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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
// 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.
#include <memory>
#include <string>
#include <vector>
#include "api/asyncresolverfactory.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/call/callfactoryinterface.h"
#include "api/crypto/cryptooptions.h"
#include "api/datachannelinterface.h"
#include "api/fec_controller.h"
#include "api/jsep.h"
#include "api/media_transport_interface.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/transport/bitrate_settings.h"
#include "api/transport/network_control.h"
#include "api/turncustomizer.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/base/mediaconfig.h"
// TODO( cricket::VideoCapturer is deprecated and should
// be deleted from the PeerConnection api.
#include "media/base/videocapturer.h" // nogncheck
// TODO( We plan to provide a way to let applications
// inject a PacketSocketFactory and/or NetworkManager, and not expose
// PortAllocator in the PeerConnection api.
#include "media/base/mediaengine.h" // nogncheck
#include "p2p/base/portallocator.h" // nogncheck
// TODO(nisse): The interface for bitrate allocation strategy belongs in api/.
#include "rtc_base/bitrateallocationstrategy.h"
#include "rtc_base/network.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/rtccertificate.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/socketaddress.h"
#include "rtc_base/sslcertificate.h"
#include "rtc_base/sslstreamadapter.h"
#include "rtc_base/system/rtc_export.h"
namespace rtc {
class SSLIdentity;
class Thread;
} // namespace rtc
namespace cricket {
class WebRtcVideoDecoderFactory;
class WebRtcVideoEncoderFactory;
} // namespace cricket
namespace webrtc {
class AudioDeviceModule;
class AudioMixer;
class AudioProcessing;
class DtlsTransportInterface;
class MediaConstraintsInterface;
class VideoDecoderFactory;
class VideoEncoderFactory;
// MediaStream container interface.
class StreamCollectionInterface : public rtc::RefCountInterface {
// 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;
// Dtor protected as objects shouldn't be deleted via this interface.
~StreamCollectionInterface() override = default;
class StatsObserver : public rtc::RefCountInterface {
virtual void OnComplete(const StatsReports& reports) = 0;
~StatsObserver() override = default;
enum class SdpSemantics { kPlanB, kUnifiedPlan };
class PeerConnectionInterface : public rtc::RefCountInterface {
// See
enum SignalingState {
// See
enum IceGatheringState {
// See
enum class PeerConnectionState {
// See
enum IceConnectionState {
// TLS certificate policy.
enum TlsCertPolicy {
// For TLS based protocols, ensure the connection is secure by not
// circumventing certificate validation.
// 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.
struct IceServer {
IceServer(const 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.
enum BundlePolicy {
enum RtcpMuxPolicy {
enum TcpCandidatePolicy {
enum CandidateNetworkPolicy {
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.
// An aggressive configuration that has better performance, although it
// may be riskier and may need extra support in the application.
// 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 RTC_EXPORT 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(const RTCConfiguration&);
explicit RTCConfiguration(RTCConfigurationType type);
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 {
void set_cpu_adaptation(bool enable) { = enable;
bool suspend_below_min_bitrate() const {
void set_suspend_below_min_bitrate(bool enable) { = enable;
bool prerenderer_smoothing() const {
void set_prerenderer_smoothing(bool enable) { = enable;
bool experiment_cpu_load_estimator() const {
void set_experiment_cpu_load_estimator(bool enable) { = enable;
int audio_rtcp_report_interval_ms() const {
void set_audio_rtcp_report_interval_ms(int audio_rtcp_report_interval_ms) { =
int video_rtcp_report_interval_ms() const {
void set_video_rtcp_report_interval_ms(int video_rtcp_report_interval_ms) { =
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:
// 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.
