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/*
* 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 <stddef.h>
#include <stdint.h>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/call/audio_sink.h"
#include "api/crypto/crypto_options.h"
#include "api/function_view.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "call/rtp_demuxer.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 "modules/rtp_rtcp/source/rtp_packet_received.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/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "pc/srtp_transport.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/async_udp_socket.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/location.h"
#include "rtc_base/network.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_message.h"
#include "rtc_base/unique_id_generator.h"
namespace webrtc {
class AudioSinkInterface;
} // 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,
// TODO(tommi): Remove has_slots inheritance.
public sigslot::has_slots<>,
// TODO(tommi): Consider implementing these interfaces
// via composition.
public MediaChannel::NetworkInterface,
public webrtc::RtpPacketSinkInterface {
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);
// 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 {
RTC_DCHECK_RUN_ON(network_thread());
if (rtp_transport_)
return rtp_transport_->transport_name();
// TODO(tommi): Delete this variable.
return transport_name_;
}
// This function returns true if using SRTP (DTLS-based keying or SDES).
bool srtp_active() const {
RTC_DCHECK_RUN_ON(network_thread());
return rtp_transport_ && rtp_transport_->IsSrtpActive();
}
// 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;
webrtc::RtpTransportInternal* rtp_transport() const {
RTC_DCHECK_RUN_ON(network_thread());
return rtp_transport_;
}
// 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;
// Controls whether this channel will receive packets on the basis of
// matching payload type alone. This is needed for legacy endpoints that
// don't signal SSRCs or use MID/RID, but doesn't make sense if there is
// more than channel of specific media type, As that creates an ambiguity.
//
// This method will also remove any existing streams that were bound to this
// channel on the basis of payload type, since one of these streams might
// actually belong to a new channel. See: crbug.com/webrtc/11477
bool SetPayloadTypeDemuxingEnabled(bool enabled) override;
void 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_;
}
// Used for latency measurements.
void SetFirstPacketReceivedCallback(std::function<void()> callback) override;
// 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;
// RtpPacketSinkInterface overrides.
void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
MediaChannel* media_channel() const override {
return media_channel_.get();
}
protected:
bool was_ever_writable() const {
RTC_DCHECK_RUN_ON(worker_thread());
return was_ever_writable_;
}
void set_local_content_direction(webrtc::RtpTransceiverDirection direction) {
RTC_DCHECK_RUN_ON(worker_thread());
local_content_direction_ = direction;
}
void set_remote_content_direction(webrtc::RtpTransceiverDirection direction) {
RTC_DCHECK_RUN_ON(worker_thread());
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 RTC_RUN_ON(worker_thread());
bool IsReadyToSendMedia_w() const RTC_RUN_ON(worker_thread());
rtc::Thread* signaling_thread() const { return signaling_thread_; }
// 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 SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options);
void EnableMedia_w() RTC_RUN_ON(worker_thread());
void DisableMedia_w() RTC_RUN_ON(worker_thread());
// 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() RTC_RUN_ON(network_thread());
void ChannelWritable_n() RTC_RUN_ON(network_thread());
void ChannelNotWritable_n() RTC_RUN_ON(network_thread());
bool AddRecvStream_w(const StreamParams& sp) RTC_RUN_ON(worker_thread());
bool RemoveRecvStream_w(uint32_t ssrc) RTC_RUN_ON(worker_thread());
void ResetUnsignaledRecvStream_w() RTC_RUN_ON(worker_thread());
bool SetPayloadTypeDemuxingEnabled_w(bool enabled)
RTC_RUN_ON(worker_thread());
bool AddSendStream_w(const StreamParams& sp) RTC_RUN_ON(worker_thread());
bool RemoveSendStream_w(uint32_t ssrc) RTC_RUN_ON(worker_thread());
// Should be called whenever the conditions for
// IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
// Updates the send/recv state of the media channel.
virtual void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) = 0;
bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
webrtc::SdpType type,
std::string* error_desc)
RTC_RUN_ON(worker_thread());
bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
webrtc::SdpType type,
std::string* error_desc)
RTC_RUN_ON(worker_thread());
virtual bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc)
RTC_RUN_ON(worker_thread()) = 0;
virtual bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string* error_desc)
RTC_RUN_ON(worker_thread()) = 0;
// Returns a list of RTP header extensions where any extension URI is unique.
