#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/statstypes.h"
#include "talk/app/webrtc/umametrics.h"
-#include "talk/base/fileutils.h"
-#include "talk/base/socketaddress.h"
+#include "webrtc/base/fileutils.h"
+#include "webrtc/base/socketaddress.h"
-namespace talk_base {
+namespace rtc {
class Thread;
}
class MediaConstraintsInterface;
// MediaStream container interface.
-class StreamCollectionInterface : public talk_base::RefCountInterface {
+class StreamCollectionInterface : public rtc::RefCountInterface {
public:
// TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
virtual size_t count() = 0;
~StreamCollectionInterface() {}
};
-class StatsObserver : public talk_base::RefCountInterface {
+class StatsObserver : public rtc::RefCountInterface {
public:
virtual void OnComplete(const std::vector<StatsReport>& reports) = 0;
virtual ~StatsObserver() {}
};
-class UMAObserver : public talk_base::RefCountInterface {
+class UMAObserver : public rtc::RefCountInterface {
public:
virtual void IncrementCounter(PeerConnectionUMAMetricsCounter type) = 0;
virtual void AddHistogramSample(PeerConnectionUMAMetricsName type,
virtual ~UMAObserver() {}
};
-class PeerConnectionInterface : public talk_base::RefCountInterface {
+class PeerConnectionInterface : public rtc::RefCountInterface {
public:
// See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
enum SignalingState {
explicit RTCConfiguration(IceTransportsType type) : type(type) {}
};
+ struct RTCOfferAnswerOptions {
+ static const int kUndefined = -1;
+ static const int kMaxOfferToReceiveMedia = 1;
+
+ // The default value for constraint offerToReceiveX:true.
+ static const int kOfferToReceiveMediaTrue = 1;
+
+ int offer_to_receive_video;
+ int offer_to_receive_audio;
+ bool voice_activity_detection;
+ bool ice_restart;
+ bool use_rtp_mux;
+
+ RTCOfferAnswerOptions()
+ : offer_to_receive_video(kUndefined),
+ offer_to_receive_audio(kUndefined),
+ voice_activity_detection(true),
+ ice_restart(false),
+ use_rtp_mux(true) {}
+
+ 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
};
// Accessor methods to active local streams.
- virtual talk_base::scoped_refptr<StreamCollectionInterface>
+ virtual rtc::scoped_refptr<StreamCollectionInterface>
local_streams() = 0;
// Accessor methods to remote streams.
- virtual talk_base::scoped_refptr<StreamCollectionInterface>
+ virtual rtc::scoped_refptr<StreamCollectionInterface>
remote_streams() = 0;
// Add a new MediaStream to be sent on this PeerConnection.
// Returns pointer to the created DtmfSender on success.
// Otherwise returns NULL.
- virtual talk_base::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
+ virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
AudioTrackInterface* track) = 0;
virtual bool GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) = 0;
- virtual talk_base::scoped_refptr<DataChannelInterface> CreateDataChannel(
+ virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
const DataChannelInit* config) = 0;
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
- const MediaConstraintsInterface* constraints) = 0;
+ 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,
// Factory class used for creating cricket::PortAllocator that is used
// for ICE negotiation.
-class PortAllocatorFactoryInterface : public talk_base::RefCountInterface {
+class PortAllocatorFactoryInterface : public rtc::RefCountInterface {
public:
struct StunConfiguration {
StunConfiguration(const std::string& address, int port)
: server(address, port) {}
// STUN server address and port.
- talk_base::SocketAddress server;
+ rtc::SocketAddress server;
};
struct TurnConfiguration {
password(password),
transport_type(transport_type),
secure(secure) {}
- talk_base::SocketAddress server;
+ rtc::SocketAddress server;
std::string username;
std::string password;
std::string transport_type;
};
// Used to receive callbacks of DTLS identity requests.
-class DTLSIdentityRequestObserver : public talk_base::RefCountInterface {
+class DTLSIdentityRequestObserver : public rtc::RefCountInterface {
public:
virtual void OnFailure(int error) = 0;
virtual void OnSuccess(const std::string& der_cert,
// CreatePeerConnectionFactory method which accepts threads as input and use the
// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
// argument.
-class PeerConnectionFactoryInterface : public talk_base::RefCountInterface {
+class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
public:
class Options {
public:
virtual void SetOptions(const Options& options) = 0;
- virtual
talk_base::scoped_refptr<PeerConnectionInterface>
+ virtual
rtc::scoped_refptr<PeerConnectionInterface>
CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
// and not IceServers. RTCConfiguration is made up of ice servers and
// ice transport type.
// http://dev.w3.org/2011/webrtc/editor/webrtc.html
- inline
talk_base::scoped_refptr<PeerConnectionInterface>
+ inline
rtc::scoped_refptr<PeerConnectionInterface>
CreatePeerConnection(
const PeerConnectionInterface::IceServers& configuration,
const MediaConstraintsInterface* constraints,
dtls_identity_service, observer);
}
- virtual talk_base::scoped_refptr<MediaStreamInterface>
+ virtual rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) = 0;
// Creates a AudioSourceInterface.
// |constraints| decides audio processing settings but can be NULL.
- virtual
talk_base::scoped_refptr<AudioSourceInterface> CreateAudioSource(
+ virtual
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const MediaConstraintsInterface* constraints) = 0;
// Creates a VideoSourceInterface. The new source take ownership of
// |capturer|. |constraints| decides video resolution and frame rate but can
// be NULL.
- virtual talk_base::scoped_refptr<VideoSourceInterface> CreateVideoSource(
+ virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
cricket::VideoCapturer* capturer,
const MediaConstraintsInterface* constraints) = 0;
// Creates a new local VideoTrack. The same |source| can be used in several
// tracks.
- virtual talk_base::scoped_refptr<VideoTrackInterface>
+ virtual rtc::scoped_refptr<VideoTrackInterface>
CreateVideoTrack(const std::string& label,
VideoSourceInterface* source) = 0;
// Creates an new AudioTrack. At the moment |source| can be NULL.
- virtual
talk_base::scoped_refptr<AudioTrackInterface>
+ virtual
rtc::scoped_refptr<AudioTrackInterface>
CreateAudioTrack(const std::string& label,
AudioSourceInterface* source) = 0;
// the ownerhip. If the operation fails, the file will be closed.
// TODO(grunell): Remove when Chromium has started to use AEC in each source.
// http://crbug.com/264611.
- virtual bool StartAecDump(talk_base::PlatformFile file) = 0;
+ virtual bool StartAecDump(rtc::PlatformFile file) = 0;
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
};
// Create a new instance of PeerConnectionFactoryInterface.
-
talk_base::scoped_refptr<PeerConnectionFactoryInterface>
+
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreatePeerConnectionFactory();
// Create a new instance of PeerConnectionFactoryInterface.
// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
// |decoder_factory| transferred to the returned factory.
-
talk_base::scoped_refptr<PeerConnectionFactoryInterface>
+
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreatePeerConnectionFactory(
- talk_base::Thread* worker_thread,
- talk_base::Thread* signaling_thread,
+ rtc::Thread* worker_thread,
+ rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
cricket::WebRtcVideoEncoderFactory* encoder_factory,
cricket::WebRtcVideoDecoderFactory* decoder_factory);
projects / platform / framework / web / crosswalk.git / blobdiff