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511 lines
20 KiB
511 lines
20 KiB
/*
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* Copyright 2004 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_CHANNEL_H_
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#define PC_CHANNEL_H_
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <utility>
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#include <vector>
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#include "api/call/audio_sink.h"
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#include "api/function_view.h"
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#include "api/jsep.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/video/video_sink_interface.h"
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#include "api/video/video_source_interface.h"
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#include "call/rtp_packet_sink_interface.h"
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#include "media/base/media_channel.h"
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#include "media/base/media_engine.h"
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#include "media/base/stream_params.h"
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#include "p2p/base/dtls_transport_internal.h"
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#include "p2p/base/packet_transport_internal.h"
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#include "pc/channel_interface.h"
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#include "pc/dtls_srtp_transport.h"
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#include "pc/media_session.h"
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#include "pc/rtp_transport.h"
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#include "pc/srtp_filter.h"
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#include "pc/srtp_transport.h"
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#include "rtc_base/async_invoker.h"
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#include "rtc_base/async_udp_socket.h"
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#include "rtc_base/network.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/unique_id_generator.h"
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namespace webrtc {
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class AudioSinkInterface;
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} // namespace webrtc
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namespace cricket {
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struct CryptoParams;
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// BaseChannel contains logic common to voice and video, including enable,
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// marshaling calls to a worker and network threads, and connection and media
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// monitors.
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//
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// BaseChannel assumes signaling and other threads are allowed to make
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// synchronous calls to the worker thread, the worker thread makes synchronous
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// calls only to the network thread, and the network thread can't be blocked by
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// other threads.
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// All methods with _n suffix must be called on network thread,
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// methods with _w suffix on worker thread
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// and methods with _s suffix on signaling thread.
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// Network and worker threads may be the same thread.
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//
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// WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
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// This is required to avoid a data race between the destructor modifying the
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// vtable, and the media channel's thread using BaseChannel as the
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// NetworkInterface.
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class BaseChannel : public ChannelInterface,
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public rtc::MessageHandler,
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public sigslot::has_slots<>,
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public MediaChannel::NetworkInterface,
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public webrtc::RtpPacketSinkInterface {
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public:
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// If |srtp_required| is true, the channel will not send or receive any
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// RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
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// The BaseChannel does not own the UniqueRandomIdGenerator so it is the
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// responsibility of the user to ensure it outlives this object.
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// TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists
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// which will make it easier to change the constructor.
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BaseChannel(rtc::Thread* worker_thread,
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rtc::Thread* network_thread,
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rtc::Thread* signaling_thread,
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std::unique_ptr<MediaChannel> media_channel,
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const std::string& content_name,
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bool srtp_required,
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webrtc::CryptoOptions crypto_options,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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virtual ~BaseChannel();
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virtual void Init_w(webrtc::RtpTransportInternal* rtp_transport);
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// Deinit may be called multiple times and is simply ignored if it's already
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// done.
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void Deinit();
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rtc::Thread* worker_thread() const { return worker_thread_; }
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rtc::Thread* network_thread() const { return network_thread_; }
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const std::string& content_name() const override { return content_name_; }
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// TODO(deadbeef): This is redundant; remove this.
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const std::string& transport_name() const override { return transport_name_; }
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bool enabled() const override { return enabled_; }
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// This function returns true if using SRTP (DTLS-based keying or SDES).
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bool srtp_active() const {
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return rtp_transport_ && rtp_transport_->IsSrtpActive();
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}
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bool writable() const { return writable_; }
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// Set an RTP level transport which could be an RtpTransport without
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// encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP.
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// This can be called from any thread and it hops to the network thread
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// internally. It would replace the |SetTransports| and its variants.
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bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) override;
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webrtc::RtpTransportInternal* rtp_transport() const { return rtp_transport_; }
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// Channel control
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bool SetLocalContent(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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bool SetRemoteContent(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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bool Enable(bool enable) override;
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const std::vector<StreamParams>& local_streams() const override {
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return local_streams_;
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}
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const std::vector<StreamParams>& remote_streams() const override {
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return remote_streams_;
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}
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sigslot::signal2<BaseChannel*, bool> SignalDtlsSrtpSetupFailure;
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void SignalDtlsSrtpSetupFailure_n(bool rtcp);
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void SignalDtlsSrtpSetupFailure_s(bool rtcp);
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// Used for latency measurements.
