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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 <stdint.h>
#include <functional>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/crypto/crypto_options.h"
#include "api/jsep.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "call/rtp_demuxer.h"
#include "call/rtp_packet_sink_interface.h"
#include "media/base/media_channel.h"
#include "media/base/media_channel_impl.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "pc/channel_interface.h"
#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/async_packet_socket.h"
#include "rtc_base/checks.h"
#include "rtc_base/containers/flat_set.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
#include "rtc_base/socket.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/unique_id_generator.h"
namespace cricket {
// 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.
//
class VideoChannel;
class VoiceChannel;
class BaseChannel : public ChannelInterface,
// TODO(tommi): Remove has_slots inheritance.
public sigslot::has_slots<>,
// TODO(tommi): Consider implementing these interfaces
// via composition.
public MediaChannelNetworkInterface,
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.
// Constructor for use when the MediaChannels are split
BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_send_channel_impl,
std::unique_ptr<MediaChannel> media_receive_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Constructor for use when the MediaChannel is not split
// TODO(bugs.webrtc.org/13931): Delete when split channel project is complete.
BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<MediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
virtual ~BaseChannel();
rtc::Thread* worker_thread() const { return worker_thread_; }
rtc::Thread* network_thread() const { return network_thread_; }
const std::string& mid() const override { return demuxer_criteria_.mid(); }
// TODO(deadbeef): This is redundant; remove this.
absl::string_view transport_name() const override {
RTC_DCHECK_RUN_ON(network_thread());
if (rtp_transport_)
return rtp_transport_->transport_name();
return "";
}
// 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;
VideoMediaSendChannelInterface* video_media_send_channel() override {
RTC_CHECK(false) << "Attempt to fetch video channel from non-video";
return nullptr;
}
VoiceMediaSendChannelInterface* voice_media_send_channel() override {
RTC_CHECK(false) << "Attempt to fetch voice channel from non-voice";
return nullptr;
}
VideoMediaReceiveChannelInterface* video_media_receive_channel() override {
RTC_CHECK(false) << "Attempt to fetch video channel from non-video";
return nullptr;
}
VoiceMediaReceiveChannelInterface* voice_media_receive_channel() override {
RTC_CHECK(false) << "Attempt to fetch voice channel from non-voice";
return nullptr;
}
protected:
void set_local_content_direction(webrtc::RtpTransceiverDirection direction)
RTC_RUN_ON(worker_thread()) {
local_content_direction_ = direction;
}
webrtc::RtpTransceiverDirection local_content_direction() const
RTC_RUN_ON(worker_thread()) {
return local_content_direction_;
}
void set_remote_content_direction(webrtc::RtpTransceiverDirection direction)
RTC_RUN_ON(worker_thread()) {
remote_content_direction_ = direction;
}
webrtc::RtpTransceiverDirection remote_content_direction() const
RTC_RUN_ON(worker_thread()) {
return remote_content_direction_;
}
webrtc::RtpExtension::Filter extensions_filter() const {
return extensions_filter_;
}
bool network_initialized() RTC_RUN_ON(network_thread()) {
return media_send_channel()->HasNetworkInterface();
}
bool enabled() const RTC_RUN_ON(worker_thread()) { return enabled_; }
rtc::Thread* signaling_thread() const { return signaling_thread_; }
// Call to verify that:
// * The required content description directions have been set.
// * The channel is enabled.
// * 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 IsReadyToSendMedia_w() const RTC_RUN_ON(worker_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 SetPayloadTypeDemuxingEnabled_w(bool enabled)
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 MediaContentDescription* content,
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.
// Returns true if the demuxer payload type changed and a re-registration
// is needed.
bool MaybeAddHandledPayloadType(int payload_type) RTC_RUN_ON(worker_thread());
// Returns true if the demuxer payload type criteria was non-empty before
// clearing.
bool ClearHandledPayloadTypes() RTC_RUN_ON(worker_thread());
// Hops to the network thread to update the transport if an update is
// requested. If `update_demuxer` is false and `extensions` is not set, the
// function simply returns. If either of these is set, the function updates
// the transport with either or both of the demuxer criteria and the supplied
// rtp header extensions.
// Returns `true` if either an update wasn't needed or one was successfully
// applied. If the return value is `false`, then updating the demuxer criteria
// failed, which needs to be treated as an error.
bool MaybeUpdateDemuxerAndRtpExtensions_w(
bool update_demuxer,
absl::optional<RtpHeaderExtensions> extensions,
std::string& error_desc) RTC_RUN_ON(worker_thread());
bool RegisterRtpDemuxerSink_w() RTC_RUN_ON(worker_thread());
// Return description of media channel to facilitate logging
std::string ToString() const;
// MediaChannel related members that should be accessed from the worker
// thread. These are used in initializing the subclasses and deleting
// the channels when exiting; they have no accessors.
// Either the media_channel_impl_ is set, or the media_send_channel_impl_
// and the media_receive_channel_impl_ is set.
