| /* |
| * 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 <stddef.h> |
| #include <stdint.h> |
| |
| #include <functional> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #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<const uint8_t> packet, |
| const webrtc::PacketOptions& options) override; |
| bool SendRtcp(rtc::ArrayView<const uint8_t> 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<webrtc::PendingTaskSafetyFlag> 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_ |