// absl::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.
absl::optional<int> screencast_min_bitrate;
// Use new combined audio/video bandwidth estimation?
absl::optional<bool> combined_audio_video_bwe;
// TODO( - Move to crypto_options
// 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.
absl::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 =
// 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;
// The minimum delay in milliseconds for the audio jitter buffer.
int audio_jitter_buffer_min_delay_ms = 0;
// 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:
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;
// The following fields define intervals in milliseconds at which ICE
// connectivity checks are sent.
// We consider ICE is "strongly connected" for an agent when there is at
// least one candidate pair that currently succeeds in connectivity check
// from its direction i.e. sending a STUN ping and receives a STUN ping
// response, AND all candidate pairs have sent a minimum number of pings for
// connectivity (this number is implementation-specific). Otherwise, ICE is
// considered in "weak connectivity".
// Note that the above notion of strong and weak connectivity is not defined
// in RFC 5245, and they apply to our current ICE implementation only.
// 1) ice_check_interval_strong_connectivity defines the interval applied to
// ALL candidate pairs when ICE is strongly connected, and it overrides the
// default value of this interval in the ICE implementation;
// 2) ice_check_interval_weak_connectivity defines the counterpart for ALL
// pairs when ICE is weakly connected, and it overrides the default value of
// this interval in the ICE implementation;
// 3) ice_check_min_interval defines the minimal interval (equivalently the
// maximum rate) that overrides the above two intervals when either of them
// is less.
absl::optional<int> ice_check_interval_strong_connectivity;
absl::optional<int> ice_check_interval_weak_connectivity;
absl::optional<int> ice_check_min_interval;
// The min time period for which a candidate pair must wait for response to
// connectivity checks before it becomes unwritable. This parameter
// overrides the default value in the ICE implementation if set.
absl::optional<int> ice_unwritable_timeout;
// The min number of connectivity checks that a candidate pair must sent
// without receiving response before it becomes unwritable. This parameter
// overrides the default value in the ICE implementation if set.
absl::optional<int> ice_unwritable_min_checks;
// The min time period for which a candidate pair must wait for response to
// connectivity checks it becomes inactive. This parameter overrides the
// default value in the ICE implementation if set.
absl::optional<int> ice_inactive_timeout;
// The interval in milliseconds at which STUN candidates will resend STUN
// binding requests to keep NAT bindings open.
absl::optional<int> stun_candidate_keepalive_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].
absl::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.
absl::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 ignore all but the first m= section 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. This
// will also cause PeerConnection to ignore all but the first a=ssrc lines
// that form a Plan B stream.
// For users who wish to send multiple audio/video streams and need to stay
// interoperable with legacy WebRTC implementations or use legacy APIs,
// specify kPlanB.
// For all other users, specify kUnifiedPlan.
SdpSemantics sdp_semantics = SdpSemantics::kPlanB;
// TODO( - Move to crypto_options or remove.
// Actively reset the SRTP parameters whenever the DTLS transports
// underneath are reset for every offer/answer negotiation.
// This is only intended to be a workaround for
// WARNING: This would cause RTP/RTCP packets decryption failure if not used
// correctly. This flag will be deprecated soon. Do not rely on it.
bool active_reset_srtp_params = false;
// If MediaTransportFactory is provided in PeerConnectionFactory, this flag
// informs PeerConnection that it should use the MediaTransportInterface.
// It's invalid to set it to |true| if the MediaTransportFactory wasn't
// provided.
bool use_media_transport = false;
// If MediaTransportFactory is provided in PeerConnectionFactory, this flag
// informs PeerConnection that it should use the MediaTransportInterface for
// data channels. It's invalid to set it to |true| if the
// MediaTransportFactory wasn't provided. Data channels over media
// transport are not compatible with RTP or SCTP data channels. Setting
// both |use_media_transport_for_data_channels| and
// |enable_rtp_data_channel| is invalid.
bool use_media_transport_for_data_channels = false;
// Defines advanced optional cryptographic settings related to SRTP and
// frame encryption for native WebRTC. Setting this will overwrite any
// settings set in PeerConnectionFactory (which is deprecated).
absl::optional<CryptoOptions> crypto_options;
// Configure if we should include the SDP attribute extmap-allow-mixed in
// our offer. Although we currently do support this, it's not included in
// our offer by default due to a previous bug that caused the SDP parser to
// abort parsing if this attribute was present. This is fixed in Chrome 71.
