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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
commit43a97878ce14b72f0981164f87f2e35e14151312 (patch)
tree620249daf56c0258faa40cbdcf9cfba06de2a846 /media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
parentInitial commit. (diff)
downloadfirefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz
firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r--media/webrtc/signaling/gtest/mediapipeline_unittest.cpp701
1 files changed, 701 insertions, 0 deletions
diff --git a/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp b/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
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+++ b/media/webrtc/signaling/gtest/mediapipeline_unittest.cpp
@@ -0,0 +1,701 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include "logging.h"
+#include "nss.h"
+#include "ssl.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/scoped_refptr.h"
+#include "AudioSegment.h"
+#include "ConcreteConduitControl.h"
+#include "modules/audio_device/include/fake_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/SpinEventLoopUntil.h"
+#include "MediaPipeline.h"
+#include "MediaPipelineFilter.h"
+#include "MediaTrackGraph.h"
+#include "MediaTrackListener.h"
+#include "TaskQueueWrapper.h"
+#include "mtransport_test_utils.h"
+#include "SharedBuffer.h"
+#include "MediaTransportHandler.h"
+#include "WebrtcCallWrapper.h"
+#include "PeerConnectionCtx.h"
+#include "WaitFor.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+
+using namespace mozilla;
+MOZ_MTLOG_MODULE("transportbridge")
+
+static MtransportTestUtils* test_utils;
+
+namespace {
+class MainAsCurrent : public TaskQueueWrapper<DeletionPolicy::NonBlocking> {
+ public:
+ MainAsCurrent()
+ : TaskQueueWrapper(
+ TaskQueue::Create(do_AddRef(GetMainThreadEventTarget()),
+ "MainAsCurrentTaskQueue"),
+ "MainAsCurrent"_ns),
+ mSetter(this) {
+ MOZ_RELEASE_ASSERT(NS_IsMainThread());
+ }
+
+ ~MainAsCurrent() = default;
+
+ private:
+ CurrentTaskQueueSetter mSetter;
+};
+
+class FakeAudioTrack : public ProcessedMediaTrack {
+ public:
+ FakeAudioTrack()
+ : ProcessedMediaTrack(44100, MediaSegment::AUDIO, nullptr),
+ mMutex("Fake AudioTrack") {
+ NS_NewTimerWithFuncCallback(
+ getter_AddRefs(mTimer), FakeAudioTrackGenerateData, this, 20,
+ nsITimer::TYPE_REPEATING_SLACK,
+ "FakeAudioTrack::FakeAudioTrackGenerateData", test_utils->sts_target());
+ }
+
+ void Destroy() override {
+ MutexAutoLock lock(mMutex);
+ MOZ_ASSERT(!mMainThreadDestroyed);
+ mMainThreadDestroyed = true;
+ mTimer->Cancel();
+ mTimer = nullptr;
+ }
+
+ void QueueSetAutoend(bool) override {}
+
+ void AddListener(MediaTrackListener* aListener) override {
+ MutexAutoLock lock(mMutex);
+ MOZ_ASSERT(!mListener);
+ mListener = aListener;
+ }
+
+ RefPtr<GenericPromise> RemoveListener(
+ MediaTrackListener* aListener) override {
+ MutexAutoLock lock(mMutex);
+ MOZ_ASSERT(mListener == aListener);
+ mListener = nullptr;
+ return GenericPromise::CreateAndResolve(true, __func__);
+ }
+
+ void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override {}
+
+ uint32_t NumberOfChannels() const override { return NUM_CHANNELS; }
