The Tor Browser: media/webrtc/signaling/test/mediapipeline

Implement a real Private Browsing Mode condition by changing the API/ABI;
This solves Tor bug #9701, complying with disk avoidance documented in
https://www.torproject.org/projects/torbrowser/design/#disk-avoidance.

     1 /* This Source Code Form is subject to the terms of the Mozilla Public

     2  * License, v. 2.0. If a copy of the MPL was not distributed with this file,

     3  * You can obtain one at http://mozilla.org/MPL/2.0/. */

     5 // Original author: ekr@rtfm.com

     7 #include <iostream>

     9 #include "sigslot.h"

    11 #include "logging.h"

    12 #include "nsThreadUtils.h"

    13 #include "nsXPCOM.h"

    14 #include "nss.h"

    15 #include "ssl.h"

    16 #include "sslproto.h"

    18 #include "dtlsidentity.h"

    19 #include "mozilla/RefPtr.h"

    20 #include "FakeMediaStreams.h"

    21 #include "FakeMediaStreamsImpl.h"

    22 #include "MediaConduitErrors.h"

    23 #include "MediaConduitInterface.h"

    24 #include "MediaPipeline.h"

    25 #include "MediaPipelineFilter.h"

    26 #include "runnable_utils.h"

    27 #include "transportflow.h"

    28 #include "transportlayerloopback.h"

    29 #include "transportlayerdtls.h"

    30 #include "mozilla/SyncRunnable.h"

    33 #include "mtransport_test_utils.h"

    34 #include "runnable_utils.h"

    36 #include "webrtc/modules/interface/module_common_types.h"

    38 #define GTEST_HAS_RTTI 0

    39 #include "gtest/gtest.h"

    40 #include "gtest_utils.h"

    42 using namespace mozilla;

    43 MOZ_MTLOG_MODULE("mediapipeline")

    45 MtransportTestUtils *test_utils;

    47 namespace {

    49 class TransportInfo {

    50  public:

    51   TransportInfo() :

    52     flow_(nullptr),

    53     loopback_(nullptr),

    54     dtls_(nullptr) {}

    56   static void InitAndConnect(TransportInfo &client, TransportInfo &server) {

    57     client.Init(true);

    58     server.Init(false);

    59     client.PushLayers();

    60     server.PushLayers();

    61     client.Connect(&server);

    62     server.Connect(&client);

    63   }

    65   void Init(bool client) {

    66     nsresult res;

    68     flow_ = new TransportFlow();

    69     loopback_ = new TransportLayerLoopback();

    70     dtls_ = new TransportLayerDtls();

    72     res = loopback_->Init();

    73     if (res != NS_OK) {

    74       FreeLayers();

    75     }

    76     ASSERT_EQ((nsresult)NS_OK, res);

    78     std::vector<uint16_t> ciphers;

    79     ciphers.push_back(SRTP_AES128_CM_HMAC_SHA1_80);

    80     dtls_->SetSrtpCiphers(ciphers);

    81     dtls_->SetIdentity(DtlsIdentity::Generate());

    82     dtls_->SetRole(client ? TransportLayerDtls::CLIENT :

    83       TransportLayerDtls::SERVER);

    84     dtls_->SetVerificationAllowAll();

    85   }

    87   void PushLayers() {

    88     nsresult res;

    90     nsAutoPtr<std::queue<TransportLayer *> > layers(

    91       new std::queue<TransportLayer *>);

    92     layers->push(loopback_);

    93     layers->push(dtls_);

    94     res = flow_->PushLayers(layers);

    95     if (res != NS_OK) {

    96       FreeLayers();

    97     }

    98     ASSERT_EQ((nsresult)NS_OK, res);

    99   }

   101   void Connect(TransportInfo* peer) {

   102     MOZ_ASSERT(loopback_);

   103     MOZ_ASSERT(peer->loopback_);

   105     loopback_->Connect(peer->loopback_);

   106   }

   108   // Free the memory allocated at the beginning of Init

   109   // if failure occurs before layers setup.

   110   void FreeLayers() {

   111     delete loopback_;

   112     loopback_ = nullptr;

   113     delete dtls_;

   114     dtls_ = nullptr;

   115   }

   117   void Shutdown() {

   118     if (loopback_) {

   119       loopback_->Disconnect();

   120     }

   121     loopback_ = nullptr;

   122     dtls_ = nullptr;

   123     flow_ = nullptr;

   124   }

   126   mozilla::RefPtr<TransportFlow> flow_;

   127   TransportLayerLoopback *loopback_;

   128   TransportLayerDtls *dtls_;

   129 };

   131 class TestAgent {

   132  public:

   133   TestAgent() :

   134       audio_config_(109, "opus", 48000, 960, 2, 64000),

   135       audio_conduit_(mozilla::AudioSessionConduit::Create(nullptr)),

   136       audio_(),

   137       audio_pipeline_() {

   138   }

   140   static void ConnectRtp(TestAgent *client, TestAgent *server) {

   141     TransportInfo::InitAndConnect(client->audio_rtp_transport_,

   142                                   server->audio_rtp_transport_);

