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/* 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/. */
#ifndef peerconnectionctx_h___h__
#define peerconnectionctx_h___h__
#include <map>
#include <string>
#include "api/field_trials_view.h"
#include "call/audio_state.h"
#include "MediaTransportHandler.h" // Mostly for IceLogPromise
#include "mozIGeckoMediaPluginService.h"
#include "mozilla/StaticPtr.h"
#include "nsIRunnable.h"
#include "PeerConnectionImpl.h"
class WebrtcLogSinkHandle;
namespace webrtc {
class AudioDecoderFactory;
// Used for testing in mediapipeline_unittest.cpp, MockCall.h
class NoTrialsConfig : public FieldTrialsView {
public:
NoTrialsConfig() = default;
std::string Lookup(absl::string_view key) const override {
// Upstream added a new default field trial string for
// CongestionWindow, that we don't want. In
// third_party/libwebrtc/rtc_base/experiments/rate_control_settings.cc
// they set kCongestionWindowDefaultFieldTrialString to
// "QueueSize:350,MinBitrate:30000,DropFrame:true". With QueueSize
// set, GoogCcNetworkController::UpdateCongestionWindowSize is
// called. Because negative values are calculated in
// feedback_rtt, an assert fires when calculating data_window in
// GoogCcNetworkController::UpdateCongestionWindowSize. We probably
// need to figure out why we're calculating negative feedback_rtt.
// See Bug 1780620.
if ("WebRTC-CongestionWindow" == key) {
return std::string("MinBitrate:30000,DropFrame:true");
}
return std::string();
}
};
} // namespace webrtc
namespace mozilla {
class PeerConnectionCtxObserver;
namespace dom {
class WebrtcGlobalInformation;
}
/**
* Refcounted class containing state shared across all PeerConnections and all
* Call instances. Managed by PeerConnectionCtx, and kept around while there are
* registered peer connections.
*/
class SharedWebrtcState {
public:
NS_INLINE_DECL_THREADSAFE_REFCOUNTING(SharedWebrtcState)
SharedWebrtcState(RefPtr<AbstractThread> aCallWorkerThread,
webrtc::AudioState::Config&& aAudioStateConfig,
RefPtr<webrtc::AudioDecoderFactory> aAudioDecoderFactory,
UniquePtr<webrtc::FieldTrialsView> aTrials);
// A global Call worker thread shared between all Call instances. Implements
// AbstractThread for running tasks that call into a Call instance through its
// webrtc::TaskQueue member, and for using AbstractThread-specific higher
// order constructs like StateMirroring.
const RefPtr<AbstractThread> mCallWorkerThread;
// AudioState config containing dummy implementations of the audio stack,
// since we use our own audio stack instead. Shared across all Call instances.
const webrtc::AudioState::Config mAudioStateConfig;
// AudioDecoderFactory instance shared between calls, to limit the number of
// instances in large calls.
const RefPtr<webrtc::AudioDecoderFactory> mAudioDecoderFactory;
// Trials instance shared between calls, to limit the number of instances in
// large calls.
const UniquePtr<webrtc::FieldTrialsView> mTrials;
private:
virtual ~SharedWebrtcState();
};
// A class to hold some of the singleton objects we need:
// * The global PeerConnectionImpl table and its associated lock.
// * Stats report objects for PCs that are gone
// * GMP related state
// * Upstream webrtc state shared across all Calls (processing thread)
class PeerConnectionCtx {
public:
static nsresult InitializeGlobal();
static PeerConnectionCtx* GetInstance();
static bool isActive();
static void Destroy();
bool isReady() {
// If mGMPService is not set, we aren't using GMP.
if (mGMPService) {
return mGMPReady;
}
return true;
}
void queueJSEPOperation(nsIRunnable* aJSEPOperation);
void onGMPReady();
bool gmpHasH264();
static void UpdateNetworkState(bool online);
RefPtr<MediaTransportHandler> GetTransportHandler() const {
return mTransportHandler;
}
SharedWebrtcState* GetSharedWebrtcState() const;
void RemovePeerConnection(const std::string& aKey);
void AddPeerConnection(const std::string& aKey,
PeerConnectionImpl* aPeerConnection);
PeerConnectionImpl* GetPeerConnection(const std::string& aKey) const;
template <typename Function>
void ForEachPeerConnection(Function&& aFunction) const {
MOZ_ASSERT(NS_IsMainThread());
for (const auto& pair : mPeerConnections) {
aFunction(pair.second);
}
}
void ClearClosedStats();
private:
std::map<const std::string, PeerConnectionImpl*> mPeerConnections;
PeerConnectionCtx();
// This is a singleton, so don't copy construct it, etc.
PeerConnectionCtx(const PeerConnectionCtx& other) = delete;
void operator=(const PeerConnectionCtx& other) = delete;
virtual ~PeerConnectionCtx() = default;
nsresult Initialize();
nsresult StartTelemetryTimer();
void StopTelemetryTimer();
nsresult Cleanup();
void initGMP();
static void EverySecondTelemetryCallback_m(nsITimer* timer, void*);
nsCOMPtr<nsITimer> mTelemetryTimer;
private:
void DeliverStats(UniquePtr<dom::RTCStatsReportInternal>&& aReport);
std::map<nsString, UniquePtr<dom::RTCStatsReportInternal>> mLastReports;
// We cannot form offers/answers properly until the Gecko Media Plugin stuff
// has been initted, which is a complicated mess of thread dispatches,
// including sync dispatches to main. So, we need to be able to queue up
// offer creation (or SetRemote, when we're the answerer) until all of this is
// ready to go, since blocking on this init is just begging for deadlock.
nsCOMPtr<mozIGeckoMediaPluginService> mGMPService;
bool mGMPReady;
nsTArray<nsCOMPtr<nsIRunnable>> mQueuedJSEPOperations;
const RefPtr<WebrtcLogSinkHandle> mLogHandle;
// Not initted, just for ICE logging stuff
RefPtr<MediaTransportHandler> mTransportHandler;
// State used by libwebrtc that needs to be shared across all PeerConnections
// and all Call instances. Set while there is at least one peer connection
// registered. CallWrappers can hold a ref to this object to be sure members
// are alive long enough.
RefPtr<SharedWebrtcState> mSharedWebrtcState;
static PeerConnectionCtx* gInstance;
public:
static mozilla::StaticRefPtr<mozilla::PeerConnectionCtxObserver>
gPeerConnectionCtxObserver;
};
} // namespace mozilla
#endif
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