+
+ private:
+ Mutex mMutex MOZ_UNANNOTATED;
+ MediaTrackListener* mListener = nullptr;
+ nsCOMPtr<nsITimer> mTimer;
+ int mCount = 0;
+
+ static const int AUDIO_BUFFER_SIZE = 1600;
+ static const int NUM_CHANNELS = 2;
+ static void FakeAudioTrackGenerateData(nsITimer* timer, void* closure) {
+ auto t = static_cast<FakeAudioTrack*>(closure);
+ MutexAutoLock lock(t->mMutex);
+ if (t->mMainThreadDestroyed) {
+ return;
+ }
+ CheckedInt<size_t> bufferSize(sizeof(int16_t));
+ bufferSize *= NUM_CHANNELS;
+ bufferSize *= AUDIO_BUFFER_SIZE;
+ RefPtr<SharedBuffer> samples = SharedBuffer::Create(bufferSize);
+ int16_t* data = reinterpret_cast<int16_t*>(samples->Data());
+ for (int i = 0; i < (AUDIO_BUFFER_SIZE * NUM_CHANNELS); i++) {
+ // saw tooth audio sample
+ data[i] = ((t->mCount % 8) * 4000) - (7 * 4000) / 2;
+ t->mCount++;
+ }
+
+ AudioSegment segment;
+ AutoTArray<const int16_t*, 1> channels;
+ channels.AppendElement(data);
+ segment.AppendFrames(samples.forget(), channels, AUDIO_BUFFER_SIZE,
+ PRINCIPAL_HANDLE_NONE);
+
+ if (t->mListener) {
+ t->mListener->NotifyQueuedChanges(nullptr, 0, segment);
+ }
+ }
+};
+
+template <typename Function>
+void RunOnSts(Function&& aFunction) {
+ MOZ_ALWAYS_SUCCEEDS(test_utils->sts_target()->Dispatch(
+ NS_NewRunnableFunction(__func__, [&] { aFunction(); }),
+ nsISerialEventTarget::DISPATCH_SYNC));
+}
+
+class LoopbackTransport : public MediaTransportHandler {
+ public:
+ LoopbackTransport() : MediaTransportHandler(nullptr) {
+ RunOnSts([&] {
+ SetState("mux", TransportLayer::TS_INIT);
+ SetRtcpState("mux", TransportLayer::TS_INIT);
+ SetState("non-mux", TransportLayer::TS_INIT);
+ SetRtcpState("non-mux", TransportLayer::TS_INIT);
+ });
+ }
+
+ static void InitAndConnect(LoopbackTransport& client,
+ LoopbackTransport& server) {
+ client.Connect(&server);
+ server.Connect(&client);
+ }
+
+ void Connect(LoopbackTransport* peer) { peer_ = peer; }
+
+ void Shutdown() { peer_ = nullptr; }
+
+ RefPtr<IceLogPromise> GetIceLog(const nsCString& aPattern) override {
+ return nullptr;
+ }
+
+ void ClearIceLog() override {}
+ void EnterPrivateMode() override {}
+ void ExitPrivateMode() override {}
+
+ void CreateIceCtx(const std::string& aName) override {}
+
+ nsresult SetIceConfig(const nsTArray<dom::RTCIceServer>& aIceServers,
+ dom::RTCIceTransportPolicy aIcePolicy) override {
+ return NS_OK;
+ }
+
+ void Destroy() override {}
+
+ // We will probably be able to move the proxy lookup stuff into
+ // this class once we move mtransport to its own process.
+ void SetProxyConfig(NrSocketProxyConfig&& aProxyConfig) override {}
+
+ void EnsureProvisionalTransport(const std::string& aTransportId,
+ const std::string& aLocalUfrag,
+ const std::string& aLocalPwd,
+ int aComponentCount) override {}
+
+ void SetTargetForDefaultLocalAddressLookup(const std::string& aTargetIp,
+ uint16_t aTargetPort) override {}
+
+ // We set default-route-only as late as possible because it depends on what
+ // capture permissions have been granted on the window, which could easily
+ // change between Init (ie; when the PC is created) and StartIceGathering
+ // (ie; when we set the local description).
