blob: 9747ec27cb42da221f502758388a6b27cae5b286 [file] [log] [blame]
/*
* Copyright 2004 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 PC_CHANNEL_H_
#define PC_CHANNEL_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "api/call/audio_sink.h"
#include "api/jsep.h"
#include "api/media_transport_interface.h"
#include "api/rtp_receiver_interface.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "call/rtp_packet_sink_interface.h"
#include "media/base/media_channel.h"
#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
#include "p2p/base/dtls_transport_internal.h"
#include "p2p/base/packet_transport_internal.h"
#include "pc/channel_interface.h"
#include "pc/dtls_srtp_transport.h"
#include "pc/media_session.h"
#include "pc/rtp_transport.h"
#include "pc/srtp_filter.h"
#include "pc/srtp_transport.h"
#include "rtc_base/async_invoker.h"
#include "rtc_base/async_udp_socket.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/network.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/unique_id_generator.h"
namespace webrtc {
class AudioSinkInterface;
class MediaTransportInterface;
} // namespace webrtc
namespace cricket {
struct CryptoParams;
// BaseChannel contains logic common to voice and video, including enable,
// marshaling calls to a worker and network threads, and connection and media
// monitors.
//
// BaseChannel assumes signaling and other threads are allowed to make
// synchronous calls to the worker thread, the worker thread makes synchronous
// calls only to the network thread, and the network thread can't be blocked by
// other threads.
// All methods with _n suffix must be called on network thread,
// methods with _w suffix on worker thread
// and methods with _s suffix on signaling thread.
// Network and worker threads may be the same thread.
//
// WARNING! SUBCLASSES MUST CALL Deinit() IN THEIR DESTRUCTORS!
// This is required to avoid a data race between the destructor modifying the
// vtable, and the media channel's thread using BaseChannel as the
// NetworkInterface.
class BaseChannel : public ChannelInterface,
public rtc::MessageHandler,
public sigslot::has_slots<>,
public MediaChannel::NetworkInterface,
public webrtc::RtpPacketSinkInterface,
public webrtc::MediaTransportNetworkChangeCallback {
public:
// If |srtp_required| is true, the channel will not send or receive any
// RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
// The BaseChannel does not own the UniqueRandomIdGenerator so it is the
// responsibility of the user to ensure it outlives this object.
// TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists
// which will make it easier to change the constructor.
BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
virtual ~BaseChannel();
virtual void Init_w(webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport);
// Deinit may be called multiple times and is simply ignored if it's already
// done.
void Deinit();
rtc::Thread* worker_thread() const { return worker_thread_; }
rtc::Thread* network_thread() const { return network_thread_; }
const std::string& content_name() const override { return content_name_; }
// TODO(deadbeef): This is redundant; remove this.
const std::string& transport_name() const override { return transport_name_; }
bool enabled() const override { return enabled_; }
// This function returns true if using SRTP (DTLS-based keying or SDES).
bool srtp_active() const {
return rtp_transport_ && rtp_transport_->IsSrtpActive();
}
bool writable() const { return writable_; }
// Set an RTP level transport which could be an RtpTransport without
// encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP.
// This can be called from any thread and it hops to the network thread
// internally. It would replace the |SetTransports| and its variants.
bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) override;
// Channel control
bool SetLocalContent(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
bool SetRemoteContent(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
bool Enable(bool enable) override;
const std::vector<StreamParams>& local_streams() const override {
return local_streams_;
}
const std::vector<StreamParams>& remote_streams() const override {
return remote_streams_;
}
sigslot::signal2<BaseChannel*, bool> SignalDtlsSrtpSetupFailure;
void SignalDtlsSrtpSetupFailure_n(bool rtcp);
void SignalDtlsSrtpSetupFailure_s(bool rtcp);
// Used for latency measurements.
sigslot::signal1<ChannelInterface*>& SignalFirstPacketReceived() override {
return SignalFirstPacketReceived_;
}
// Forward SignalSentPacket to worker thread.
sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
// Emitted whenever rtcp-mux is fully negotiated and the rtcp-transport can
// be destroyed.
