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개요
It pairs classic signal processing with a small recurrent neural network of about 88,000 weights, so it runs on an ordinary CPU core and became a common building block in streaming and voice-call software.
주요 시사점
- RNNoise is a BSD-licensed C library from Xiph.Org and Mozilla, released in 2017, that suppresses background noise in 48 kHz speech in real time.
- It predicts 22 band gains with a small gated recurrent network instead of estimating every frequency bin, which keeps computation low.
- The 2018 paper reports 87,503 weights and about 1.3 percent of one x86 core, with no GPU required.
- A pitch-tuned comb filter removes noise between voice harmonics that the coarse band gains cannot reach.
- OBS Studio added RNNoise as a noise suppression method in version 26.0 in September 2020.
심층 분석
RNNoise is a speech enhancement library: software that takes audio containing someone talking over background noise, such as a fan, traffic or keyboard clicks, and returns a cleaner version in which the voice is kept and the noise is reduced. It was written by Jean-Marc Valin and published in September 2017 as a Xiph.Org and Mozilla project with a browser demo and a technical paper. The paper, A Hybrid DSP/Deep Learning Approach to Real-Time Full-Band Speech Enhancement, appeared at the 2018 IEEE Workshop on Multimedia Signal Processing. The code is written in C and distributed under a BSD license.
The design goal was practical: run in real time on modest hardware at full-band 48 kHz audio, the sample rate used in high-quality voice calls, without a GPU. Many deep learning denoisers of the time estimated every frequency bin of a spectrogram, or even every audio sample, which takes large networks. RNNoise instead keeps the parts of a traditional noise suppressor that signal processing already does well, such as framing, spectral analysis and pitch tracking, and asks a neural network to do only the hardest part, which is estimating how much each part of the spectrum should be turned down.
The processing happens in short, overlapping frames. Audio is cut into 20 millisecond windows that advance every 10 milliseconds, and each window is converted to the frequency domain with a Fourier transform. Rather than handle hundreds of individual frequency bins, RNNoise groups them into 22 bands spaced on the Bark scale, a scale that roughly matches how human hearing resolves frequency, using the band layout of the Opus codec. For each frame it computes 42 input features: 22 Bark-frequency cepstral coefficients that summarize the spectral shape, the first and second time derivatives of the first six, six coefficients describing how strongly each band follows the voice's pitch, the pitch period itself and a measure of how stationary the spectrum is.
Those features go into a recurrent neural network, a network that keeps an internal state from frame to frame so it can track how sounds evolve over time. The network uses gated recurrent units (GRUs), a type of recurrent layer that learns what to remember and what to forget, combined with fully connected layers. The paper reports four hidden layers with 215 units in total, the largest having 96, and 87,503 weights. The network produces 22 gains between 0 and 1, one per band, plus a voice activity probability. Each gain multiplies the energy in its band: a value near 1 lets the band through, near 0 suppresses it. Because 22 band gains are too coarse to remove noise sitting between the harmonics of a voice, a final pitch filter, a comb filter tuned to the detected pitch, attenuates energy between harmonics. The result is converted back to a waveform by overlap-add.
Training uses synthetic mixtures. Clean speech and noise are rarely recorded together with a clean reference, so they are recorded separately and mixed at random levels with random filtering. The paper reports about 6 hours of speech and 4 hours of noise expanded this way into about 140 hours of noisy training audio. The target for each band is the square root of the ratio between clean speech energy and noisy energy, and the loss compares gains after a further square root, which penalizes mistakes on quiet bands more. Because the system does not normalize levels, the training data must include speech at every realistic volume.
The efficiency numbers explain its adoption. The paper reports that the network needs about 175,000 floating point operations per frame, or 17.5 million per second at 100 frames per second, and about 40 million per second for the whole algorithm including transforms and pitch search. That used about 1.3 percent of one core of an Intel Haswell laptop processor and about 14 percent of a Raspberry Pi 3 core. Weights constrained to plus or minus 0.5 during training can be stored as 8-bit values, giving a model of about 85 kB according to the demo page. In a PESQ evaluation, an objective speech quality score, the paper reports clearly better quality than the conventional suppressor in SpeexDSP, particularly for non-stationary noise such as babble, car and street noise.