+ void StartIceGathering(bool aDefaultRouteOnly, bool aObfuscateAddresses,
+ // TODO: It probably makes sense to look
+ // this up internally
+ const nsTArray<NrIceStunAddr>& aStunAddrs) override {}
+
+ void ActivateTransport(
+ const std::string& aTransportId, const std::string& aLocalUfrag,
+ const std::string& aLocalPwd, size_t aComponentCount,
+ const std::string& aUfrag, const std::string& aPassword,
+ const nsTArray<uint8_t>& aKeyDer, const nsTArray<uint8_t>& aCertDer,
+ SSLKEAType aAuthType, bool aDtlsClient, const DtlsDigestList& aDigests,
+ bool aPrivacyRequested) override {}
+
+ void RemoveTransportsExcept(
+ const std::set<std::string>& aTransportIds) override {}
+
+ void StartIceChecks(bool aIsControlling,
+ const std::vector<std::string>& aIceOptions) override {}
+
+ void AddIceCandidate(const std::string& aTransportId,
+ const std::string& aCandidate, const std::string& aUfrag,
+ const std::string& aObfuscatedAddress) override {}
+
+ void UpdateNetworkState(bool aOnline) override {}
+
+ RefPtr<dom::RTCStatsPromise> GetIceStats(const std::string& aTransportId,
+ DOMHighResTimeStamp aNow) override {
+ return nullptr;
+ }
+
+ void SendPacket(const std::string& aTransportId,
+ MediaPacket&& aPacket) override {
+ peer_->SignalPacketReceived(aTransportId, aPacket);
+ }
+
+ void SetState(const std::string& aTransportId, TransportLayer::State aState) {
+ MediaTransportHandler::OnStateChange(aTransportId, aState);
+ }
+
+ void SetRtcpState(const std::string& aTransportId,
+ TransportLayer::State aState) {
+ MediaTransportHandler::OnRtcpStateChange(aTransportId, aState);
+ }
+
+ private:
+ RefPtr<MediaTransportHandler> peer_;
+};
+
+class TestAgent {
+ public:
+ explicit TestAgent(const RefPtr<SharedWebrtcState>& aSharedState)
+ : conduit_control_(aSharedState->mCallWorkerThread),
+ audio_config_(109, "opus", 48000, 2, false),
+ call_(WebrtcCallWrapper::Create(mozilla::dom::RTCStatsTimestampMaker(),
+ nullptr, aSharedState)),
+ audio_conduit_(
+ AudioSessionConduit::Create(call_, test_utils->sts_target())),
+ audio_pipeline_(),
+ transport_(new LoopbackTransport) {
+ Unused << WaitFor(InvokeAsync(call_->mCallThread, __func__, [&] {
+ audio_conduit_->InitControl(&conduit_control_);
+ return GenericPromise::CreateAndResolve(true, "TestAgent()");
+ }));
+ }
+
+ static void Connect(TestAgent* client, TestAgent* server) {
+ LoopbackTransport::InitAndConnect(*client->transport_, *server->transport_);
+ }
+
+ virtual void CreatePipeline(const std::string& aTransportId) = 0;
+
+ void SetState_s(const std::string& aTransportId,
+ TransportLayer::State aState) {
+ transport_->SetState(aTransportId, aState);
+ }
+
+ void SetRtcpState_s(const std::string& aTransportId,
+ TransportLayer::State aState) {
+ transport_->SetRtcpState(aTransportId, aState);
+ }
+
+ void UpdateTransport_s(const std::string& aTransportId,
+ UniquePtr<MediaPipelineFilter>&& aFilter) {
+ audio_pipeline_->UpdateTransport_s(aTransportId, std::move(aFilter));
+ }
+
+ void Stop() {
+ MOZ_MTLOG(ML_DEBUG, "Stopping");
+
+ if (audio_pipeline_) {
+ audio_pipeline_->Stop();
+ }
+ if (audio_conduit_) {
+ conduit_control_.Update([](auto& aControl) {
+ aControl.mTransmitting = false;
+ aControl.mReceiving = false;
+ });
+ }
+ }
+
+ void Shutdown_s() { transport_->Shutdown(); }
+
+ void Shutdown() {
+ if (audio_pipeline_) {
+ audio_pipeline_->Shutdown();
+ }
+ if (audio_conduit_) {
+ Unused << WaitFor(audio_conduit_->Shutdown());
+ }
+ if (call_) {
+ call_->Destroy();
+ }
+ if (audio_track_) {
+ audio_track_->Destroy();
+ audio_track_ = nullptr;
+ }
+
+ test_utils->sts_target()->Dispatch(
+ WrapRunnable(this, &TestAgent::Shutdown_s),
+ nsISerialEventTarget::DISPATCH_SYNC);
+ }
+
+ uint32_t GetRemoteSSRC() {
+ return audio_conduit_->GetRemoteSSRC().valueOr(0);
+ }
+
+ uint32_t GetLocalSSRC() {
+ std::vector<uint32_t> res;
+ res = audio_conduit_->GetLocalSSRCs();
+ return res.empty() ? 0 : res[0];
+ }
+
+ int GetAudioRtpCountSent() { return audio_pipeline_->RtpPacketsSent(); }
+
+ int GetAudioRtpCountReceived() {
+ return audio_pipeline_->RtpPacketsReceived();
+ }
+
+ int GetAudioRtcpCountSent() { return audio_pipeline_->RtcpPacketsSent(); }
+
+ int GetAudioRtcpCountReceived() {
+ return audio_pipeline_->RtcpPacketsReceived();
+ }
+
+ protected:
+ ConcreteConduitControl conduit_control_;
+ AudioCodecConfig audio_config_;
+ RefPtr<WebrtcCallWrapper> call_;
+ RefPtr<AudioSessionConduit> audio_conduit_;
+ RefPtr<FakeAudioTrack> audio_track_;
+ // TODO(bcampen@mozilla.com): Right now this does not let us test RTCP in
+ // both directions; only the sender's RTCP is sent, but the receiver should
+ // be sending it too.
