2016-01-21 11:44:55 -08:00
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/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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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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*/
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#ifndef WEBRTC_API_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
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#define WEBRTC_API_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
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2016-04-27 01:54:20 -07:00
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#import <AVFoundation/AVFoundation.h>
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2016-01-21 11:44:55 -08:00
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#include "webrtc/base/scoped_ptr.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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2016-01-21 11:44:55 -08:00
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#include "webrtc/video_frame.h"
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@class RTCAVFoundationVideoCapturerInternal;
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2016-03-17 12:20:41 +01:00
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namespace rtc {
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class Thread;
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} // namespace rtc
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2016-01-21 11:44:55 -08:00
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namespace webrtc {
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2016-03-31 17:14:04 -07:00
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class AVFoundationVideoCapturer : public cricket::VideoCapturer,
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public rtc::MessageHandler {
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2016-01-21 11:44:55 -08:00
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public:
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AVFoundationVideoCapturer();
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~AVFoundationVideoCapturer();
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cricket::CaptureState Start(const cricket::VideoFormat& format) override;
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void Stop() override;
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bool IsRunning() override;
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bool IsScreencast() const override {
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return false;
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}
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bool GetPreferredFourccs(std::vector<uint32_t> *fourccs) override {
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fourccs->push_back(cricket::FOURCC_NV12);
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return true;
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}
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2016-03-31 17:14:04 -07:00
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// Returns the active capture session. Calls to the capture session should
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// occur on the RTCDispatcherTypeCaptureSession queue in RTCDispatcher.
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2016-01-21 11:44:55 -08:00
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AVCaptureSession* GetCaptureSession();
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2016-03-31 17:14:04 -07:00
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// Returns whether the rear-facing camera can be used.
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// e.g. It can't be used because it doesn't exist.
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2016-03-14 20:55:22 -07:00
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bool CanUseBackCamera() const;
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2016-03-31 17:14:04 -07:00
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// Switches the camera being used (either front or back).
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2016-01-21 11:44:55 -08:00
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void SetUseBackCamera(bool useBackCamera);
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bool GetUseBackCamera() const;
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2016-03-31 17:14:04 -07:00
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// Converts the sample buffer into a cricket::CapturedFrame and signals the
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// frame for capture.
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2016-01-21 11:44:55 -08:00
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void CaptureSampleBuffer(CMSampleBufferRef sampleBuffer);
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2016-03-31 17:14:04 -07:00
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// Handles messages from posts.
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void OnMessage(rtc::Message *msg) override;
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2016-01-21 11:44:55 -08:00
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private:
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2016-03-31 17:14:04 -07:00
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void OnFrameMessage(CVImageBufferRef image_buffer, int64_t capture_time);
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2016-01-21 11:44:55 -08:00
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RTCAVFoundationVideoCapturerInternal *_capturer;
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rtc::Thread *_startThread; // Set in Start(), unset in Stop().
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}; // AVFoundationVideoCapturer
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} // namespace webrtc
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2016-03-31 17:14:04 -07:00
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#endif // WEBRTC_API_OBJC_AVFOUNDATION_VIDEO_CAPTURER_H_
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