   143   }

   145   static void ConnectRtcp(TestAgent *client, TestAgent *server) {

   146     TransportInfo::InitAndConnect(client->audio_rtcp_transport_,

   147                                   server->audio_rtcp_transport_);

   148   }

   150   static void ConnectBundle(TestAgent *client, TestAgent *server) {

   151     TransportInfo::InitAndConnect(client->bundle_transport_,

   152                                   server->bundle_transport_);

   153   }

   155   virtual void CreatePipelines_s(bool aIsRtcpMux) = 0;

   157   void Start() {

   158     nsresult ret;

   160     MOZ_MTLOG(ML_DEBUG, "Starting");

   162     mozilla::SyncRunnable::DispatchToThread(

   163       test_utils->sts_target(),

   164       WrapRunnableRet(audio_->GetStream(), &Fake_MediaStream::Start, &ret));

   166     ASSERT_TRUE(NS_SUCCEEDED(ret));

   167   }

   169   void StopInt() {

   170     audio_->GetStream()->Stop();

   171   }

   173   void Stop() {

   174     MOZ_MTLOG(ML_DEBUG, "Stopping");

   176     if (audio_pipeline_)

   177       audio_pipeline_->ShutdownMedia_m();

   179     mozilla::SyncRunnable::DispatchToThread(

   180       test_utils->sts_target(),

   181       WrapRunnable(this, &TestAgent::StopInt));

   182   }

   184   void Shutdown_s() {

   185     audio_rtp_transport_.Shutdown();

   186     audio_rtcp_transport_.Shutdown();

   187     bundle_transport_.Shutdown();

   188     if (audio_pipeline_)

   189       audio_pipeline_->ShutdownTransport_s();

   190   }

   192   void Shutdown() {

   193     if (audio_pipeline_)

   194       audio_pipeline_->ShutdownMedia_m();

   196     mozilla::SyncRunnable::DispatchToThread(

   197       test_utils->sts_target(),

   198       WrapRunnable(this, &TestAgent::Shutdown_s));

   199   }

   201   uint32_t GetRemoteSSRC() {

   202     uint32_t res = 0;

   203     audio_conduit_->GetRemoteSSRC(&res);

   204     return res;

   205   }

   207   uint32_t GetLocalSSRC() {

   208     uint32_t res = 0;

   209     audio_conduit_->GetLocalSSRC(&res);

   210     return res;

   211   }

   213   int GetAudioRtpCountSent() {

   214     return audio_pipeline_->rtp_packets_sent();

   215   }

   217   int GetAudioRtpCountReceived() {

   218     return audio_pipeline_->rtp_packets_received();

   219   }

   221   int GetAudioRtcpCountSent() {

   222     return audio_pipeline_->rtcp_packets_sent();

   223   }

   225   int GetAudioRtcpCountReceived() {

   226     return audio_pipeline_->rtcp_packets_received();

   227   }

   229  protected:

   230   mozilla::AudioCodecConfig audio_config_;

   231   mozilla::RefPtr<mozilla::MediaSessionConduit> audio_conduit_;

   232   nsRefPtr<DOMMediaStream> audio_;

   233   // TODO(bcampen@mozilla.com): Right now this does not let us test RTCP in

   234   // both directions; only the sender's RTCP is sent, but the receiver should

   235   // be sending it too.

   236   mozilla::RefPtr<mozilla::MediaPipeline> audio_pipeline_;

   237   TransportInfo audio_rtp_transport_;

   238   TransportInfo audio_rtcp_transport_;

   239   TransportInfo bundle_transport_;

   240 };

   242 class TestAgentSend : public TestAgent {

   243  public:

   244   TestAgentSend() : use_bundle_(false) {}

   246   virtual void CreatePipelines_s(bool aIsRtcpMux) {

   247     audio_ = new Fake_DOMMediaStream(new Fake_AudioStreamSource());

   249     mozilla::MediaConduitErrorCode err =

   250         static_cast<mozilla::AudioSessionConduit *>(audio_conduit_.get())->

   251         ConfigureSendMediaCodec(&audio_config_);

   252     EXPECT_EQ(mozilla::kMediaConduitNoError, err);

   254     std::string test_pc("PC");

   256     if (aIsRtcpMux) {

   257       ASSERT_FALSE(audio_rtcp_transport_.flow_);

   258     }

   260     RefPtr<TransportFlow> rtp(audio_rtp_transport_.flow_);

   261     RefPtr<TransportFlow> rtcp(audio_rtcp_transport_.flow_);

   263     if (use_bundle_) {

   264       rtp = bundle_transport_.flow_;

   265       rtcp = nullptr;

   266     }

   268     audio_pipeline_ = new mozilla::MediaPipelineTransmit(

   269         test_pc,

   270         nullptr,

   271         test_utils->sts_target(),

   272         audio_,

   273         1,

   274         1,

   275         audio_conduit_,

   276         rtp,

   277         rtcp);

   279     audio_pipeline_->Init();

   280   }

   282   void SetUsingBundle(bool use_bundle) {

   283     use_bundle_ = use_bundle;

   284   }

   286  private:

   287   bool use_bundle_;

   288 };