Compared with the alternatives, RNNoise sits between old and new. Classical suppressors such as the Speex preprocessor estimate the noise spectrum statistically and subtract it; they are cheap but struggle with noise that changes quickly. Large neural denoisers trained for the Microsoft-run Deep Noise Suppression Challenge at INTERSPEECH 2020 can remove more noise but cost more compute; the challenge organizers ranked entries with crowdsourced listening tests under the ITU-T P.808 standard because objective metrics often disagreed with human judgments. Valin's own follow-up with co-authors, PercepNet, presented at INTERSPEECH 2020, kept the band-and-pitch design but reported higher quality while still using less than 5 percent of a CPU core.
The limitations follow from the design. A band gain cannot separate a voice from noise that occupies the same band at the same moment, so strong noise leaves a muffled or watery voice, a distortion often called musical noise. RNNoise is trained to separate speech from background sounds, not one talker from another, so a competing voice in the background is treated as speech. Its quality is bounded by the training mixtures, and a 2017-era model can misjudge noise types it never heard. The 10 millisecond look-ahead keeps latency low for calls but limits how much future context the network can use.
Using RNNoise in practice requires little. The repository builds with a standard configure and make process, downloads the model files during setup, and includes a demo that reads headerless 16-bit mono audio at 48 kHz. Developers who want a custom model can generate features from their own speech and noise files with the provided tools, train with the included Python script and compile the quantized weights back into the library. The README recommends at least 10,000 training sequences and preferably 200,000 or more.
04실제 사례
Denoise a recording with the RNNoise demo
Record about 30 seconds of speech near a running fan, or mix clean speech with a noise clip.
Convert the file to raw headerless 16-bit mono PCM at 48 kHz, for example with a command-line audio converter.
Build RNNoise from the official repository with autogen.sh, configure and make, which also downloads the model files.
Run examples/rnnoise_demo with the noisy file as input and a new file name as output.
Convert the output back to WAV and listen to the original and denoised versions back to back.
Open both files in a spectrogram viewer and compare the energy between speech harmonics and during pauses.
그것이 보여주는 것
The learner ends with a denoised file in which steady fan noise is strongly reduced, especially in pauses. The spectrogram should show darker gaps between voice harmonics, while very loud noise overlapping the voice may leave some muffling, illustrating the band-gain limitation.
실제 구현
OBS Studio 26.0, released in September 2020, added RNNoise as a method in its Noise Suppression filter, giving streamers a neural option alongside the older Speex-based suppressor.
GamingOnLinux reported that the Linux tools NoiseTorch and Cadmus use the same RNNoise technology to clean up microphone input system-wide for voice chat and calls.
The rnnoise_demo program in the official repository denoises a raw 16-bit, 48 kHz mono recording from the command line, which makes it easy to test on podcast or meeting audio.
Valin's Noise Donation project invited the public to record a minute of their own environment's noise, and the collected recordings were released as a 6.4 GB downloadable archive for training.
RNNoise 0.2, released in April 2024, shipped models retrained only on publicly available datasets and added AVX2 and SSE4.1 optimizations with run-time CPU detection.
RNNoise를 통한 음성 잡음 제거의 미래
RNNoise's 2024 release, version 0.2, changed training and engineering rather than the core idea: Phoronix reported in April 2024 that the distributed models were retrained using only publicly available datasets and that the code gained SSE4.1 and AVX2 optimizations with run-time CPU detection. The repository now also offers a sparser little model and a way to load weights from a file at run time, which makes swapping in custom models easier. The broader research direction, visible in PercepNet (INTERSPEECH 2020) and the Deep Noise Suppression Challenge series that began in 2020, is toward higher perceived quality at similar low compute, judged by human listening tests rather than objective scores alone, since the challenge organizers found the two often disagree. Open problems include suppressing competing talkers and reverberation without damaging the target voice, robustness to noise types absent from training data, and evaluation that reflects real calls rather than synthetic mixtures. Whether small hybrid designs like RNNoise or larger fully neural models running on dedicated hardware become the default for voice calls remains an open question.
