2013-07-10 00:45:36 +00:00
|
|
|
/*
|
2016-02-10 07:54:43 -08:00
|
|
|
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
|
2013-07-10 00:45:36 +00:00
|
|
|
*
|
2016-02-10 07:54:43 -08:00
|
|
|
* 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.
|
2013-07-10 00:45:36 +00:00
|
|
|
*/
|
|
|
|
|
|
2016-02-10 10:53:12 +01:00
|
|
|
#ifndef WEBRTC_API_VIDEOSOURCEINTERFACE_H_
|
|
|
|
|
#define WEBRTC_API_VIDEOSOURCEINTERFACE_H_
|
2013-07-10 00:45:36 +00:00
|
|
|
|
2016-02-10 10:53:12 +01:00
|
|
|
#include "webrtc/api/mediastreaminterface.h"
|
Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
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 in order to not
break Git history.
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.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media 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/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-04 23:52:28 -08:00
|
|
|
#include "webrtc/media/base/mediachannel.h"
|
|
|
|
|
#include "webrtc/media/base/videorenderer.h"
|
2013-07-10 00:45:36 +00:00
|
|
|
|
|
|
|
|
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;
|
2015-02-17 13:53:56 +00:00
|
|
|
|
|
|
|
|
// Stop the video capturer.
|
2015-03-02 11:33:20 +00:00
|
|
|
virtual void Stop() = 0;
|
|
|
|
|
virtual void Restart() = 0;
|
2015-02-17 13:53:56 +00:00
|
|
|
|
2013-07-10 00:45:36 +00:00
|
|
|
// Adds |output| to the source to receive frames.
|
2016-02-04 01:01:54 -08:00
|
|
|
virtual void AddSink(
|
|
|
|
|
rtc::VideoSinkInterface<cricket::VideoFrame>* output) = 0;
|
2016-01-28 04:47:08 -08:00
|
|
|
virtual void RemoveSink(
|
2016-02-04 01:01:54 -08:00
|
|
|
rtc::VideoSinkInterface<cricket::VideoFrame>* output) = 0;
|
2013-07-10 00:45:36 +00:00
|
|
|
virtual const cricket::VideoOptions* options() const = 0;
|
2016-02-05 01:52:15 -08:00
|
|
|
// TODO(nisse): Dummy implementation. Delete as soon as chrome's
|
|
|
|
|
// MockVideoSource is updated.
|
|
|
|
|
virtual cricket::VideoRenderer* FrameInput() { return nullptr; }
|
2013-07-10 00:45:36 +00:00
|
|
|
|
|
|
|
|
protected:
|
|
|
|
|
virtual ~VideoSourceInterface() {}
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
} // namespace webrtc
|
|
|
|
|
|
2016-02-10 10:53:12 +01:00
|
|
|
#endif // WEBRTC_API_VIDEOSOURCEINTERFACE_H_
|