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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 WEBRTC_PC_CHANNEL_H_
#define WEBRTC_PC_CHANNEL_H_
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/asyncinvoker.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/network.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/window.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/p2p/base/transportcontroller.h"
#include "webrtc/p2p/client/socketmonitor.h"
#include "webrtc/pc/audiomonitor.h"
#include "webrtc/pc/bundlefilter.h"
#include "webrtc/pc/mediamonitor.h"
#include "webrtc/pc/mediasession.h"
#include "webrtc/pc/rtcpmuxfilter.h"
#include "webrtc/pc/srtpfilter.h"
namespace rtc {
class PacketTransportInterface;
}
namespace webrtc {
class AudioSinkInterface;
} // namespace webrtc
namespace cricket {
struct CryptoParams;
class MediaContentDescription;
// 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 rtc::MessageHandler, public sigslot::has_slots<>,
public MediaChannel::NetworkInterface,
public ConnectionStatsGetter {
public:
// |rtcp| represents whether or not this channel uses RTCP.
// 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).
BaseChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
MediaChannel* channel,
TransportController* transport_controller,
const std::string& content_name,
bool rtcp,
bool srtp_required);
virtual ~BaseChannel();
bool Init_w(const std::string* bundle_transport_name);
// 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 { return content_name_; }
const std::string& transport_name() const { return transport_name_; }
bool enabled() const { return enabled_; }
// This function returns true if we are using SRTP.
bool secure() const { return srtp_filter_.IsActive(); }
// The following function returns true if we are using
// DTLS-based keying. If you turned off SRTP later, however
// you could have secure() == false and dtls_secure() == true.
bool secure_dtls() const { return dtls_keyed_; }
bool writable() const { return writable_; }
// Activate RTCP mux, regardless of the state so far. Once
// activated, it can not be deactivated, and if the remote
// description doesn't support RTCP mux, setting the remote
// description will fail.
void ActivateRtcpMux();
bool SetTransport(const std::string& transport_name);
bool PushdownLocalDescription(const SessionDescription* local_desc,
ContentAction action,
std::string* error_desc);
bool PushdownRemoteDescription(const SessionDescription* remote_desc,
ContentAction action,
std::string* error_desc);
// Channel control
bool SetLocalContent(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc);
bool SetRemoteContent(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc);
bool Enable(bool enable);
// Multiplexing
bool AddRecvStream(const StreamParams& sp);
bool RemoveRecvStream(uint32_t ssrc);
bool AddSendStream(const StreamParams& sp);
bool RemoveSendStream(uint32_t ssrc);
// Monitoring
void StartConnectionMonitor(int cms);
void StopConnectionMonitor();
// For ConnectionStatsGetter, used by ConnectionMonitor
bool GetConnectionStats(ConnectionInfos* infos) override;
BundleFilter* bundle_filter() { return &bundle_filter_; }
const std::vector<StreamParams>& local_streams() const {
return local_streams_;
}
const std::vector<StreamParams>& remote_streams() const {
return remote_streams_;
}
sigslot::signal2<BaseChannel*, bool> SignalDtlsSetupFailure;
void SignalDtlsSetupFailure_n(bool rtcp);
void SignalDtlsSetupFailure_s(bool rtcp);
// Used for latency measurements.
sigslot::signal1<BaseChannel*> SignalFirstPacketReceived;
// Forward TransportChannel SignalSentPacket to worker thread.
sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
// Only public for unit tests. Otherwise, consider private.
TransportChannel* transport_channel() const { return transport_channel_; }
TransportChannel* rtcp_transport_channel() const {
return rtcp_transport_channel_;
}
// Made public for easier testing.
//
// Updates "ready to send" for an individual channel, and informs the media
// channel that the transport is ready to send if each channel (in use) is
// ready to send. This is more specific than just "writable"; it means the
// last send didn't return ENOTCONN.
//
// This should be called whenever a channel's ready-to-send state changes,
// or when RTCP muxing becomes active/inactive.
void SetTransportChannelReadyToSend(bool rtcp, 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);
SrtpFilter* srtp_filter() { return &srtp_filter_; }
virtual cricket::MediaType media_type() = 0;
bool SetCryptoOptions(const rtc::CryptoOptions& crypto_options);
// This function returns true if we require SRTP for call setup.
bool srtp_required_for_testing() const { return srtp_required_; }
protected:
virtual MediaChannel* media_channel() const { return media_channel_; }
// Sets the |transport_channel_| (and |rtcp_transport_channel_|, if
// |rtcp_enabled_| is true). Gets the transport channels from
// |transport_controller_|.
