2013-07-10 00:45:36 +00:00
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/*
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* libjingle
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2015-01-20 21:36:13 +00:00
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* Copyright 2012 Google Inc.
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2013-07-10 00:45:36 +00:00
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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2016-02-10 10:53:12 +01:00
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#ifndef WEBRTC_API_VIDEOSOURCEINTERFACE_H_
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#define WEBRTC_API_VIDEOSOURCEINTERFACE_H_
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2013-07-10 00:45:36 +00:00
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2016-02-10 10:53:12 +01:00
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#include "webrtc/api/mediastreaminterface.h"
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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
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#include "webrtc/media/base/mediachannel.h"
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#include "webrtc/media/base/videorenderer.h"
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2013-07-10 00:45:36 +00:00
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namespace webrtc {
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// VideoSourceInterface is a reference counted source used for VideoTracks.
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// The same source can be used in multiple VideoTracks.
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// The methods are only supposed to be called by the PeerConnection
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// implementation.
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class VideoSourceInterface : public MediaSourceInterface {
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public:
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// Get access to the source implementation of cricket::VideoCapturer.
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// This can be used for receiving frames and state notifications.
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// But it should not be used for starting or stopping capturing.
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virtual cricket::VideoCapturer* GetVideoCapturer() = 0;
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2015-02-17 13:53:56 +00:00
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// Stop the video capturer.
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2015-03-02 11:33:20 +00:00
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virtual void Stop() = 0;
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virtual void Restart() = 0;
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2015-02-17 13:53:56 +00:00
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2013-07-10 00:45:36 +00:00
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// Adds |output| to the source to receive frames.
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2016-02-04 01:01:54 -08:00
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virtual void AddSink(
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rtc::VideoSinkInterface<cricket::VideoFrame>* output) = 0;
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2016-01-28 04:47:08 -08:00
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virtual void RemoveSink(
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2016-02-04 01:01:54 -08:00
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rtc::VideoSinkInterface<cricket::VideoFrame>* output) = 0;
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2013-07-10 00:45:36 +00:00
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virtual const cricket::VideoOptions* options() const = 0;
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2016-02-05 01:52:15 -08:00
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// TODO(nisse): Dummy implementation. Delete as soon as chrome's
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// MockVideoSource is updated.
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virtual cricket::VideoRenderer* FrameInput() { return nullptr; }
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2013-07-10 00:45:36 +00:00
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protected:
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virtual ~VideoSourceInterface() {}
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};
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} // namespace webrtc
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2016-02-10 10:53:12 +01:00
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#endif // WEBRTC_API_VIDEOSOURCEINTERFACE_H_
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