| Index: webrtc/api/ortc/ortcrtpsenderinterface.h
|
| diff --git a/webrtc/api/ortc/ortcrtpsenderinterface.h b/webrtc/api/ortc/ortcrtpsenderinterface.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..0e344828da23037909aecd5b5a77202f78dcd7b4
|
| --- /dev/null
|
| +++ b/webrtc/api/ortc/ortcrtpsenderinterface.h
|
| @@ -0,0 +1,61 @@
|
| +/*
|
| + * Copyright 2017 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 interfaces for RtpSenders
|
| +// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
|
| +
|
| +#ifndef WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
|
| +#define WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
|
| +
|
| +#include "webrtc/api/mediatypes.h"
|
| +#include "webrtc/api/mediastreaminterface.h"
|
| +#include "webrtc/api/ortc/rtptransportinterface.h"
|
| +#include "webrtc/api/rtcerror.h"
|
| +#include "webrtc/api/rtpparameters.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +// Note: Since sender capabilities may depend on how the OrtcFactory was
|
| +// created, instead of a static "GetCapabilities" method on this interface,
|
| +// there is a "GetRtpSenderCapabilities" method on the OrtcFactory.
|
| +class OrtcRtpSenderInterface {
|
| + public:
|
| + virtual ~OrtcRtpSenderInterface() {}
|
| +
|
| + // Returns INVALID_PARAMETER error if an audio track is set on a video
|
| + // RtpSender, or vice-versa.
|
| + virtual RTCError SetTrack(MediaStreamTrackInterface* track) = 0;
|
| + // Returns previously set (or constructed-with) track.
|
| + virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
|
| +
|
| + // Switches to sending media on a new transport.
|
| + virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
|
| + // Returns previously set (or constructed-with) transport.
|
| + virtual RtpTransportInterface* GetTransport() const = 0;
|
| +
|
| + // Allows the parameters of a sender to be changed after being constructed.
|
| + // There are no limitations to how the parameters can be changed after
|
| + // construction, as long as they're valid (for example, they can't use the
|
| + // same payload type for two codecs).
|
| + //
|
| + // Equivalent to "send" in the ORTC API.
|
| + virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
|
| + // Returns previously set (or constructed-with) parameters.
|
| + virtual RtpParameters GetParameters() const = 0;
|
| +
|
| + // Audio or video sender?
|
| + virtual cricket::MediaType GetKind() const = 0;
|
| +
|
| + // TODO(deadbeef): SSRC conflict signal.
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
|
|
|