Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/test/testAPI/test_api.h

@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_types.h"
+#include "rtp_rtcp.h"
+#include "rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class FakeRtpRtcpClock : public RtpRtcpClock {
+ public:
+  FakeRtpRtcpClock() {
+    time_in_ms_ = 123456;
+  }
+  // Return a timestamp in milliseconds relative to some arbitrary
+  // source; the source is fixed for this clock.
+  virtual WebRtc_Word64 GetTimeInMS() {
+    return time_in_ms_;
+  }
+  // Retrieve an NTP absolute timestamp.
+  virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
+    secs = time_in_ms_ / 1000;
+    frac = (time_in_ms_ % 1000) * 4294967;
+  }
+  void IncrementTime(WebRtc_UWord32 time_increment_ms) {
+    time_in_ms_ += time_increment_ms;
+  }
+ private:
+  WebRtc_Word64 time_in_ms_;
+};
+
+// This class sends all its packet straight to the provided RtpRtcp module.
+// with optional packet loss.
+class LoopBackTransport : public webrtc::Transport {
+ public:
+  LoopBackTransport()
+    : _count(0),
+      _packetLoss(0),
+      _rtpRtcpModule(NULL) {
+  }
+  void SetSendModule(RtpRtcp* rtpRtcpModule) {
+    _rtpRtcpModule = rtpRtcpModule;
+  }
+  void DropEveryNthPacket(int n) {
+    _packetLoss = n;
+  }
+  virtual int SendPacket(int channel, const void *data, int len) {
+    _count++;
+    if (_packetLoss > 0) {
+      if ((_count % _packetLoss) == 0) {
+        return len;
+      }
+    }
+    if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
+      return len;
+    }
+    return -1;
+  }
+  virtual int SendRTCPPacket(int channel, const void *data, int len) {
+    if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
+      return len;
+    }
+    return -1;
+  }
+ private:
+  int _count;
+  int _packetLoss;
+  RtpRtcp* _rtpRtcpModule;
+};
+
+class RtpReceiver : public RtpData {
+ public:
+   virtual WebRtc_Word32 OnReceivedPayloadData(
+       const WebRtc_UWord8* payloadData,
+       const WebRtc_UWord16 payloadSize,
+       const webrtc::WebRtcRTPHeader* rtpHeader) {
+    return 0;
+  }
+};
+
+}  // namespace webrtc
+