// Fired on the network thread.
sigslot::signal1<const std::string&> SignalRtcpMuxFullyActive;
rtc::PacketTransportInternal* rtp_packet_transport() {
if (rtp_transport_) {
return rtp_transport_->rtp_packet_transport();
}
return nullptr;
}
rtc::PacketTransportInternal* rtcp_packet_transport() {
if (rtp_transport_) {
return rtp_transport_->rtcp_packet_transport();
}
return nullptr;
}
// Returns media transport, can be null if media transport is not available.
webrtc::MediaTransportInterface* media_transport() {
return media_transport_;
}
// From RtpTransport - public for testing only
void OnTransportReadyToSend(bool ready);
// Only public for unit tests. Otherwise, consider protected.
int SetOption(SocketType type, rtc::Socket::Option o, int val) override;
int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
// RtpPacketSinkInterface overrides.
void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
// Used by the RTCStatsCollector tests to set the transport name without
// creating RtpTransports.
void set_transport_name_for_testing(const std::string& transport_name) {
transport_name_ = transport_name;
}
MediaChannel* media_channel() const override { return media_channel_.get(); }
protected:
bool was_ever_writable() const { return was_ever_writable_; }
void set_local_content_direction(webrtc::RtpTransceiverDirection direction) {
local_content_direction_ = direction;
}
void set_remote_content_direction(webrtc::RtpTransceiverDirection direction) {
remote_content_direction_ = direction;
}
// These methods verify that:
// * The required content description directions have been set.
// * The channel is enabled.
// * And for sending:
// - The SRTP filter is active if it's needed.
// - The transport has been writable before, meaning it should be at least
// possible to succeed in sending a packet.
//
// When any of these properties change, UpdateMediaSendRecvState_w should be
// called.
bool IsReadyToReceiveMedia_w() const;
bool IsReadyToSendMedia_w() const;
rtc::Thread* signaling_thread() { return signaling_thread_; }
void FlushRtcpMessages_n();
// NetworkInterface implementation, called by MediaEngine
bool SendPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) override;
bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options) override;
// From RtpTransportInternal
void OnWritableState(bool writable);
void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
bool PacketIsRtcp(const rtc::PacketTransportInternal* transport,
const char* data,
size_t len);
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options);
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
int64_t packet_time_us);
void OnPacketReceived(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us);
void ProcessPacket(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us);
void EnableMedia_w();
void DisableMedia_w();
// Performs actions if the RTP/RTCP writable state changed. This should
// be called whenever a channel's writable state changes or when RTCP muxing
// becomes active/inactive.
void UpdateWritableState_n();
void ChannelWritable_n();
void ChannelNotWritable_n();
bool AddRecvStream_w(const StreamParams& sp);
bool RemoveRecvStream_w(uint32_t ssrc);
bool AddSendStream_w(const StreamParams& sp);
bool RemoveSendStream_w(uint32_t ssrc);
// Should be called whenever the conditions for
// IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
// Updates the send/recv state of the media channel.
void UpdateMediaSendRecvState();
virtual void UpdateMediaSendRecvState_w() = 0;
bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
webrtc::SdpType type,
std::string* error_desc);
bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
webrtc::SdpType type,
std::string* error_desc);
virtual bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) = 0;
virtual bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) = 0;
// Return a list of RTP header extensions with the non-encrypted extensions
// removed depending on the current crypto_options_ and only if both the
// non-encrypted and encrypted extension is present for the same URI.
RtpHeaderExtensions GetFilteredRtpHeaderExtensions(
const RtpHeaderExtensions& extensions);
// From MessageHandler
void OnMessage(rtc::Message* pmsg) override;
// Helper function template for invoking methods on the worker thread.
template <class T, class FunctorT>
T InvokeOnWorker(const rtc::Location& posted_from, const FunctorT& functor) {
return worker_thread_->Invoke<T>(posted_from, functor);
}
void AddHandledPayloadType(int payload_type);
void UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions);
bool RegisterRtpDemuxerSink();
bool has_received_packet_ = false;
private:
bool ConnectToRtpTransport();
void DisconnectFromRtpTransport();
void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
bool IsReadyToSendMedia_n() const;
// MediaTransportNetworkChangeCallback override.
void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route) override;
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
rtc::Thread* const signaling_thread_;
rtc::AsyncInvoker invoker_;
sigslot::signal1<ChannelInterface*> SignalFirstPacketReceived_;
const std::string content_name_;
// Won't be set when using raw packet transports. SDP-specific thing.
std::string transport_name_;
webrtc::RtpTransportInternal* rtp_transport_ = nullptr;
// Optional media transport (experimental).
// If provided, audio and video will be sent through media_transport instead
// of RTP/RTCP. Currently media_transport can co-exist with rtp_transport.
webrtc::MediaTransportInterface* media_transport_ = nullptr;
std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
bool writable_ = false;
bool was_ever_writable_ = false;
const bool srtp_required_ = true;
webrtc::CryptoOptions crypto_options_;
// MediaChannel related members that should be accessed from the worker
// thread.
std::unique_ptr<MediaChannel> media_channel_;
// Currently the |enabled_| flag is accessed from the signaling thread as
// well, but it can be changed only when signaling thread does a synchronous
// call to the worker thread, so it should be safe.
bool enabled_ = false;
std::vector<StreamParams> local_streams_;
std::vector<StreamParams> remote_streams_;
webrtc::RtpTransceiverDirection local_content_direction_ =
webrtc::RtpTransceiverDirection::kInactive;
webrtc::RtpTransceiverDirection remote_content_direction_ =
webrtc::RtpTransceiverDirection::kInactive;
webrtc::RtpDemuxerCriteria demuxer_criteria_;
// This generator is used to generate SSRCs for local streams.