// Encrypted extensions will be either preferred or discarded, depending on
// the current crypto_options_.
RtpHeaderExtensions GetDeduplicatedRtpHeaderExtensions(
const RtpHeaderExtensions& extensions);
// Add `payload_type` to `demuxer_criteria_` if payload type demuxing is
// enabled.
void MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread());
void ClearHandledPayloadTypes() RTC_RUN_ON(worker_thread());
void UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions);
bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());
// Return description of media channel to facilitate logging
std::string ToString() const;
private:
bool ConnectToRtpTransport() RTC_RUN_ON(network_thread());
void DisconnectFromRtpTransport() RTC_RUN_ON(network_thread());
void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
rtc::Thread* const signaling_thread_;
rtc::scoped_refptr<webrtc::PendingTaskSafetyFlag> alive_;
const std::string content_name_;
std::function<void()> on_first_packet_received_
RTC_GUARDED_BY(network_thread());
// Won't be set when using raw packet transports. SDP-specific thing.
// TODO(bugs.webrtc.org/12230): Written on network thread, read on
// worker thread (at least).
// TODO(tommi): Remove this variable and instead use rtp_transport_ to
// return the transport name. This variable is currently required for
// "for_test" methods.
std::string transport_name_;
webrtc::RtpTransportInternal* rtp_transport_
RTC_GUARDED_BY(network_thread()) = nullptr;
std::vector<std::pair<rtc::Socket::Option, int> > socket_options_
RTC_GUARDED_BY(network_thread());
std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_
RTC_GUARDED_BY(network_thread());
bool writable_ RTC_GUARDED_BY(network_thread()) = false;
bool was_ever_writable_n_ RTC_GUARDED_BY(network_thread()) = false;
bool was_ever_writable_ RTC_GUARDED_BY(worker_thread()) = false;
const bool srtp_required_ = true;
// TODO(tommi): This field shouldn't be necessary. It's a copy of
// PeerConnection::GetCryptoOptions(), which is const state. It's also only
// used to filter header extensions when calling
// `rtp_transport_->UpdateRtpHeaderExtensionMap()` when the local/remote
// content description is updated. Since the transport is actually owned
// by the transport controller that also gets updated whenever the content
// description changes, it seems we have two paths into the transports, along
// with several thread hops via various classes (such as the Channel classes)
// that only serve as additional layers and store duplicate state. The Jsep*
// family of classes already apply session description updates on the network
// thread every time it changes.
// For the Channel classes, we should be able to get rid of:
// * crypto_options (and fewer construction parameters)_
// * UpdateRtpHeaderExtensionMap
// * GetFilteredRtpHeaderExtensions
// * Blocking thread hop to the network thread for every call to set
// local/remote content is updated.
const webrtc::CryptoOptions crypto_options_;
// MediaChannel related members that should be accessed from the worker
// thread.
const 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_ RTC_GUARDED_BY(worker_thread()) = false;
bool enabled_s_ RTC_GUARDED_BY(signaling_thread()) = false;
bool payload_type_demuxing_enabled_ RTC_GUARDED_BY(worker_thread()) = true;
std::vector<StreamParams> local_streams_ RTC_GUARDED_BY(worker_thread());
std::vector<StreamParams> remote_streams_ RTC_GUARDED_BY(worker_thread());
// TODO(bugs.webrtc.org/12230): local_content_direction and
// remote_content_direction are set on the worker thread, but accessed on the
// network thread.
webrtc::RtpTransceiverDirection local_content_direction_ =
webrtc::RtpTransceiverDirection::kInactive;
webrtc::RtpTransceiverDirection remote_content_direction_ =
webrtc::RtpTransceiverDirection::kInactive;
// Cached list of payload types, used if payload type demuxing is re-enabled.
std::set<uint8_t> payload_types_ RTC_GUARDED_BY(worker_thread());
// TODO(bugs.webrtc.org/12239): Modified on worker thread, accessed
// on network thread in RegisterRtpDemuxerSink_n (called from Init_w)
webrtc::RtpDemuxerCriteria demuxer_criteria_;
// Accessed on the worker thread, modified on the network thread from
// RegisterRtpDemuxerSink_w's Invoke.
webrtc::RtpDemuxerCriteria previous_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:
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;
}
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 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_;
};
} // namespace cricket
#endif // PC_CHANNEL_H_