753eada3aa
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
60 lines
1.9 KiB
C++
60 lines
1.9 KiB
C++
/*
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* Copyright (c) 2014 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_RMS_LEVEL_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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#include <cstddef>
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Computes the root mean square (RMS) level in dBFs (decibels from digital
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// full-scale) of audio data. The computation follows RFC 6465:
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// https://tools.ietf.org/html/rfc6465
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// with the intent that it can provide the RTP audio level indication.
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//
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// The expected approach is to provide constant-sized chunks of audio to
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// Process(). When enough chunks have been accumulated to form a packet, call
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// RMS() to get the audio level indicator for the RTP header.
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class RMSLevel {
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public:
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static const int kMinLevel = 127;
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RMSLevel();
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~RMSLevel();
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// Can be called to reset internal states, but is not required during normal
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// operation.
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void Reset();
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// Pass each chunk of audio to Process() to accumulate the level.
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void Process(const int16_t* data, size_t length);
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// If all samples with the given |length| have a magnitude of zero, this is
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// a shortcut to avoid some computation.
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void ProcessMuted(size_t length);
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// Computes the RMS level over all data passed to Process() since the last
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// call to RMS(). The returned value is positive but should be interpreted as
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// negative as per the RFC. It is constrained to [0, 127].
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int RMS();
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private:
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float sum_square_;
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size_t sample_count_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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