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/source/video_coding_test.gyp

@@ -31,6 +31,7 @@
         '../test/generic_codec_test.h',
         '../test/jitter_estimate_test.h',
         '../test/media_opt_test.h',
+        '../test/mt_test_common.h',
         '../test/normal_test.h',
         '../test/quality_modes_test.h',
         '../test/receiver_tests.h',
@@ -46,6 +47,7 @@
         '../test/generic_codec_test.cc',
         '../test/jitter_buffer_test.cc',
         '../test/media_opt_test.cc',
+        '../test/mt_test_common.cc',
         '../test/mt_rx_tx_test.cc',
         '../test/normal_test.cc',
         '../test/quality_modes_test.cc',

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

@@ -14,16 +14,17 @@
  *
  **************************************************/
 
-#include "receiver_tests.h" // shared RTP state and receive side threads
-#include "video_coding.h"
-#include "rtp_rtcp.h"
-#include "thread_wrapper.h"
+#include <string.h>
+
 #include "../source/event.h"
+#include "media_opt_test.h"
+#include "mt_test_common.h"
+#include "receiver_tests.h" // shared RTP state and receive side threads
+#include "rtp_rtcp.h"
 #include "test_macros.h"
 #include "test_util.h" // send side callback
-#include "media_opt_test.h"
-
-#include <string.h>
+#include "thread_wrapper.h"
+#include "video_coding.h"
 
 using namespace webrtc;
 

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;
+}

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

@@ -0,0 +1,86 @@
+/*
+ *  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.
+ */
+
+/*
+ * Common multi-thread functionality across video coding module tests
+ */
+
+#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
+
+#include "rtp_rtcp.h"
+#include "test_util.h"
+#include "video_coding.h"
+
+using namespace webrtc;
+
+class SendSharedState
+{
+public:
+    SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp,
+            CmdArgs args) :
+            _vcm(vcm),
+            _rtp(rtp),
+            _args(args),
+            _sourceFile(NULL),
+            _frameCnt(0),
+            _timestamp(0) {}
+
+    webrtc::VideoCodingModule&  _vcm;
+    webrtc::RtpRtcp&            _rtp;
+    CmdArgs                     _args;
+    FILE*                       _sourceFile;
+    WebRtc_Word32               _frameCnt;
+    WebRtc_Word32               _timestamp;
+};
+
+// MT implementation of the RTPSendCompleteCallback (Transport)
+class TransportCallback:public RTPSendCompleteCallback
+{
+ public:
+    // constructor input: (receive side) rtp module to send encoded data to
+    TransportCallback(webrtc::RtpRtcp* rtp,
+                      const char* filename = NULL);
+     virtual ~TransportCallback();
+     // Add packets to list
+     // Incorporate network conditions - delay and packet loss
+     // Actual transmission will occur on a separate thread
+     int SendPacket(int channel, const void *data, int len);
+     // Send to the receiver packets which are ready to be submitted
+     int TransportPackets();
+};
+
+class SharedRTPState
+{
+public:
+    SharedRTPState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp) :
+        _vcm(vcm),
+        _rtp(rtp) {}
+    webrtc::VideoCodingModule&  _vcm;
+    webrtc::RtpRtcp&            _rtp;
+};
+
+
+class SharedTransportState
+{
+public:
+    SharedTransportState(webrtc::RtpRtcp& rtp, TransportCallback& transport):
+        _rtp(rtp),
+        _transport(transport) {}
+    webrtc::RtpRtcp&            _rtp;
+    TransportCallback&          _transport;
+};
+
+bool VCMProcessingThread(void* obj);
+bool VCMDecodeThread(void* obj);
+bool TransportThread(void *obj);
+
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_

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

@@ -63,15 +63,6 @@ public:
     RTPPlayer&              _rtpPlayer;
 };
 
-class SharedRTPState
-{
-public:
-    SharedRTPState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp) :
-        _vcm(vcm),
-        _rtp(rtp) {}
-    webrtc::VideoCodingModule&  _vcm;
-    webrtc::RtpRtcp&            _rtp;
-};
 
 int RtpPlay(CmdArgs& args);
 int RtpPlayMT(CmdArgs& args,

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

@@ -249,65 +249,6 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
 {
     _sendCount++;
     _totalSentLength += len;
-    bool transmitPacket = true;
-
-    // Packet Loss
-
-    if (_burstLength <= 1.0)
-    {
-        // Random loss: if _burstLength parameter is not set, or <=1
-        if (PacketLoss(_lossPct))
-        {
-            // drop
-            transmitPacket = false;
-        }
-    }
-    else
-    {
-        // Simulate bursty channel (Gilbert model)
-        // (1st order) Markov chain model with memory of the previous/last
-        // packet state (loss or received)
-
-        // 0 = received state
-        // 1 = loss state
-
-        // probTrans10: if previous packet is lost, prob. to -> received state
-        // probTrans11: if previous packet is lost, prob. to -> loss state
-
-        // probTrans01: if previous packet is received, prob. to -> loss state
-        // probTrans00: if previous packet is received, prob. to -> received
-
-        // Map the two channel parameters (average loss rate and burst length)
-        // to the transition probabilities:
-        double probTrans10 = 100 * (1.0 / _burstLength);
-        double probTrans11 = (100.0 - probTrans10);
-        double probTrans01 = (probTrans10 * ( _lossPct / (100.0 - _lossPct)));
-
-        // Note: Random loss (Bernoulli) model is a special case where:
-        // burstLength = 100.0 / (100.0 - _lossPct) (i.e., p10 + p01 = 100)
-
-        if (_prevLossState == 0 )
-        {
-            // previous packet was received
-            if (PacketLoss(probTrans01))
-            {
-                // drop, update previous state to loss
-                _prevLossState = 1;
-                transmitPacket = false;
-            }
-        }
-        else if (_prevLossState == 1)
-        {
-            _prevLossState = 0;
-            // previous packet was lost
-            if (PacketLoss(probTrans11))
-            {
-                // drop, update previous state to loss
-                _prevLossState = 1;
-                transmitPacket = false;
-            }
-        }
-    }
 
     if (_rtpDump != NULL)
     {
@@ -317,6 +258,8 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
         }
     }
 