+ RefPtr<MediaPipeline> audio_pipeline_;
+ RefPtr<LoopbackTransport> transport_;
+};
+
+class TestAgentSend : public TestAgent {
+ public:
+ explicit TestAgentSend(const RefPtr<SharedWebrtcState>& aSharedState)
+ : TestAgent(aSharedState) {
+ conduit_control_.Update([&](auto& aControl) {
+ aControl.mAudioSendCodec = Some(audio_config_);
+ });
+ audio_track_ = new FakeAudioTrack();
+ }
+
+ virtual void CreatePipeline(const std::string& aTransportId) {
+ std::string test_pc;
+
+ RefPtr<MediaPipelineTransmit> audio_pipeline = new MediaPipelineTransmit(
+ test_pc, transport_, AbstractThread::MainThread(),
+ test_utils->sts_target(), false, audio_conduit_);
+
+ audio_pipeline->SetSendTrackOverride(audio_track_);
+ audio_pipeline->Start();
+ conduit_control_.Update(
+ [](auto& aControl) { aControl.mTransmitting = true; });
+
+ audio_pipeline_ = audio_pipeline;
+
+ audio_pipeline_->UpdateTransport_m(aTransportId, nullptr);
+ }
+};
+
+class TestAgentReceive : public TestAgent {
+ public:
+ explicit TestAgentReceive(const RefPtr<SharedWebrtcState>& aSharedState)
+ : TestAgent(aSharedState) {
+ conduit_control_.Update([&](auto& aControl) {
+ std::vector<AudioCodecConfig> codecs;
+ codecs.push_back(audio_config_);
+ aControl.mAudioRecvCodecs = codecs;
+ });
+ }
+
+ virtual void CreatePipeline(const std::string& aTransportId) {
+ std::string test_pc;
+
+ audio_pipeline_ = new MediaPipelineReceiveAudio(
+ test_pc, transport_, AbstractThread::MainThread(),
+ test_utils->sts_target(),
+ static_cast<AudioSessionConduit*>(audio_conduit_.get()), nullptr,
+ PRINCIPAL_HANDLE_NONE);
+
+ audio_pipeline_->Start();
+ conduit_control_.Update([](auto& aControl) { aControl.mReceiving = true; });
+
+ audio_pipeline_->UpdateTransport_m(aTransportId, std::move(bundle_filter_));
+ }
+
+ void SetBundleFilter(UniquePtr<MediaPipelineFilter>&& filter) {
+ bundle_filter_ = std::move(filter);
+ }
+
+ void UpdateTransport_s(const std::string& aTransportId,
+ UniquePtr<MediaPipelineFilter>&& filter) {
+ audio_pipeline_->UpdateTransport_s(aTransportId, std::move(filter));
+ }
+
+ private:
+ UniquePtr<MediaPipelineFilter> bundle_filter_;
+};
+
+void WaitFor(TimeDuration aDuration) {
+ bool done = false;
+ NS_DelayedDispatchToCurrentThread(
+ NS_NewRunnableFunction(__func__, [&] { done = true; }),
+ aDuration.ToMilliseconds());
+ SpinEventLoopUntil<ProcessFailureBehavior::IgnoreAndContinue>(
+ "WaitFor(TimeDuration aDuration)"_ns, [&] { return done; });
+}
+
+webrtc::AudioState::Config CreateAudioStateConfig() {
+ webrtc::AudioState::Config audio_state_config;
+ audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
+ webrtc::AudioProcessingBuilder audio_processing_builder;
+ audio_state_config.audio_processing = audio_processing_builder.Create();