// TODO(webrtc:9985): Change default to true once sufficient time has
// passed.
bool offer_extmap_allow_mixed = false;
// Don't forget to update operator== if adding something.
// See:
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 options are left as backwards compatibility for clients who need
// "Plan B" semantics. Clients who have switched to "Unified Plan" semantics
// should use the RtpTransceiver API (AddTransceiver) instead.
// 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;
// This will apply to all video tracks with a Plan B SDP offer/answer.
int num_simulcast_layers = 1;
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),
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 {
// Accessor methods to active local streams.
// This method is not supported with kUnifiedPlan semantics. Please use
// GetSenders() instead.
virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams() = 0;
// Accessor methods to remote streams.
// This method is not supported with kUnifiedPlan semantics. Please use
// GetReceivers() instead.
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.
// This method is not supported with kUnifiedPlan semantics. Please use
// AddTrack instead.
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.
// This method is not supported with kUnifiedPlan semantics. Please use
// RemoveTrack instead.
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_ids| 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.
virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids);
// Remove an RtpSender from this PeerConnection.
// Returns true on success.
// TODO(steveanton): Replace with signature that returns RTCError.
virtual bool RemoveTrack(RtpSenderInterface* sender);
// Plan B semantics: Removes the RtpSender from this PeerConnection.
// Unified Plan semantics: Stop sending on the RtpSender and mark the
// corresponding RtpTransceiver direction as no longer sending.
// Errors:
// - INVALID_PARAMETER: |sender| is null or (Plan B only) the sender is not
// associated with this PeerConnection.
// - INVALID_STATE: PeerConnection is closed.
// TODO( Rename to RemoveTrack once the other signature
// is removed.
virtual RTCError RemoveTrackNew(
rtc::scoped_refptr<RtpSenderInterface> sender);
// 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.
// 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);
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init);
// Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
// Errors:
// - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type);
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type, const RtpTransceiverInit& init);
// 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.
// This method is not supported with kUnifiedPlan semantics. Please use
// AddTransceiver instead.
virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
const std::string& kind,
const std::string& stream_id);
// If Plan B semantics are specified, gets all RtpSenders, created either
// through AddStream, AddTrack, or CreateSender. All senders of a specific
// media type share the same media description.
// If Unified Plan semantics are specified, gets the RtpSender for each
// RtpTransceiver.
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
// If Plan B semantics are specified, gets all RtpReceivers created when a
// remote description is applied. 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.
// If Unified Plan semantics are specified, gets the RtpReceiver for each
// RtpTransceiver.
virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
// 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;
// The legacy non-compliant GetStats() API. This correspond to the
// callback-based version of getStats() in JavaScript. The returned metrics
// are UNDOCUMENTED and many of them rely on implementation-specific details.
// The goal is to DELETE THIS VERSION but we can't today because it is heavily
// relied upon by third parties. See
// This version is wired up into Chrome. Any stats implemented are
// automatically exposed to the Web Platform. This has BYPASSED the Chrome
// release processes for years and lead to cross-browser incompatibility
// issues and web application reliance on Chrome-only behavior.
// This API is in "maintenance mode", serious regressions should be fixed but
// adding new stats is highly discouraged.
// TODO(hbos): Deprecate and remove this when third parties have migrated to
// the spec-compliant GetStats() API.
virtual bool GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track, // Optional
StatsOutputLevel level) = 0;
// The spec-compliant GetStats() API. This correspond to the promise-based
// version of getStats() in JavaScript. Implementation status is described in
// api/stats/rtcstats_objects.h. For more details on stats, see spec:
// TODO(hbos): Takes shared ownership, use rtc::scoped_refptr<> instead. This
// requires stop overriding the current version in third party or making third
// party calls explicit to avoid ambiguity during switch. Make the future
// version abstract as soon as third party projects implement it.
virtual void GetStats(RTCStatsCollectorCallback* callback) {}
// Spec-compliant getStats() performing the stats selection algorithm with the
// sender.
