webrtc_m130/webrtc/api/androidvideocapturer.h

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/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ANDROIDVIDEOCAPTURER_H_
#define WEBRTC_API_ANDROIDVIDEOCAPTURER_H_
#include <string>
#include <vector>
#include "webrtc/base/thread_checker.h"
#include "webrtc/common_video/include/video_frame_buffer.h"
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
#include "webrtc/media/base/videocapturer.h"
namespace webrtc {
class AndroidVideoCapturer;
class AndroidVideoCapturerDelegate : public rtc::RefCountInterface {
public:
virtual ~AndroidVideoCapturerDelegate() {}
// Start capturing. The implementation of the delegate must call
// AndroidVideoCapturer::OnCapturerStarted with the result of this request.
virtual void Start(int width, int height, int framerate,
AndroidVideoCapturer* capturer) = 0;
// Stops capturing.
// The delegate may not call into AndroidVideoCapturer after this call.
virtual void Stop() = 0;
virtual std::vector<cricket::VideoFormat> GetSupportedFormats() = 0;
};
// Android implementation of cricket::VideoCapturer for use with WebRtc
// PeerConnection.
class AndroidVideoCapturer : public cricket::VideoCapturer {
public:
explicit AndroidVideoCapturer(
const rtc::scoped_refptr<AndroidVideoCapturerDelegate>& delegate);
virtual ~AndroidVideoCapturer();
// Called from JNI when the capturer has been started.
void OnCapturerStarted(bool success);
// Called from JNI when a new frame has been captured.
// Argument |buffer| is intentionally by value, for use with rtc::Bind.
void OnIncomingFrame(
const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
int rotation,
int64_t time_stamp);
// Called from JNI to request a new video format.
void OnOutputFormatRequest(int width, int height, int fps);
AndroidVideoCapturerDelegate* delegate() { return delegate_.get(); }
// cricket::VideoCapturer implementation.
bool GetBestCaptureFormat(const cricket::VideoFormat& desired,
cricket::VideoFormat* best_format) override;
private:
// cricket::VideoCapturer implementation.
// Video frames will be delivered using
// cricket::VideoCapturer::SignalFrameCaptured on the thread that calls Start.
cricket::CaptureState Start(
const cricket::VideoFormat& capture_format) override;
void Stop() override;
bool IsRunning() override;
bool IsScreencast() const override { return false; }
bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) override;
bool running_;
rtc::scoped_refptr<AndroidVideoCapturerDelegate> delegate_;
rtc::ThreadChecker thread_checker_;
class FrameFactory;
FrameFactory* frame_factory_; // Owned by cricket::VideoCapturer.
cricket::CaptureState current_state_;
};
} // namespace webrtc
#endif // WEBRTC_API_ANDROIDVIDEOCAPTURER_H_