Add unittests for ReceiverFec.

Also added mock for RtpReceiverVideo and did appropriate changes to allow for mocking. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/367017 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1566 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-01-30 09:03:37 +00:00 · 2012-01-30 09:03:37 +00:00 · 2fd1e1e40d
commit 2fd1e1e40d
parent 31ba407ac2
5 changed files with 420 additions and 8 deletions
--- a/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
+++ b/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
@ -0,0 +1,41 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtp_receiver_video.h"
+
+namespace webrtc {
+
+class MockRTPReceiverVideo : public RTPReceiverVideo {
+ public:
+  MOCK_METHOD1(ChangeUniqueId,
+      void(const WebRtc_Word32 id));
+  MOCK_METHOD3(ReceiveRecoveredPacketCallback,
+      WebRtc_Word32(WebRtcRTPHeader* rtpHeader,
+                    const WebRtc_UWord8* payloadData,
+                    const WebRtc_UWord16 payloadDataLength));
+  MOCK_METHOD3(CallbackOfReceivedPayloadData,
+      WebRtc_Word32(const WebRtc_UWord8* payloadData,
+                    const WebRtc_UWord16 payloadSize,
+                    const WebRtcRTPHeader* rtpHeader));
+  MOCK_CONST_METHOD0(TimeStamp,
+      WebRtc_UWord32());
+  MOCK_CONST_METHOD0(SequenceNumber,
+      WebRtc_UWord16());
+  MOCK_CONST_METHOD2(PayloadTypeToPayload,
+      WebRtc_UWord32(const WebRtc_UWord8 payloadType,
+                     ModuleRTPUtility::Payload*& payload));
+  MOCK_CONST_METHOD2(RetransmitOfOldPacket,
+      bool(const WebRtc_UWord16 sequenceNumber,
+           const WebRtc_UWord32 rtpTimeStamp));
+  MOCK_CONST_METHOD0(REDPayloadType,
+      WebRtc_Word8());
+};
+
+}  // namespace webrtc
--- a/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc
+++ b/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc
@ -0,0 +1,345 @@
+/*
+ *  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 <string.h>
+#include <list>
+
+#include "gmock/gmock.h"
+#include "gtest/gtest.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h"
+#include "modules/rtp_rtcp/source/receiver_fec.h"
+
+using ::testing::_;
+using ::testing::Args;
+using ::testing::ElementsAreArray;
+using ::testing::InSequence;
+
+namespace webrtc {
+
+typedef ForwardErrorCorrection::Packet Packet;
+
+enum { kRtpHeaderSize = 12 };
+enum { kFecPayloadType = 96 };
+enum { kRedPayloadType = 97 };
+enum { kVp8PayloadType = 120 };
+
+struct RtpPacket : public Packet {
+  WebRtcRTPHeader header;
+};
+
+class FrameGenerator {
+ public:
+  FrameGenerator() : num_packets_(0), seq_num_(0), timestamp_(0) {}
+
+  void NewFrame(int num_packets) {
+    num_packets_ = num_packets;
+    timestamp_ += 3000;
+  }
+
+  RtpPacket* NextPacket(int offset, size_t length) {
+    RtpPacket* rtp_packet = new RtpPacket;
+    for (size_t i = 0; i < length; ++i)
+      rtp_packet->data[i] = offset + i;
+    rtp_packet->length = length;
+    memset(&rtp_packet->header, 0, sizeof(WebRtcRTPHeader));
+    rtp_packet->header.frameType = kVideoFrameDelta;
+    rtp_packet->header.header.headerLength = kRtpHeaderSize;
+    rtp_packet->header.header.markerBit = (num_packets_ == 1);
+    rtp_packet->header.header.sequenceNumber = seq_num_;
+    rtp_packet->header.header.timestamp = timestamp_;
+    rtp_packet->header.header.payloadType = kVp8PayloadType;
+    BuildRtpHeader(rtp_packet->data, rtp_packet->header.header);
+    ++seq_num_;
+    --num_packets_;
+    return rtp_packet;
+  }
+
+  // Creates a new RtpPacket with the RED header added to the packet.
+  RtpPacket* BuildMediaRedPacket(const RtpPacket* packet) {
+    const int kHeaderLength = packet->header.header.headerLength;
+    RtpPacket* red_packet = new RtpPacket;
+    red_packet->header = packet->header;
+    red_packet->length = packet->length + 1;  // 1 byte RED header.
+    memset(red_packet->data, 0, red_packet->length);
+    // Copy RTP header.
+    memcpy(red_packet->data, packet->data, kHeaderLength);
+    SetRedHeader(red_packet, red_packet->data[1] & 0x7f, kHeaderLength);
+    memcpy(red_packet->data + kHeaderLength + 1, packet->data + kHeaderLength,
+           packet->length - kHeaderLength);
+    return red_packet;
+  }
+
+  // Creates a new RtpPacket with FEC payload and red header. Does this by
+  // creating a new fake media RtpPacket, clears the marker bit and adds a RED
+  // header. Finally replaces the payload with the content of |packet->data|.
+  RtpPacket* BuildFecRedPacket(const Packet* packet) {
+    // Create a fake media packet to get a correct header. 1 byte RED header.
+    ++num_packets_;
+    RtpPacket* red_packet = NextPacket(0, packet->length + 1);
+    red_packet->data[1] &= ~0x80;  // Clear marker bit.
+    const int kHeaderLength = red_packet->header.header.headerLength;
+    SetRedHeader(red_packet, kFecPayloadType, kHeaderLength);
