Delete some unused header files.
NOTRY=True # two of the androids bots are not cooperating BUG= Review-Url: https://codereview.webrtc.org/2057533002 Cr-Commit-Position: refs/heads/master@{#13099}
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/*
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* Copyright (c) 2012 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_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_INTERNAL_H_
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#define WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_INTERNAL_H_
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#ifdef ANDROID
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#include <jni.h>
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namespace webrtc {
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// In order to be able to use the internal webrtc video capture
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// for android, the jvm objects must be set via this method.
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int32_t SetCaptureAndroidVM(JavaVM* javaVM, jobject context);
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} // namespace webrtc
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#endif // ANDROID
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#endif // WEBRTC_MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_INTERNAL_H_
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/*
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* Copyright (c) 2011 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_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
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#define WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
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int ReleaseTest();
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int ReleaseTestPart2();
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#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RELEASE_TEST_H_
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/*
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* Copyright (c) 2012 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_TOOLS_AGC_FAKE_AGC_H_
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#define WEBRTC_TOOLS_AGC_FAKE_AGC_H_
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#include "webrtc/modules/audio_processing/agc/agc.h"
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namespace webrtc {
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class FakeAgc : public Agc {
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public:
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FakeAgc()
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: counter_(0),
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volume_(kMaxVolume / 2) {
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}
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virtual int Process(const AudioFrame& audio_frame) {
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const int kUpdateIntervalFrames = 10;
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const int kMaxVolume = 255;
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if (counter_ % kUpdateIntervalFrames == 0) {
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volume_ = (++volume_) % kMaxVolume;
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}
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counter_++;
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return 0;
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}
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virtual int FakeAgc::MicVolume() {
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return volume_;
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}
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private:
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int counter_;
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int volume_;
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};
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} // namespace webrtc
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#endif // WEBRTC_TOOLS_AGC_FAKE_AGC_H_
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