diff options
Diffstat (limited to 'dom/media/webrtc/jsapi/PeerConnectionCtx.cpp')
| -rw-r--r-- | dom/media/webrtc/jsapi/PeerConnectionCtx.cpp | 650 |
1 files changed, 650 insertions, 0 deletions
diff --git a/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp b/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp new file mode 100644 index 0000000000..9a8f27fb59 --- /dev/null +++ b/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp @@ -0,0 +1,650 @@ +/* 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/. */ + +#include "PeerConnectionCtx.h" + +#include "WebrtcGlobalStatsHistory.h" +#include "api/audio/audio_mixer.h" +#include "api/audio_codecs/builtin_audio_decoder_factory.h" +#include "modules/rtp_rtcp/source/rtp_header_extensions.h" +#include "call/audio_state.h" +#include "common/browser_logging/CSFLog.h" +#include "common/browser_logging/WebRtcLog.h" +#include "gmp-video-decode.h" // GMP_API_VIDEO_DECODER +#include "gmp-video-encode.h" // GMP_API_VIDEO_ENCODER +#include "libwebrtcglue/CallWorkerThread.h" +#include "modules/audio_device/include/fake_audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "modules/audio_processing/include/aec_dump.h" +#include "mozilla/UniquePtr.h" +#include "mozilla/dom/RTCPeerConnectionBinding.h" +#include "mozilla/Preferences.h" +#include "mozilla/Services.h" +#include "mozilla/StaticPtr.h" +#include "mozilla/Telemetry.h" +#include "mozilla/Types.h" +#include "mozilla/dom/RTCStatsReportBinding.h" +#include "nsCRTGlue.h" +#include "nsIIOService.h" +#include "nsIObserver.h" +#include "nsIObserverService.h" +#include "nsNetCID.h" // NS_SOCKETTRANSPORTSERVICE_CONTRACTID +#include "nsServiceManagerUtils.h" // do_GetService +#include "PeerConnectionImpl.h" +#include "prcvar.h" +#include "transport/runnable_utils.h" +#include "WebrtcGlobalChild.h" +#include "WebrtcGlobalInformation.h" + +static const char* pccLogTag = "PeerConnectionCtx"; +#ifdef LOGTAG +# undef LOGTAG +#endif +#define LOGTAG pccLogTag + +using namespace webrtc; + +namespace { +class DummyAudioMixer : public AudioMixer { + public: + bool AddSource(Source*) override { return true; } + void RemoveSource(Source*) override {} + void Mix(size_t, AudioFrame*) override { MOZ_CRASH("Unexpected call"); } +}; + +class DummyAudioProcessing : public AudioProcessing { + public: + int Initialize() override { + MOZ_CRASH("Unexpected call"); + return kNoError; + } + int Initialize(const ProcessingConfig&) override { return Initialize(); } + void ApplyConfig(const Config&) override { MOZ_CRASH("Unexpected call"); } + int proc_sample_rate_hz() const override { + MOZ_CRASH("Unexpected call"); + return 0; + } + int proc_split_sample_rate_hz() const override { + MOZ_CRASH("Unexpected call"); + return 0; + } + size_t num_input_channels() const override { + MOZ_CRASH("Unexpected call"); + return 0; + } + size_t num_proc_channels() const override { + MOZ_CRASH("Unexpected call"); + return 0; + } + size_t num_output_channels() const override { + MOZ_CRASH("Unexpected call"); + return 0; + } + size_t num_reverse_channels() const override { + MOZ_CRASH("Unexpected call"); + return 0; + } + void set_output_will_be_muted(bool) override { MOZ_CRASH("Unexpected call"); } + void SetRuntimeSetting(RuntimeSetting) override { + MOZ_CRASH("Unexpected call"); + } + bool PostRuntimeSetting(RuntimeSetting setting) override { return false; } + int ProcessStream(const int16_t* const, const StreamConfig&, + const