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Removing noise and graininess from videos is easier than ever. Development as dom ebook pdf reader. Noise and graininess usually occur when a video is shot with poor lighting, when the ISO of the camera was unusually high when the video was being recorded or when the camera used in recording the video does not have the right specs.Video and multimedia content editors like Premiere Pro have proprietary algorithms and filters for fast noise reduction. And when these built-in filters are not adequate or convenient, you can use plugins or standalone noise reduction software to improve the quality and clarity of your videos.In this post, we show you how to remove noise from videos using Premiere Pro, Premiere Pro Plugins and free noise reduction software.Part 1: How to Reduce Noise in Premiere ProUse media effectPremiere Pro has a noise reduction filter in the effects panel called MEDIAN. To apply this filter, follow the steps below:.Go to EFFECTS VIDEO EFFECTS NOISE & GRAIN MEDIAN. Drag MEDIAN onto the timeline for your noisy video. Note that this will not apply noise reduction to your video immediately.Check the EFFECTS CONTROL window to see a dropdown menu where you can adjust the MEDIAN effect for best results. Adjust the RADIUS parameter under MEDIAN to immediately preview the effect applied.Alternatively, you could draw masks around noisy portions of your video to preview the denoising effect on those areas only.Be sure to feather the edges of your masks to avoid hard edges around areas where the MEDIAN effect is applied.Part 2: Another Way to Reduce Noise by After EffectsNoise reduction in Premiere Pro does not always yield the best results.

• Denoiser Premiere Pro

Even when the MEDIAN effect is applied with masks, you may find that the denoising effect is applied to parts of the video where noise reduction is not needed. This is because while the masks are static, the contents of the video are not.In this case, you could open your video in ADOBE AFTER EFFECTS using the following steps:.Right-click the timeline for the noisy video is PREMIERE PRO and click REPLACE WITH AFTER EFFECTS COMPOSITION.When AFTER EFFECTS opens with your video loaded, go to EFFECTS & PRESETS, select REMOVE GRAIN and drag it onto the timeline for your video.Check the EFFECTS CONTROL panel to see settings for the REMOVE GRAIN effect. Adjust preview region, noise reduction setting and temporal filtering for best results.Part 3: Noise reduction plugins for Premiere Pro1.Neat Video is arguably the most popular denoising plugin for Premiere Pro. It is both powerful and fast and is used by many professionals. It works with other popular video editing software like Final Cut and Avid and has versions compatible with PCs and MacOS.

Neat Video also has a free version and a premium version.FeaturesNeat Video works by creating a noise profile. It also allows for users to select areas of the footage from which a noise profile can be created or to load their own noise profile footage.Once installed in Premier Pro, NEAT VIDEO will be listed in the Video Effects panel where it can be selected when needed.

Its default settings are powerful and can be further customized through the setup menu visible in the EFFECTS CONTROL panel.2.Magic Bullet Denoiser III is another powerful denoising plugin that’s compatible with PC and MacOS. It has a fully functional trial version and a paid version. However, its paid version is nearly double the price of Neat Video. Professionals may find it more cost effective to buy the complete suite of Magic Bullet plugins.FeaturesLike Neat Video, Magic Bullet Denoiser III works by creating a noise profile. Unlike Neat Video however, Magic Bullet Denoiser III samples every frame of the video to create its noise profile.

Its settings are also very simple and intuitive, with sliders for reduce noise, smooth colors and preserve details.Part 4: 3 Free Video Noise Reduction Software1.Aiseesoft Video Enhancer is a standalone application with PC and MacOS compatible versions that has good noise reduction and video conversion features. It is very light, feature specific, and very easy to use. It also has a free version and an affordable paid version.FeaturesNoise reduction is one of three (3) enhancement features available in Aiseesoft Video Enhancer. To remove noise from a video, simply open it in Aiseesoft, check the NOISE REDUCTION box under the ENHANCE menu and click APPLY. This application also lets users export their enhanced videos in many formats and with no compression.2.Cinemartin Denoiser is a professional video denoising application that offers powerful features for a small fee. A free version with few locked features is available for a limited time.

The full version delivers excellent noise removal with options to export to multiple video formats.FeaturesThis application has simple noise removal settings as well as built-in presets for fast and easy noise removal. It works with 2 proprietary noise removal mechanisms to remove the most common kinds of noise generated by DSLR cameras.3.Tipard Video Enhancer looks and feels a lot like AISEESOFT VIDEO ENHANCER.

