2015-02-11 11:26:56 +00:00
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/*
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2016-02-10 07:54:43 -08:00
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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2015-02-11 11:26:56 +00:00
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*
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2016-02-10 07:54:43 -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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2015-02-11 11:26:56 +00:00
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*/
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2016-02-10 10:53:12 +01:00
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#ifndef WEBRTC_API_ANDROIDVIDEOCAPTURER_H_
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#define WEBRTC_API_ANDROIDVIDEOCAPTURER_H_
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2015-02-11 11:26:56 +00:00
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#include <string>
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#include <vector>
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2015-04-02 12:30:51 +02:00
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#include "webrtc/base/thread_checker.h"
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2015-11-16 13:52:24 -08:00
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#include "webrtc/common_video/include/video_frame_buffer.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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2015-02-11 11:26:56 +00:00
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namespace webrtc {
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class AndroidVideoCapturer;
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2015-04-02 12:30:51 +02:00
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class AndroidVideoCapturerDelegate : public rtc::RefCountInterface {
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2015-02-11 11:26:56 +00:00
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public:
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virtual ~AndroidVideoCapturerDelegate() {}
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// Start capturing. The implementation of the delegate must call
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// AndroidVideoCapturer::OnCapturerStarted with the result of this request.
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virtual void Start(int width, int height, int framerate,
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AndroidVideoCapturer* capturer) = 0;
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2015-02-19 08:43:38 +00:00
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// Stops capturing.
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2015-02-11 11:26:56 +00:00
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// The delegate may not call into AndroidVideoCapturer after this call.
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2015-02-19 08:43:38 +00:00
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virtual void Stop() = 0;
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2015-02-11 11:26:56 +00:00
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2016-02-18 13:09:54 +01:00
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virtual std::vector<cricket::VideoFormat> GetSupportedFormats() = 0;
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2015-02-11 11:26:56 +00:00
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};
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// Android implementation of cricket::VideoCapturer for use with WebRtc
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// PeerConnection.
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class AndroidVideoCapturer : public cricket::VideoCapturer {
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public:
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explicit AndroidVideoCapturer(
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2015-04-02 12:30:51 +02:00
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const rtc::scoped_refptr<AndroidVideoCapturerDelegate>& delegate);
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2015-02-11 11:26:56 +00:00
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virtual ~AndroidVideoCapturer();
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2015-02-25 09:20:07 +00:00
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// Called from JNI when the capturer has been started.
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2015-02-11 11:26:56 +00:00
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void OnCapturerStarted(bool success);
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2015-02-25 09:20:07 +00:00
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// Called from JNI when a new frame has been captured.
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2015-08-25 23:22:08 +02:00
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// Argument |buffer| is intentionally by value, for use with rtc::Bind.
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2015-10-20 11:04:56 -07:00
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void OnIncomingFrame(
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const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
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int rotation,
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int64_t time_stamp);
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2015-02-11 11:26:56 +00:00
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2015-06-15 09:53:05 +02:00
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// Called from JNI to request a new video format.
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void OnOutputFormatRequest(int width, int height, int fps);
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2015-02-11 11:26:56 +00:00
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AndroidVideoCapturerDelegate* delegate() { return delegate_.get(); }
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2015-08-28 11:40:59 +02:00
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// cricket::VideoCapturer implementation.
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bool GetBestCaptureFormat(const cricket::VideoFormat& desired,
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cricket::VideoFormat* best_format) override;
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2015-02-11 11:26:56 +00:00
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private:
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// cricket::VideoCapturer implementation.
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// Video frames will be delivered using
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// cricket::VideoCapturer::SignalFrameCaptured on the thread that calls Start.
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cricket::CaptureState Start(
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const cricket::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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Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 12:23:21 +02:00
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bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) override;
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2015-02-11 11:26:56 +00:00
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bool running_;
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2015-04-02 12:30:51 +02:00
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rtc::scoped_refptr<AndroidVideoCapturerDelegate> delegate_;
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2015-02-11 11:26:56 +00:00
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2015-04-02 12:30:51 +02:00
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rtc::ThreadChecker thread_checker_;
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2015-02-11 11:26:56 +00:00
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class FrameFactory;
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FrameFactory* frame_factory_; // Owned by cricket::VideoCapturer.
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2015-02-17 13:53:56 +00:00
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cricket::CaptureState current_state_;
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2015-02-11 11:26:56 +00:00
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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_ANDROIDVIDEOCAPTURER_H_
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