// This is needed in cases where SSRCs are not negotiated or set explicitly
// like in Simulcast.
// This object is not owned by the channel so it must outlive it.
rtc::UniqueRandomIdGenerator* const ssrc_generator_;
};
// VoiceChannel is a specialization that adds support for early media, DTMF,
// and input/output level monitoring.
class VoiceChannel : public BaseChannel,
public webrtc::AudioPacketReceivedObserver {
public:
VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
~VoiceChannel();
// downcasts a MediaChannel
VoiceMediaChannel* media_channel() const override {
return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
void Init_w(webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport) override;
private:
// overrides from BaseChannel
void UpdateMediaSendRecvState_w() override;
bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
void OnFirstAudioPacketReceived(int64_t channel_id) override;
// Last AudioSendParameters sent down to the media_channel() via
// SetSendParameters.
AudioSendParameters last_send_params_;
// Last AudioRecvParameters sent down to the media_channel() via
// SetRecvParameters.
AudioRecvParameters last_recv_params_;
};
// VideoChannel is a specialization for video.
class VideoChannel : public BaseChannel {
public:
VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
~VideoChannel();
// downcasts a MediaChannel
VideoMediaChannel* media_channel() const override {
return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
}
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
private:
// overrides from BaseChannel
void UpdateMediaSendRecvState_w() override;
bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
// Last VideoSendParameters sent down to the media_channel() via
// SetSendParameters.
VideoSendParameters last_send_params_;
// Last VideoRecvParameters sent down to the media_channel() via
// SetRecvParameters.
VideoRecvParameters last_recv_params_;
};
// RtpDataChannel is a specialization for data.
class RtpDataChannel : public BaseChannel {
public:
RtpDataChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<DataMediaChannel> channel,
const std::string& content_name,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
~RtpDataChannel();
// TODO(zhihuang): Remove this once the RtpTransport can be shared between
// BaseChannels.
void Init_w(DtlsTransportInternal* rtp_dtls_transport,
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport);
void Init_w(
webrtc::RtpTransportInternal* rtp_transport,
webrtc::MediaTransportInterface* media_transport = nullptr) override;
virtual bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result);
// Should be called on the signaling thread only.
bool ready_to_send_data() const { return ready_to_send_data_; }
sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&>
SignalDataReceived;
// Signal for notifying when the channel becomes ready to send data.
// That occurs when the channel is enabled, the transport is writable,
// both local and remote descriptions are set, and the channel is unblocked.
sigslot::signal1<bool> SignalReadyToSendData;
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_DATA;
}
protected:
// downcasts a MediaChannel.
DataMediaChannel* media_channel() const override {
return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
}
private:
struct SendDataMessageData : public rtc::MessageData {
SendDataMessageData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer* payload,
SendDataResult* result)
: params(params), payload(payload), result(result), succeeded(false) {}
const SendDataParams& params;
const rtc::CopyOnWriteBuffer* payload;
SendDataResult* result;
bool succeeded;
};
struct DataReceivedMessageData : public rtc::MessageData {
// We copy the data because the data will become invalid after we
// handle DataMediaChannel::SignalDataReceived but before we fire
// SignalDataReceived.
DataReceivedMessageData(const ReceiveDataParams& params,
const char* data,
size_t len)
: params(params), payload(data, len) {}
const ReceiveDataParams params;
const rtc::CopyOnWriteBuffer payload;
};
typedef rtc::TypedMessageData<bool> DataChannelReadyToSendMessageData;
// overrides from BaseChannel
// Checks that data channel type is RTP.
bool CheckDataChannelTypeFromContent(const RtpDataContentDescription* content,
std::string* error_desc);
bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc) override;
void UpdateMediaSendRecvState_w() override;
void OnMessage(rtc::Message* pmsg) override;
void OnDataReceived(const ReceiveDataParams& params,
const char* data,
size_t len);
void OnDataChannelReadyToSend(bool writable);
bool ready_to_send_data_ = false;
// Last DataSendParameters sent down to the media_channel() via
// SetSendParameters.
DataSendParameters last_send_params_;
// Last DataRecvParameters sent down to the media_channel() via
// SetRecvParameters.
DataRecvParameters last_recv_params_;
};
} // namespace cricket
#endif // PC_CHANNEL_H_