webrtc_m130/src/modules/audio_processing/level_estimator_impl.h
andrew@webrtc.org 755b04a06e Add RMS computation for the RTP level indicator.
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
  AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.

TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/279003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-15 16:57:56 +00:00

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/*
* Copyright (c) 2011 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_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
#include "audio_processing.h"
#include "processing_component.h"
namespace webrtc {
class AudioProcessingImpl;
class AudioBuffer;
class LevelEstimatorImpl : public LevelEstimator,
public ProcessingComponent {
public:
explicit LevelEstimatorImpl(const AudioProcessingImpl* apm);
virtual ~LevelEstimatorImpl();
int ProcessStream(AudioBuffer* audio);
// LevelEstimator implementation.
virtual bool is_enabled() const;
// ProcessingComponent implementation.
virtual int get_version(char* version, int version_len_bytes) const;
private:
// LevelEstimator implementation.
virtual int Enable(bool enable);
virtual int RMS();
// ProcessingComponent implementation.
virtual void* CreateHandle() const;
virtual int InitializeHandle(void* handle) const;
virtual int ConfigureHandle(void* handle) const;
virtual int DestroyHandle(void* handle) const;
virtual int num_handles_required() const;
virtual int GetHandleError(void* handle) const;
const AudioProcessingImpl* apm_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_