   291 class TestAgentReceive : public TestAgent {

   292  public:

   293   virtual void CreatePipelines_s(bool aIsRtcpMux) {

   294     mozilla::SourceMediaStream *audio = new Fake_SourceMediaStream();

   295     audio->SetPullEnabled(true);

   297     mozilla::AudioSegment* segment= new mozilla::AudioSegment();

   298     audio->AddTrack(0, 100, 0, segment);

   299     audio->AdvanceKnownTracksTime(mozilla::STREAM_TIME_MAX);

   301     audio_ = new Fake_DOMMediaStream(audio);

   303     std::vector<mozilla::AudioCodecConfig *> codecs;

   304     codecs.push_back(&audio_config_);

   306     mozilla::MediaConduitErrorCode err =

   307         static_cast<mozilla::AudioSessionConduit *>(audio_conduit_.get())->

   308         ConfigureRecvMediaCodecs(codecs);

   309     EXPECT_EQ(mozilla::kMediaConduitNoError, err);

   311     std::string test_pc("PC");

   313     if (aIsRtcpMux) {

   314       ASSERT_FALSE(audio_rtcp_transport_.flow_);

   315     }

   317     // For now, assume bundle always uses rtcp mux

   318     RefPtr<TransportFlow> dummy;

   319     RefPtr<TransportFlow> bundle_transport;

   320     if (bundle_filter_) {

   321       bundle_transport = bundle_transport_.flow_;

   322       bundle_filter_->AddLocalSSRC(GetLocalSSRC());

   323     }

   325     audio_pipeline_ = new mozilla::MediaPipelineReceiveAudio(

   326         test_pc,

   327         nullptr,

   328         test_utils->sts_target(),

   329         audio_->GetStream(), 1, 1,

   330         static_cast<mozilla::AudioSessionConduit *>(audio_conduit_.get()),

   331         audio_rtp_transport_.flow_,

   332         audio_rtcp_transport_.flow_,

   333         bundle_transport,

   334         dummy,

   335         bundle_filter_);

   337     audio_pipeline_->Init();

   338   }

   340   void SetBundleFilter(nsAutoPtr<MediaPipelineFilter> filter) {

   341     bundle_filter_ = filter;

   342   }

   344   void SetUsingBundle_s(bool decision) {

   345     audio_pipeline_->SetUsingBundle_s(decision);

   346   }

   348   void UpdateFilterFromRemoteDescription_s(

   349       nsAutoPtr<MediaPipelineFilter> filter) {

   350     audio_pipeline_->UpdateFilterFromRemoteDescription_s(filter);

   351   }

   353  private:

   354   nsAutoPtr<MediaPipelineFilter> bundle_filter_;

   355 };

   358 class MediaPipelineTest : public ::testing::Test {

   359  public:

   360   ~MediaPipelineTest() {

   361     p1_.Stop();

   362     p2_.Stop();

   363     p1_.Shutdown();

   364     p2_.Shutdown();

   365   }

   367   // Setup transport.

   368   void InitTransports(bool aIsRtcpMux) {

   369     // RTP, p1_ is server, p2_ is client

   370     mozilla::SyncRunnable::DispatchToThread(

   371       test_utils->sts_target(),

   372       WrapRunnableNM(&TestAgent::ConnectRtp, &p2_, &p1_));

   374     // Create RTCP flows separately if we are not muxing them.

   375     if(!aIsRtcpMux) {

   376       // RTCP, p1_ is server, p2_ is client

   377       mozilla::SyncRunnable::DispatchToThread(

   378         test_utils->sts_target(),

   379         WrapRunnableNM(&TestAgent::ConnectRtcp, &p2_, &p1_));

   380     }

   382     // BUNDLE, p1_ is server, p2_ is client

   383     mozilla::SyncRunnable::DispatchToThread(

   384       test_utils->sts_target(),

   385       WrapRunnableNM(&TestAgent::ConnectBundle, &p2_, &p1_));

   386   }

   388   // Verify RTP and RTCP

   389   void TestAudioSend(bool aIsRtcpMux,

   390                      bool bundle = false,

   391                      nsAutoPtr<MediaPipelineFilter> localFilter =

   392                         nsAutoPtr<MediaPipelineFilter>(nullptr),

   393                      nsAutoPtr<MediaPipelineFilter> remoteFilter =

   394                         nsAutoPtr<MediaPipelineFilter>(nullptr),

   395                      unsigned int ms_until_answer = 500,

   396                      unsigned int ms_of_traffic_after_answer = 10000) {

   398     // We do not support testing bundle without rtcp mux, since that doesn't

   399     // make any sense.