+    bool transmitPacket = PacketLoss();
+
     WebRtc_UWord64 now = VCMTickTime::MillisecondTimestamp();
     // Insert outgoing packet into list
     if (transmitPacket)
@@ -367,11 +310,8 @@ RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
 int
 RTPSendCompleteCallback::SendRTCPPacket(int channel, const void *data, int len)
 {
-    if (_rtp->IncomingPacket((const WebRtc_UWord8*)data, len) == 0)
-    {
-        return len;
-    }
-    return -1;
+    // Incorporate network conditions
+    return SendPacket(channel, data, len);
 }
 
 void
@@ -389,7 +329,70 @@ RTPSendCompleteCallback::SetBurstLength(double burstLength)
 }
 
 bool
-RTPSendCompleteCallback::PacketLoss(double lossPct)
+RTPSendCompleteCallback::PacketLoss()
+{
+    bool transmitPacket = true;
+    if (_burstLength <= 1.0)
+    {
+        // Random loss: if _burstLength parameter is not set, or <=1
+        if (UnifomLoss(_lossPct))
+        {
+            // drop
+            transmitPacket = false;
+        }
+    }
+    else
+    {
+        // Simulate bursty channel (Gilbert model)
+        // (1st order) Markov chain model with memory of the previous/last
+        // packet state (loss or received)
+
+        // 0 = received state
+        // 1 = loss state
+
+        // probTrans10: if previous packet is lost, prob. to -> received state
+        // probTrans11: if previous packet is lost, prob. to -> loss state
+
+        // probTrans01: if previous packet is received, prob. to -> loss state
+        // probTrans00: if previous packet is received, prob. to -> received
+
+        // Map the two channel parameters (average loss rate and burst length)
+        // to the transition probabilities:
+        double probTrans10 = 100 * (1.0 / _burstLength);
+        double probTrans11 = (100.0 - probTrans10);
+        double probTrans01 = (probTrans10 * ( _lossPct / (100.0 - _lossPct)));
+
+        // Note: Random loss (Bernoulli) model is a special case where:
+        // burstLength = 100.0 / (100.0 - _lossPct) (i.e., p10 + p01 = 100)
+
+        if (_prevLossState == 0 )
+        {
+            // previous packet was received
+            if (UnifomLoss(probTrans01))
+            {
+                // drop, update previous state to loss
+                _prevLossState = 1;
+                transmitPacket = false;
+            }
+        }
+        else if (_prevLossState == 1)
+        {
+            _prevLossState = 0;
+            // previous packet was lost
+            if (UnifomLoss(probTrans11))
+            {
+                // drop, update previous state to loss
+                _prevLossState = 1;
+                transmitPacket = false;
+             }
+        }
+    }
+    return transmitPacket;
+}
+
+
+bool
+RTPSendCompleteCallback::UnifomLoss(double lossPct)
 {
     double randVal = (std::rand() + 1.0)/(RAND_MAX + 1.0);
     return randVal < lossPct/100;
@@ -402,7 +405,6 @@ PacketRequester::ResendPackets(const WebRtc_UWord16* sequenceNumbers,
     return _rtp.SendNACK(sequenceNumbers, length);
 }
 
-
 RTPVideoCodecTypes
 ConvertCodecType(const char* plname)
 {

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

@@ -51,6 +51,7 @@ public:
 
 // forward declaration
 int MTRxTxTest(CmdArgs& args);
+double NormalDist(double mean, double stdDev);
 namespace webrtc
 {
     class RtpDump;
@@ -64,7 +65,7 @@ namespace webrtc
  2. EncodeComplete callback:
  2a. Transfer encoded data directly to the decoder
  2b. Pass encoded data via the RTP module
- 3. Caluclate PSNR from file function (for now: does not deal with frame drops)
+ 3. Calculate PSNR from file function (for now: does not deal with frame drops)
  */
 
 // Send Side - Packetization callback - send an encoded frame to the VCMReceiver
@@ -215,9 +216,11 @@ public:
     int SendCount() {return _sendCount; }
     // Return accumulated length in bytes of transmitted packets
     WebRtc_UWord32 TotalSentLength() {return _totalSentLength;}
-private:
+protected:
     // Randomly decide whether to drop packets, based on the channel model
-    bool PacketLoss(double lossPct);
+    bool PacketLoss();
+    // Random uniform loss model
+    bool UnifomLoss(double lossPct);
 
     WebRtc_UWord32          _sendCount;
     webrtc::RtpRtcp*        _rtp;
@@ -231,27 +234,6 @@ private:
     webrtc::RtpDump*        _rtpDump;
 };
 
-// Used in multi thread test
-class SendSharedState
-{
-public:
-    SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp,
-            CmdArgs args) :
-            _vcm(vcm),
-            _rtp(rtp),
-            _args(args),
-            _sourceFile(NULL),
-            _frameCnt(0),
-            _timestamp(0) {}
-
-    webrtc::VideoCodingModule&  _vcm;
-    webrtc::RtpRtcp&            _rtp;
-    CmdArgs                     _args;
-    FILE*                       _sourceFile;
-    WebRtc_Word32               _frameCnt;
-    WebRtc_Word32               _timestamp;
-};
-
 class PacketRequester: public webrtc::VCMPacketRequestCallback
 {
 public: