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/*
* Copyright 2012 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.
*/
// This file contains the PeerConnection interface as defined in
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
//
// The PeerConnectionFactory class provides factory methods to create
// PeerConnection, MediaStream and MediaStreamTrack objects.
//
// The following steps are needed to setup a typical call using WebRTC:
//
// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
// information about input parameters.
//
// 2. Create a PeerConnection object. Provide a configuration struct which
// points to STUN and/or TURN servers used to generate ICE candidates, and
// provide an object that implements the PeerConnectionObserver interface,
// which is used to receive callbacks from the PeerConnection.
//
// 3. Create local MediaStreamTracks using the PeerConnectionFactory and add
// them to PeerConnection by calling AddTrack (or legacy method, AddStream).
//
// 4. Create an offer, call SetLocalDescription with it, serialize it, and send
// it to the remote peer
//
// 5. Once an ICE candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. The candidates must also be serialized and
// sent to the remote peer.
//
// 6. Once an answer is received from the remote peer, call
// SetRemoteDescription with the remote answer.
//
// 7. Once a remote candidate is received from the remote peer, provide it to
// the PeerConnection by calling AddIceCandidate.
//
// The receiver of a call (assuming the application is "call"-based) can decide
// to accept or reject the call; this decision will be taken by the application,
// not the PeerConnection.
//
// If the application decides to accept the call, it should:
//
// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
//
// 2. Create a new PeerConnection.
//
// 3. Provide the remote offer to the new PeerConnection object by calling
// SetRemoteDescription.
//
// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
// back to the remote peer.
//
// 5. Provide the local answer to the new PeerConnection by calling
// SetLocalDescription with the answer.
//
// 6. Provide the remote ICE candidates by calling AddIceCandidate.
//
// 7. Once a candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. Send these candidates to the remote peer.
#ifndef API_PEERCONNECTIONINTERFACE_H_
#define API_PEERCONNECTIONINTERFACE_H_
// TODO(sakal): Remove this define after migration to virtual PeerConnection
// observer is complete.
#define VIRTUAL_PEERCONNECTION_OBSERVER_DESTRUCTOR
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/datachannelinterface.h"
#include "api/dtmfsenderinterface.h"
#include "api/jsep.h"
#include "api/mediastreaminterface.h"
#include "api/rtcerror.h"
#include "api/rtceventlogoutput.h"
#include "api/rtpreceiverinterface.h"
#include "api/rtpsenderinterface.h"
#include "api/rtptransceiverinterface.h"
#include "api/setremotedescriptionobserverinterface.h"
#include "api/stats/rtcstatscollectorcallback.h"
#include "api/statstypes.h"
#include "api/turncustomizer.h"
#include "api/umametrics.h"
#include "call/callfactoryinterface.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/base/mediaconfig.h"
#include "media/base/videocapturer.h"
#include "p2p/base/portallocator.h"
#include "rtc_base/network.h"
#include "rtc_base/rtccertificate.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/socketaddress.h"
#include "rtc_base/sslstreamadapter.h"
namespace rtc {
class SSLIdentity;
class Thread;
}
namespace cricket {
class MediaEngineInterface;
class WebRtcVideoDecoderFactory;
class WebRtcVideoEncoderFactory;
}
namespace webrtc {
class AudioDeviceModule;
class AudioMixer;
class CallFactoryInterface;
class MediaConstraintsInterface;
class VideoDecoderFactory;
class VideoEncoderFactory;
// MediaStream container interface.
class StreamCollectionInterface : public rtc::RefCountInterface {
public:
// TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
virtual size_t count() = 0;
virtual MediaStreamInterface* at(size_t index) = 0;
virtual MediaStreamInterface* find(const std::string& label) = 0;
virtual MediaStreamTrackInterface* FindAudioTrack(
const std::string& id) = 0;
virtual MediaStreamTrackInterface* FindVideoTrack(
const std::string& id) = 0;
protected:
// Dtor protected as objects shouldn't be deleted via this interface.
~StreamCollectionInterface() {}
};
class StatsObserver : public rtc::RefCountInterface {
public:
virtual void OnComplete(const StatsReports& reports) = 0;
protected:
virtual ~StatsObserver() {}
};
// For now, kDefault is interpreted as kPlanB.
// TODO(bugs.webrtc.org/8530): Switch default to kUnifiedPlan.
enum class SdpSemantics { kDefault, kPlanB, kUnifiedPlan };
class PeerConnectionInterface : public rtc::RefCountInterface {
public:
// See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
enum SignalingState {
kStable,
kHaveLocalOffer,
kHaveLocalPrAnswer,
kHaveRemoteOffer,
kHaveRemotePrAnswer,
kClosed,
};
enum IceGatheringState {
kIceGatheringNew,
kIceGatheringGathering,
kIceGatheringComplete
};
enum IceConnectionState {
kIceConnectionNew,
kIceConnectionChecking,
kIceConnectionConnected,
kIceConnectionCompleted,
kIceConnectionFailed,
kIceConnectionDisconnected,
kIceConnectionClosed,
kIceConnectionMax,
};
// TLS certificate policy.
enum TlsCertPolicy {
// For TLS based protocols, ensure the connection is secure by not
// circumventing certificate validation.
kTlsCertPolicySecure,
// For TLS based protocols, disregard security completely by skipping
// certificate validation. This is insecure and should never be used unless
// security is irrelevant in that particular context.
kTlsCertPolicyInsecureNoCheck,
};
struct IceServer {
// TODO(jbauch): Remove uri when all code using it has switched to urls.
// List of URIs associated with this server. Valid formats are described
// in RFC7064 and RFC7065, and more may be added in the future. The "host"
// part of the URI may contain either an IP address or a hostname.
std::string uri;
std::vector<std::string> urls;
std::string username;
std::string password;
TlsCertPolicy tls_cert_policy = kTlsCertPolicySecure;
// If the URIs in |urls| only contain IP addresses, this field can be used
// to indicate the hostname, which may be necessary for TLS (using the SNI
// extension). If |urls| itself contains the hostname, this isn't
// necessary.
std::string hostname;
// List of protocols to be used in the TLS ALPN extension.
std::vector<std::string> tls_alpn_protocols;
// List of elliptic curves to be used in the TLS elliptic curves extension.
std::vector<std::string> tls_elliptic_curves;
bool operator==(const IceServer& o) const {
return uri == o.uri && urls == o.urls && username == o.username &&
password == o.password && tls_cert_policy == o.tls_cert_policy &&
hostname == o.hostname &&
tls_alpn_protocols == o.tls_alpn_protocols &&
tls_elliptic_curves == o.tls_elliptic_curves;
}
bool operator!=(const IceServer& o) const { return !(*this == o); }
};
typedef std::vector<IceServer> IceServers;
enum IceTransportsType {
// TODO(pthatcher): Rename these kTransporTypeXXX, but update
// Chromium at the same time.
kNone,
kRelay,
kNoHost,
kAll
};
// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
enum BundlePolicy {
kBundlePolicyBalanced,
kBundlePolicyMaxBundle,
kBundlePolicyMaxCompat
};
// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
enum RtcpMuxPolicy {
kRtcpMuxPolicyNegotiate,
kRtcpMuxPolicyRequire,
};
enum TcpCandidatePolicy {
kTcpCandidatePolicyEnabled,
kTcpCandidatePolicyDisabled
};
enum CandidateNetworkPolicy {
kCandidateNetworkPolicyAll,
kCandidateNetworkPolicyLowCost
};
enum ContinualGatheringPolicy {
GATHER_ONCE,
GATHER_CONTINUALLY
};
enum class RTCConfigurationType {
// A configuration that is safer to use, despite not having the best
// performance. Currently this is the default configuration.
kSafe,
// An aggressive configuration that has better performance, although it
// may be riskier and may need extra support in the application.
kAggressive
};
// TODO(hbos): Change into class with private data and public getters.
// TODO(nisse): In particular, accessing fields directly from an
// application is brittle, since the organization mirrors the
// organization of the implementation, which isn't stable. So we
// need getters and setters at least for fields which applications
// are interested in.
struct RTCConfiguration {
// This struct is subject to reorganization, both for naming
// consistency, and to group settings to match where they are used
// in the implementation. To do that, we need getter and setter
// methods for all settings which are of interest to applications,
// Chrome in particular.
RTCConfiguration() = default;
explicit RTCConfiguration(RTCConfigurationType type) {
if (type == RTCConfigurationType::kAggressive) {
// These parameters are also defined in Java and IOS configurations,
// so their values may be overwritten by the Java or IOS configuration.
bundle_policy = kBundlePolicyMaxBundle;
rtcp_mux_policy = kRtcpMuxPolicyRequire;
ice_connection_receiving_timeout =
kAggressiveIceConnectionReceivingTimeout;
// These parameters are not defined in Java or IOS configuration,
// so their values will not be overwritten.
enable_ice_renomination = true;
redetermine_role_on_ice_restart = false;
}
}
bool operator==(const RTCConfiguration& o) const;
bool operator!=(const RTCConfiguration& o) const;
bool dscp() const { return media_config.enable_dscp; }
void set_dscp(bool enable) { media_config.enable_dscp = enable; }
bool cpu_adaptation() const {
return media_config.video.enable_cpu_adaptation;
}
void set_cpu_adaptation(bool enable) {
media_config.video.enable_cpu_adaptation = enable;
}
bool suspend_below_min_bitrate() const {
return media_config.video.suspend_below_min_bitrate;
}
void set_suspend_below_min_bitrate(bool enable) {
media_config.video.suspend_below_min_bitrate = enable;
}
bool prerenderer_smoothing() const {
return media_config.video.enable_prerenderer_smoothing;
}
void set_prerenderer_smoothing(bool enable) {
media_config.video.enable_prerenderer_smoothing = enable;
}
bool experiment_cpu_load_estimator() const {
return media_config.video.experiment_cpu_load_estimator;
}
void set_experiment_cpu_load_estimator(bool enable) {
media_config.video.experiment_cpu_load_estimator = enable;
}
static const int kUndefined = -1;
// Default maximum number of packets in the audio jitter buffer.
static const int kAudioJitterBufferMaxPackets = 50;
// ICE connection receiving timeout for aggressive configuration.
static const int kAggressiveIceConnectionReceivingTimeout = 1000;
////////////////////////////////////////////////////////////////////////
// The below few fields mirror the standard RTCConfiguration dictionary:
// https://www.w3.org/TR/webrtc/#rtcconfiguration-dictionary
////////////////////////////////////////////////////////////////////////
// TODO(pthatcher): Rename this ice_servers, but update Chromium
// at the same time.
IceServers servers;
// TODO(pthatcher): Rename this ice_transport_type, but update
// Chromium at the same time.
IceTransportsType type = kAll;
BundlePolicy bundle_policy = kBundlePolicyBalanced;
RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyRequire;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
int ice_candidate_pool_size = 0;
//////////////////////////////////////////////////////////////////////////
// The below fields correspond to constraints from the deprecated
// constraints interface for constructing a PeerConnection.
//
// rtc::Optional fields can be "missing", in which case the implementation
// default will be used.
//////////////////////////////////////////////////////////////////////////
// If set to true, don't gather IPv6 ICE candidates.
// TODO(deadbeef): Remove this? IPv6 support has long stopped being
// experimental
bool disable_ipv6 = false;
// If set to true, don't gather IPv6 ICE candidates on Wi-Fi.
// Only intended to be used on specific devices. Certain phones disable IPv6
// when the screen is turned off and it would be better to just disable the
// IPv6 ICE candidates on Wi-Fi in those cases.
bool disable_ipv6_on_wifi = false;
// By default, the PeerConnection will use a limited number of IPv6 network
// interfaces, in order to avoid too many ICE candidate pairs being created
// and delaying ICE completion.
//
// Can be set to INT_MAX to effectively disable the limit.
int max_ipv6_networks = cricket::kDefaultMaxIPv6Networks;
// Exclude link-local network interfaces
// from considertaion for gathering ICE candidates.
bool disable_link_local_networks = false;
// If set to true, use RTP data channels instead of SCTP.
// TODO(deadbeef): Remove this. We no longer commit to supporting RTP data
// channels, though some applications are still working on moving off of
// them.
bool enable_rtp_data_channel = false;
// Minimum bitrate at which screencast video tracks will be encoded at.
// This means adding padding bits up to this bitrate, which can help
// when switching from a static scene to one with motion.
rtc::Optional<int> screencast_min_bitrate;
// Use new combined audio/video bandwidth estimation?
rtc::Optional<bool> combined_audio_video_bwe;
// Can be used to disable DTLS-SRTP. This should never be done, but can be
// useful for testing purposes, for example in setting up a loopback call
// with a single PeerConnection.
rtc::Optional<bool> enable_dtls_srtp;
/////////////////////////////////////////////////
// The below fields are not part of the standard.
/////////////////////////////////////////////////
// Can be used to disable TCP candidate generation.
TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
// Can be used to avoid gathering candidates for a "higher cost" network,
// if a lower cost one exists. For example, if both Wi-Fi and cellular
// interfaces are available, this could be used to avoid using the cellular
// interface.
CandidateNetworkPolicy candidate_network_policy =
kCandidateNetworkPolicyAll;
// The maximum number of packets that can be stored in the NetEq audio
// jitter buffer. Can be reduced to lower tolerated audio latency.
int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;
// Whether to use the NetEq "fast mode" which will accelerate audio quicker
// if it falls behind.
bool audio_jitter_buffer_fast_accelerate = false;
// Timeout in milliseconds before an ICE candidate pair is considered to be
// "not receiving", after which a lower priority candidate pair may be
// selected.
int ice_connection_receiving_timeout = kUndefined;
// Interval in milliseconds at which an ICE "backup" candidate pair will be
// pinged. This is a candidate pair which is not actively in use, but may
// be switched to if the active candidate pair becomes unusable.
//
// This is relevant mainly to Wi-Fi/cell handoff; the application may not
// want this backup cellular candidate pair pinged frequently, since it
// consumes data/battery.
int ice_backup_candidate_pair_ping_interval = kUndefined;
// Can be used to enable continual gathering, which means new candidates
// will be gathered as network interfaces change. Note that if continual
// gathering is used, the candidate removal API should also be used, to
// avoid an ever-growing list of candidates.
ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
// If set to true, candidate pairs will be pinged in order of most likely
// to work (which means using a TURN server, generally), rather than in
// standard priority order.
bool prioritize_most_likely_ice_candidate_pairs = false;
// Implementation defined settings. A public member only for the benefit of
// the implementation. Applications must not access it directly, and should
// instead use provided accessor methods, e.g., set_cpu_adaptation.
struct cricket::MediaConfig media_config;
// If set to true, only one preferred TURN allocation will be used per
// network interface. UDP is preferred over TCP and IPv6 over IPv4. This
// can be used to cut down on the number of candidate pairings.
bool prune_turn_ports = false;
// If set to true, this means the ICE transport should presume TURN-to-TURN
// candidate pairs will succeed, even before a binding response is received.
// This can be used to optimize the initial connection time, since the DTLS
// handshake can begin immediately.
bool presume_writable_when_fully_relayed = false;
// If true, "renomination" will be added to the ice options in the transport
// description.
// See: https://tools.ietf.org/html/draft-thatcher-ice-renomination-00
bool enable_ice_renomination = false;
// If true, the ICE role is re-determined when the PeerConnection sets a
// local transport description that indicates an ICE restart.
//
// This is standard RFC5245 ICE behavior, but causes unnecessary role
// thrashing, so an application may wish to avoid it. This role
// re-determining was removed in ICEbis (ICE v2).
bool redetermine_role_on_ice_restart = true;
// If set, the min interval (max rate) at which we will send ICE checks
// (STUN pings), in milliseconds.
rtc::Optional<int> ice_check_min_interval;
// ICE Periodic Regathering
// If set, WebRTC will periodically create and propose candidates without
// starting a new ICE generation. The regathering happens continuously with
// interval specified in milliseconds by the uniform distribution [a, b].
rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
// Optional TurnCustomizer.
// With this class one can modify outgoing TURN messages.
// The object passed in must remain valid until PeerConnection::Close() is
// called.
webrtc::TurnCustomizer* turn_customizer = nullptr;
// Preferred network interface.
// A candidate pair on a preferred network has a higher precedence in ICE
// than one on an un-preferred network, regardless of priority or network
// cost.
rtc::Optional<rtc::AdapterType> network_preference;
// Configure the SDP semantics used by this PeerConnection. Note that the
// WebRTC 1.0 specification requires kUnifiedPlan semantics. The
// RtpTransceiver API is only available with kUnifiedPlan semantics.
//
// kPlanB will cause PeerConnection to create offers and answers with at
// most one audio and one video m= section with multiple RtpSenders and
// RtpReceivers specified as multiple a=ssrc lines within the section. This
// will also cause PeerConnection to reject offers/answers with multiple m=
// sections of the same media type.
//
// kUnifiedPlan will cause PeerConnection to create offers and answers with
// multiple m= sections where each m= section maps to one RtpSender and one
// RtpReceiver (an RtpTransceiver), either both audio or both video. Plan B
// style offers or answers will be rejected in calls to SetLocalDescription
// or SetRemoteDescription.
//
// For users who only send at most one audio and one video track, this
// choice does not matter and should be left as kDefault.
//
// For users who wish to send multiple audio/video streams and need to stay
// interoperable with legacy WebRTC implementations, specify kPlanB.
//
// For users who wish to send multiple audio/video streams and/or wish to
// use the new RtpTransceiver API, specify kUnifiedPlan.
//
// TODO(steveanton): Implement support for kUnifiedPlan.
SdpSemantics sdp_semantics = SdpSemantics::kDefault;
//
// Don't forget to update operator== if adding something.
//
};
// See: https://www.w3.org/TR/webrtc/#idl-def-rtcofferansweroptions
struct RTCOfferAnswerOptions {
static const int kUndefined = -1;
static const int kMaxOfferToReceiveMedia = 1;
// The default value for constraint offerToReceiveX:true.
static const int kOfferToReceiveMediaTrue = 1;
// These have been removed from the standard in favor of the "transceiver"
// API, but given that we don't support that API, we still have them here.
//
// offer_to_receive_X set to 1 will cause a media description to be
// generated in the offer, even if no tracks of that type have been added.
// Values greater than 1 are treated the same.
//
// If set to 0, the generated directional attribute will not include the
// "recv" direction (meaning it will be "sendonly" or "inactive".
int offer_to_receive_video = kUndefined;
int offer_to_receive_audio = kUndefined;
bool voice_activity_detection = true;
bool ice_restart = false;
// If true, will offer to BUNDLE audio/video/data together. Not to be
// confused with RTCP mux (multiplexing RTP and RTCP together).
bool use_rtp_mux = true;
RTCOfferAnswerOptions() = default;
RTCOfferAnswerOptions(int offer_to_receive_video,
int offer_to_receive_audio,
bool voice_activity_detection,
bool ice_restart,
bool use_rtp_mux)
: offer_to_receive_video(offer_to_receive_video),
offer_to_receive_audio(offer_to_receive_audio),
voice_activity_detection(voice_activity_detection),
ice_restart(ice_restart),
use_rtp_mux(use_rtp_mux) {}
};
// Used by GetStats to decide which stats to include in the stats reports.
// |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
// |kStatsOutputLevelDebug| includes both the standard stats and additional
// stats for debugging purposes.
enum StatsOutputLevel {
kStatsOutputLevelStandard,
kStatsOutputLevelDebug,
};
// Accessor methods to active local streams.
virtual rtc::scoped_refptr<StreamCollectionInterface>
local_streams() = 0;
// Accessor methods to remote streams.
virtual rtc::scoped_refptr<StreamCollectionInterface>
remote_streams() = 0;
// Add a new MediaStream to be sent on this PeerConnection.
// Note that a SessionDescription negotiation is needed before the
// remote peer can receive the stream.
//
// This has been removed from the standard in favor of a track-based API. So,
// this is equivalent to simply calling AddTrack for each track within the
// stream, with the one difference that if "stream->AddTrack(...)" is called
// later, the PeerConnection will automatically pick up the new track. Though
// this functionality will be deprecated in the future.
virtual bool AddStream(MediaStreamInterface* stream) = 0;
// Remove a MediaStream from this PeerConnection.
// Note that a SessionDescription negotiation is needed before the
// remote peer is notified.
virtual void RemoveStream(MediaStreamInterface* stream) = 0;
// Add a new MediaStreamTrack to be sent on this PeerConnection, and return
// the newly created RtpSender. The RtpSender will be associated with the
// streams specified in the |stream_labels| list.
//
// Errors:
// - INVALID_PARAMETER: |track| is null, has a kind other than audio or video,
// or a sender already exists for the track.
// - INVALID_STATE: The PeerConnection is closed.
// TODO(steveanton): Remove default implementation once downstream
// implementations have been updated.
virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_labels) {
return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
}
// |streams| indicates which stream labels the track should be associated
// with.
// TODO(steveanton): Remove this overload once callers have moved to the
// signature with stream labels.
virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
MediaStreamTrackInterface* track,
std::vector<MediaStreamInterface*> streams) = 0;
// Remove an RtpSender from this PeerConnection.