Their interfaces, colors and menus are very identical.Noise removal is an enhancement feature in the Tipard Video Enhancer software and the Tipard Video Converter Ultimate. So, you can download any of these two Tipard applications to enjoy noise removal features. They both have free and affordable paid versions.FeaturesThe Tipard Video Enhancer enables users enhance videos through noise reduction and increased video resolution. Unlike Aiseesoft Video Enhancer, Tipard also lets users enhance their videos by reducing the perceived shaking or unsteadiness of the video footage.ConclusionAs with audio noise removal, video noise removal is rarely perfect.

Users need to ensure that desirable details of a video footage are not lost in the denoising process.

. Mode: Choose Broadband to uniformly compress all frequencies or Multiband to only compress the sibilance range.

Multiband is best for most audio content but slightly increases processing time. Threshold: Sets the amplitude above which compression occurs. Center Frequency: Specifies the frequency at which sibilance is most intense. To verify, adjust this setting while playing audio. Bandwidth: Determines the frequency range that triggers the compressor. Output Sibilance Only: Lets you hear detected sibilance. Start playback, and fine-tune settings above.

Gain Reduction: Shows the compression level of the processed frequencies. Auto Gate: Removes noise below a certain amplitude threshold. The LED meter is green when audio passes through the gate. The meter turns red when there is no audio passing, and yellow during the attack, release, and hold times. Compressor: Reduces the dynamic range of the audio signal by attenuating audio that exceeds a specific threshold.

The Ratio parameter can be used control the change in dynamic range while Attack and Release parameter changes the temporal behavior. Use the Gain parameter to increase the audio level after compressing the signal. The Gain Reduction meter shows how much the audio level is reduced. Expander: Increases the dynamic range of the audio signal by attenuating audio below the specified threshold. The ratio parameter can be used to control the change in dynamic range. The Gain Reduction meter shows the level of reduction in audio level.

Limiter: Attenuate audio that exceeds a specified threshold. The meter LED turns on when the signal is limited. Dynamics. Graph: Depicts input level along the horizontal ruler (x‑axis) and the new output level along the vertical ruler (y‑axis). The default graph, with a straight line from the lower left to the upper right, depicts a signal that has been left untouched.

Every input level has the same output level. Adjusting the graph changes the relationship between input and output levels, altering dynamic range. For example, if a desirable sonic element occurs around ‑20 dB, you can boost the input signal at that level, but leave everything else unchanged.

You can also draw an inverse line (from the upper left to the lower right) that boosts quiet sounds and suppress loud ones. Add point: Adds control point in graph using numerical input and output levels you specify. This method is more precise than clicking the graph to add points.

Delete point: Removes selected point from the graph. Invert: Flips the graph, converting compression into expansion, or the other way around. Reset: Resets the graph to its default state. Spline Curves: Creates smoother, curved transitions between control points, rather than more abrupt, linear transitions. For more information, see. Make-up Gain: Boosts the processed signal.

SettingsGeneral: Provides overall settings. Look Ahead Time: Addresses transient spikes that can occur at the onset of loud signals that extend beyond the compressor’s Attack Time settings.

Extending Look-Ahead Time causes compression to attack before the audio gets loud, ensuring that amplitude never exceeds a certain level. Conversely, reducing Look-Ahead Time is desirable to enhance the impact of percussive music like drum hits. Noise Gating: Completely silences signals that are expanded below a 50-to-1 ratio.Level Detector: Determines the original input amplitude.

Input Gain: Applies gain to the signal before it enters the Level Detector. Attack Time: Determines how many milliseconds it takes for the input signal to register a changed amplitude level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the input registers an amplitude change. This selection avoids erroneous amplitude readings due to temporary changes. Release Time: Determines how many milliseconds the current amplitude level is maintained before another amplitude change can register.

Peak mode: Determines levels based on amplitude peaks. This mode is a bit more difficult to use than RMS, because peaks aren’t precisely reflected in the Dynamics graph. However, it can be helpful when audio has loud transient peaks you want to subdue. RMS mode: Determines levels based on the root-mean-square formula, an averaging method that more closely matches the way people perceive volume.

This mode precisely reflects amplitudes in the Dynamics graph. For example, a limiter (flat horizontal line) at ‑10 dB reflects an average RMS amplitude of ‑10 dB.Gain Processor: Amplifies or attenuates the signal depending on the amplitude detected. Output Gain: Applies gain to the output signal after all dynamics processing.

Attack Time: Determines how many milliseconds it takes for the output signal to reach the specified level. For example, if audio suddenly drops 30 dB, the specified attack time passes before the output level changes.

Release Time: Determines how many milliseconds the current output level is maintained. Link Channels: Processes all channels equally, preserving the stereo or surround balance. For example, a compressed drum beat on the left channel reduces the right channel level by an equal amount.Band Limiting: Restricts dynamics processing to a specific frequency range.