+    memcpy(red_packet->data + kHeaderLength + 1, packet->data,
+           packet->length);
+    red_packet->length = kHeaderLength + 1 + packet->length;
+    return red_packet;
+  }
+
+  void SetRedHeader(Packet* red_packet, uint8_t payload_type,
+                    int header_length) const {
+    // Replace pltype.
+    red_packet->data[1] &= 0x80;  // Reset.
+    red_packet->data[1] += kRedPayloadType;  // Replace.
+
+    // Add RED header, f-bit always 0.
+    red_packet->data[header_length] = payload_type;
+  }
+
+ private:
+  void BuildRtpHeader(uint8_t* data, RTPHeader header) {
+    data[0] = 0x80;  // Version 2.
+    data[1] = header.payloadType;
+    data[1] |= (header.markerBit ? kRtpMarkerBitMask : 0);
+    ModuleRTPUtility::AssignUWord16ToBuffer(data+2, header.sequenceNumber);
+    ModuleRTPUtility::AssignUWord32ToBuffer(data+4, header.timestamp);
+    ModuleRTPUtility::AssignUWord32ToBuffer(data+8, header.ssrc);
+  }
+
+  int num_packets_;
+  uint16_t seq_num_;
+  uint32_t timestamp_;
+};
+
+class ReceiverFecTest : public ::testing::Test {
+ protected:
+  virtual void SetUp() {
+    fec_ = new ForwardErrorCorrection(0);
+    receiver_fec_ = new ReceiverFEC(0, &rtp_receiver_video_);
+    generator_ = new FrameGenerator();
+  }
+
+  virtual void TearDown() {
+    delete fec_;
+    delete receiver_fec_;
+    delete generator_;
+  }
+
+  void GenerateFrame(int num_media_packets,
+                     int frame_offset,
+                     std::list<RtpPacket*>* media_rtp_packets,
+                     std::list<Packet*>* media_packets) {
+    generator_->NewFrame(num_media_packets);
+    for (int i = 0; i < num_media_packets; ++i) {
+      media_rtp_packets->push_back(generator_->NextPacket(frame_offset + i,
+                                                          kRtpHeaderSize + 10));
+      media_packets->push_back(media_rtp_packets->back());
+    }
+  }
+
+  void VerifyReconstructedMediaPacket(const RtpPacket* packet, int times) {
+    // Verify that the content of the reconstructed packet is equal to the
+    // content of |packet|, and that the same content is received |times| number
+    // of times in a row.
+    EXPECT_CALL(rtp_receiver_video_,
+                ReceiveRecoveredPacketCallback(_, _,
+                                               packet->length - kRtpHeaderSize))
+        .With(Args<1, 2>(ElementsAreArray(packet->data + kRtpHeaderSize,
+                                          packet->length - kRtpHeaderSize)))
+        .Times(times);
+  }
+
+  void BuildAndAddRedMediaPacket(RtpPacket* packet) {
+    RtpPacket* red_packet = generator_->BuildMediaRedPacket(packet);
+    bool is_fec = false;
+    EXPECT_EQ(0, receiver_fec_->AddReceivedFECPacket(&red_packet->header,
+                                                     red_packet->data,
+                                                     red_packet->length -
+                                                     kRtpHeaderSize,
+                                                     is_fec,
+                                                     false));
+    delete red_packet;
+    EXPECT_FALSE(is_fec);
+  }
+
+  void BuildAndAddRedFecPacket(Packet* packet) {
+    RtpPacket* red_packet = generator_->BuildFecRedPacket(packet);
+    bool is_fec = false;
+    EXPECT_EQ(0, receiver_fec_->AddReceivedFECPacket(&red_packet->header,
+                                                     red_packet->data,
+                                                     red_packet->length -
+                                                     kRtpHeaderSize,
+                                                     is_fec,
+                                                     false));
+    delete red_packet;
+    EXPECT_TRUE(is_fec);
+  }
+
+  ForwardErrorCorrection* fec_;
+  MockRTPReceiverVideo rtp_receiver_video_;
+  ReceiverFEC* receiver_fec_;
+  FrameGenerator* generator_;
+};
+
+void DeletePackets(std::list<Packet*>* packets) {
+  while (!packets->empty()) {
+    delete packets->front();
+    packets->pop_front();
+  }
+}
+
+TEST_F(ReceiverFecTest, TwoMediaOneFec) {
+  const unsigned int kNumFecPackets = 1u;
+  const unsigned int kNumMediaPackets = 2u;
+  std::list<RtpPacket*> media_rtp_packets;
+  std::list<Packet*> media_packets;
+  GenerateFrame(2, 0, &media_rtp_packets, &media_packets);
+  std::list<Packet*> fec_packets;
+  EXPECT_EQ(0, fec_->GenerateFEC(media_packets,
+                                 kNumFecPackets * 255 / kNumMediaPackets,
+                                 0,
+                                 false,
+                                 &fec_packets));
+  ASSERT_EQ(kNumFecPackets, fec_packets.size());
+
+  // Recovery
+  receiver_fec_->SetPayloadTypeFEC(kFecPayloadType);
+  std::list<RtpPacket*>::iterator media_it = media_rtp_packets.begin();
+  BuildAndAddRedMediaPacket(*media_it);