// Returns true on success.
virtual bool RemoveTrack(RtpSenderInterface* sender) = 0;
// AddTransceiver creates a new RtpTransceiver and adds it to the set of
// transceivers. Adding a transceiver will cause future calls to CreateOffer
// to add a media description for the corresponding transceiver.
//
// The initial value of |mid| in the returned transceiver is null. Setting a
// new session description may change it to a non-null value.
//
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
//
// Optionally, an RtpTransceiverInit structure can be specified to configure
// the transceiver from construction. If not specified, the transceiver will
// default to having a direction of kSendRecv and not be part of any streams.
//
// These methods are only available when Unified Plan is enabled (see
// RTCConfiguration).
//
// Common errors:
// - INTERNAL_ERROR: The configuration does not have Unified Plan enabled.
// TODO(steveanton): Make these pure virtual once downstream projects have
// updated.
// Adds a transceiver with a sender set to transmit the given track. The kind
// of the transceiver (and sender/receiver) will be derived from the kind of
// the track.
// Errors:
// - INVALID_PARAMETER: |track| is null.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
// Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
// Errors:
// - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
// Returns pointer to a DtmfSender on success. Otherwise returns null.
//
// This API is no longer part of the standard; instead DtmfSenders are
// obtained from RtpSenders. Which is what the implementation does; it finds
// an RtpSender for |track| and just returns its DtmfSender.
virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
AudioTrackInterface* track) = 0;
// TODO(deadbeef): Make these pure virtual once all subclasses implement them.
// Creates a sender without a track. Can be used for "early media"/"warmup"
// use cases, where the application may want to negotiate video attributes
// before a track is available to send.
//
// The standard way to do this would be through "addTransceiver", but we
// don't support that API yet.
//
// |kind| must be "audio" or "video".
//
// |stream_id| is used to populate the msid attribute; if empty, one will
// be generated automatically.
virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
const std::string& kind,
const std::string& stream_id) {
return rtc::scoped_refptr<RtpSenderInterface>();
}
// Get all RtpSenders, created either through AddStream, AddTrack, or
// CreateSender. Note that these are "Plan B SDP" RtpSenders, not "Unified
// Plan SDP" RtpSenders, which means that all senders of a specific media
// type share the same media description.
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const {
return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
}
// Get all RtpReceivers, created when a remote description is applied.
// Note that these are "Plan B SDP" RtpReceivers, not "Unified Plan SDP"
// RtpReceivers, which means that all receivers of a specific media type
// share the same media description.
//
// It is also possible to have a media description with no associated
// RtpReceivers, if the directional attribute does not indicate that the
// remote peer is sending any media.
virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
const {
return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
}
// Get all RtpTransceivers, created either through AddTransceiver, AddTrack or
// by a remote description applied with SetRemoteDescription.
// Note: This method is only available when Unified Plan is enabled (see
// RTCConfiguration).
virtual std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
GetTransceivers() const {
return {};
}
virtual bool GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) = 0;
// Gets stats using the new stats collection API, see webrtc/api/stats/. These
// will replace old stats collection API when the new API has matured enough.
// TODO(hbos): Default implementation that does nothing only exists as to not
// break third party projects. As soon as they have been updated this should
// be changed to "= 0;".
virtual void GetStats(RTCStatsCollectorCallback* callback) {}
// Clear cached stats in the rtcstatscollector.
// Exposed for testing while waiting for automatic cache clear to work.
// https://bugs.webrtc.org/8693
virtual void ClearStatsCache() {}
// Create a data channel with the provided config, or default config if none
// is provided. Note that an offer/answer negotiation is still necessary
// before the data channel can be used.
//
// Also, calling CreateDataChannel is the only way to get a data "m=" section
// in SDP, so it should be done before CreateOffer is called, if the
// application plans to use data channels.
virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
const DataChannelInit* config) = 0;
// Returns the more recently applied description; "pending" if it exists, and
// otherwise "current". See below.
virtual const SessionDescriptionInterface* local_description() const = 0;
virtual const SessionDescriptionInterface* remote_description() const = 0;
// A "current" description the one currently negotiated from a complete
// offer/answer exchange.
virtual const SessionDescriptionInterface* current_local_description() const {
return nullptr;
}
virtual const SessionDescriptionInterface* current_remote_description()
const {
return nullptr;
}
// A "pending" description is one that's part of an incomplete offer/answer
// exchange (thus, either an offer or a pranswer). Once the offer/answer
// exchange is finished, the "pending" description will become "current".
virtual const SessionDescriptionInterface* pending_local_description() const {
return nullptr;
}
virtual const SessionDescriptionInterface* pending_remote_description()
const {
return nullptr;
}
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {}
// TODO(jiayl): remove the default impl and the old interface when chromium
// code is updated.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {}
// Create an answer to an offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {}
// Deprecated - use version above.
// TODO(hta): Remove and remove default implementations when all callers
// are updated.
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {}
// Sets the local session description.
// The PeerConnection takes the ownership of |desc| even if it fails.
// The |observer| callback will be called when done.
// TODO(deadbeef): Change |desc| to be a unique_ptr, to make it clear
// that this method always takes ownership of it.
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) = 0;
// Sets the remote session description.
// The PeerConnection takes the ownership of |desc| even if it fails.
// The |observer| callback will be called when done.
// TODO(hbos): Remove when Chrome implements the new signature.
virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {}
// TODO(hbos): Make pure virtual when Chrome has updated its signature.
virtual void SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {}
// Deprecated; Replaced by SetConfiguration.
// TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
virtual bool UpdateIce(const IceServers& configuration,
const MediaConstraintsInterface* constraints) {
return false;
}
virtual bool UpdateIce(const IceServers& configuration) { return false; }
// TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
// PeerConnectionInterface implement it.
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() {
return PeerConnectionInterface::RTCConfiguration();
}
// Sets the PeerConnection's global configuration to |config|.
//
// The members of |config| that may be changed are |type|, |servers|,
// |ice_candidate_pool_size| and |prune_turn_ports| (though the candidate
// pool size can't be changed after the first call to SetLocalDescription).
// Note that this means the BUNDLE and RTCP-multiplexing policies cannot be
// changed with this method.
//
// Any changes to STUN/TURN servers or ICE candidate policy will affect the
// next gathering phase, and cause the next call to createOffer to generate
// new ICE credentials, as described in JSEP. This also occurs when
// |prune_turn_ports| changes, for the same reasoning.
//
// If an error occurs, returns false and populates |error| if non-null:
// - INVALID_MODIFICATION if |config| contains a modified parameter other
// than one of the parameters listed above.
// - INVALID_RANGE if |ice_candidate_pool_size| is out of range.
// - SYNTAX_ERROR if parsing an ICE server URL failed.
// - INVALID_PARAMETER if a TURN server is missing |username| or |password|.
// - INTERNAL_ERROR if an unexpected error occurred.
//
// TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
// PeerConnectionInterface implement it.
virtual bool SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& config,
RTCError* error) {
return false;
}
// Version without error output param for backwards compatibility.
// TODO(deadbeef): Remove once chromium is updated.
virtual bool SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& config) {
return false;
}
// Provides a remote candidate to the ICE Agent.
// A copy of the |candidate| will be created and added to the remote
// description. So the caller of this method still has the ownership of the
// |candidate|.
virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
// Removes a group of remote candidates from the ICE agent. Needed mainly for
// continual gathering, to avoid an ever-growing list of candidates as
// networks come and go.
virtual bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
return false;
}
// Register a metric observer (used by chromium). It's reference counted, and
// this method takes a reference. RegisterUMAObserver(nullptr) will release
// the reference.
// TODO(deadbeef): Take argument as scoped_refptr?
virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
// 0 <= min <= current <= max should hold for set parameters.
struct BitrateParameters {
rtc::Optional<int> min_bitrate_bps;
rtc::Optional<int> current_bitrate_bps;
rtc::Optional<int> max_bitrate_bps;
};
// SetBitrate limits the bandwidth allocated for all RTP streams sent by
// this PeerConnection. Other limitations might affect these limits and
// are respected (for example "b=AS" in SDP).
//
// Setting |current_bitrate_bps| will reset the current bitrate estimate
// to the provided value.
virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0;
// Sets current strategy. If not set default WebRTC allocator will be used.
// May be changed during an active session. The strategy
// ownership is passed with std::unique_ptr
// TODO(alexnarest): Make this pure virtual when tests will be updated
virtual void SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) {}
// Enable/disable playout of received audio streams. Enabled by default. Note
// that even if playout is enabled, streams will only be played out if the
// appropriate SDP is also applied. Setting |playout| to false will stop
// playout of the underlying audio device but starts a task which will poll
// for audio data every 10ms to ensure that audio processing happens and the
// audio statistics are updated.
// TODO(henrika): deprecate and remove this.
virtual void SetAudioPlayout(bool playout) {}
// Enable/disable recording of transmitted audio streams. Enabled by default.
// Note that even if recording is enabled, streams will only be recorded if
// the appropriate SDP is also applied.
// TODO(henrika): deprecate and remove this.
virtual void SetAudioRecording(bool recording) {}
// Returns the current SignalingState.
virtual SignalingState signaling_state() = 0;
// Returns the aggregate state of all ICE *and* DTLS transports.
// TODO(deadbeef): Implement "PeerConnectionState" according to the standard,
// to aggregate ICE+DTLS state, and change the scope of IceConnectionState to
// be just the ICE layer. See: crbug.com/webrtc/6145
virtual IceConnectionState ice_connection_state() = 0;
virtual IceGatheringState ice_gathering_state() = 0;
// Starts RtcEventLog using existing file. Takes ownership of |file| and
// passes it on to Call, which will take the ownership. If the
// operation fails the file will be closed. The logging will stop
// automatically after 10 minutes have passed, or when the StopRtcEventLog
// function is called.
// TODO(eladalon): Deprecate and remove this.
virtual bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) {
return false;
}
// Start RtcEventLog using an existing output-sink. Takes ownership of
// |output| and passes it on to Call, which will take the ownership. If the
// operation fails the output will be closed and deallocated. The event log
// will send serialized events to the output object every |output_period_ms|.
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
return false;
}
// Stops logging the RtcEventLog.
// TODO(ivoc): Make this pure virtual when Chrome is updated.
virtual void StopRtcEventLog() {}
// Terminates all media, closes the transports, and in general releases any
// resources used by the PeerConnection. This is an irreversible operation.
//
// Note that after this method completes, the PeerConnection will no longer
// use the PeerConnectionObserver interface passed in on construction, and
// thus the observer object can be safely destroyed.
virtual void Close() = 0;
protected:
// Dtor protected as objects shouldn't be deleted via this interface.
~PeerConnectionInterface() {}
};
// PeerConnection callback interface, used for RTCPeerConnection events.