+ audio_state_config.audio_device_module = new webrtc::FakeAudioDeviceModule();
+ return audio_state_config;
+}
+
+class MediaPipelineTest : public ::testing::Test {
+ public:
+ MediaPipelineTest()
+ : main_task_queue_(
+ WrapUnique<TaskQueueWrapper<DeletionPolicy::NonBlocking>>(
+ new MainAsCurrent())),
+ shared_state_(MakeAndAddRef<SharedWebrtcState>(
+ AbstractThread::MainThread(), CreateAudioStateConfig(),
+ already_AddRefed(
+ webrtc::CreateBuiltinAudioDecoderFactory().release()),
+ WrapUnique(new webrtc::NoTrialsConfig()))),
+ p1_(shared_state_),
+ p2_(shared_state_) {}
+
+ ~MediaPipelineTest() {
+ p1_.Shutdown();
+ p2_.Shutdown();
+ }
+
+ static void SetUpTestCase() {
+ test_utils = new MtransportTestUtils();
+ NSS_NoDB_Init(nullptr);
+ NSS_SetDomesticPolicy();
+ }
+
+ // Setup transport.
+ void InitTransports() {
+ test_utils->sts_target()->Dispatch(
+ WrapRunnableNM(&TestAgent::Connect, &p2_, &p1_),
+ nsISerialEventTarget::DISPATCH_SYNC);
+ }
+
+ // Verify RTP and RTCP
+ void TestAudioSend(bool aIsRtcpMux,
+ UniquePtr<MediaPipelineFilter>&& initialFilter = nullptr,
+ UniquePtr<MediaPipelineFilter>&& refinedFilter = nullptr,
+ unsigned int ms_until_filter_update = 500,
+ unsigned int ms_of_traffic_after_answer = 10000) {
+ bool bundle = !!(initialFilter);
+ // We do not support testing bundle without rtcp mux, since that doesn't
+ // make any sense.
+ ASSERT_FALSE(!aIsRtcpMux && bundle);
+
+ p2_.SetBundleFilter(std::move(initialFilter));
+
+ // Setup transport flows
+ InitTransports();
+
+ std::string transportId = aIsRtcpMux ? "mux" : "non-mux";
+ p1_.CreatePipeline(transportId);
+ p2_.CreatePipeline(transportId);
+
+ // Set state of transports to CONNECTING. MediaPipeline doesn't really care
+ // about this transition, but we're trying to simluate what happens in a
+ // real case.
+ RunOnSts([&] {
+ p1_.SetState_s(transportId, TransportLayer::TS_CONNECTING);
+ p1_.SetRtcpState_s(transportId, TransportLayer::TS_CONNECTING);
+ p2_.SetState_s(transportId, TransportLayer::TS_CONNECTING);
+ p2_.SetRtcpState_s(transportId, TransportLayer::TS_CONNECTING);
+ });
+
+ WaitFor(TimeDuration::FromMilliseconds(10));
+
+ // Set state of transports to OPEN (ie; connected). This should result in
+ // media flowing.
+ RunOnSts([&] {
+ p1_.SetState_s(transportId, TransportLayer::TS_OPEN);
+ p1_.SetRtcpState_s(transportId, TransportLayer::TS_OPEN);
+ p2_.SetState_s(transportId, TransportLayer::TS_OPEN);
+ p2_.SetRtcpState_s(transportId, TransportLayer::TS_OPEN);
+ });
+
+ if (bundle) {
+ WaitFor(TimeDuration::FromMilliseconds(ms_until_filter_update));
+
+ // Leaving refinedFilter not set implies we want to just update with
+ // the other side's SSRC
+ if (!refinedFilter) {
+ refinedFilter = MakeUnique<MediaPipelineFilter>();
+ // Might not be safe, strictly speaking.