   400     ASSERT_FALSE(!aIsRtcpMux && bundle);

   402     p1_.SetUsingBundle(bundle);

   403     p2_.SetBundleFilter(localFilter);

   405     // Setup transport flows

   406     InitTransports(aIsRtcpMux);

   408     mozilla::SyncRunnable::DispatchToThread(

   409       test_utils->sts_target(),

   410       WrapRunnable(&p1_, &TestAgent::CreatePipelines_s, aIsRtcpMux));

   412     mozilla::SyncRunnable::DispatchToThread(

   413       test_utils->sts_target(),

   414       WrapRunnable(&p2_, &TestAgent::CreatePipelines_s, aIsRtcpMux));

   416     p2_.Start();

   417     p1_.Start();

   419     // Simulate pre-answer traffic

   420     PR_Sleep(ms_until_answer);

   422     mozilla::SyncRunnable::DispatchToThread(

   423       test_utils->sts_target(),

   424       WrapRunnable(&p2_, &TestAgentReceive::SetUsingBundle_s, bundle));

   426     if (bundle) {

   427       // Leaving remoteFilter not set implies we want to test sunny-day

   428       if (!remoteFilter) {

   429         remoteFilter = new MediaPipelineFilter;

   430         // Might not be safe, strictly speaking.

   431         remoteFilter->AddRemoteSSRC(p1_.GetLocalSSRC());

   432       }

   434       mozilla::SyncRunnable::DispatchToThread(

   435           test_utils->sts_target(),

   436           WrapRunnable(&p2_,

   437                        &TestAgentReceive::UpdateFilterFromRemoteDescription_s,

   438                        remoteFilter));

   439     }

   442     // wait for some RTP/RTCP tx and rx to happen

   443     PR_Sleep(ms_of_traffic_after_answer);

   445     p1_.Stop();

   446     p2_.Stop();

   448     // wait for any packets in flight to arrive

   449     PR_Sleep(100);

   451     p1_.Shutdown();

   452     p2_.Shutdown();

   454     if (!bundle) {

   455       // If we are doing bundle, allow the test-case to do this checking.

   456       ASSERT_GE(p1_.GetAudioRtpCountSent(), 40);

   457       ASSERT_EQ(p1_.GetAudioRtpCountReceived(), p2_.GetAudioRtpCountSent());

   458       ASSERT_EQ(p1_.GetAudioRtpCountSent(), p2_.GetAudioRtpCountReceived());

   460       // Calling ShutdownMedia_m on both pipelines does not stop the flow of

   461       // RTCP. So, we might be off by one here.

   462       ASSERT_LE(p2_.GetAudioRtcpCountReceived(), p1_.GetAudioRtcpCountSent());

   463       ASSERT_GE(p2_.GetAudioRtcpCountReceived() + 1, p1_.GetAudioRtcpCountSent());

   464     }

   466   }

   468   void TestAudioReceiverOffersBundle(bool bundle_accepted,

   469       nsAutoPtr<MediaPipelineFilter> localFilter,

   470       nsAutoPtr<MediaPipelineFilter> remoteFilter =

   471           nsAutoPtr<MediaPipelineFilter>(nullptr),

   472       unsigned int ms_until_answer = 500,

   473       unsigned int ms_of_traffic_after_answer = 10000) {

   474     TestAudioSend(true,

   475                   bundle_accepted,

   476                   localFilter,

   477                   remoteFilter,

   478                   ms_until_answer,

   479                   ms_of_traffic_after_answer);

   480   }

   481 protected:

   482   TestAgentSend p1_;

   483   TestAgentReceive p2_;

   484 };

   486 class MediaPipelineFilterTest : public ::testing::Test {

   487   public:

   488     bool Filter(MediaPipelineFilter& filter,

   489                 int32_t correlator,

   490                 uint32_t ssrc,

   491                 uint8_t payload_type) {

   493       webrtc::RTPHeader header;

   494       header.ssrc = ssrc;

   495       header.payloadType = payload_type;

   496       return filter.Filter(header, correlator);

   497     }

   498 };

   500 TEST_F(MediaPipelineFilterTest, TestConstruct) {

   501   MediaPipelineFilter filter;

   502 }

   504 TEST_F(MediaPipelineFilterTest, TestDefault) {

   505   MediaPipelineFilter filter;

   506   ASSERT_FALSE(Filter(filter, 0, 233, 110));

   507 }

   509 TEST_F(MediaPipelineFilterTest, TestSSRCFilter) {

   510   MediaPipelineFilter filter;

   511   filter.AddRemoteSSRC(555);

   512   ASSERT_TRUE(Filter(filter, 0, 555, 110));

   513   ASSERT_FALSE(Filter(filter, 0, 556, 110));

   514 }

   516 #define SSRC(ssrc) \

   517   ((ssrc >> 24) & 0xFF), \

   518   ((ssrc >> 16) & 0xFF), \

   519   ((ssrc >> 8 ) & 0xFF), \

   520   (ssrc         & 0xFF)

   522 #define REPORT_FRAGMENT(ssrc) \

   523   SSRC(ssrc), \

   524   0,0,0,0, \

   525   0,0,0,0, \

   526   0,0,0,0, \

   527   0,0,0,0, \

   528   0,0,0,0

   530 #define RTCP_TYPEINFO(num_rrs, type, size) \

   531   0x80 + num_rrs, type, 0, size

   533 const unsigned char rtcp_rr_s16[] = {

   534   // zero rrs, size 1 words

   535   RTCP_TYPEINFO(0, MediaPipelineFilter::RECEIVER_REPORT_T, 1),

   536   SSRC(16)

   537 };

