a96e2d77cb
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}
66 lines
2.0 KiB
Objective-C
66 lines
2.0 KiB
Objective-C
/*
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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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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/media/base/videocapturer.h"
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#include "webrtc/video_frame.h"
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#import <AVFoundation/AVFoundation.h>
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@class RTCAVFoundationVideoCapturerInternal;
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namespace webrtc {
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class AVFoundationVideoCapturer : public cricket::VideoCapturer {
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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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/** Returns the active capture session. */
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AVCaptureSession* GetCaptureSession();
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/** Switches the camera being used (either front or back). */
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void SetUseBackCamera(bool useBackCamera);
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bool GetUseBackCamera() const;
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/**
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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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*/
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void CaptureSampleBuffer(CMSampleBufferRef sampleBuffer);
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private:
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/**
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* Used to signal frame capture on the thread that capturer was started on.
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*/
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void SignalFrameCapturedOnStartThread(const cricket::CapturedFrame *frame);
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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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#endif // TALK_APP_WEBRTC_OBJC_AVFOUNDATION_CAPTURER_H_
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