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sigslot::signal1<ChannelInterface*>& SignalFirstPacketReceived() override {
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return SignalFirstPacketReceived_;
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}
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// Forward SignalSentPacket to worker thread.
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sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
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// Emitted whenever rtcp-mux is fully negotiated and the rtcp-transport can
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// be destroyed.
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// Fired on the network thread.
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sigslot::signal1<const std::string&> SignalRtcpMuxFullyActive;
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// From RtpTransport - public for testing only
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void OnTransportReadyToSend(bool ready);
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// Only public for unit tests. Otherwise, consider protected.
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int SetOption(SocketType type, rtc::Socket::Option o, int val) override;
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int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
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// RtpPacketSinkInterface overrides.
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void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
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// Used by the RTCStatsCollector tests to set the transport name without
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// creating RtpTransports.
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void set_transport_name_for_testing(const std::string& transport_name) {
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transport_name_ = transport_name;
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}
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MediaChannel* media_channel() const override { return media_channel_.get(); }
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protected:
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bool was_ever_writable() const { return was_ever_writable_; }
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void set_local_content_direction(webrtc::RtpTransceiverDirection direction) {
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local_content_direction_ = direction;
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}
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void set_remote_content_direction(webrtc::RtpTransceiverDirection direction) {
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remote_content_direction_ = direction;
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}
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// These methods verify that:
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// * The required content description directions have been set.
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// * The channel is enabled.
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// * And for sending:
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// - The SRTP filter is active if it's needed.
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// - The transport has been writable before, meaning it should be at least
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// possible to succeed in sending a packet.
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//
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// When any of these properties change, UpdateMediaSendRecvState_w should be
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// called.
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bool IsReadyToReceiveMedia_w() const;
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bool IsReadyToSendMedia_w() const;
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rtc::Thread* signaling_thread() { return signaling_thread_; }
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void FlushRtcpMessages_n();
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// NetworkInterface implementation, called by MediaEngine
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bool SendPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options) override;
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bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options) override;
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// From RtpTransportInternal
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void OnWritableState(bool writable);
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void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
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bool PacketIsRtcp(const rtc::PacketTransportInternal* transport,
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const char* data,
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size_t len);
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bool SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options);
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void EnableMedia_w();
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void DisableMedia_w();
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// Performs actions if the RTP/RTCP writable state changed. This should
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// be called whenever a channel's writable state changes or when RTCP muxing
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// becomes active/inactive.
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void UpdateWritableState_n();
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void ChannelWritable_n();
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void ChannelNotWritable_n();
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bool AddRecvStream_w(const StreamParams& sp);
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bool RemoveRecvStream_w(uint32_t ssrc);
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void ResetUnsignaledRecvStream_w();
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bool AddSendStream_w(const StreamParams& sp);
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bool RemoveSendStream_w(uint32_t ssrc);
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// Should be called whenever the conditions for
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// IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
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// Updates the send/recv state of the media channel.
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void UpdateMediaSendRecvState();
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virtual void UpdateMediaSendRecvState_w() = 0;
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bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
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webrtc::SdpType type,
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std::string* error_desc);
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bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
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webrtc::SdpType type,
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std::string* error_desc);
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virtual bool SetLocalContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) = 0;
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virtual bool SetRemoteContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) = 0;
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// Return a list of RTP header extensions with the non-encrypted extensions
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// removed depending on the current crypto_options_ and only if both the
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// non-encrypted and encrypted extension is present for the same URI.
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RtpHeaderExtensions GetFilteredRtpHeaderExtensions(
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const RtpHeaderExtensions& extensions);
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// From MessageHandler
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void OnMessage(rtc::Message* pmsg) override;
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// Helper function template for invoking methods on the worker thread.
