382c0c209d
Break out the computation to a separate class, and call directly into this from channel.cc rather than going through AudioProcessing. This circumvents AudioProcessing's sample rate limitations. We now compute the RMS over all samples rather than downmixing to a single channel. This makes the call point in channel.cc easier, is more "correct" and should have similar (negligible) complexity. This caused slight changes in the RMS output, so the ApmTest.Process reference has been updated. Snippet of the failing output: [ RUN ] ApmTest.Process Running test 4 of 12... Value of: rms_level Actual: 27 Expected: test->rms_level() Which is: 28 Running test 5 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 6 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 10 of 12... Value of: rms_level Actual: 27 Expected: test->rms_level() Which is: 28 Running test 11 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 Running test 12 of 12... Value of: rms_level Actual: 26 Expected: test->rms_level() Which is: 27 BUG=3290 TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc produce reasonable printed out results from RMS(). R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
54 lines
1.8 KiB
C++
54 lines
1.8 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_LEVEL_ESTIMATOR_IMPL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/audio_processing/processing_component.h"
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#include "webrtc/modules/audio_processing/rms_level.h"
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namespace webrtc {
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class AudioBuffer;
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class CriticalSectionWrapper;
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class LevelEstimatorImpl : public LevelEstimator,
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public ProcessingComponent {
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public:
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LevelEstimatorImpl(const AudioProcessing* apm,
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CriticalSectionWrapper* crit);
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virtual ~LevelEstimatorImpl();
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int ProcessStream(AudioBuffer* audio);
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// LevelEstimator implementation.
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virtual bool is_enabled() const OVERRIDE;
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private:
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// LevelEstimator implementation.
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virtual int Enable(bool enable) OVERRIDE;
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virtual int RMS() OVERRIDE;
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// ProcessingComponent implementation.
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virtual void* CreateHandle() const OVERRIDE;
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virtual int InitializeHandle(void* handle) const OVERRIDE;
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virtual int ConfigureHandle(void* handle) const OVERRIDE;
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virtual void DestroyHandle(void* handle) const OVERRIDE;
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virtual int num_handles_required() const OVERRIDE;
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virtual int GetHandleError(void* handle) const OVERRIDE;
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CriticalSectionWrapper* crit_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
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