// TODO(hbos): Make abstract as soon as third party projects implement it.
virtual void GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
// Spec-compliant getStats() performing the stats selection algorithm with the
// receiver.
// TODO(hbos): Make abstract as soon as third party projects implement it.
virtual void GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
// Clear cached stats in the RTCStatsCollector.
// Exposed for testing while waiting for automatic cache clear to work.
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;
virtual const SessionDescriptionInterface* current_remote_description() const;
// 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;
virtual const SessionDescriptionInterface* pending_remote_description() const;
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) = 0;
// Create an answer to an offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) = 0;
// 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) {}
// TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
// PeerConnectionInterface implement it.
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration();
// 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);
// Version without error output param for backwards compatibility.
// TODO(deadbeef): Remove once chromium is updated.
virtual bool SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& config);
// 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);
// 0 <= min <= current <= max should hold for set parameters.
struct BitrateParameters {
absl::optional<int> min_bitrate_bps;
absl::optional<int> current_bitrate_bps;
absl::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 BitrateSettings& bitrate);
// TODO(nisse): Deprecated - use version above. These two default
// implementations require subclasses to implement one or the other
// of the methods.
virtual RTCError SetBitrate(const BitrateParameters& bitrate_parameters);
// 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(
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) {}
// Looks up the DtlsTransport associated with a MID value.
// In the Javascript API, DtlsTransport is a property of a sender, but
// because the PeerConnection owns the DtlsTransport in this implementation,
// it is better to look them up on the PeerConnection.
// TODO(hta): Remove default implementation after updating Chrome.
virtual rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
const std::string& mid);
// Returns the current SignalingState.
virtual SignalingState signaling_state() = 0;
// Returns an aggregate state of all ICE *and* DTLS transports.
// This is left in place to avoid breaking native clients who expect our old,
// nonstandard behavior.
// TODO(jonasolsson): deprecate and remove this.
virtual IceConnectionState ice_connection_state() = 0;
// Returns an aggregated state of all ICE transports.
virtual IceConnectionState standardized_ice_connection_state();
// Returns an aggregated state of all ICE and DTLS transports.
virtual PeerConnectionState peer_connection_state();
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);
// 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);
// 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;
// Dtor protected as objects shouldn't be deleted via this interface.
~PeerConnectionInterface() override = default;
// PeerConnection callback interface, used for RTCPeerConnection events.
// Application should implement these methods.
class PeerConnectionObserver {
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.
virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
// Triggered when a remote peer closes a stream.
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 legacy 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.
// TODO(jonasolsson): deprecate and remove this.
virtual void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) = 0;
// Called any time the standards-compliant IceConnectionState changes.
virtual void OnStandardizedIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {}
// Called any time the PeerConnectionState changes.
virtual void OnConnectionChange(
PeerConnectionInterface::PeerConnectionState new_state) {}
// 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 are created.
// TODO(zhihuang): Make this pure virtual when all subclasses implement it.
// Note: This is called with both Plan B and Unified Plan semantics. Unified
// Plan users should prefer OnTrack, OnAddTrack is only called as backwards
// compatibility (and is called in the exact same situations as OnTrack).
virtual void OnAddTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
// This is called when signaling indicates a transceiver will be receiving
// media from the remote endpoint. This is fired during a call to
// SetRemoteDescription. The receiving track can be accessed by:
// |transceiver->receiver()->track()| and its associated streams by
// |transceiver->receiver()->streams()|.
// Note: This will only be called if Unified Plan semantics are specified.