// Application should implement these methods.
class PeerConnectionObserver {
public:
virtual ~PeerConnectionObserver() = default;
// Triggered when the SignalingState changed.
virtual void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) = 0;
// Triggered when media is received on a new stream from remote peer.
// Deprecated: This callback will no longer be fired with Unified Plan
// semantics. Consider switching to OnAddTrack.
virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
// Triggered when a remote peer close a stream.
// Deprecated: This callback will no longer be fired with Unified Plan
// semantics.
virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
}
// Triggered when a remote peer opens a data channel.
virtual void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) = 0;
// Triggered when renegotiation is needed. For example, an ICE restart
// has begun.
virtual void OnRenegotiationNeeded() = 0;
// Called any time the IceConnectionState changes.
//
// Note that our ICE states lag behind the standard slightly. The most
// notable differences include the fact that "failed" occurs after 15
// seconds, not 30, and this actually represents a combination ICE + DTLS
// state, so it may be "failed" if DTLS fails while ICE succeeds.
virtual void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) = 0;
// Called any time the IceGatheringState changes.
virtual void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) = 0;
// A new ICE candidate has been gathered.
virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
// Ice candidates have been removed.
// TODO(honghaiz): Make this a pure virtual method when all its subclasses
// implement it.
virtual void OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {}
// Called when the ICE connection receiving status changes.
virtual void OnIceConnectionReceivingChange(bool receiving) {}
// This is called when a receiver and its track is created.
// TODO(zhihuang): Make this pure virtual when all subclasses implement it.
virtual void OnAddTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
// TODO(hbos,deadbeef): Add |OnAssociatedStreamsUpdated| with |receiver| and
// |streams| as arguments. This should be called when an existing receiver its
// associated streams updated. https://crbug.com/webrtc/8315
// This may be blocked on supporting multiple streams per sender or else
// this may count as the removal and addition of a track?
// https://crbug.com/webrtc/7932
// Called when a receiver is completely removed. This is current (Plan B SDP)
// behavior that occurs when processing the removal of a remote track, and is
// called when the receiver is removed and the track is muted. When Unified
// Plan SDP is supported, transceivers can change direction (and receivers
// stopped) but receivers are never removed.
// https://w3c.github.io/webrtc-pc/#process-remote-track-removal
// TODO(hbos,deadbeef): When Unified Plan SDP is supported and receivers are
// no longer removed, deprecate and remove this callback.
// TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
virtual void OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
};
// PeerConnectionFactoryInterface is the factory interface used for creating
// PeerConnection, MediaStream and MediaStreamTrack objects.
//
// The simplest method for obtaiing one, CreatePeerConnectionFactory will
// create the required libjingle threads, socket and network manager factory
// classes for networking if none are provided, though it requires that the
// application runs a message loop on the thread that called the method (see
// explanation below)
//
// If an application decides to provide its own threads and/or implementation
// of networking classes, it should use the alternate
// CreatePeerConnectionFactory method which accepts threads as input, and use
// the CreatePeerConnection version that takes a PortAllocator as an argument.
class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
public:
class Options {
public:
Options() : crypto_options(rtc::CryptoOptions::NoGcm()) {}
// If set to true, created PeerConnections won't enforce any SRTP
// requirement, allowing unsecured media. Should only be used for
// testing/debugging.
bool disable_encryption = false;
// Deprecated. The only effect of setting this to true is that
// CreateDataChannel will fail, which is not that useful.
bool disable_sctp_data_channels = false;
// If set to true, any platform-supported network monitoring capability
// won't be used, and instead networks will only be updated via polling.
//
// This only has an effect if a PeerConnection is created with the default
// PortAllocator implementation.
bool disable_network_monitor = false;
// Sets the network types to ignore. For instance, calling this with
// ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
// loopback interfaces.
int network_ignore_mask = rtc::kDefaultNetworkIgnoreMask;
// Sets the maximum supported protocol version. The highest version
// supported by both ends will be used for the connection, i.e. if one
// party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
// Sets crypto related options, e.g. enabled cipher suites.
rtc::CryptoOptions crypto_options;
};
// Set the options to be used for subsequently created PeerConnections.
virtual void SetOptions(const Options& options) = 0;
// |allocator| and |cert_generator| may be null, in which case default
// implementations will be used.
//
// |observer| must not be null.
//
// Note that this method does not take ownership of |observer|; it's the
// responsibility of the caller to delete it. It can be safely deleted after
// Close has been called on the returned PeerConnection, which ensures no
// more observer callbacks will be invoked.
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) = 0;
// Deprecated; should use RTCConfiguration for everything that previously
// used constraints.
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) = 0;
virtual rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) = 0;
// Creates an AudioSourceInterface.
// |options| decides audio processing settings.
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) = 0;
// Deprecated - use version above.
// Can use CopyConstraintsIntoAudioOptions to bridge the gap.
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const MediaConstraintsInterface* constraints) = 0;
// Creates a VideoTrackSourceInterface from |capturer|.
// TODO(deadbeef): We should aim to remove cricket::VideoCapturer from the
// API. It's mainly used as a wrapper around webrtc's provided
// platform-specific capturers, but these should be refactored to use
// VideoTrackSourceInterface directly.
// TODO(deadbeef): Make pure virtual once downstream mock PC factory classes
// are updated.
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer> capturer) {
return nullptr;
}
// A video source creator that allows selection of resolution and frame rate.
// |constraints| decides video resolution and frame rate but can be null.
// In the null case, use the version above.
