webrtc_m130/call/degraded_call.h
Erik Språng eea605deeb Make fake network degradation work also for sent audio
Previously this functionality only worked correctly with a single
Transport instance, meaning a single video track.
This CL moves the transport pointer from being a member in
FakeNetworkPipe to being set on each packet, so that e.g. audio packets
point to the audio transport and video packet to the video transport.
This means we need a separate adapter per stream in DegradedCall.
Additionally, since Transport instances can potentially be destroyed
before it's time to forward the message to it, we need to keep track
of which instance that are live and ignore packets we can't forward.

Bug: None
Change-Id: I314d431c04ff81c3859cf661e2722c99342f785e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148586
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28831}
2019-08-12 15:20:18 +00:00

172 lines
6.1 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 "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/test/simulated_network.h"
#include "api/video_codecs/video_encoder_config.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 "modules/utility/include/process_thread.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"
namespace webrtc {
class DegradedCall : public Call, private PacketReceiver {
public:
explicit DegradedCall(
std::unique_ptr<Call> call,
absl::optional<BuiltInNetworkBehaviorConfig> send_config,
absl::optional<BuiltInNetworkBehaviorConfig> receive_config,
TaskQueueFactory* task_queue_factory);
~DegradedCall() override;
// Implements Call.
AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) override;
void DestroyAudioSendStream(AudioSendStream* send_stream) override;
AudioReceiveStream* CreateAudioReceiveStream(
const AudioReceiveStream::Config& config) override;
void DestroyAudioReceiveStream(AudioReceiveStream* 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;
VideoReceiveStream* CreateVideoReceiveStream(
VideoReceiveStream::Config configuration) override;
void DestroyVideoReceiveStream(VideoReceiveStream* receive_stream) override;
FlexfecReceiveStream* CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) override;
void DestroyFlexfecReceiveStream(
FlexfecReceiveStream* receive_stream) override;
PacketReceiver* Receiver() override;
RtpTransportControllerSendInterface* GetTransportControllerSend() override;
Stats GetStats() const override;
void SignalChannelNetworkState(MediaType media, NetworkState state) override;
void OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) 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(
TaskQueueFactory* task_queue_factory,
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_;
rtc::TaskQueue task_queue_;
FakeNetworkPipe pipe_;
absl::optional<int64_t> next_process_ms_ RTC_GUARDED_BY(&task_queue_);
};
// 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_;
};
Clock* const clock_;
const std::unique_ptr<Call> call_;
TaskQueueFactory* const task_queue_factory_;
void SetClientBitratePreferences(
const webrtc::BitrateSettings& preferences) override {}
const absl::optional<BuiltInNetworkBehaviorConfig> send_config_;
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_;
const absl::optional<BuiltInNetworkBehaviorConfig> receive_config_;
SimulatedNetwork* receive_simulated_network_;
std::unique_ptr<FakeNetworkPipe> receive_pipe_;
};
} // namespace webrtc
#endif // CALL_DEGRADED_CALL_H_