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template <class T>
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T InvokeOnWorker(const rtc::Location& posted_from,
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rtc::FunctionView<T()> functor) {
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return worker_thread_->Invoke<T>(posted_from, functor);
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}
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void AddHandledPayloadType(int payload_type);
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void ClearHandledPayloadTypes();
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void UpdateRtpHeaderExtensionMap(
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const RtpHeaderExtensions& header_extensions);
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bool RegisterRtpDemuxerSink();
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// Return description of media channel to facilitate logging
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std::string ToString() const;
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bool has_received_packet_ = false;
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private:
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bool ConnectToRtpTransport();
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void DisconnectFromRtpTransport();
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void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
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bool IsReadyToSendMedia_n() const;
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rtc::Thread* const worker_thread_;
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rtc::Thread* const network_thread_;
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rtc::Thread* const signaling_thread_;
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rtc::AsyncInvoker invoker_;
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sigslot::signal1<ChannelInterface*> SignalFirstPacketReceived_;
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const std::string content_name_;
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// Won't be set when using raw packet transports. SDP-specific thing.
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std::string transport_name_;
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webrtc::RtpTransportInternal* rtp_transport_ = nullptr;
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std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
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std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
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bool writable_ = false;
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bool was_ever_writable_ = false;
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const bool srtp_required_ = true;
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webrtc::CryptoOptions crypto_options_;
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// MediaChannel related members that should be accessed from the worker
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// thread.
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std::unique_ptr<MediaChannel> media_channel_;
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// Currently the |enabled_| flag is accessed from the signaling thread as
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// well, but it can be changed only when signaling thread does a synchronous
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// call to the worker thread, so it should be safe.
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bool enabled_ = false;
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std::vector<StreamParams> local_streams_;
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std::vector<StreamParams> remote_streams_;
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webrtc::RtpTransceiverDirection local_content_direction_ =
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webrtc::RtpTransceiverDirection::kInactive;
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webrtc::RtpTransceiverDirection remote_content_direction_ =
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webrtc::RtpTransceiverDirection::kInactive;
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webrtc::RtpDemuxerCriteria demuxer_criteria_;
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// This generator is used to generate SSRCs for local streams.
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// This is needed in cases where SSRCs are not negotiated or set explicitly
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// like in Simulcast.
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// This object is not owned by the channel so it must outlive it.
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rtc::UniqueRandomIdGenerator* const ssrc_generator_;
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};
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// VoiceChannel is a specialization that adds support for early media, DTMF,
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// and input/output level monitoring.
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class VoiceChannel : public BaseChannel {
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public:
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VoiceChannel(rtc::Thread* worker_thread,
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rtc::Thread* network_thread,
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rtc::Thread* signaling_thread,
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std::unique_ptr<VoiceMediaChannel> channel,
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const std::string& content_name,
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bool srtp_required,
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webrtc::CryptoOptions crypto_options,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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~VoiceChannel();
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// downcasts a MediaChannel
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VoiceMediaChannel* media_channel() const override {
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return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
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}
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_AUDIO;
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}
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void Init_w(webrtc::RtpTransportInternal* rtp_transport) override;
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private:
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// overrides from BaseChannel
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void UpdateMediaSendRecvState_w() override;
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bool SetLocalContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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bool SetRemoteContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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// Last AudioSendParameters sent down to the media_channel() via
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// SetSendParameters.
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AudioSendParameters last_send_params_;
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// Last AudioRecvParameters sent down to the media_channel() via
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// SetRecvParameters.
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AudioRecvParameters last_recv_params_;
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};
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// VideoChannel is a specialization for video.
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class VideoChannel : public BaseChannel {
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public:
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VideoChannel(rtc::Thread* worker_thread,
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rtc::Thread* network_thread,
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rtc::Thread* signaling_thread,
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std::unique_ptr<VideoMediaChannel> media_channel,
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const std::string& content_name,
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bool srtp_required,
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webrtc::CryptoOptions crypto_options,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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~VideoChannel();
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// downcasts a MediaChannel
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VideoMediaChannel* media_channel() const override {
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return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
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}
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void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_VIDEO;
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}
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private:
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// overrides from BaseChannel
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void UpdateMediaSendRecvState_w() override;
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bool SetLocalContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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bool SetRemoteContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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// Last VideoSendParameters sent down to the media_channel() via
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// SetSendParameters.
