2015-01-20 21:36:13 +00:00
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/*
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2016-02-07 20:46:45 -08:00
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* Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
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2015-01-20 21:36:13 +00:00
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*
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2016-02-07 20:46:45 -08:00
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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2015-01-20 21:36:13 +00:00
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*/
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2013-07-10 00:45:36 +00:00
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2016-02-12 06:39:40 +01:00
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#ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOCAPTURER_H_
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#define WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOCAPTURER_H_
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2013-07-10 00:45:36 +00:00
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2016-02-26 03:00:35 -08:00
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#include <memory>
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2013-07-10 00:45:36 +00:00
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#include <string>
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#include <vector>
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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2016-03-16 09:34:56 -07:00
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#include "webrtc/media/base/device.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/videocapturer.h"
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2015-11-12 12:46:09 +01:00
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#include "webrtc/modules/video_capture/video_capture.h"
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2017-07-06 19:44:34 +02:00
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#include "webrtc/rtc_base/asyncinvoker.h"
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#include "webrtc/rtc_base/messagehandler.h"
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#include "webrtc/rtc_base/scoped_ref_ptr.h"
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2013-07-10 00:45:36 +00:00
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namespace cricket {
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// Factory to allow injection of a VCM impl into WebRtcVideoCapturer.
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// DeviceInfos do not have a Release() and therefore need an explicit Destroy().
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class WebRtcVcmFactoryInterface {
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public:
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virtual ~WebRtcVcmFactoryInterface() {}
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2016-03-22 17:17:39 +01:00
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virtual rtc::scoped_refptr<webrtc::VideoCaptureModule> Create(
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const char* device) = 0;
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2016-12-12 00:22:56 -08:00
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virtual webrtc::VideoCaptureModule::DeviceInfo* CreateDeviceInfo() = 0;
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2013-07-10 00:45:36 +00:00
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virtual void DestroyDeviceInfo(
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webrtc::VideoCaptureModule::DeviceInfo* info) = 0;
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};
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// WebRTC-based implementation of VideoCapturer.
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class WebRtcVideoCapturer : public VideoCapturer,
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public rtc::VideoSinkInterface<webrtc::VideoFrame> {
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public:
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WebRtcVideoCapturer();
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explicit WebRtcVideoCapturer(WebRtcVcmFactoryInterface* factory);
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virtual ~WebRtcVideoCapturer();
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bool Init(const Device& device);
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2016-03-22 17:17:39 +01:00
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bool Init(const rtc::scoped_refptr<webrtc::VideoCaptureModule>& module);
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2013-07-10 00:45:36 +00:00
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// Override virtual methods of the parent class VideoCapturer.
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2016-02-12 13:30:57 +01:00
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bool GetBestCaptureFormat(const VideoFormat& desired,
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VideoFormat* best_format) override;
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CaptureState Start(const VideoFormat& capture_format) override;
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void Stop() override;
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bool IsRunning() override;
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bool IsScreencast() const override { return false; }
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2013-07-10 00:45:36 +00:00
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protected:
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void OnSinkWantsChanged(const rtc::VideoSinkWants& wants) override;
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// Override virtual methods of the parent class VideoCapturer.
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bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) override;
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2013-07-10 00:45:36 +00:00
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private:
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// Callback when a frame is captured by camera.
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void OnFrame(const webrtc::VideoFrame& frame) override;
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2013-07-10 00:45:36 +00:00
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2015-02-19 17:43:25 +00:00
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// Used to signal captured frames on the same thread as invoked Start().
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// With WebRTC's current VideoCapturer implementations, this will mean a
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// thread hop, but in other implementations (e.g. Chrome) it will be called
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// directly from OnIncomingCapturedFrame.
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// TODO(tommi): Remove this workaround when we've updated the WebRTC capturers
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// to follow the same contract.
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2015-10-24 11:14:46 -07:00
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void SignalFrameCapturedOnStartThread(const webrtc::VideoFrame& frame);
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2015-02-19 17:43:25 +00:00
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2016-02-26 03:00:35 -08:00
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std::unique_ptr<WebRtcVcmFactoryInterface> factory_;
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2016-03-22 17:17:39 +01:00
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rtc::scoped_refptr<webrtc::VideoCaptureModule> module_;
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2013-07-10 00:45:36 +00:00
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int captured_frames_;
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2013-11-05 23:45:14 +00:00
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std::vector<uint8_t> capture_buffer_;
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2015-02-19 17:43:25 +00:00
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rtc::Thread* start_thread_; // Set in Start(), unset in Stop();
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2014-01-03 22:08:47 +00:00
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2016-02-26 03:00:35 -08:00
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std::unique_ptr<rtc::AsyncInvoker> async_invoker_;
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2013-07-10 00:45:36 +00:00
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};
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} // namespace cricket
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2016-02-12 13:30:57 +01:00
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#endif // WEBRTC_MEDIA_WEBRTC_WEBRTCVIDEOCAPTURER_H_
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