+  // Drop one media packet.
+  std::list<Packet*>::iterator fec_it = fec_packets.begin();
+  BuildAndAddRedFecPacket(*fec_it);
+  {
+    InSequence s;
+    std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+    VerifyReconstructedMediaPacket(*it, 1);
+    ++it;
+    VerifyReconstructedMediaPacket(*it, 1);
+  }
+  EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC(false));
+
+  DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, TwoMediaTwoFec) {
+  const unsigned int kNumFecPackets = 2u;
+  const unsigned int kNumMediaPackets = 2u;
+  std::list<RtpPacket*> media_rtp_packets;
+  std::list<Packet*> media_packets;
+  GenerateFrame(2, 0, &media_rtp_packets, &media_packets);
+  std::list<Packet*> fec_packets;
+  EXPECT_EQ(0, fec_->GenerateFEC(media_packets,
+                                 kNumFecPackets * 255 / kNumMediaPackets,
+                                 0,
+                                 false, &fec_packets));
+  ASSERT_EQ(kNumFecPackets, fec_packets.size());
+
+  // Recovery
+  // Drop both media packets.
+  receiver_fec_->SetPayloadTypeFEC(kFecPayloadType);
+  std::list<Packet*>::iterator fec_it = fec_packets.begin();
+  BuildAndAddRedFecPacket(*fec_it);
+  ++fec_it;
+  BuildAndAddRedFecPacket(*fec_it);
+  {
+    InSequence s;
+    std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+    VerifyReconstructedMediaPacket(*it, 1);
+    ++it;
+    VerifyReconstructedMediaPacket(*it, 1);
+  }
+  EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC(false));
+
+  DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, TwoFramesOneFec) {
+  const unsigned int kNumFecPackets = 1u;
+  const unsigned int kNumMediaPackets = 2u;
+  std::list<RtpPacket*> media_rtp_packets;
+  std::list<Packet*> media_packets;
+  GenerateFrame(1, 0, &media_rtp_packets, &media_packets);
+  GenerateFrame(1, 1, &media_rtp_packets, &media_packets);
+  std::list<Packet*> fec_packets;
+  EXPECT_EQ(0, fec_->GenerateFEC(media_packets,
+                                 kNumFecPackets * 255 / kNumMediaPackets,
+                                 0,
+                                 false,
+                                 &fec_packets));
+  ASSERT_EQ(kNumFecPackets, fec_packets.size());
+
+  // Recovery
+  receiver_fec_->SetPayloadTypeFEC(kFecPayloadType);
+  BuildAndAddRedMediaPacket(media_rtp_packets.front());
+  // Drop one media packet.
+  BuildAndAddRedFecPacket(fec_packets.front());
+  {
+    InSequence s;
+    std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+    VerifyReconstructedMediaPacket(*it, 1);
+    ++it;
+    VerifyReconstructedMediaPacket(*it, 1);
+  }
+  EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC(false));
+
+  DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, MaxFramesOneFec) {
+  const unsigned int kNumFecPackets = 1u;
+  const unsigned int kNumMediaPackets = 48u;
+  std::list<RtpPacket*> media_rtp_packets;
+  std::list<Packet*> media_packets;
+  for (unsigned int i = 0; i < kNumMediaPackets; ++i)
+    GenerateFrame(1, i, &media_rtp_packets, &media_packets);
+  std::list<Packet*> fec_packets;
+  EXPECT_EQ(0, fec_->GenerateFEC(media_packets,
+                                 kNumFecPackets * 255 / kNumMediaPackets,
+                                 0,
+                                 false,
+                                 &fec_packets));
+  ASSERT_EQ(kNumFecPackets, fec_packets.size());
+
+  // Recovery
+  receiver_fec_->SetPayloadTypeFEC(kFecPayloadType);
+  std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+  ++it;  // Drop first packet.
+  for (; it != media_rtp_packets.end(); ++it)
+    BuildAndAddRedMediaPacket(*it);
+  BuildAndAddRedFecPacket(fec_packets.front());
+  {
+    InSequence s;
+    std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
+    for (; it != media_rtp_packets.end(); ++it)
+      VerifyReconstructedMediaPacket(*it, 1);
+  }
+  EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC(false));
+
+  DeletePackets(&media_packets);
+}
+
+TEST_F(ReceiverFecTest, TooManyFrames) {
+  const unsigned int kNumFecPackets = 1u;
+  const unsigned int kNumMediaPackets = 49u;
+  std::list<RtpPacket*> media_rtp_packets;
+  std::list<Packet*> media_packets;
+  for (unsigned int i = 0; i < kNumMediaPackets; ++i)
+    GenerateFrame(1, i, &media_rtp_packets, &media_packets);
+  std::list<Packet*> fec_packets;
+  EXPECT_EQ(-1, fec_->GenerateFEC(media_packets,
+                                  kNumFecPackets * 255 / kNumMediaPackets,
+                                  0,
+                                  false,
+                                  &fec_packets));
+
+  DeletePackets(&media_packets);
+}
+
+}  // namespace webrtc
--- a/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
@ -25,10 +25,31 @@ WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
    return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
 }