// This behavior is specified in section of the "Set the
// RTCSessionDescription" algorithm:
virtual void OnTrack(
rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {}
// Called when signaling indicates that media will no longer be received on a
// track.
// With Plan B semantics, the given receiver will have been removed from the
// PeerConnection and the track muted.
// With Unified Plan semantics, the receiver will remain but the transceiver
// will have changed direction to either sendonly or inactive.
// TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
virtual void OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
// Called when an interesting usage is detected by WebRTC.
// An appropriate action is to add information about the context of the
// PeerConnection and write the event to some kind of "interesting events"
// log function.
// The heuristics for defining what constitutes "interesting" are
// implementation-defined.
virtual void OnInterestingUsage(int usage_pattern) {}
// PeerConnectionDependencies holds all of PeerConnections dependencies.
// A dependency is distinct from a configuration as it defines significant
// executable code that can be provided by a user of the API.
// All new dependencies should be added as a unique_ptr to allow the
// PeerConnection object to be the definitive owner of the dependencies
// lifetime making injection safer.
struct PeerConnectionDependencies final {
explicit PeerConnectionDependencies(PeerConnectionObserver* observer_in);
// This object is not copyable or assignable.
PeerConnectionDependencies(const PeerConnectionDependencies&) = delete;
PeerConnectionDependencies& operator=(const PeerConnectionDependencies&) =
// This object is only moveable.
PeerConnectionDependencies& operator=(PeerConnectionDependencies&&) = default;
// Mandatory dependencies
PeerConnectionObserver* observer = nullptr;
// Optional dependencies
std::unique_ptr<cricket::PortAllocator> allocator;
std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory;
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
// PeerConnectionFactoryDependencies holds all of the PeerConnectionFactory
// dependencies. All new dependencies should be added here instead of
// overloading the function. This simplifies dependency injection and makes it
// clear which are mandatory and optional. If possible please allow the peer
// connection factory to take ownership of the dependency by adding a unique_ptr
// to this structure.
struct PeerConnectionFactoryDependencies final {
// This object is not copyable or assignable.
PeerConnectionFactoryDependencies(const PeerConnectionFactoryDependencies&) =
PeerConnectionFactoryDependencies& operator=(
const PeerConnectionFactoryDependencies&) = delete;
// This object is only moveable.
PeerConnectionFactoryDependencies& operator=(
PeerConnectionFactoryDependencies&&) = default;
// Optional dependencies
rtc::Thread* network_thread = nullptr;
rtc::Thread* worker_thread = nullptr;
rtc::Thread* signaling_thread = nullptr;
std::unique_ptr<cricket::MediaEngineInterface> media_engine;
std::unique_ptr<CallFactoryInterface> call_factory;
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
std::unique_ptr<MediaTransportFactory> media_transport_factory;
// 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 {
class Options {
Options() {}
// 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
// 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.
CryptoOptions crypto_options = CryptoOptions::NoGcm();
// Set the options to be used for subsequently created PeerConnections.
virtual void SetOptions(const Options& options) = 0;
// The preferred way to create a new peer connection. Simply provide the
// configuration and a PeerConnectionDependencies structure.
// TODO(benwright): Make pure virtual once downstream mock PC factory classes
// are updated.
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
// Deprecated; |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);
// Returns the capabilities of an RTP sender of type |kind|.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpSenderCapabilities(
cricket::MediaType kind) const;
// Returns the capabilities of an RTP receiver of type |kind|.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpReceiverCapabilities(
cricket::MediaType kind) const;
virtual rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id) = 0;
// Creates an AudioSourceInterface.
// |options| decides audio processing settings.
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) = 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);
// 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);
// 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);
// 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;
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.
PeerConnectionFactoryInterface() {}
~PeerConnectionFactoryInterface() override = default;
// 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::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);
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,
std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
network_controller_factory = nullptr);
PeerConnectionFactoryDependencies dependencies);
} // namespace webrtc