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7f969c0eff
To quote rfc6464: The audio level for digital silence -- for a muted audio source, for example -- MUST be represented as 127 (-127 dBov), regardless of the dynamic range of the encoded audio format. The behavior in webrtc is correct that digital silence is represented with 127, but it is also possible to get a value of 127 for not quite digitally silent audio buffer (as in, not strictly 0s). Bug: webrtc:14029 Change-Id: I7ff8698a7e4d5c0960c667fd1cc961838e269456 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261244 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36793}
77 lines
2.5 KiB
C++
77 lines
2.5 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 MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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#define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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#include <stddef.h>
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "api/array_view.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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// Analyze(). When enough chunks have been accumulated to form a packet, call
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// Average() 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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struct Levels {
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int average;
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int peak;
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};
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enum : int { kMinLevelDb = 127, kInaudibleButNotMuted = 126 };
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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 Analyze() to accumulate the level.
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void Analyze(rtc::ArrayView<const int16_t> data);
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void Analyze(rtc::ArrayView<const float> data);
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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 AnalyzeMuted(size_t length);
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// Computes the RMS level over all data passed to Analyze() since the last
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// call to Average(). The returned value is positive but should be interpreted
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// as negative as per the RFC. It is constrained to [0, 127]. Resets the
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// internal state to start a new measurement period.
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int Average();
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// Like Average() above, but also returns the RMS peak value. Resets the
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// internal state to start a new measurement period.
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Levels AverageAndPeak();
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private:
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// Compares `block_size` with `block_size_`. If they are different, calls
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// Reset() and stores the new size.
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void CheckBlockSize(size_t block_size);
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float sum_square_;
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size_t sample_count_;
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float max_sum_square_;
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absl::optional<size_t> block_size_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
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