21299d4e00
We want to remove energy_ entirely as we've seen that carrying around this potentially invalid value is dangerous. Results in the removal of AudioBuffer::is_muted(). This wasn't used in practice any longer, after the level calculation moved directly to channel.cc Instead, now use ProcessMuted() in channel.cc, to shortcut the level computation when the signal is muted. BUG=3315 TESTED=Muting the channel in voe_cmd_test results in rms=127. R=bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
58 lines
1.8 KiB
C++
58 lines
1.8 KiB
C++
/*
|
|
* Copyright (c) 2014 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_RMS_LEVEL_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
|
|
|
|
#include "webrtc/typedefs.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Computes the root mean square (RMS) level in dBFs (decibels from digital
|
|
// full-scale) of audio data. The computation follows RFC 6465:
|
|
// https://tools.ietf.org/html/rfc6465
|
|
// with the intent that it can provide the RTP audio level indication.
|
|
//
|
|
// The expected approach is to provide constant-sized chunks of audio to
|
|
// Process(). When enough chunks have been accumulated to form a packet, call
|
|
// RMS() to get the audio level indicator for the RTP header.
|
|
class RMSLevel {
|
|
public:
|
|
static const int kMinLevel = 127;
|
|
|
|
RMSLevel();
|
|
~RMSLevel();
|
|
|
|
// Can be called to reset internal states, but is not required during normal
|
|
// operation.
|
|
void Reset();
|
|
|
|
// Pass each chunk of audio to Process() to accumulate the level.
|
|
void Process(const int16_t* data, int length);
|
|
|
|
// If all samples with the given |length| have a magnitude of zero, this is
|
|
// a shortcut to avoid some computation.
|
|
void ProcessMuted(int length);
|
|
|
|
// Computes the RMS level over all data passed to Process() since the last
|
|
// call to RMS(). The returned value is positive but should be interpreted as
|
|
// negative as per the RFC. It is constrained to [0, 127].
|
|
int RMS();
|
|
|
|
private:
|
|
float sum_square_;
|
|
int sample_count_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
|
|
|