   539 const unsigned char rtcp_rr_s16_r17[] = {

   540   // one rr, 7 words

   541   RTCP_TYPEINFO(1, MediaPipelineFilter::RECEIVER_REPORT_T, 7),

   542   SSRC(16),

   543   REPORT_FRAGMENT(17)

   544 };

   546 const unsigned char rtcp_rr_s16_r17_18[] = {

   547   // two rrs, size 13 words

   548   RTCP_TYPEINFO(2, MediaPipelineFilter::RECEIVER_REPORT_T, 13),

   549   SSRC(16),

   550   REPORT_FRAGMENT(17),

   551   REPORT_FRAGMENT(18)

   552 };

   554 const unsigned char rtcp_sr_s16[] = {

   555   // zero rrs, size 6 words

   556   RTCP_TYPEINFO(0, MediaPipelineFilter::SENDER_REPORT_T, 6),

   557   REPORT_FRAGMENT(16)

   558 };

   560 const unsigned char rtcp_sr_s16_r17[] = {

   561   // one rr, size 12 words

   562   RTCP_TYPEINFO(1, MediaPipelineFilter::SENDER_REPORT_T, 12),

   563   REPORT_FRAGMENT(16),

   564   REPORT_FRAGMENT(17)

   565 };

   567 const unsigned char rtcp_sr_s16_r17_18[] = {

   568   // two rrs, size 18 words

   569   RTCP_TYPEINFO(2, MediaPipelineFilter::SENDER_REPORT_T, 18),

   570   REPORT_FRAGMENT(16),

   571   REPORT_FRAGMENT(17),

   572   REPORT_FRAGMENT(18)

   573 };

   575 const unsigned char unknown_type[] = {

   576   RTCP_TYPEINFO(1, 222, 0)

   577 };

   579 TEST_F(MediaPipelineFilterTest, TestEmptyFilterReport0) {

   580   MediaPipelineFilter filter;

   581   ASSERT_EQ(MediaPipelineFilter::FAIL,

   582             filter.FilterRTCP(rtcp_sr_s16, sizeof(rtcp_sr_s16)));

   583   ASSERT_EQ(MediaPipelineFilter::FAIL,

   584             filter.FilterRTCP(rtcp_rr_s16, sizeof(rtcp_rr_s16)));

   585 }

   587 TEST_F(MediaPipelineFilterTest, TestFilterReport0) {

   588   MediaPipelineFilter filter;

   589   filter.AddRemoteSSRC(16);

   590   ASSERT_EQ(MediaPipelineFilter::PASS,

   591             filter.FilterRTCP(rtcp_sr_s16, sizeof(rtcp_sr_s16)));

   592   ASSERT_EQ(MediaPipelineFilter::PASS,

   593             filter.FilterRTCP(rtcp_rr_s16, sizeof(rtcp_rr_s16)));

   594 }

   596 TEST_F(MediaPipelineFilterTest, TestFilterReport0SSRCTruncated) {

   597   MediaPipelineFilter filter;

   598   filter.AddRemoteSSRC(16);

   599   const unsigned char data[] = {

   600     RTCP_TYPEINFO(0, MediaPipelineFilter::RECEIVER_REPORT_T, 1),

   601     0,0,0

   602   };

   603   ASSERT_EQ(MediaPipelineFilter::FAIL,

   604             filter.FilterRTCP(data, sizeof(data)));

   605 }

   607 TEST_F(MediaPipelineFilterTest, TestFilterReport0PTTruncated) {

   608   MediaPipelineFilter filter;

   609   filter.AddRemoteSSRC(16);

   610   const unsigned char data[] = {0x80};

   611   ASSERT_EQ(MediaPipelineFilter::FAIL,

   612             filter.FilterRTCP(data, sizeof(data)));

   613 }

   615 TEST_F(MediaPipelineFilterTest, TestFilterReport0CountTruncated) {

   616   MediaPipelineFilter filter;

   617   filter.AddRemoteSSRC(16);

   618   const unsigned char data[] = {};

   619   ASSERT_EQ(MediaPipelineFilter::FAIL,

   620             filter.FilterRTCP(data, sizeof(data)));

   621 }

   623 TEST_F(MediaPipelineFilterTest, TestFilterReport1BothMatch) {

   624   MediaPipelineFilter filter;

   625   filter.AddRemoteSSRC(16);

   626   filter.AddLocalSSRC(17);

   627   ASSERT_EQ(MediaPipelineFilter::PASS,

   628             filter.FilterRTCP(rtcp_sr_s16_r17, sizeof(rtcp_sr_s16_r17)));

   629   ASSERT_EQ(MediaPipelineFilter::PASS,

   630             filter.FilterRTCP(rtcp_rr_s16_r17, sizeof(rtcp_rr_s16_r17)));

   631 }

   633 TEST_F(MediaPipelineFilterTest, TestFilterReport1SSRCTruncated) {

   634   MediaPipelineFilter filter;

   635   filter.AddRemoteSSRC(16);

   636   filter.AddLocalSSRC(17);

   637   const unsigned char rr[] = {

   638     RTCP_TYPEINFO(1, MediaPipelineFilter::RECEIVER_REPORT_T, 7),

   639     SSRC(16),

   640     0,0,0

   641   };

