Nagram/TMessagesProj/jni/voip/webrtc/call/degraded_call.h
2023-02-19 01:24:25 +04:00

221 lines
8.1 KiB
C++

/*
* Copyright (c) 2018 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.
*/
#ifndef CALL_DEGRADED_CALL_H_
#define CALL_DEGRADED_CALL_H_
#include <stddef.h>
#include <stdint.h>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/fec_controller.h"
#include "api/media_types.h"
#include "api/rtp_headers.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/test/simulated_network.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/simulated_network.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/task_queue.h"
#include "system_wrappers/include/clock.h"
#include "video/config/video_encoder_config.h"
namespace webrtc {
class DegradedCall : public Call, private PacketReceiver {
public:
struct TimeScopedNetworkConfig : public BuiltInNetworkBehaviorConfig {
TimeDelta duration = TimeDelta::PlusInfinity();
};
explicit DegradedCall(
std::unique_ptr<Call> call,
const std::vector<TimeScopedNetworkConfig>& send_configs,
const std::vector<TimeScopedNetworkConfig>& receive_configs);
~DegradedCall() override;
// Implements Call.
AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) override;
void DestroyAudioSendStream(AudioSendStream* send_stream) override;
AudioReceiveStreamInterface* CreateAudioReceiveStream(
const AudioReceiveStreamInterface::Config& config) override;
void DestroyAudioReceiveStream(
AudioReceiveStreamInterface* receive_stream) override;
VideoSendStream* CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config) override;
VideoSendStream* CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller) override;
void DestroyVideoSendStream(VideoSendStream* send_stream) override;
VideoReceiveStreamInterface* CreateVideoReceiveStream(
VideoReceiveStreamInterface::Config configuration) override;
void DestroyVideoReceiveStream(
VideoReceiveStreamInterface* receive_stream) override;
FlexfecReceiveStream* CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config config) override;
void DestroyFlexfecReceiveStream(
FlexfecReceiveStream* receive_stream) override;
void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) override;
PacketReceiver* Receiver() override;
RtpTransportControllerSendInterface* GetTransportControllerSend() override;
Stats GetStats() const override;
const FieldTrialsView& trials() const override;
TaskQueueBase* network_thread() const override;
TaskQueueBase* worker_thread() const override;
void SignalChannelNetworkState(MediaType media, NetworkState state) override;
void OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) override;
void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream,
uint32_t local_ssrc) override;
void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
uint32_t local_ssrc) override;
void OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
uint32_t local_ssrc) override;
void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream,
absl::string_view sync_group) override;
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
protected:
// Implements PacketReceiver.
DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
private:
class FakeNetworkPipeOnTaskQueue {
public:
FakeNetworkPipeOnTaskQueue(
TaskQueueBase* task_queue,
rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive,
Clock* clock,
std::unique_ptr<NetworkBehaviorInterface> network_behavior);
void SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options,
Transport* transport);
void SendRtcp(const uint8_t* packet, size_t length, Transport* transport);
void AddActiveTransport(Transport* transport);
void RemoveActiveTransport(Transport* transport);
private:
// Try to process packets on the fake network queue.
// Returns true if call resulted in a delayed process, false if queue empty.
bool Process();
Clock* const clock_;
TaskQueueBase* const task_queue_;
rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive_;
FakeNetworkPipe pipe_;
absl::optional<int64_t> next_process_ms_ RTC_GUARDED_BY(&task_queue_);
};
class ThreadedPacketReceiver : public PacketReceiver {
public:
ThreadedPacketReceiver(webrtc::TaskQueueBase* worker_thread,
webrtc::TaskQueueBase* network_thread,
rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive,
PacketReceiver* receiver);
~ThreadedPacketReceiver() override;
DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
private:
webrtc::TaskQueueBase* const worker_thread_;
webrtc::TaskQueueBase* const network_thread_;
rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive_;
webrtc::PacketReceiver* const receiver_;
};
// For audio/video send stream, a TransportAdapter instance is used to
// intercept packets to be sent, and put them into a common FakeNetworkPipe
// in such as way that they will eventually (unless dropped) be forwarded to
// the correct Transport for that stream.
class FakeNetworkPipeTransportAdapter : public Transport {
public:
FakeNetworkPipeTransportAdapter(FakeNetworkPipeOnTaskQueue* fake_network,
Call* call,
Clock* clock,
Transport* real_transport);
~FakeNetworkPipeTransportAdapter();
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
private:
FakeNetworkPipeOnTaskQueue* const network_pipe_;
Call* const call_;
Clock* const clock_;
Transport* const real_transport_;
};
void SetClientBitratePreferences(
const webrtc::BitrateSettings& preferences) override;
void UpdateSendNetworkConfig();
void UpdateReceiveNetworkConfig();
Clock* const clock_;
const std::unique_ptr<Call> call_;
// For cancelling tasks on the network thread when DegradedCall is destroyed
rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive_;
size_t send_config_index_;
const std::vector<TimeScopedNetworkConfig> send_configs_;
SimulatedNetwork* send_simulated_network_;
std::unique_ptr<FakeNetworkPipeOnTaskQueue> send_pipe_;
std::map<AudioSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>>
audio_send_transport_adapters_;
std::map<VideoSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>>
video_send_transport_adapters_;
size_t receive_config_index_;
const std::vector<TimeScopedNetworkConfig> receive_configs_;
SimulatedNetwork* receive_simulated_network_;
std::unique_ptr<FakeNetworkPipe> receive_pipe_;
std::unique_ptr<ThreadedPacketReceiver> packet_receiver_;
};
} // namespace webrtc
#endif // CALL_DEGRADED_CALL_H_