webrtc_m130/webrtc/api/videosourceinterface.h
Henrik Kjellander 15583c19d7 Move talk/app/webrtc to webrtc/api
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc

The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.

I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002

BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1610243002 .

Cr-Commit-Position: refs/heads/master@{#11545}
2016-02-10 09:53:26 +00:00

69 lines
2.9 KiB
C++

/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef WEBRTC_API_VIDEOSOURCEINTERFACE_H_
#define WEBRTC_API_VIDEOSOURCEINTERFACE_H_
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/base/videorenderer.h"
namespace webrtc {
// VideoSourceInterface is a reference counted source used for VideoTracks.
// The same source can be used in multiple VideoTracks.
// The methods are only supposed to be called by the PeerConnection
// implementation.
class VideoSourceInterface : public MediaSourceInterface {
public:
// Get access to the source implementation of cricket::VideoCapturer.
// This can be used for receiving frames and state notifications.
// But it should not be used for starting or stopping capturing.
virtual cricket::VideoCapturer* GetVideoCapturer() = 0;
// Stop the video capturer.
virtual void Stop() = 0;
virtual void Restart() = 0;
// Adds |output| to the source to receive frames.
virtual void AddSink(
rtc::VideoSinkInterface<cricket::VideoFrame>* output) = 0;
virtual void RemoveSink(
rtc::VideoSinkInterface<cricket::VideoFrame>* output) = 0;
virtual const cricket::VideoOptions* options() const = 0;
// TODO(nisse): Dummy implementation. Delete as soon as chrome's
// MockVideoSource is updated.
virtual cricket::VideoRenderer* FrameInput() { return nullptr; }
protected:
virtual ~VideoSourceInterface() {}
};
} // namespace webrtc
#endif // WEBRTC_API_VIDEOSOURCEINTERFACE_H_