video coding test: Adding MT functionality

Review URL: http://webrtc-codereview.appspot.com/135008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@570 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/video_coding/main/test/mt_test_common.cc

@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2011 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 "mt_test_common.h"
+
+#include <cmath>
+
+#include "rtp_dump.h"
+
+
+TransportCallback::TransportCallback(webrtc::RtpRtcp* rtp,
+                                     const char* filename):
+RTPSendCompleteCallback(rtp, filename)
+{
+    //
+}
+
+TransportCallback::~TransportCallback()
+{
+    //
+}
+
+int
+TransportCallback::SendPacket(int channel, const void *data, int len)
+{
+    _sendCount++;
+    _totalSentLength += len;
+
+    if (_rtpDump != NULL)
+    {
+        if (_rtpDump->DumpPacket((const WebRtc_UWord8*)data, len) != 0)
+        {
+            return -1;
+        }
+    }
+
+    bool transmitPacket = true;
+    // Off-line tests, don't drop first Key frame (approx.)
+    if (_sendCount > 20)
+    {
+        transmitPacket = PacketLoss();
+    }
+
+    WebRtc_UWord64 now = VCMTickTime::MillisecondTimestamp();
+    // Insert outgoing packet into list
+    if (transmitPacket)
+    {
+        rtpPacket* newPacket = new rtpPacket();
+        memcpy(newPacket->data, data, len);
+        newPacket->length = len;
+        // Simulate receive time = network delay + packet jitter
+        // simulated as a Normal distribution random variable with
+        // mean = networkDelay and variance = jitterVar
+        WebRtc_Word32
+        simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
+                                                   sqrt(_jitterVar));
+        newPacket->receiveTime = now + simulatedDelay;
+        _rtpPackets.PushBack(newPacket);
+    }
+    return 0;
+}
+
+int
+TransportCallback::TransportPackets()
+{
+    // Are we ready to send packets to the receiver?
+    rtpPacket* packet = NULL;
+    WebRtc_UWord64 now = VCMTickTime::MillisecondTimestamp();
+
+    while (!_rtpPackets.Empty())
+    {
+        // Take first packet in list
+        packet = static_cast<rtpPacket*>((_rtpPackets.First())->GetItem());
+        WebRtc_Word64 timeToReceive = packet->receiveTime - now;
+        if (timeToReceive > 0)
+        {
+            // No available packets to send
+            break;
+        }
+
+        _rtpPackets.PopFront();
+        // Send to receive side
+        if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
+                                     packet->length) < 0)
+        {
+            delete packet;
+            packet = NULL;
+            // Will return an error after the first packet that goes wrong
+            return -1;
+        }
+        delete packet;
+        packet = NULL;
+    }
+    return 0; // OK
+}
+
+
+
+bool VCMProcessingThread(void* obj)
+{
+    SharedRTPState* state = static_cast<SharedRTPState*>(obj);
+    if (state->_vcm.TimeUntilNextProcess() <= 0)
+    {
+        if (state->_vcm.Process() < 0)
+        {
+            return false;
+        }
+    }
+    return true;
+}
+
+
+bool VCMDecodeThread(void* obj)
+{
+    SharedRTPState* state = static_cast<SharedRTPState*>(obj);
+    state->_vcm.Decode();
+    return true;
+}
+
+bool TransportThread(void *obj)
+{
+    SharedTransportState* state = static_cast<SharedTransportState*>(obj);
+    state->_transport.TransportPackets();
+    return true;
+}