   642   ASSERT_EQ(MediaPipelineFilter::FAIL,

   643             filter.FilterRTCP(rr, sizeof(rr)));

   644   const unsigned char sr[] = {

   645     RTCP_TYPEINFO(1, MediaPipelineFilter::RECEIVER_REPORT_T, 12),

   646     REPORT_FRAGMENT(16),

   647     0,0,0

   648   };

   649   ASSERT_EQ(MediaPipelineFilter::FAIL,

   650             filter.FilterRTCP(sr, sizeof(rr)));

   651 }

   653 TEST_F(MediaPipelineFilterTest, TestFilterReport1BigSSRC) {

   654   MediaPipelineFilter filter;

   655   filter.AddRemoteSSRC(0x01020304);

   656   filter.AddLocalSSRC(0x11121314);

   657   const unsigned char rr[] = {

   658     RTCP_TYPEINFO(1, MediaPipelineFilter::RECEIVER_REPORT_T, 7),

   659     SSRC(0x01020304),

   660     REPORT_FRAGMENT(0x11121314)

   661   };

   662   ASSERT_EQ(MediaPipelineFilter::PASS,

   663             filter.FilterRTCP(rr, sizeof(rr)));

   664   const unsigned char sr[] = {

   665     RTCP_TYPEINFO(1, MediaPipelineFilter::RECEIVER_REPORT_T, 12),

   666     SSRC(0x01020304),

   667     REPORT_FRAGMENT(0x11121314)

   668   };

   669   ASSERT_EQ(MediaPipelineFilter::PASS,

   670             filter.FilterRTCP(sr, sizeof(rr)));

   671 }

   673 TEST_F(MediaPipelineFilterTest, TestFilterReport1LocalMatch) {

   674   MediaPipelineFilter filter;

   675   filter.AddLocalSSRC(17);

   676   ASSERT_EQ(MediaPipelineFilter::PASS,

   677             filter.FilterRTCP(rtcp_sr_s16_r17, sizeof(rtcp_sr_s16_r17)));

   678   ASSERT_EQ(MediaPipelineFilter::PASS,

   679             filter.FilterRTCP(rtcp_rr_s16_r17, sizeof(rtcp_rr_s16_r17)));

   680 }

   682 TEST_F(MediaPipelineFilterTest, TestFilterReport1Inconsistent) {

   683   MediaPipelineFilter filter;

   684   filter.AddRemoteSSRC(16);

   685   // We assume that the filter is exactly correct in terms of local ssrcs.

   686   // So, when RTCP shows up with a remote SSRC that matches, and a local

   687   // ssrc that doesn't, we assume the other end has messed up and put ssrcs

   688   // from more than one m-line in the packet.

   689   ASSERT_EQ(MediaPipelineFilter::FAIL,

   690             filter.FilterRTCP(rtcp_sr_s16_r17, sizeof(rtcp_sr_s16_r17)));

   691   ASSERT_EQ(MediaPipelineFilter::FAIL,

   692             filter.FilterRTCP(rtcp_rr_s16_r17, sizeof(rtcp_rr_s16_r17)));

   693 }

   695 TEST_F(MediaPipelineFilterTest, TestFilterReport1NeitherMatch) {

   696   MediaPipelineFilter filter;

   697   filter.AddRemoteSSRC(17);

   698   filter.AddLocalSSRC(18);

   699   ASSERT_EQ(MediaPipelineFilter::PASS,

   700             filter.FilterRTCP(rtcp_sr_s16_r17, sizeof(rtcp_sr_s16_r17)));

   701   ASSERT_EQ(MediaPipelineFilter::PASS,

   702             filter.FilterRTCP(rtcp_rr_s16_r17, sizeof(rtcp_rr_s16_r17)));

   703 }

   705 TEST_F(MediaPipelineFilterTest, TestFilterReport2AllMatch) {

   706   MediaPipelineFilter filter;

   707   filter.AddRemoteSSRC(16);

   708   filter.AddLocalSSRC(17);

   709   filter.AddLocalSSRC(18);

   710   ASSERT_EQ(MediaPipelineFilter::PASS,

   711             filter.FilterRTCP(rtcp_sr_s16_r17_18,

   712                               sizeof(rtcp_sr_s16_r17_18)));

   713 }

   715 TEST_F(MediaPipelineFilterTest, TestFilterReport2LocalMatch) {

   716   MediaPipelineFilter filter;

   717   filter.AddLocalSSRC(17);

   718   filter.AddLocalSSRC(18);

   719   ASSERT_EQ(MediaPipelineFilter::PASS,

   720             filter.FilterRTCP(rtcp_sr_s16_r17_18,

   721                               sizeof(rtcp_sr_s16_r17_18)));

   722   ASSERT_EQ(MediaPipelineFilter::PASS,

   723             filter.FilterRTCP(rtcp_rr_s16_r17_18,

   724                               sizeof(rtcp_rr_s16_r17_18)));

   725 }

   727 TEST_F(MediaPipelineFilterTest, TestFilterReport2Inconsistent101) {

   728   MediaPipelineFilter filter;

   729   filter.AddRemoteSSRC(16);