+RTPReceiverVideo::RTPReceiverVideo():
+    _id(0),
+    _rtpRtcp(NULL),
+    _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
+    _cbVideoFeedback(NULL),
+    _criticalSectionReceiverVideo(
+        CriticalSectionWrapper::CreateCriticalSection()),
+    _completeFrame(false),
+    _packetStartTimeMs(0),
+    _receivedBW(),
+    _estimatedBW(0),
+    _currentFecFrameDecoded(false),
+    _receiveFEC(NULL),
+    _overUseDetector(),
+    _videoBitRate(),
+    _lastBitRateChange(0),
+    _packetOverHead(28)
+{
+    memset(_receivedBW, 0,sizeof(_receivedBW));
+}
+
 RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
                                   ModuleRtpRtcpImpl* owner):
    _id(id),
-    _rtpRtcp(*owner),
+    _rtpRtcp(owner),
    _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
    _cbVideoFeedback(NULL),
    _criticalSectionReceiverVideo(
@ -317,11 +338,13 @@ RTPReceiverVideo::ParseVideoCodecSpecific(WebRtcRTPHeader* rtpHeader,
    _criticalSectionReceiverVideo->Leave();

    // Call the callback outside critical section
-    const RateControlRegion region = _rtpRtcp.OnOverUseStateUpdate(input);
+    if (_rtpRtcp) {
+      const RateControlRegion region = _rtpRtcp->OnOverUseStateUpdate(input);