//
// |constraints| is only used for the invocation of this method, and can
// safely be destroyed afterwards.
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer> capturer,
const MediaConstraintsInterface* constraints) {
return nullptr;
}
// Deprecated; please use the versions that take unique_ptrs above.
// TODO(deadbeef): Remove these once safe to do so.
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
cricket::VideoCapturer* capturer) {
return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer));
}
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
cricket::VideoCapturer* capturer,
const MediaConstraintsInterface* constraints) {
return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer),
constraints);
}
// Creates a new local VideoTrack. The same |source| can be used in several
// tracks.
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label,
VideoTrackSourceInterface* source) = 0;
// Creates an new AudioTrack. At the moment |source| can be null.
virtual rtc::scoped_refptr<AudioTrackInterface>
CreateAudioTrack(const std::string& label,
AudioSourceInterface* source) = 0;
// Starts AEC dump using existing file. Takes ownership of |file| and passes
// it on to VoiceEngine (via other objects) immediately, which will take
// the ownerhip. If the operation fails, the file will be closed.
// A maximum file size in bytes can be specified. When the file size limit is
// reached, logging is stopped automatically. If max_size_bytes is set to a
// value <= 0, no limit will be used, and logging will continue until the
// StopAecDump function is called.
virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
// Stops logging the AEC dump.
virtual void StopAecDump() = 0;
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.
PeerConnectionFactoryInterface() {}
~PeerConnectionFactoryInterface() {} // NOLINT
};
// Create a new instance of PeerConnectionFactoryInterface.
//
// This method relies on the thread it's called on as the "signaling thread"
// for the PeerConnectionFactory it creates.
//
// As such, if the current thread is not already running an rtc::Thread message
// loop, an application using this method must eventually either call
// rtc::Thread::Current()->Run(), or call
// rtc::Thread::Current()->ProcessMessages() within the application's own
// message loop.
rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
// Create a new instance of PeerConnectionFactoryInterface.
//
// |network_thread|, |worker_thread| and |signaling_thread| are
// the only mandatory parameters.
//
// If non-null, a reference is added to |default_adm|, and ownership of
// |video_encoder_factory| and |video_decoder_factory| is transferred to the
// returned factory.
// TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
// ownership transfer and ref counting more obvious.
rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
// Create a new instance of PeerConnectionFactoryInterface with optional
// external audio mixed and audio processing modules.
//
// If |audio_mixer| is null, an internal audio mixer will be created and used.
// If |audio_processing| is null, an internal audio processing module will be
// created and used.
rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
rtc::scoped_refptr<AudioMixer> audio_mixer,
rtc::scoped_refptr<AudioProcessing> audio_processing);
// Create a new instance of PeerConnectionFactoryInterface with optional video
// codec factories. These video factories represents all video codecs, i.e. no
// extra internal video codecs will be added.
rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
rtc::scoped_refptr<AudioDeviceModule> default_adm,
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
rtc::scoped_refptr<AudioMixer> audio_mixer,
rtc::scoped_refptr<AudioProcessing> audio_processing);
// Create a new instance of PeerConnectionFactoryInterface with external audio
// mixer.
//
// If |audio_mixer| is null, an internal audio mixer will be created and used.
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreatePeerConnectionFactoryWithAudioMixer(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
rtc::scoped_refptr<AudioMixer> audio_mixer);
// Create a new instance of PeerConnectionFactoryInterface.
// Same thread is used as worker and network thread.
inline rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreatePeerConnectionFactory(
rtc::Thread* worker_and_network_thread,
rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory) {
return CreatePeerConnectionFactory(
worker_and_network_thread, worker_and_network_thread, signaling_thread,
default_adm, audio_encoder_factory, audio_decoder_factory,
video_encoder_factory, video_decoder_factory);
}
// This is a lower-level version of the CreatePeerConnectionFactory functions
// above. It's implemented in the "peerconnection" build target, whereas the
// above methods are only implemented in the broader "libjingle_peerconnection"
// build target, which pulls in the implementations of every module webrtc may
// use.
//
// If an application knows it will only require certain modules, it can reduce
// webrtc's impact on its binary size by depending only on the "peerconnection"
// target and the modules the application requires, using
// CreateModularPeerConnectionFactory instead of one of the
// CreatePeerConnectionFactory methods above. For example, if an application
// only uses WebRTC for audio, it can pass in null pointers for the
// video-specific interfaces, and omit the corresponding modules from its
// build.
//
// If |network_thread| or |worker_thread| are null, the PeerConnectionFactory
// will create the necessary thread internally. If |signaling_thread| is null,
// the PeerConnectionFactory will use the thread on which this method is called
// as the signaling thread, wrapping it in an rtc::Thread object if needed.
//
// If non-null, a reference is added to |default_adm|, and ownership of
// |video_encoder_factory| and |video_decoder_factory| is transferred to the
// returned factory.
//
// If |audio_mixer| is null, an internal audio mixer will be created and used.
//
// TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
// ownership transfer and ref counting more obvious.
//
// TODO(deadbeef): Encapsulate these modules in a struct, so that when a new
// module is inevitably exposed, we can just add a field to the struct instead
// of adding a whole new CreateModularPeerConnectionFactory overload.
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreateModularPeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<cricket::MediaEngineInterface> media_engine,
std::unique_ptr<CallFactoryInterface> call_factory,
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
} // namespace webrtc
#endif // API_PEERCONNECTIONINTERFACE_H_