1ece0cbbec
And the follow-up fix in r8198 that was not sufficient. Reason: breaks Chromium bots runhooks (GYP). I will have to try some more to make sure I don't include test code, since include_tests==0 in Chromium. TBR=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37039004 Cr-Commit-Position: refs/heads/master@{#8200} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8200 4adac7df-926f-26a2-2b94-8c16560cd09d
82 lines
2.9 KiB
C++
82 lines
2.9 KiB
C++
/*
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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_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
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#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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namespace webrtc {
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class Agc;
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class AudioProcessing;
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class CriticalSectionWrapper;
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class MediaCallback;
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class PreprocCallback;
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class VoEExternalMedia;
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class VoEVolumeControl;
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class VoiceEngine;
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class VolumeCallbacks;
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// Handles the interaction between VoiceEngine and the internal AGC. It hooks
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// into the capture stream through VoiceEngine's external media interface and
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// sends the audio to the AGC for analysis. It forwards requests for a capture
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// volume change from the AGC to the VoiceEngine volume interface.
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class AgcManager {
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public:
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explicit AgcManager(VoiceEngine* voe);
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// Dependency injection for testing. Don't delete |agc| or |audioproc| as the
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// memory is owned by the manager. If |media| or |volume| are non-fake
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// reference counted classes, don't release them as this is handled by the
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// manager.
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AgcManager(VoEExternalMedia* media, VoEVolumeControl* volume, Agc* agc,
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AudioProcessing* audioproc);
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virtual ~AgcManager();
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// When enabled, registers external media processing callbacks with
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// VoiceEngine to hook into the capture stream. Disabling deregisters the
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// callbacks.
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virtual int Enable(bool enable);
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virtual bool enabled() const { return enabled_; }
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// Call when the capture device has changed. This will trigger a retrieval of
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// the initial capture volume on the next audio frame.
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virtual void CaptureDeviceChanged();
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// Call when the capture stream has been muted/unmuted. This causes the
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// manager to disregard all incoming audio; chances are good it's background
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// noise to which we'd like to avoid adapting.
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virtual void SetCaptureMuted(bool muted);
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virtual bool capture_muted() const { return direct_->capture_muted(); }
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protected:
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// Provide a default constructor for testing.
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AgcManager();
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private:
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int DeregisterCallbacks();
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int CheckVolumeAndReset();
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VoEExternalMedia* media_;
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scoped_ptr<VolumeCallbacks> volume_callbacks_;
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scoped_ptr<CriticalSectionWrapper> crit_;
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scoped_ptr<AudioProcessing> audioproc_;
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scoped_ptr<AgcManagerDirect> direct_;
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scoped_ptr<MediaCallback> media_callback_;
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scoped_ptr<PreprocCallback> preproc_callback_;
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bool enabled_;
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bool initialized_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_TEST_AGC_MANAGER_H_
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