   730   filter.AddLocalSSRC(18);

   731   ASSERT_EQ(MediaPipelineFilter::FAIL,

   732             filter.FilterRTCP(rtcp_sr_s16_r17_18,

   733                               sizeof(rtcp_sr_s16_r17_18)));

   734   ASSERT_EQ(MediaPipelineFilter::FAIL,

   735             filter.FilterRTCP(rtcp_rr_s16_r17_18,

   736                               sizeof(rtcp_rr_s16_r17_18)));

   737 }

   739 TEST_F(MediaPipelineFilterTest, TestFilterReport2Inconsistent001) {

   740   MediaPipelineFilter filter;

   741   filter.AddLocalSSRC(18);

   742   ASSERT_EQ(MediaPipelineFilter::FAIL,

   743             filter.FilterRTCP(rtcp_sr_s16_r17_18,

   744                               sizeof(rtcp_sr_s16_r17_18)));

   745   ASSERT_EQ(MediaPipelineFilter::FAIL,

   746             filter.FilterRTCP(rtcp_rr_s16_r17_18,

   747                               sizeof(rtcp_rr_s16_r17_18)));

   748 }

   750 TEST_F(MediaPipelineFilterTest, TestFilterUnknownRTCPType) {

   751   MediaPipelineFilter filter;

   752   filter.AddLocalSSRC(18);

   753   ASSERT_EQ(MediaPipelineFilter::UNSUPPORTED,

   754             filter.FilterRTCP(unknown_type, sizeof(unknown_type)));

   755 }

   757 TEST_F(MediaPipelineFilterTest, TestCorrelatorFilter) {

   758   MediaPipelineFilter filter;

   759   filter.SetCorrelator(7777);

   760   ASSERT_TRUE(Filter(filter, 7777, 16, 110));

   761   ASSERT_FALSE(Filter(filter, 7778, 17, 110));

   762   // This should also have resulted in the SSRC 16 being added to the filter

   763   ASSERT_TRUE(Filter(filter, 0, 16, 110));

   764   ASSERT_FALSE(Filter(filter, 0, 17, 110));

   766   // rtcp_sr_s16 has 16 as an SSRC

   767   ASSERT_EQ(MediaPipelineFilter::PASS,

   768             filter.FilterRTCP(rtcp_sr_s16, sizeof(rtcp_sr_s16)));

   769   ASSERT_EQ(MediaPipelineFilter::PASS,

   770             filter.FilterRTCP(rtcp_rr_s16, sizeof(rtcp_rr_s16)));

   771 }

   773 TEST_F(MediaPipelineFilterTest, TestPayloadTypeFilter) {

   774   MediaPipelineFilter filter;

   775   filter.AddUniquePT(110);

   776   ASSERT_TRUE(Filter(filter, 0, 555, 110));

   777   ASSERT_FALSE(Filter(filter, 0, 556, 111));

   778 }

   780 TEST_F(MediaPipelineFilterTest, TestPayloadTypeFilterSSRCUpdate) {

   781   MediaPipelineFilter filter;

   782   filter.AddUniquePT(110);

   783   ASSERT_TRUE(Filter(filter, 0, 16, 110));

   785   // rtcp_sr_s16 has 16 as an SSRC

   786   ASSERT_EQ(MediaPipelineFilter::PASS,

   787             filter.FilterRTCP(rtcp_sr_s16, sizeof(rtcp_sr_s16)));

   788 }

   790 TEST_F(MediaPipelineFilterTest, TestAnswerAddsSSRCs) {

   791   MediaPipelineFilter filter;

   792   filter.SetCorrelator(7777);

   793   ASSERT_TRUE(Filter(filter, 7777, 555, 110));

   794   ASSERT_FALSE(Filter(filter, 7778, 556, 110));

   795   // This should also have resulted in the SSRC 555 being added to the filter

   796   ASSERT_TRUE(Filter(filter, 0, 555, 110));

   797   ASSERT_FALSE(Filter(filter, 0, 556, 110));

   799   // This sort of thing can happen when getting an answer with SSRC attrs

   800   // The answer will not contain the correlator.

   801   MediaPipelineFilter filter2;

   802   filter2.AddRemoteSSRC(555);

   803   filter2.AddRemoteSSRC(556);

   804   filter2.AddRemoteSSRC(557);

   806   filter.IncorporateRemoteDescription(filter2);

   808   // Ensure that the old SSRC still works.

   809   ASSERT_TRUE(Filter(filter, 0, 555, 110));

   811   // Ensure that the new SSRCs work.

   812   ASSERT_TRUE(Filter(filter, 0, 556, 110));

   813   ASSERT_TRUE(Filter(filter, 0, 557, 110));

   815   // Ensure that the correlator continues to work

   816   ASSERT_TRUE(Filter(filter, 7777, 558, 110));

   817 }

   819 TEST_F(MediaPipelineFilterTest, TestSSRCMovedWithSDP) {

   820   MediaPipelineFilter filter;

   821   filter.SetCorrelator(7777);

   822   filter.AddUniquePT(111);

   823   ASSERT_TRUE(Filter(filter, 7777, 555, 110));

   825   MediaPipelineFilter filter2;

   826   filter2.AddRemoteSSRC(556);

   828   filter.IncorporateRemoteDescription(filter2);

   830   // Ensure that the old SSRC has been removed.