// TODO(bugs.webrtc.org/13931): Delete when split channel project is complete.
const std::unique_ptr<MediaChannel> media_channel_impl_;
const std::unique_ptr<MediaChannel> media_send_channel_impl_;
const std::unique_ptr<MediaChannel> media_receive_channel_impl_;
private:
bool ConnectToRtpTransport_n() RTC_RUN_ON(network_thread());
void DisconnectFromRtpTransport_n() 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_;
std::function<void()> on_first_packet_received_
RTC_GUARDED_BY(network_thread());
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;
// Set to either kPreferEncryptedExtension or kDiscardEncryptedExtension
// based on the supplied CryptoOptions.
const webrtc::RtpExtension::Filter extensions_filter_;
// 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());
webrtc::RtpTransceiverDirection local_content_direction_ RTC_GUARDED_BY(
worker_thread()) = webrtc::RtpTransceiverDirection::kInactive;
webrtc::RtpTransceiverDirection remote_content_direction_ RTC_GUARDED_BY(
worker_thread()) = webrtc::RtpTransceiverDirection::kInactive;
// Cached list of payload types, used if payload type demuxing is re-enabled.
webrtc::flat_set<uint8_t> payload_types_ RTC_GUARDED_BY(worker_thread());
// A stored copy of the rtp header extensions as applied to the transport.
RtpHeaderExtensions rtp_header_extensions_ 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_;
// 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> send_channel_impl,
std::unique_ptr<VoiceMediaChannel> receive_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Constructor for use when the MediaChannel is not split
// TODO(bugs.webrtc.org/13931): Delete when split channel project is complete.
VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VoiceMediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
~VoiceChannel();
VideoChannel* AsVideoChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
VoiceChannel* AsVoiceChannel() override { return this; }
VoiceMediaSendChannelInterface* media_send_channel() override {
return &send_channel_;
}
VoiceMediaSendChannelInterface* voice_media_send_channel() override {
return &send_channel_;
}
VoiceMediaReceiveChannelInterface* media_receive_channel() override {
return &receive_channel_;
}
VoiceMediaReceiveChannelInterface* voice_media_receive_channel() override {
return &receive_channel_;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
private:
// overrides from BaseChannel
void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override;
bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string& error_desc)
RTC_RUN_ON(worker_thread()) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string& error_desc)
RTC_RUN_ON(worker_thread()) override;
VoiceMediaSendChannel send_channel_ RTC_GUARDED_BY(worker_thread());
VoiceMediaReceiveChannel receive_channel_ RTC_GUARDED_BY(worker_thread());
// Last AudioSendParameters sent down to the media_channel() via
// SetSendParameters.
AudioSendParameters last_send_params_ RTC_GUARDED_BY(worker_thread());
// Last AudioRecvParameters sent down to the media_channel() via
// SetRecvParameters.
AudioRecvParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
};
// 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_send_channel_impl,
std::unique_ptr<VideoMediaChannel> media_receive_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
// Constructor for use when the MediaChannel is not split
// TODO(bugs.webrtc.org/13931): Delete when split channel project is complete.
VideoChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<VideoMediaChannel> media_channel_impl,
absl::string_view mid,
bool srtp_required,
webrtc::CryptoOptions crypto_options,
rtc::UniqueRandomIdGenerator* ssrc_generator);
~VideoChannel();
VideoChannel* AsVideoChannel() override { return this; }
VoiceChannel* AsVoiceChannel() override {
RTC_CHECK_NOTREACHED();
return nullptr;
}
VideoMediaSendChannelInterface* media_send_channel() override {
return &send_channel_;
}
VideoMediaSendChannelInterface* video_media_send_channel() override {
return &send_channel_;
}
VideoMediaReceiveChannelInterface* media_receive_channel() override {
return &receive_channel_;
}
VideoMediaReceiveChannelInterface* video_media_receive_channel() override {
return &receive_channel_;
}
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_VIDEO;
}
private:
// overrides from BaseChannel
void UpdateMediaSendRecvState_w() RTC_RUN_ON(worker_thread()) override;
bool SetLocalContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string& error_desc)
RTC_RUN_ON(worker_thread()) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
webrtc::SdpType type,
std::string& error_desc)
RTC_RUN_ON(worker_thread()) override;
VideoMediaSendChannel send_channel_ RTC_GUARDED_BY(worker_thread());
VideoMediaReceiveChannel receive_channel_ RTC_GUARDED_BY(worker_thread());
// Last VideoSendParameters sent down to the media_channel() via
// SetSendParameters.
VideoSendParameters last_send_params_ RTC_GUARDED_BY(worker_thread());
// Last VideoRecvParameters sent down to the media_channel() via
// SetRecvParameters.
VideoRecvParameters last_recv_params_ RTC_GUARDED_BY(worker_thread());
};
} // namespace cricket
#endif // PC_CHANNEL_H_