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VideoSendParameters last_send_params_;
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// Last VideoRecvParameters sent down to the media_channel() via
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// SetRecvParameters.
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VideoRecvParameters last_recv_params_;
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};
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// RtpDataChannel is a specialization for data.
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class RtpDataChannel : public BaseChannel {
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public:
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RtpDataChannel(rtc::Thread* worker_thread,
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rtc::Thread* network_thread,
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rtc::Thread* signaling_thread,
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std::unique_ptr<DataMediaChannel> channel,
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const std::string& content_name,
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bool srtp_required,
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webrtc::CryptoOptions crypto_options,
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rtc::UniqueRandomIdGenerator* ssrc_generator);
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~RtpDataChannel();
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// TODO(zhihuang): Remove this once the RtpTransport can be shared between
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// BaseChannels.
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void Init_w(DtlsTransportInternal* rtp_dtls_transport,
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DtlsTransportInternal* rtcp_dtls_transport,
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rtc::PacketTransportInternal* rtp_packet_transport,
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rtc::PacketTransportInternal* rtcp_packet_transport);
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void Init_w(webrtc::RtpTransportInternal* rtp_transport) override;
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virtual bool SendData(const SendDataParams& params,
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const rtc::CopyOnWriteBuffer& payload,
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SendDataResult* result);
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// Should be called on the signaling thread only.
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bool ready_to_send_data() const { return ready_to_send_data_; }
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sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&>
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SignalDataReceived;
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// Signal for notifying when the channel becomes ready to send data.
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// That occurs when the channel is enabled, the transport is writable,
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// both local and remote descriptions are set, and the channel is unblocked.
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sigslot::signal1<bool> SignalReadyToSendData;
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cricket::MediaType media_type() const override {
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return cricket::MEDIA_TYPE_DATA;
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}
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protected:
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// downcasts a MediaChannel.
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DataMediaChannel* media_channel() const override {
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return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
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}
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private:
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struct SendDataMessageData : public rtc::MessageData {
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SendDataMessageData(const SendDataParams& params,
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const rtc::CopyOnWriteBuffer* payload,
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SendDataResult* result)
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: params(params), payload(payload), result(result), succeeded(false) {}
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const SendDataParams& params;
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const rtc::CopyOnWriteBuffer* payload;
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SendDataResult* result;
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bool succeeded;
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};
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struct DataReceivedMessageData : public rtc::MessageData {
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// We copy the data because the data will become invalid after we
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// handle DataMediaChannel::SignalDataReceived but before we fire
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// SignalDataReceived.
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DataReceivedMessageData(const ReceiveDataParams& params,
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const char* data,
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size_t len)
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: params(params), payload(data, len) {}
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const ReceiveDataParams params;
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const rtc::CopyOnWriteBuffer payload;
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};
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typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData;
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// overrides from BaseChannel
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// Checks that data channel type is RTP.
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bool CheckDataChannelTypeFromContent(const MediaContentDescription* content,
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std::string* error_desc);
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bool SetLocalContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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bool SetRemoteContent_w(const MediaContentDescription* content,
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webrtc::SdpType type,
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std::string* error_desc) override;
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void UpdateMediaSendRecvState_w() override;
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void OnMessage(rtc::Message* pmsg) override;
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void OnDataReceived(const ReceiveDataParams& params,
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const char* data,
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size_t len);
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void OnDataChannelReadyToSend(bool writable);
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bool ready_to_send_data_ = false;
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// Last DataSendParameters sent down to the media_channel() via
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// SetSendParameters.
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DataSendParameters last_send_params_;
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// Last DataRecvParameters sent down to the media_channel() via
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// SetRecvParameters.
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DataRecvParameters last_recv_params_;
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};
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} // namespace cricket
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#endif // PC_CHANNEL_H_
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