// This method also updates writability and "ready-to-send" state.
bool SetTransport_n(const std::string& transport_name);
// This does not update writability or "ready-to-send" state; it just
// disconnects from the old channel and connects to the new one.
void SetTransportChannel_n(bool rtcp, TransportChannel* new_channel);
bool was_ever_writable() const { return was_ever_writable_; }
void set_local_content_direction(MediaContentDirection direction) {
local_content_direction_ = direction;
}
void set_remote_content_direction(MediaContentDirection 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 transport_controller_->signaling_thread();
}
void ConnectToTransportChannel(TransportChannel* tc);
void DisconnectFromTransportChannel(TransportChannel* tc);
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 TransportChannel
void OnWritableState(rtc::PacketTransportInterface* transport);
virtual void OnPacketRead(rtc::PacketTransportInterface* transport,
const char* data,
size_t len,
const rtc::PacketTime& packet_time,
int flags);
void OnReadyToSend(rtc::PacketTransportInterface* transport);
void OnDtlsState(TransportChannel* channel, DtlsTransportState state);
void OnSelectedCandidatePairChanged(
TransportChannel* channel,
CandidatePairInterface* selected_candidate_pair,
int last_sent_packet_id,
bool ready_to_send);
bool PacketIsRtcp(const rtc::PacketTransportInterface* transport,
const char* data,
size_t len);
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options);
virtual bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void OnPacketReceived(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
const rtc::PacketTime& packet_time);
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);
virtual bool ShouldSetupDtlsSrtp_n() const;
// Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
// |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
bool SetupDtlsSrtp_n(bool rtcp_channel);
void MaybeSetupDtlsSrtp_n();
// Set the DTLS-SRTP cipher policy on this channel as appropriate.
bool SetDtlsSrtpCryptoSuites_n(TransportChannel* tc, bool rtcp);
// 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;
// Gets the content info appropriate to the channel (audio or video).
virtual const ContentInfo* GetFirstContent(
const SessionDescription* sdesc) = 0;
bool UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
ContentAction action,
std::string* error_desc);
bool UpdateRemoteStreams_w(const std::vector<StreamParams>& streams,
ContentAction action,
std::string* error_desc);
virtual bool SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) = 0;
virtual bool SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) = 0;
bool SetRtpTransportParameters(const MediaContentDescription* content,
ContentAction action,
ContentSource src,
std::string* error_desc);
bool SetRtpTransportParameters_n(const MediaContentDescription* content,
ContentAction action,
ContentSource src,
std::string* error_desc);
// Helper method to get RTP Absoulute SendTime extension header id if
// present in remote supported extensions list.
void MaybeCacheRtpAbsSendTimeHeaderExtension_w(
const std::vector<webrtc::RtpExtension>& extensions);
bool CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
bool* dtls,
std::string* error_desc);
bool SetSrtp_n(const std::vector<CryptoParams>& params,
ContentAction action,
ContentSource src,
std::string* error_desc);
void ActivateRtcpMux_n();
bool SetRtcpMux_n(bool enable,
ContentAction action,
ContentSource src,
std::string* error_desc);
// From MessageHandler
void OnMessage(rtc::Message* pmsg) override;
const rtc::CryptoOptions& crypto_options() const {
return crypto_options_;
}
// Handled in derived classes
// Get the SRTP crypto suites to use for RTP media
virtual void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const = 0;
virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
const std::vector<ConnectionInfo>& infos) = 0;
// Helper function for invoking bool-returning methods on the worker thread.
template <class FunctorT>
bool InvokeOnWorker(const rtc::Location& posted_from,
const FunctorT& functor) {
return worker_thread_->Invoke<bool>(posted_from, functor);
}
private:
bool InitNetwork_n(const std::string* bundle_transport_name);
void DisconnectTransportChannels_n();
void DestroyTransportChannels_n();
void SignalSentPacket_n(rtc::PacketTransportInterface* transport,
const rtc::SentPacket& sent_packet);
void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
bool IsReadyToSendMedia_n() const;
void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
int GetTransportOverheadPerPacket() const;
void UpdateTransportOverhead();
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
rtc::AsyncInvoker invoker_;
const std::string content_name_;
std::unique_ptr<ConnectionMonitor> connection_monitor_;
// Transport related members that should be accessed from network thread.
