback : public webrtc::RtpData {
   RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {}
   virtual ~RtpDataCallback() {}
 
-  virtual WebRtc_Word32 OnReceivedPayloadData(
+  virtual int32_t OnReceivedPayloadData(
-      const WebRtc_UWord8* payload_data,
+      const uint8_t* payload_data,
-      const WebRtc_UWord16 payload_size,
+      const uint16_t payload_size,
       const webrtc::WebRtcRTPHeader* rtp_header) {
     return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
   }

webrtc/modules/video_coding/main/test/test_util.cc

@@ -111,7 +111,7 @@ FileOutputFrameReceiver::~FileOutputFrameReceiver() {
   }
 }
 
-WebRtc_Word32 FileOutputFrameReceiver::FrameToRender(
+int32_t FileOutputFrameReceiver::FrameToRender(
     webrtc::I420VideoFrame& video_frame) {
   if (timing_file_ == NULL) {
     std::string basename;

webrtc/modules/video_coding/main/test/test_util.h

@@ -87,7 +87,7 @@ class FileOutputFrameReceiver : public webrtc::VCMReceiveCallback {
   virtual ~FileOutputFrameReceiver();
 
   // VCMReceiveCallback
-  virtual WebRtc_Word32 FrameToRender(webrtc::I420VideoFrame& video_frame);
+  virtual int32_t FrameToRender(webrtc::I420VideoFrame& video_frame);
 
  private:
   std::string out_filename_;

webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc

@@ -60,21 +60,22 @@ class VcmPayloadSinkFactory::VcmPayloadSink
   }
 
   // PayloadSinkInterface
-  virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payload_data,
+  virtual int32_t OnReceivedPayloadData(
-      const WebRtc_UWord16 payload_size,
+      const uint8_t* payload_data,
+      const uint16_t payload_size,
       const WebRtcRTPHeader* rtp_header) {
     return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
   }
 
   // VCMPacketRequestCallback
-  virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequence_numbers,
+  virtual int32_t ResendPackets(const uint16_t* sequence_numbers,
-                                      WebRtc_UWord16 length) {
+                                uint16_t length) {
     stream_->ResendPackets(sequence_numbers, length);
     return 0;
   }
 
   // VCMFrameStorageCallback
-  virtual WebRtc_Word32 StoreReceivedFrame(
+  virtual int32_t StoreReceivedFrame(
       const EncodedVideoData& frame_to_store) {
     vcm_playback_->DecodeFromStorage(frame_to_store);
     return VCM_OK;

webrtc/typedefs.h

@@ -75,17 +75,6 @@ typedef unsigned int        uint32_t;
 typedef unsigned __int64    uint64_t;
 #endif
 
-// TODO(andrew): remove WebRtc_ types:
-// http://code.google.com/p/webrtc/issues/detail?id=314
-typedef int8_t              WebRtc_Word8;
-typedef int16_t             WebRtc_Word16;
-typedef int32_t             WebRtc_Word32;
-typedef int64_t             WebRtc_Word64;
-typedef uint8_t             WebRtc_UWord8;
-typedef uint16_t            WebRtc_UWord16;
-typedef uint32_t            WebRtc_UWord32;
-typedef uint64_t            WebRtc_UWord64;
-
 // Borrowed from Chromium's base/compiler_specific.h.
 // Annotate a virtual method indicating it must be overriding a virtual
 // method in the parent class.