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Erik Språng09708512018-03-14 15:16:50 +01001/*
2 * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef CALL_DEGRADED_CALL_H_
12#define CALL_DEGRADED_CALL_H_
13
Yves Gerey3e707812018-11-28 16:47:49 +010014#include <stddef.h>
15#include <stdint.h>
Erik Språng09708512018-03-14 15:16:50 +010016#include <memory>
17
Danil Chapovalovb9b146c2018-06-15 12:28:07 +020018#include "absl/types/optional.h"
Erik Språng09708512018-03-14 15:16:50 +010019#include "api/call/transport.h"
Yves Gerey3e707812018-11-28 16:47:49 +010020#include "api/fec_controller.h"
Steve Anton10542f22019-01-11 09:11:00 -080021#include "api/media_types.h"
Yves Gerey3e707812018-11-28 16:47:49 +010022#include "api/rtp_headers.h"
Artem Titov3229d652018-08-17 13:00:54 +020023#include "api/test/simulated_network.h"
Yves Gerey3e707812018-11-28 16:47:49 +010024#include "api/video_codecs/video_encoder_config.h"
25#include "call/audio_receive_stream.h"
26#include "call/audio_send_stream.h"
Erik Språng09708512018-03-14 15:16:50 +010027#include "call/call.h"
28#include "call/fake_network_pipe.h"
Yves Gerey3e707812018-11-28 16:47:49 +010029#include "call/flexfec_receive_stream.h"
30#include "call/packet_receiver.h"
31#include "call/rtp_transport_controller_send_interface.h"
Artem Titov3229d652018-08-17 13:00:54 +020032#include "call/simulated_network.h"
Yves Gerey3e707812018-11-28 16:47:49 +010033#include "call/video_receive_stream.h"
34#include "call/video_send_stream.h"
Sebastian Jansson836fee12019-02-08 16:08:10 +010035#include "modules/include/module.h"
Erik Språng09708512018-03-14 15:16:50 +010036#include "modules/utility/include/process_thread.h"
Steve Anton10542f22019-01-11 09:11:00 -080037#include "rtc_base/bitrate_allocation_strategy.h"
38#include "rtc_base/copy_on_write_buffer.h"
Yves Gerey3e707812018-11-28 16:47:49 +010039#include "rtc_base/network/sent_packet.h"
Erik Språng09708512018-03-14 15:16:50 +010040#include "system_wrappers/include/clock.h"
41
42namespace webrtc {
Sebastian Jansson836fee12019-02-08 16:08:10 +010043class FakeNetworkPipeModule : public Module {
44 public:
45 FakeNetworkPipeModule(
46 Clock* clock,
47 std::unique_ptr<NetworkBehaviorInterface> network_behavior,
48 Transport* transport);
49 ~FakeNetworkPipeModule() override;
50 void SendRtp(const uint8_t* packet,
51 size_t length,
52 const PacketOptions& options);
53 void SendRtcp(const uint8_t* packet, size_t length);
54
55 // Implements Module interface
56 int64_t TimeUntilNextProcess() override;
57 void ProcessThreadAttached(ProcessThread* process_thread) override;
58 void Process() override;
59
60 private:
61 void MaybeResumeProcess();
62 FakeNetworkPipe pipe_;
63 rtc::CriticalSection process_thread_lock_;
64 ProcessThread* process_thread_ RTC_GUARDED_BY(process_thread_lock_) = nullptr;
65 bool pending_process_ RTC_GUARDED_BY(process_thread_lock_) = false;
66};
Erik Språng09708512018-03-14 15:16:50 +010067
68class DegradedCall : public Call, private Transport, private PacketReceiver {
69 public:
Artem Titov3229d652018-08-17 13:00:54 +020070 explicit DegradedCall(
71 std::unique_ptr<Call> call,
Artem Titov75e36472018-10-08 12:28:56 +020072 absl::optional<BuiltInNetworkBehaviorConfig> send_config,
73 absl::optional<BuiltInNetworkBehaviorConfig> receive_config);
Erik Språng09708512018-03-14 15:16:50 +010074 ~DegradedCall() override;
75
76 // Implements Call.