TransportController* const transport_controller_;
std::string transport_name_;
// Is RTCP used at all by this type of channel?
// Expected to be true (as of typing this) for everything except data
// channels.
const bool rtcp_enabled_;
// TODO(johan): Replace TransportChannel* with rtc::PacketTransportInterface*.
TransportChannel* transport_channel_ = nullptr;
std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
TransportChannel* rtcp_transport_channel_ = nullptr;
std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
SrtpFilter srtp_filter_;
RtcpMuxFilter rtcp_mux_filter_;
BundleFilter bundle_filter_;
bool rtp_ready_to_send_ = false;
bool rtcp_ready_to_send_ = false;
bool writable_ = false;
bool was_ever_writable_ = false;
bool has_received_packet_ = false;
bool dtls_keyed_ = false;
const bool srtp_required_ = true;
rtc::CryptoOptions crypto_options_;
int rtp_abs_sendtime_extn_id_ = -1;
// MediaChannel related members that should be accessed from the worker
// thread.
MediaChannel* const 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_;
MediaContentDirection local_content_direction_ = MD_INACTIVE;
MediaContentDirection remote_content_direction_ = MD_INACTIVE;
CandidatePairInterface* selected_candidate_pair_;
};
// 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,
MediaEngineInterface* media_engine,
VoiceMediaChannel* channel,
TransportController* transport_controller,
const std::string& content_name,
bool rtcp,
bool srtp_required);
~VoiceChannel();
bool Init_w(const std::string* bundle_transport_name);
// Configure sending media on the stream with SSRC |ssrc|
// If there is only one sending stream SSRC 0 can be used.
bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source);
// downcasts a MediaChannel
VoiceMediaChannel* media_channel() const override {
return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
}
void SetEarlyMedia(bool enable);
// This signal is emitted when we have gone a period of time without
// receiving early media. When received, a UI should start playing its
// own ringing sound
sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
// Returns if the telephone-event has been negotiated.
bool CanInsertDtmf();
// Send and/or play a DTMF |event| according to the |flags|.
// The DTMF out-of-band signal will be used on sending.
// The |ssrc| should be either 0 or a valid send stream ssrc.
// The valid value for the |event| are 0 which corresponding to DTMF
// event 0-9, *, #, A-D.
bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
bool SetOutputVolume(uint32_t ssrc, double volume);
void SetRawAudioSink(uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink);
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const;
bool SetRtpSendParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters);
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const;
bool SetRtpReceiveParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters);
// Get statistics about the current media session.
bool GetStats(VoiceMediaInfo* stats);
// Monitoring functions
sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
SignalConnectionMonitor;
void StartMediaMonitor(int cms);
void StopMediaMonitor();
sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
void StartAudioMonitor(int cms);
void StopAudioMonitor();
bool IsAudioMonitorRunning() const;
sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
int GetInputLevel_w();
int GetOutputLevel_w();
void GetActiveStreams_w(AudioInfo::StreamList* actives);
webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const;
bool SetRtpReceiveParameters_w(uint32_t ssrc,
webrtc::RtpParameters parameters);
cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; }
private:
// overrides from BaseChannel
void OnPacketRead(rtc::PacketTransportInterface* transport,
const char* data,
size_t len,
const rtc::PacketTime& packet_time,
int flags) override;
void UpdateMediaSendRecvState_w() override;
const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
bool SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) override;
void HandleEarlyMediaTimeout();
bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
bool SetOutputVolume_w(uint32_t ssrc, double volume);
bool GetStats_w(VoiceMediaInfo* stats);
void OnMessage(rtc::Message* pmsg) override;
void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
void OnConnectionMonitorUpdate(
ConnectionMonitor* monitor,
const std::vector<ConnectionInfo>& infos) override;
void OnMediaMonitorUpdate(VoiceMediaChannel* media_channel,
const VoiceMediaInfo& info);
void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
static const int kEarlyMediaTimeout = 1000;
MediaEngineInterface* media_engine_;
bool received_media_;
std::unique_ptr<VoiceMediaMonitor> media_monitor_;
std::unique_ptr<AudioMonitor> audio_monitor_;
// 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* netwokr_thread,
VideoMediaChannel* channel,
TransportController* transport_controller,
const std::string& content_name,
bool rtcp,
bool srtp_required);
~VideoChannel();
bool Init_w(const std::string* bundle_transport_name);
// downcasts a MediaChannel
VideoMediaChannel* media_channel() const override {
return static_cast<VideoMediaChannel*>(BaseChannel::media_channel());
}
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink);
// Get statistics about the current media session.