-    _criticalSectionReceiverVideo->Enter();
-    _overUseDetector.SetRateControlRegion(region);
-    _criticalSectionReceiverVideo->Leave();
+      _criticalSectionReceiverVideo->Enter();
+      _overUseDetector.SetRateControlRegion(region);
+      _criticalSectionReceiverVideo->Leave();
+    }

    return retVal;
 }
--- a/src/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_video.h
@ -28,6 +28,7 @@ class CriticalSectionWrapper;
 class RTPReceiverVideo
 {
 public:
+    RTPReceiverVideo();
    RTPReceiverVideo(const WebRtc_Word32 id, ModuleRtpRtcpImpl* owner);

    virtual ~RTPReceiverVideo();
@ -58,7 +59,7 @@ public:
        const WebRtc_UWord16 incomingRtpPacketSize,
        const WebRtc_Word64 nowMS);

-    WebRtc_Word32 ReceiveRecoveredPacketCallback(
+    virtual WebRtc_Word32 ReceiveRecoveredPacketCallback(
        WebRtcRTPHeader* rtpHeader,
        const WebRtc_UWord8* payloadData,
        const WebRtc_UWord16 payloadDataLength);
@ -110,7 +111,7 @@ protected:

 private:
    WebRtc_Word32             _id;
-    ModuleRtpRtcpImpl&        _rtpRtcp;
+    ModuleRtpRtcpImpl*        _rtpRtcp;

    CriticalSectionWrapper*   _criticalSectionFeedback;
    RtpVideoFeedback*         _cbVideoFeedback;
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
@ -1,4 +1,4 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+# 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
@ -13,6 +13,7 @@
      'type': 'executable',
      'dependencies': [
        'rtp_rtcp',
+        '<(webrtc_root)/../testing/gmock.gyp:gmock',
        '<(webrtc_root)/../testing/gtest.gyp:gtest',
        '<(webrtc_root)/../test/test.gyp:test_support_main',
        '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
@ -21,6 +22,7 @@
        '../../../',
      ],
      'sources': [
+        'receiver_fec_unittest.cc',
        'rtp_format_vp8_unittest.cc',
        'rtp_format_vp8_test_helper.cc',
        'rtp_format_vp8_test_helper.h',