   831   ASSERT_FALSE(Filter(filter, 0, 555, 110));

   833   // Ensure that the new SSRC works.

   834   ASSERT_TRUE(Filter(filter, 0, 556, 110));

   836   // Ensure that the correlator continues to work

   837   ASSERT_TRUE(Filter(filter, 7777, 558, 110));

   839   // Ensure that the payload type mapping continues to work

   840   ASSERT_TRUE(Filter(filter, 0, 559, 111));

   841 }

   843 TEST_F(MediaPipelineFilterTest, TestSSRCMovedWithCorrelator) {

   844   MediaPipelineFilter filter;

   845   filter.SetCorrelator(7777);

   846   ASSERT_TRUE(Filter(filter, 7777, 555, 110));

   847   ASSERT_TRUE(Filter(filter, 0, 555, 110));

   848   ASSERT_FALSE(Filter(filter, 7778, 555, 110));

   849   ASSERT_FALSE(Filter(filter, 0, 555, 110));

   850 }

   852 TEST_F(MediaPipelineFilterTest, TestRemoteSDPNoSSRCs) {

   853   // If the remote SDP doesn't have SSRCs, right now this is a no-op and

   854   // there is no point of even incorporating a filter, but we make the

   855   // behavior consistent to avoid confusion.

   856   MediaPipelineFilter filter;

   857   filter.SetCorrelator(7777);

   858   filter.AddUniquePT(111);

   859   ASSERT_TRUE(Filter(filter, 7777, 555, 110));

   861   MediaPipelineFilter filter2;

   863   filter.IncorporateRemoteDescription(filter2);

   865   // Ensure that the old SSRC still works.

   866   ASSERT_TRUE(Filter(filter, 7777, 555, 110));

   867 }

   869 TEST_F(MediaPipelineTest, TestAudioSendNoMux) {

   870   TestAudioSend(false);

   871 }

   873 TEST_F(MediaPipelineTest, TestAudioSendMux) {

   874   TestAudioSend(true);

   875 }

   877 TEST_F(MediaPipelineTest, TestAudioSendBundleOfferedAndDeclined) {

   878   nsAutoPtr<MediaPipelineFilter> filter(new MediaPipelineFilter);

   879   TestAudioReceiverOffersBundle(false, filter);

   880 }

   882 TEST_F(MediaPipelineTest, TestAudioSendBundleOfferedAndAccepted) {

   883   nsAutoPtr<MediaPipelineFilter> filter(new MediaPipelineFilter);

   884   // These durations have to be _extremely_ long to have any assurance that

   885   // some RTCP will be sent at all. This is because the first RTCP packet

   886   // is sometimes sent before the transports are ready, which causes it to

   887   // be dropped.

   888   TestAudioReceiverOffersBundle(true,

   889                                 filter,

   890   // We do not specify the filter for the remote description, so it will be

   891   // set to something sane after a short time.

   892                                 nsAutoPtr<MediaPipelineFilter>(),

   893                                 10000,

   894                                 10000);

   896   // Some packets should have been dropped, but not all

   897   ASSERT_GT(p1_.GetAudioRtpCountSent(), p2_.GetAudioRtpCountReceived());

   898   ASSERT_GT(p2_.GetAudioRtpCountReceived(), 40);

   899   ASSERT_GT(p1_.GetAudioRtcpCountSent(), 1);

   900   ASSERT_GT(p1_.GetAudioRtcpCountSent(), p2_.GetAudioRtcpCountReceived());

   901   ASSERT_GT(p2_.GetAudioRtcpCountReceived(), 0);

   902 }

   904 TEST_F(MediaPipelineTest, TestAudioSendBundleOfferedAndAcceptedEmptyFilter) {

   905   nsAutoPtr<MediaPipelineFilter> filter(new MediaPipelineFilter);

   906   nsAutoPtr<MediaPipelineFilter> bad_answer_filter(new MediaPipelineFilter);

   907   TestAudioReceiverOffersBundle(true, filter, bad_answer_filter);

   908   // Filter is empty, so should drop everything.

   909   ASSERT_EQ(0, p2_.GetAudioRtpCountReceived());

   910   ASSERT_EQ(0, p2_.GetAudioRtcpCountReceived());

   911 }

   913 }  // end namespace

   916 int main(int argc, char **argv) {

   917   test_utils = new MtransportTestUtils();

   918   // Start the tests

   919   NSS_NoDB_Init(nullptr);

   920   NSS_SetDomesticPolicy();

   921   ::testing::InitGoogleTest(&argc, argv);

   923   int rv = RUN_ALL_TESTS();

   924   delete test_utils;

   925   return rv;

   926 }

The Tor Browser / file revision

media/webrtc/signaling/test/mediapipeline_unittest.cpp@ac0c01689b40

media/webrtc/signaling/test/mediapipeline_unittest.cpp