실천에 옮기기
왜 중요한가요?
- 접근 및 도달. 전사, 내레이션, 음성 인터페이스를 통해 접근성을 향상시킵니다.
- 비용 및 예산. 미디어 팀은 더 적은 예산으로 세련된 오디오를 더 빠르게 출시할 수 있습니다.
- 속도와 규모. 고객 대면 시스템은 음성 상호 작용을 더 큰 규모로 처리할 수 있습니다.
조심하세요
- 동의가 없으면 음성 오용 및 명의 도용 위험이 높아집니다.
- 악센트, 방언 또는 시끄러운 환경에서는 정확도가 떨어질 수 있습니다.
- 합성 오디오는 명확한 라벨링이 없으면 실제 음성으로 오인될 수 있습니다.
다음 단계
- 음성 캡처, 복제 및 재사용에 대한 명시적인 동의를 얻습니다.
- 다양한 화자와 배경 조건에서 품질을 테스트합니다.
- 사람이 출력을 검토하거나 승인해야 하는 시기를 정의합니다.
- 합성 오디오에 라벨을 붙이고 책임을 묻기 위해 출처 기록을 보관하세요.
출처 및 추가 자료
- arXiv / IEEE MMSP 2018A Hybrid DSP/Deep Learning Approach to Real-Time Full-Band Speech Enhancement
- Jean-Marc Valin (Xiph.Org / Mozilla)RNNoise: Learning Noise Suppression
- Xiph.Org Foundation (GitHub)xiph/rnnoise: Recurrent neural network for audio noise reduction
- PhoronixRNNoise 0.2 Released With AVX2 Optimizations For Neural Network Noise Suppression
- GamingOnLinuxOBS Studio adds in better noise suppression thanks to RNNoise in the 26.0 release
- arXiv / INTERSPEECH 2020A Perceptually-Motivated Approach for Low-Complexity, Real-Time Enhancement of Fullband Speech
- arXiv / INTERSPEECH 2020The INTERSPEECH 2020 Deep Noise Suppression Challenge: Datasets, Subjective Testing Framework, and Challenge Results
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자주 묻는 질문
What is Speech Denoising with RNNoise?
RNNoise is an open-source noise suppression library, released by Jean-Marc Valin of Xiph.Org and Mozilla in 2017, that removes background noise from speech in real time. It pairs classic signal processing with a small recurrent neural network of about 88,000 weights, so it runs on an ordinary CPU core and became a common building block in streaming and voice-call software.
RNNoise는 각각의 짧은 오디오 프레임에 대해 주로 무엇을 예측합니까?
RNNoise는 모든 스펙트럼 빈에서 작업하는 대신 음성 중심 대역을 보존하면서 잡음 중심 대역을 감쇠시키는 대역별 게인을 추정합니다.
RNNoise의 핵심은 어떤 종류의 신경망인가요?
RNNoise는 Gated Recurrent Units로 구축된 소형 순환 신경망을 사용하여 작은 크기를 유지하면서 임시 메모리를 제공합니다.
RNNoise가 원시 스펙트럼 저장소 대신 지각 주파수 대역을 사용하는 이유는 무엇입니까?
~22 대역 이득만 예측하면 네트워크가 작고 빠르며 빈별 방법을 괴롭히는 음악 잡음 아티팩트에 덜 취약하게 유지됩니다.
RNNoise 모델의 크기는 대략 얼마나 됩니까?
RNNoise는 약 85,000개의 가중치를 갖고 있으며, 이는 단일 CPU 코어에서 실시간보다 빠르게 실행될 수 있을 만큼 작습니다.
RNNoise에서 피치 필터의 역할은 무엇인가요?
콤/피치 필터는 음성 음성의 고조파 구조를 활용하여 고조파 사이에 있는 잡음을 억제하고 대역 이득을 보완합니다.