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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/call/degraded_call.h
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/call/degraded_call.h')
-rw-r--r--third_party/libwebrtc/call/degraded_call.h202
1 files changed, 202 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/degraded_call.h b/third_party/libwebrtc/call/degraded_call.h
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+/*
+ * 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(rtc::ArrayView<const uint8_t> packet,
+ const PacketOptions& options,
+ Transport* transport);
+ void SendRtcp(rtc::ArrayView<const uint8_t> packet, 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(rtc::ArrayView<const uint8_t> packet,
+ const PacketOptions& options) override;
+ bool SendRtcp(rtc::ArrayView<const uint8_t> packet) 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_