+ refinedFilter->AddRemoteSSRC(p1_.GetLocalSSRC());
+ }
+
+ RunOnSts([&] {
+ p2_.UpdateTransport_s(transportId, std::move(refinedFilter));
+ });
+ }
+
+ // wait for some RTP/RTCP tx and rx to happen
+ WaitFor(TimeDuration::FromMilliseconds(ms_of_traffic_after_answer));
+
+ p1_.Stop();
+ p2_.Stop();
+
+ // wait for any packets in flight to arrive
+ WaitFor(TimeDuration::FromMilliseconds(200));
+
+ p1_.Shutdown();
+ p2_.Shutdown();
+
+ if (!bundle) {
+ // If we are filtering, allow the test-case to do this checking.
+ ASSERT_GE(p1_.GetAudioRtpCountSent(), 40);
+ ASSERT_EQ(p1_.GetAudioRtpCountReceived(), p2_.GetAudioRtpCountSent());
+ ASSERT_EQ(p1_.GetAudioRtpCountSent(), p2_.GetAudioRtpCountReceived());
+ }
+
+ // No RTCP packets should have been dropped, because we do not filter them.
+ // Calling ShutdownMedia_m on both pipelines does not stop the flow of
+ // RTCP. So, we might be off by one here.
+ ASSERT_LE(p2_.GetAudioRtcpCountReceived(), p1_.GetAudioRtcpCountSent());
+ ASSERT_GE(p2_.GetAudioRtcpCountReceived() + 1, p1_.GetAudioRtcpCountSent());
+ }
+
+ void TestAudioReceiverBundle(
+ bool bundle_accepted, UniquePtr<MediaPipelineFilter>&& initialFilter,
+ UniquePtr<MediaPipelineFilter>&& refinedFilter = nullptr,
+ unsigned int ms_until_answer = 500,
+ unsigned int ms_of_traffic_after_answer = 10000) {
+ TestAudioSend(true, std::move(initialFilter), std::move(refinedFilter),
+ ms_until_answer, ms_of_traffic_after_answer);
+ }
+
+ protected:
+ // main_task_queue_ has this type to make sure it goes through Delete() when
+ // we're destroyed.
+ UniquePtr<TaskQueueWrapper<DeletionPolicy::NonBlocking>> main_task_queue_;
+ const RefPtr<SharedWebrtcState> shared_state_;
+ TestAgentSend p1_;
+ TestAgentReceive p2_;
+};
+
+class MediaPipelineFilterTest : public ::testing::Test {
+ public:
+ bool Filter(MediaPipelineFilter& filter, uint32_t ssrc, uint8_t payload_type,
+ const Maybe<std::string>& mid = Nothing()) {
+ webrtc::RTPHeader header;
+ header.ssrc = ssrc;
+ header.payloadType = payload_type;
+ mid.apply([&](const auto& mid) { header.extension.mid = mid; });
+ return filter.Filter(header);
+ }
+};
+
+TEST_F(MediaPipelineFilterTest, TestConstruct) { MediaPipelineFilter filter; }
+
+TEST_F(MediaPipelineFilterTest, TestDefault) {
+ MediaPipelineFilter filter;
+ EXPECT_FALSE(Filter(filter, 233, 110));
+}
+
+TEST_F(MediaPipelineFilterTest, TestSSRCFilter) {
+ MediaPipelineFilter filter;
+ filter.AddRemoteSSRC(555);
+ EXPECT_TRUE(Filter(filter, 555, 110));
+ EXPECT_FALSE(Filter(filter, 556, 110));
+}
+
+#define SSRC(ssrc) \
+ ((ssrc >> 24) & 0xFF), ((ssrc >> 16) & 0xFF), ((ssrc >> 8) & 0xFF), \
+ (ssrc & 0xFF)
+#define REPORT_FRAGMENT(ssrc) \
+ SSRC(ssrc), 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
+
+#define RTCP_TYPEINFO(num_rrs, type, size) 0x80 + num_rrs, type, 0, size
+
+TEST_F(MediaPipelineFilterTest, TestMidFilter) {
+ MediaPipelineFilter filter;
+ const auto mid = Some(std::string("mid0"));
+ filter.SetRemoteMediaStreamId(mid);
+
+ EXPECT_FALSE(Filter(filter, 16, 110));
+ EXPECT_TRUE(Filter(filter, 16, 110, mid));
+ EXPECT_TRUE(Filter(filter, 16, 110));
+ EXPECT_FALSE(Filter(filter, 17, 110));
+
+ // The mid filter maintains a set of SSRCs. Adding a new SSRC should work
+ // and still allow previous SSRCs to work. Unrecognized SSRCs should still be
+ // filtered out.