bool GetStats(VideoMediaInfo* stats);
sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
SignalConnectionMonitor;
void StartMediaMonitor(int cms);
void StopMediaMonitor();
sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
// Register a source and set options.
// The |ssrc| must correspond to a registered send stream.
bool SetVideoSend(uint32_t ssrc,
bool enable,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const;
bool SetRtpSendParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters);
webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const;
bool SetRtpReceiveParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters);
cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; }
private:
// overrides from BaseChannel
void UpdateMediaSendRecvState_w() override;
const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
bool SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) override;
bool GetStats_w(VideoMediaInfo* stats);
webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
webrtc::RtpParameters GetRtpReceiveParameters_w(uint32_t ssrc) const;
bool SetRtpReceiveParameters_w(uint32_t ssrc,
webrtc::RtpParameters parameters);
void OnMessage(rtc::Message* pmsg) override;
void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
void OnConnectionMonitorUpdate(
ConnectionMonitor* monitor,
const std::vector<ConnectionInfo>& infos) override;
void OnMediaMonitorUpdate(VideoMediaChannel* media_channel,
const VideoMediaInfo& info);
std::unique_ptr<VideoMediaMonitor> media_monitor_;
// 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_;
};
// DataChannel is a specialization for data.
class DataChannel : public BaseChannel {
public:
DataChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
DataMediaChannel* media_channel,
TransportController* transport_controller,
const std::string& content_name,
bool rtcp,
bool srtp_required);
~DataChannel();
bool Init_w(const std::string* bundle_transport_name);
virtual bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result);
void StartMediaMonitor(int cms);
void StopMediaMonitor();
// Should be called on the signaling thread only.
bool ready_to_send_data() const {
return ready_to_send_data_;
}
sigslot::signal2<DataChannel*, const DataMediaInfo&> SignalMediaMonitor;
sigslot::signal2<DataChannel*, const std::vector<ConnectionInfo>&>
SignalConnectionMonitor;
sigslot::signal3<DataChannel*, 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;
// Signal for notifying that the remote side has closed the DataChannel.
sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
cricket::MediaType media_type() 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
const ContentInfo* GetFirstContent(const SessionDescription* sdesc) override;
// If data_channel_type_ is DCT_NONE, set it. Otherwise, check that
// it's the same as what was set previously. Returns false if it's
// set to one type one type and changed to another type later.
bool SetDataChannelType(DataChannelType new_data_channel_type,
std::string* error_desc);
// Same as SetDataChannelType, but extracts the type from the
// DataContentDescription.
bool SetDataChannelTypeFromContent(const DataContentDescription* content,
std::string* error_desc);
bool SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) override;
bool SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) override;
void UpdateMediaSendRecvState_w() override;
bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet) override;
void OnMessage(rtc::Message* pmsg) override;
void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;
void OnConnectionMonitorUpdate(
ConnectionMonitor* monitor,
const std::vector<ConnectionInfo>& infos) override;
void OnMediaMonitorUpdate(DataMediaChannel* media_channel,
const DataMediaInfo& info);
bool ShouldSetupDtlsSrtp_n() const override;
void OnDataReceived(
const ReceiveDataParams& params, const char* data, size_t len);
void OnDataChannelError(uint32_t ssrc, DataMediaChannel::Error error);
void OnDataChannelReadyToSend(bool writable);
void OnStreamClosedRemotely(uint32_t sid);
std::unique_ptr<DataMediaMonitor> media_monitor_;
// TODO(pthatcher): Make a separate SctpDataChannel and
// RtpDataChannel instead of using this.
DataChannelType data_channel_type_;
bool ready_to_send_data_;
// 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 // WEBRTC_PC_CHANNEL_H_