Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 204 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/degraded_call.h b/third_party/libwebrtc/call/degraded_call.h
new file mode 100644
index 0000000000..98e7891d6a
--- /dev/null
+++ b/third_party/libwebrtc/call/degraded_call.h
@@ -0,0 +1,204 @@
+/*
+ *  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.
+  void DeliverRtpPacket(
+      MediaType media_type,
+      RtpPacketReceived packet,
+      OnUndemuxablePacketHandler undemuxable_packet_handler) override;
+  void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) 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_);
+  };
+
+  // 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_;
+  SequenceChecker received_packet_sequence_checker_;
+  std::unique_ptr<FakeNetworkPipe> receive_pipe_
+      RTC_GUARDED_BY(received_packet_sequence_checker_);
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_DEGRADED_CALL_H_