+ EXPECT_TRUE(Filter(filter, 18, 111, mid));
+ EXPECT_TRUE(Filter(filter, 18, 111));
+ EXPECT_TRUE(Filter(filter, 16, 110));
+ EXPECT_FALSE(Filter(filter, 17, 110));
+}
+
+TEST_F(MediaPipelineFilterTest, TestPayloadTypeFilter) {
+ MediaPipelineFilter filter;
+ filter.AddUniquePT(110);
+ EXPECT_TRUE(Filter(filter, 555, 110));
+ EXPECT_FALSE(Filter(filter, 556, 111));
+}
+
+TEST_F(MediaPipelineFilterTest, TestSSRCMovedWithMid) {
+ MediaPipelineFilter filter;
+ const auto mid0 = Some(std::string("mid0"));
+ const auto mid1 = Some(std::string("mid1"));
+ filter.SetRemoteMediaStreamId(mid0);
+ ASSERT_TRUE(Filter(filter, 555, 110, mid0));
+ ASSERT_TRUE(Filter(filter, 555, 110));
+ // Present a new MID binding
+ ASSERT_FALSE(Filter(filter, 555, 110, mid1));
+ ASSERT_FALSE(Filter(filter, 555, 110));
+}
+
+TEST_F(MediaPipelineFilterTest, TestRemoteSDPNoSSRCs) {
+ // If the remote SDP doesn't have SSRCs, right now this is a no-op and
+ // there is no point of even incorporating a filter, but we make the
+ // behavior consistent to avoid confusion.
+ MediaPipelineFilter filter;
+ const auto mid = Some(std::string("mid0"));
+ filter.SetRemoteMediaStreamId(mid);
+ filter.AddUniquePT(111);
+ EXPECT_TRUE(Filter(filter, 555, 110, mid));
+ EXPECT_TRUE(Filter(filter, 555, 110));
+
+ // Update but remember binding./
+ MediaPipelineFilter filter2;
+
+ filter.Update(filter2);
+
+ // Ensure that the old SSRC still works.
+ EXPECT_TRUE(Filter(filter, 555, 110));
+
+ // Forget the previous binding
+ MediaPipelineFilter filter3;
+ filter3.SetRemoteMediaStreamId(Some(std::string("mid1")));
+ filter.Update(filter3);
+
+ ASSERT_FALSE(Filter(filter, 555, 110));
+}
+
+TEST_F(MediaPipelineTest, TestAudioSendNoMux) { TestAudioSend(false); }
+
+TEST_F(MediaPipelineTest, TestAudioSendMux) { TestAudioSend(true); }
+
+TEST_F(MediaPipelineTest, TestAudioSendBundle) {
+ auto filter = MakeUnique<MediaPipelineFilter>();
+ // These durations have to be _extremely_ long to have any assurance that
+ // some RTCP will be sent at all. This is because the first RTCP packet
+ // is sometimes sent before the transports are ready, which causes it to
+ // be dropped.
+ TestAudioReceiverBundle(
+ true, std::move(filter),
+ // We do not specify the filter for the remote description, so it will be
+ // set to something sane after a short time.
+ nullptr, 10000, 10000);
+
+ // Some packets should have been dropped, but not all
+ ASSERT_GT(p1_.GetAudioRtpCountSent(), p2_.GetAudioRtpCountReceived());
+ ASSERT_GT(p2_.GetAudioRtpCountReceived(), 40);
+ ASSERT_GT(p1_.GetAudioRtcpCountSent(), 1);
+}
+
+TEST_F(MediaPipelineTest, TestAudioSendEmptyBundleFilter) {
+ auto filter = MakeUnique<MediaPipelineFilter>();
+ auto bad_answer_filter = MakeUnique<MediaPipelineFilter>();
+ TestAudioReceiverBundle(true, std::move(filter),
+ std::move(bad_answer_filter));
+ // Filter is empty, so should drop everything.
+ ASSERT_EQ(0, p2_.GetAudioRtpCountReceived());
+}
+
+} // end namespace