/* * Copyright 2022 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 MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_ #define MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_ #include #include #include #include #include #include #include #include #include "absl/functional/any_invocable.h" #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/audio_options.h" #include "api/call/audio_sink.h" #include "api/call/transport.h" #include "api/crypto/frame_decryptor_interface.h" #include "api/crypto/frame_encryptor_interface.h" #include "api/frame_transformer_interface.h" #include "api/media_types.h" #include "api/rtc_error.h" #include "api/rtp_headers.h" #include "api/rtp_parameters.h" #include "api/rtp_sender_interface.h" #include "api/scoped_refptr.h" #include "api/sequence_checker.h" #include "api/task_queue/pending_task_safety_flag.h" #include "api/task_queue/task_queue_base.h" #include "api/transport/rtp/rtp_source.h" #include "api/video/recordable_encoded_frame.h" #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" #include "api/video/video_source_interface.h" #include "api/video_codecs/video_encoder_factory.h" #include "media/base/codec.h" #include "media/base/media_channel.h" #include "media/base/stream_params.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/async_packet_socket.h" #include "rtc_base/checks.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/dscp.h" #include "rtc_base/logging.h" #include "rtc_base/network/sent_packet.h" #include "rtc_base/network_route.h" #include "rtc_base/socket.h" #include "rtc_base/thread_annotations.h" // This file contains the base classes for classes that implement // the channel interfaces. // These implementation classes used to be the exposed interface names, // but this is in the process of being changed. namespace cricket { // The `MediaChannelUtil` class provides functionality that is used by // multiple MediaChannel-like objects, of both sending and receiving // types. class MediaChannelUtil { public: MediaChannelUtil(webrtc::TaskQueueBase* network_thread, bool enable_dscp = false); virtual ~MediaChannelUtil(); // Returns the absolute sendtime extension id value from media channel. virtual int GetRtpSendTimeExtnId() const; webrtc::Transport* transport() { return &transport_; } // Base methods to send packet using MediaChannelNetworkInterface. // These methods are used by some tests only. bool SendPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options); bool SendRtcp(rtc::CopyOnWriteBuffer* packet, const rtc::PacketOptions& options); int SetOption(MediaChannelNetworkInterface::SocketType type, rtc::Socket::Option opt, int option); // Functions that form part of one or more interface classes. // Not marked override, since this class does not inherit from the // interfaces. // Corresponds to the SDP attribute extmap-allow-mixed, see RFC8285. // Set to true if it's allowed to mix one- and two-byte RTP header extensions // in the same stream. The setter and getter must only be called from // worker_thread. void SetExtmapAllowMixed(bool extmap_allow_mixed); bool ExtmapAllowMixed() const; void SetInterface(MediaChannelNetworkInterface* iface); // Returns `true` if a non-null MediaChannelNetworkInterface pointer is held. // Must be called on the network thread. bool HasNetworkInterface() const; protected: bool DscpEnabled() const; void SetPreferredDscp(rtc::DiffServCodePoint new_dscp); private: // Implementation of the webrtc::Transport interface required // by Call(). class TransportForMediaChannels : public webrtc::Transport { public: TransportForMediaChannels(webrtc::TaskQueueBase* network_thread, bool enable_dscp); virtual ~TransportForMediaChannels(); // Implementation of webrtc::Transport bool SendRtp(rtc::ArrayView packet, const webrtc::PacketOptions& options) override; bool SendRtcp(rtc::ArrayView packet) override; // Not implementation of webrtc::Transport void SetInterface(MediaChannelNetworkInterface* iface); int SetOption(MediaChannelNetworkInterface::SocketType type, rtc::Socket::Option opt, int option); bool DoSendPacket(rtc::CopyOnWriteBuffer* packet, bool rtcp, const rtc::PacketOptions& options); bool HasNetworkInterface() const { RTC_DCHECK_RUN_ON(network_thread_); return network_interface_ != nullptr; } bool DscpEnabled() const { return enable_dscp_; } void SetPreferredDscp(rtc::DiffServCodePoint new_dscp); private: // This is the DSCP value used for both RTP and RTCP channels if DSCP is // enabled. It can be changed at any time via `SetPreferredDscp`. rtc::DiffServCodePoint PreferredDscp() const { RTC_DCHECK_RUN_ON(network_thread_); return preferred_dscp_; } // Apply the preferred DSCP setting to the underlying network interface RTP // and RTCP channels. If DSCP is disabled, then apply the default DSCP // value. void UpdateDscp() RTC_RUN_ON(network_thread_); int SetOptionLocked(MediaChannelNetworkInterface::SocketType type, rtc::Socket::Option opt, int option) RTC_RUN_ON(network_thread_); const rtc::scoped_refptr network_safety_ RTC_PT_GUARDED_BY(network_thread_); webrtc::TaskQueueBase* const network_thread_; const bool enable_dscp_; MediaChannelNetworkInterface* network_interface_ RTC_GUARDED_BY(network_thread_) = nullptr; rtc::DiffServCodePoint preferred_dscp_ RTC_GUARDED_BY(network_thread_) = rtc::DSCP_DEFAULT; }; bool extmap_allow_mixed_ = false; TransportForMediaChannels transport_; }; } // namespace cricket #endif // MEDIA_BASE_MEDIA_CHANNEL_IMPL_H_