77 AudioSendStream* CreateAudioSendStream(
78 const AudioSendStream::Config& config) override;
79 void DestroyAudioSendStream(AudioSendStream* send_stream) override;
80
81 AudioReceiveStream* CreateAudioReceiveStream(
82 const AudioReceiveStream::Config& config) override;
83 void DestroyAudioReceiveStream(AudioReceiveStream* receive_stream) override;
84
85 VideoSendStream* CreateVideoSendStream(
86 VideoSendStream::Config config,
87 VideoEncoderConfig encoder_config) override;
88 VideoSendStream* CreateVideoSendStream(
89 VideoSendStream::Config config,
90 VideoEncoderConfig encoder_config,
91 std::unique_ptr<FecController> fec_controller) override;
92 void DestroyVideoSendStream(VideoSendStream* send_stream) override;
93
94 VideoReceiveStream* CreateVideoReceiveStream(
95 VideoReceiveStream::Config configuration) override;
96 void DestroyVideoReceiveStream(VideoReceiveStream* receive_stream) override;
97
98 FlexfecReceiveStream* CreateFlexfecReceiveStream(
99 const FlexfecReceiveStream::Config& config) override;
100 void DestroyFlexfecReceiveStream(
101 FlexfecReceiveStream* receive_stream) override;
102
103 PacketReceiver* Receiver() override;
104
105 RtpTransportControllerSendInterface* GetTransportControllerSend() override;
106
107 Stats GetStats() const override;
108
109 void SetBitrateAllocationStrategy(
110 std::unique_ptr<rtc::BitrateAllocationStrategy>
111 bitrate_allocation_strategy) override;
112
113 void SignalChannelNetworkState(MediaType media, NetworkState state) override;
Stefan Holmer64be7fa2018-10-04 15:21:55 +0200114 void OnAudioTransportOverheadChanged(
115 int transport_overhead_per_packet) override;
Erik Språng09708512018-03-14 15:16:50 +0100116 void OnSentPacket(const rtc::SentPacket& sent_packet) override;
117
118 protected:
119 // Implements Transport.
120 bool SendRtp(const uint8_t* packet,
121 size_t length,
122 const PacketOptions& options) override;
123
124 bool SendRtcp(const uint8_t* packet, size_t length) override;
125
126 // Implements PacketReceiver.
127 DeliveryStatus DeliverPacket(MediaType media_type,
128 rtc::CopyOnWriteBuffer packet,
Niels Möller70082872018-08-07 11:03:12 +0200129 int64_t packet_time_us) override;
Erik Språng09708512018-03-14 15:16:50 +0100130
131 private:
132 Clock* const clock_;
133 const std::unique_ptr<Call> call_;
134
Piotr (Peter) Slatalacc8e8bb2018-11-15 08:26:19 -0800135 void MediaTransportChange(MediaTransportInterface* media_transport) override;
Piotr (Peter) Slatala7fbfaa42019-03-18 10:31:54 -0700136 void SetClientBitratePreferences(
137 const webrtc::BitrateSettings& preferences) override {}
Artem Titov75e36472018-10-08 12:28:56 +0200138 const absl::optional<BuiltInNetworkBehaviorConfig> send_config_;
Erik Språng09708512018-03-14 15:16:50 +0100139 const std::unique_ptr<ProcessThread> send_process_thread_;
Artem Titov3229d652018-08-17 13:00:54 +0200140 SimulatedNetwork* send_simulated_network_;
Sebastian Jansson836fee12019-02-08 16:08:10 +0100141 std::unique_ptr<FakeNetworkPipeModule> send_pipe_;
Erik Språng09708512018-03-14 15:16:50 +0100142 size_t num_send_streams_;
143
Artem Titov75e36472018-10-08 12:28:56 +0200144 const absl::optional<BuiltInNetworkBehaviorConfig> receive_config_;
Artem Titov3229d652018-08-17 13:00:54 +0200145 SimulatedNetwork* receive_simulated_network_;
Erik Språng09708512018-03-14 15:16:50 +0100146 std::unique_ptr<FakeNetworkPipe> receive_pipe_;
147};
148
149} // namespace webrtc
150
151#endif // CALL_DEGRADED_CALL_H_