StreamConfig&, int16_t* const) override { + MOZ_CRASH("Unexpected call"); + return kNoError; + } + int ProcessStream(const float* const*, const StreamConfig&, + const StreamConfig&, float* const*) override { + MOZ_CRASH("Unexpected call"); + return kNoError; + } + int ProcessReverseStream(const int16_t* const, const StreamConfig&, + const StreamConfig&, int16_t* const) override { + MOZ_CRASH("Unexpected call"); + return kNoError; + } + int ProcessReverseStream(const float* const*, const StreamConfig&, + const StreamConfig&, float* const*) override { + MOZ_CRASH("Unexpected call"); + return kNoError; + } + int AnalyzeReverseStream(const float* const*, const StreamConfig&) override { + MOZ_CRASH("Unexpected call"); + return kNoError; + } + bool GetLinearAecOutput( + rtc::ArrayView<std::array<float, 160>>) const override { + MOZ_CRASH("Unexpected call"); + return false; + } + void set_stream_analog_level(int) override { MOZ_CRASH("Unexpected call"); } + int recommended_stream_analog_level() const override { + MOZ_CRASH("Unexpected call"); + return -1; + } + int set_stream_delay_ms(int) override { + MOZ_CRASH("Unexpected call"); + return kNoError; + } + int stream_delay_ms() const override { + MOZ_CRASH("Unexpected call"); + return 0; + } + void set_stream_key_pressed(bool) override { MOZ_CRASH("Unexpected call"); } + bool CreateAndAttachAecDump(absl::string_view, int64_t, + rtc::TaskQueue*) override { + MOZ_CRASH("Unexpected call"); + return false; + } + bool CreateAndAttachAecDump(FILE*, int64_t, rtc::TaskQueue*) override { + MOZ_CRASH("Unexpected call"); + return false; + } + void AttachAecDump(std::unique_ptr<AecDump>) override { + MOZ_CRASH("Unexpected call"); + } + void DetachAecDump() override { MOZ_CRASH("Unexpected call"); } + AudioProcessingStats GetStatistics() override { + return AudioProcessingStats(); + } + AudioProcessingStats GetStatistics(bool) override { return GetStatistics(); } + AudioProcessing::Config GetConfig() const override { + MOZ_CRASH("Unexpected call"); + return Config(); + } +}; +} // namespace + +namespace mozilla { + +using namespace dom; + +SharedWebrtcState::SharedWebrtcState( + RefPtr<AbstractThread> aCallWorkerThread, + webrtc::AudioState::Config&& aAudioStateConfig, + RefPtr<webrtc::AudioDecoderFactory> aAudioDecoderFactory, + UniquePtr<webrtc::WebRtcKeyValueConfig> aTrials) + : mCallWorkerThread(std::move(aCallWorkerThread)), + mAudioStateConfig(std::move(aAudioStateConfig)), + mAudioDecoderFactory(std::move(aAudioDecoderFactory)), + mTrials(std::move(aTrials)) {} + +SharedWebrtcState::~SharedWebrtcState() = default; + +class PeerConnectionCtxObserver : public nsIObserver { + public: + NS_DECL_ISUPPORTS + + PeerConnectionCtxObserver() {} + + void Init() { + nsCOMPtr<nsIObserverService> observerService = + services::GetObserverService(); + if (!observerService) return; + + nsresult rv = NS_OK; + + rv = observerService->AddObserver(this, NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID, + false); + MOZ_ALWAYS_SUCCEEDS(rv); + rv = observerService->AddObserver(this, NS_IOSERVICE_OFFLINE_STATUS_TOPIC, + false); + MOZ_ALWAYS_SUCCEEDS(rv); + (void)rv; + } + + NS_IMETHOD Observe(nsISupports* aSubject, const char* aTopic, + const char16_t* aData) override { + if (strcmp(aTopic, NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID) == 0) { + CSFLogDebug(LOGTAG, "Shutting down PeerConnectionCtx"); + PeerConnectionCtx::Destroy(); + + nsCOMPtr<nsIObserverService> observerService = + services::GetObserverService(); + if (!observerService) return NS_ERROR_FAILURE; + + nsresult rv = observerService->RemoveObserver( + this, NS_IOSERVICE_OFFLINE_STATUS_TOPIC); + MOZ_ALWAYS_SUCCEEDS(rv); + rv = observerService->RemoveObserver(this, + NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID); + MOZ_ALWAYS_SUCCEEDS(rv); + + // Make sure we're not deleted while still inside ::Observe() + RefPtr<PeerConnectionCtxObserver> kungFuDeathGrip(this); + PeerConnectionCtx::gPeerConnectionCtxObserver = nullptr; + } + if (strcmp(aTopic, NS_IOSERVICE_OFFLINE_STATUS_TOPIC) == 0) { + if (NS_strcmp(aData, u"" NS_IOSERVICE_OFFLINE) == 0) { + CSFLogDebug(LOGTAG, "Updating network state to offline"); + PeerConnectionCtx::UpdateNetworkState(false); + } else if (NS_strcmp(aData, u"" NS_IOSERVICE_ONLINE) == 0) { + CSFLogDebug(LOGTAG, "Updating network state to online"); + PeerConnectionCtx::UpdateNetworkState(true); + } else { + CSFLogDebug(LOGTAG, "Received unsupported network state event"); + MOZ_CRASH(); + } + } + return NS_OK; + } + + private: + virtual ~PeerConnectionCtxObserver() { + nsCOMPtr<nsIObserverService> observerService = + services::GetObserverService(); + if (observerService) { + observerService->RemoveObserver(this, NS_IOSERVICE_OFFLINE_STATUS_TOPIC); + observerService->RemoveObserver(this, NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID); + } + } +}; + +NS_IMPL_ISUPPORTS(PeerConnectionCtxObserver, nsIObserver); + +PeerConnectionCtx* PeerConnectionCtx::gInstance; +StaticRefPtr<PeerConnectionCtxObserver> + PeerConnectionCtx::gPeerConnectionCtxObserver; + +nsresult PeerConnectionCtx::InitializeGlobal() { + MOZ_ASSERT(NS_IsMainThread()); + + nsresult res; + + if (!gInstance) { + CSFLogDebug(LOGTAG, "Creating PeerConnectionCtx"); + PeerConnectionCtx* ctx = new PeerConnectionCtx(); + + res = ctx->Initialize(); + PR_ASSERT(NS_SUCCEEDED(res)); + if (!NS_SUCCEEDED(res)) return res; + + gInstance = ctx; + + if (!PeerConnectionCtx::gPeerConnectionCtxObserver) { + PeerConnectionCtx::gPeerConnectionCtxObserver = + new PeerConnectionCtxObserver(); + PeerConnectionCtx::gPeerConnectionCtxObserver->Init(); + } + } + + EnableWebRtcLog(); + return NS_OK; +} + +PeerConnectionCtx* PeerConnectionCtx::GetInstance() { + MOZ_ASSERT(gInstance); + return gInstance; +} + +bool PeerConnectionCtx::isActive() { return gInstance; } + +void PeerConnectionCtx::Destroy() { + CSFLogDebug(LOGTAG, "%s", __FUNCTION__); + + if (gInstance) { + // Null out gInstance first, so PeerConnectionImpl doesn't try to use it + // in Cleanup. + auto* instance = gInstance; + gInstance = nullptr; + instance->Cleanup(); + delete instance; + } + + StopWebRtcLog(); +} + +template <typename T> +static void RecordCommonRtpTelemetry(const T& list, const T& lastList, + const bool isRemote) { + using namespace Telemetry; + for (const auto& s : list) { + const bool isAudio = s.mKind.Find(u"audio") != -1; + if (s.mPacketsLost.WasPassed() && s.mPacketsReceived.WasPassed()) { + if (const uint64_t total = + s.mPacketsLost.Value() + s.mPacketsReceived.Value()) { + HistogramID id = + isRemote ? (isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_PACKETLOSS_RATE + : WEBRTC_VIDEO_QUALITY_OUTBOUND_PACKETLOSS_RATE) + : (isAudio ? WEBRTC_AUDIO_QUALITY_INBOUND_PACKETLOSS_RATE + : WEBRTC_VIDEO_QUALITY_INBOUND_PACKETLOSS_RATE); + Accumulate(id, (s.mPacketsLost.Value() * 1000) / total); + } + } + if (s.mJitter.WasPassed()) { + HistogramID id = isRemote + ? (isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_JITTER + : WEBRTC_VIDEO_QUALITY_OUTBOUND_JITTER) + : (isAudio ? WEBRTC_AUDIO_QUALITY_INBOUND_JITTER + : WEBRTC_VIDEO_QUALITY_INBOUND_JITTER); + Accumulate(id, s.mJitter.Value() * 1000); + } + } +} + +// Telemetry reporting every second after start of first call. +// The threading model around the media pipelines is weird: +// - The pipelines are containers, +// - containers that are only safe on main thread, with members only safe on +// STS, +// - hence the there and back again approach. + +void PeerConnectionCtx::DeliverStats( + UniquePtr<dom::RTCStatsReportInternal>&& aReport) { + using namespace Telemetry; + + // First, get reports from a second ago, if any, for calculations below + UniquePtr<dom::RTCStatsReportInternal> lastReport; + { + auto i = mLastReports.find(aReport->mPcid); + if (i != mLastReports.end()) { + lastReport = std::move(i->second); + } else { + lastReport = MakeUnique<dom::RTCStatsReportInternal>(); + } + } + // Record Telemetery + RecordCommonRtpTelemetry(aReport->mInboundRtpStreamStats, + lastReport->mInboundRtpStreamStats, false); + // Record bandwidth telemetry + for (const auto& s : aReport->mInboundRtpStreamStats) { + if (s.mBytesReceived.WasPassed()) { + const bool isAudio = s.mKind.Find(u"audio") != -1; + for (const auto& lastS : lastReport->mInboundRtpStreamStats) { + if (lastS.mId == s.mId) { + int32_t deltaMs = s.mTimestamp.Value() - lastS.mTimestamp.Value(); + // In theory we're called every second, so delta *should* be in that + // range. Small deltas could cause errors due to division + if (deltaMs < 500 || deltaMs > 60000 || + !lastS.mBytesReceived.WasPassed()) { + break; + } + HistogramID id = isAudio + ? WEBRTC_AUDIO_QUALITY_INBOUND_BANDWIDTH_KBITS + : WEBRTC_VIDEO_QUALITY_INBOUND_BANDWIDTH_KBITS; + // We could accumulate values until enough time has passed + // and then Accumulate() but this isn't that important + Accumulate( + id, + ((s.mBytesReceived.Value() - lastS.mBytesReceived.Value()) * 8) / + deltaMs); + break; + } + } + } + } + RecordCommonRtpTelemetry(aReport->mRemoteInboundRtpStreamStats, + lastReport->mRemoteInboundRtpStreamStats, true); + for (const auto& s : aReport->mRemoteInboundRtpStreamStats) { + if (s.mRoundTripTime.WasPassed()) { + const bool isAudio = s.mKind.Find(u"audio") != -1; + HistogramID id = isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_RTT + : WEBRTC_VIDEO_QUALITY_OUTBOUND_RTT; + Accumulate(id, s.mRoundTripTime.Value() * 1000); + } + } + + mLastReports[aReport->mPcid] = std::move(aReport); +} + +void PeerConnectionCtx::EverySecondTelemetryCallback_m(nsITimer* timer, + void* closure) { + MOZ_ASSERT(NS_IsMainThread()); + MOZ_ASSERT(PeerConnectionCtx::isActive()); + + for (auto& idAndPc : GetInstance()->mPeerConnections) { + if (!idAndPc.second->IsClosed()) { + idAndPc.second->GetStats(nullptr, true) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [=](UniquePtr<dom::RTCStatsReportInternal>&& aReport) { + if (PeerConnectionCtx::isActive()) { + PeerConnectionCtx::GetInstance()->DeliverStats( + std::move(aReport)); + } + }, + [=](nsresult aError) {}); + idAndPc.second->CollectConduitTelemetryData(); + } + } +} + +void PeerConnectionCtx::UpdateNetworkState(bool online) { + auto ctx = GetInstance(); + if (ctx->mPeerConnections.empty()) { + return; + } + for (auto pc : ctx->mPeerConnections) { + pc.second->UpdateNetworkState(online); + } +} + +SharedWebrtcState* PeerConnectionCtx::GetSharedWebrtcState() const { + MOZ_ASSERT(NS_IsMainThread()); + return mSharedWebrtcState; +} + +void PeerConnectionCtx::RemovePeerConnection(const std::string& aKey) { + MOZ_ASSERT(NS_IsMainThread()); + auto it = mPeerConnections.find(aKey); + if (it != mPeerConnections.end()) { + if (it->second->GetFinalStats() && !it->second->LongTermStatsIsDisabled()) { + WebrtcGlobalInformation::StashStats(*(it->second->GetFinalStats())); + } + nsAutoString pcId = NS_ConvertASCIItoUTF16(it->second->GetName().c_str()); + if (XRE_IsContentProcess()) { + if (auto* child = WebrtcGlobalChild::Get(); child) { + auto pcId = NS_ConvertASCIItoUTF16(it->second->GetName().c_str()); + child->SendPeerConnectionFinalStats(*(it->second->GetFinalStats())); + child->SendPeerConnectionDestroyed(pcId); + } + } else { + using Update = WebrtcGlobalInformation::PcTrackingUpdate; + auto update = Update::Remove(pcId); + auto finalStats = + MakeUnique<RTCStatsReportInternal>(*(it->second->GetFinalStats())); + WebrtcGlobalStatsHistory::Record(std::move(finalStats)); + WebrtcGlobalInformation::PeerConnectionTracking(update); + } + + mPeerConnections.erase(it); + if (mPeerConnections.empty()) { + mSharedWebrtcState = nullptr; + StopTelemetryTimer(); + } + } +} + +void PeerConnectionCtx::AddPeerConnection(const std::string& aKey, + PeerConnectionImpl* aPeerConnection) { + MOZ_ASSERT(NS_IsMainThread()); + MOZ_ASSERT(mPeerConnections.count(aKey) == 0, + "PeerConnection with this key should not already exist"); + if (mPeerConnections.empty()) { + AudioState::Config audioStateConfig; + audioStateConfig.audio_mixer = new rtc::RefCountedObject<DummyAudioMixer>(); + AudioProcessingBuilder audio_processing_builder; + audioStateConfig.audio_processing = + new rtc::RefCountedObject<DummyAudioProcessing>(); + audioStateConfig.audio_device_module = + new rtc::RefCountedObject<FakeAudioDeviceModule>(); + + SharedThreadPoolWebRtcTaskQueueFactory taskQueueFactory; + constexpr bool supportTailDispatch = true; + // Note the NonBlocking DeletionPolicy! + // This task queue is passed into libwebrtc as a raw pointer. + // WebrtcCallWrapper guarantees that it outlives its webrtc::Call instance. + // Outside of libwebrtc we must use ref-counting to either the + // WebrtcCallWrapper or to the CallWorkerThread to keep it alive. + auto callWorkerThread = + WrapUnique(taskQueueFactory + .CreateTaskQueueWrapper<DeletionPolicy::NonBlocking>( + "CallWorker", supportTailDispatch, + webrtc::TaskQueueFactory::Priority::NORMAL, + MediaThreadType::WEBRTC_CALL_THREAD) + .release()); + + UniquePtr<webrtc::WebRtcKeyValueConfig> trials = + WrapUnique(new NoTrialsConfig()); + + mSharedWebrtcState = MakeAndAddRef<SharedWebrtcState>( + new CallWorkerThread(std::move(callWorkerThread)), + std::move(audioStateConfig), + already_AddRefed(CreateBuiltinAudioDecoderFactory().release()), + std::move(trials)); + StartTelemetryTimer(); + } + auto pcId = NS_ConvertASCIItoUTF16(aPeerConnection->GetName().c_str()); + if (XRE_IsContentProcess()) { + if (auto* child = WebrtcGlobalChild::Get(); child) { + child->SendPeerConnectionCreated( + pcId, aPeerConnection->LongTermStatsIsDisabled()); + } + } else { + using Update = WebrtcGlobalInformation::PcTrackingUpdate; + auto update = Update::Add(pcId, aPeerConnection->LongTermStatsIsDisabled()); + WebrtcGlobalInformation::PeerConnectionTracking(update); + } + mPeerConnections[aKey] = aPeerConnection; +} + +PeerConnectionImpl* PeerConnectionCtx::GetPeerConnection( + const std::string& aKey) const { + MOZ_ASSERT(NS_IsMainThread()); + auto iterator = mPeerConnections.find(aKey); + if (iterator == mPeerConnections.end()) { + return nullptr; + } + return iterator->second; +} + +void PeerConnectionCtx::ClearClosedStats() { + for (auto& [id, pc] : mPeerConnections) { + Unused << id; + if (pc->IsClosed()) { + // Rare case + pc->DisableLongTermStats(); + } + } +} + +nsresult PeerConnectionCtx::Initialize() { + MOZ_ASSERT(NS_IsMainThread()); + initGMP(); + SdpRidAttributeList::kMaxRidLength = + webrtc::BaseRtpStringExtension::kMaxValueSizeBytes; + + if (XRE_IsContentProcess()) { + WebrtcGlobalChild::Get(); + } + + return NS_OK; +} + +nsresult PeerConnectionCtx::StartTelemetryTimer() { + return NS_NewTimerWithFuncCallback(getter_AddRefs(mTelemetryTimer), + EverySecondTelemetryCallback_m, this, 1000, + nsITimer::TYPE_REPEATING_PRECISE_CAN_SKIP, + "EverySecondTelemetryCallback_m"); +} + +void PeerConnectionCtx::StopTelemetryTimer() { + if (mTelemetryTimer) { + mTelemetryTimer->Cancel(); + mTelemetryTimer = nullptr; + } +} + +static void GMPReady_m() { + if (PeerConnectionCtx::isActive()) { + PeerConnectionCtx::GetInstance()->onGMPReady(); + } +}; + +static void GMPReady() { + GetMainThreadSerialEventTarget()->Dispatch(WrapRunnableNM(&GMPReady_m), + NS_DISPATCH_NORMAL); +}; + +void PeerConnectionCtx::initGMP() { + mGMPService = do_GetService("@mozilla.org/gecko-media-plugin-service;1"); + + if (!mGMPService) { + CSFLogError(LOGTAG, "%s failed to get the gecko-media-plugin-service", + __FUNCTION__); + return; + } + + nsCOMPtr<nsIThread> thread; + nsresult rv = mGMPService->GetThread(getter_AddRefs(thread)); + + if (NS_FAILED(rv)) { + mGMPService = nullptr; + CSFLogError(LOGTAG, + "%s failed to get the gecko-media-plugin thread, err=%u", + __FUNCTION__, static_cast<unsigned>(rv)); + return; + } + + // presumes that all GMP dir scans have been queued for the GMPThread + thread->Dispatch(WrapRunnableNM(&GMPReady), NS_DISPATCH_NORMAL); +} + +nsresult PeerConnectionCtx::Cleanup() { + CSFLogDebug(LOGTAG, "%s", __FUNCTION__); + MOZ_ASSERT(NS_IsMainThread()); + + mQueuedJSEPOperations.Clear(); + mGMPService = nullptr; + mTransportHandler = nullptr; + for (auto& [id, pc] : mPeerConnections) { + (void)id; + pc->Close(); + } + mPeerConnections.clear(); + mSharedWebrtcState = nullptr; + return NS_OK; +} + +void PeerConnectionCtx::queueJSEPOperation(nsIRunnable* aOperation) { + mQueuedJSEPOperations.AppendElement(aOperation); +} + +void PeerConnectionCtx::onGMPReady() { + mGMPReady = true; + for (size_t i = 0; i < mQueuedJSEPOperations.Length(); ++i) { + mQueuedJSEPOperations[i]->Run(); + } + mQueuedJSEPOperations.Clear(); +} + +bool PeerConnectionCtx::gmpHasH264() { + if (!mGMPService) { + return false; + } + + // XXX I'd prefer if this was all known ahead of time... + + AutoTArray<nsCString, 1> tags; + tags.AppendElement("h264"_ns); + + bool has_gmp; + nsresult rv; + rv = mGMPService->HasPluginForAPI(nsLiteralCString(GMP_API_VIDEO_ENCODER), + tags, &has_gmp); + if (NS_FAILED(rv) || !has_gmp) { + return false; + } + + rv = mGMPService->HasPluginForAPI(nsLiteralCString(GMP_API_VIDEO_DECODER), + tags, &has_gmp); + if (NS_FAILED(rv) || !has_gmp) { + return false; + } + + return true; +} + +} // namespace mozilla |