Low Cutoff: Is the lowest frequency that dynamics processing affects. High Cutoff: Is the highest frequency that dynamics processing affects. Maximum Amplitude: Sets the maximum sample amplitude allowed. Input Boost: Preamplifies audio before you limit it, making a selection louder without clipping it. As you increase this level, compression increases. Try extreme settings to achieve the loud, high‑impact audio heard in contemporary pop music. Look Ahead Time: Sets the amount of time (in milliseconds) for the audio to be attenuated before the loudest peak is hit.

Release Time: Sets the time (in milliseconds) for the attenuation to rebound back 12 dB. In general, a setting of around 100 (the default) works well and preserves low bass frequencies. Link Channels: Links the loudness of all channels together, preserving the stereo or surround balance. Crossover: Sets the crossover frequencies, which determine the width of each band. Either enter specific Low, Midrange, and High frequencies, or drag the crossover markers above the graph. Solo Buttons: Let you hear specific frequency bands. Enable one Solo button at a time to hear bands in isolation, or enable multiple buttons to hear two or more bands together.

Bypass Buttons: Bypass individual bands so they pass through without processing. Thresh: Set the input level at which compression begins. Possible values range from ‑60 dB to 0 dB. The best setting depends on audio content and musical style.

To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level. Gain: Boosts or cuts amplitude after compression.

Possible values range from ‑18 dB to +18 dB, where 0 is unity gain. Ratio: Sets a compression ratio between 1‑to‑1 and 30‑to‑1. For example, a setting of 3.0 outputs 1 dB for every 3-dB increase above the compression threshold.

Typical settings range from 2.0 to 5.0; higher settings produce the compressed sound often heard in pop music. Attack: Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from 0 milliseconds to 500 milliseconds. The default, 10 milliseconds, works well for a wide range of audio.

Faster settings work better for audio with fast transients, but such settings sound unnatural for less percussive audio. Release: Determines how quickly compression stops after audio drops below the threshold. Possible values range from 0 milliseconds to 5000 milliseconds. The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio. Output Gain: Boosts or cuts overall output level after compression.

Possible values range from ‑18 dB to +18 dB, where 0 is unity gain. To reset peak and clip indicators, double‑click the meters. Gain: Boosts or cuts amplitude after compression. Possible values range from ‑18 dB to +18 dB, where 0 is unity gain. Limiter: Applies limiting after Output Gain, at the end of the signal path, optimizing overall levels. Specify Threshold, Attack, and Release settings that are less aggressive than similar band‑specific settings. Then specify a Margin setting to determine the absolute ceiling relative to 0 dBFS.Options.

Spectrum On Input: Displays the frequency spectrum of the input signal, rather than the output signal, in the multiband graph. To quickly see the amount of compression applied to each band, toggle this option on and off. Brickwall Limiter: Applies immediate, hard limiting at the current Margin setting. (Deselect this option to apply slower soft limiting, which sounds less compressed but can exceed the Margin setting.) Note: The maximum Attack time for brickwall limiting is 5 ms. Link Band Controls: Lets you globally adjust the compression settings for all bands, while retaining relative differences between bands. Threshold: Sets the input level at which compression begins.

The best setting depends on audio content and style. To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level. Ratio: Sets a compression ratio between 1‑to‑1 and 30‑to‑1.

For example, a setting of three outputs 1 dB for every 3-dB increase above the threshold. Typical settings range from 2 to 5; higher settings produce the compressed sound often heard in pop music. Attack: Determines how quickly compression starts after audio exceeds the Threshold setting. The default, 10 milliseconds, works well for a wide range of source material. Use faster settings only for audio with quick transients, such as percussion recordings. Release: Determines how quickly compression stops when audio drops below the Threshold setting.

The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio. Output Gain: Boosts or cuts amplitude after compression.

Possible values range from ‑30 dB to +30 dB, where 0 is unity gain. Threshold: Sets the input level at which compression begins. Possible values range from ‑60 dB to 0 dB. The best setting depends on audio content and musical style. To compress only extreme peaks and retain more dynamic range, try thresholds around 5 dB below the peak input level. To highly compress audio and greatly reduce dynamic range, try settings around 15 dB below the peak input level.

Output Gain: Boosts or cuts overall output level after compression. Possible values range from ‑18 dB to +18 dB, where 0 is unity gain. To reset peak and clip indicators, double‑click the meters.

Ratio: Sets a compression ratio between 1‑to‑1 and 30‑to‑1. For example, a setting of 3.0 outputs 1 dB for every 3-dB increase above the compression threshold. Typical settings range from 2.0 to 5.0; higher settings produce the compressed sound often heard in pop music. Attack: Determines how quickly compression is applied when audio exceeds the threshold. Possible values range from 0 milliseconds to 500 milliseconds. The default, 10 milliseconds, works well for a wide range of audio.

Faster settings work better for audio with fast transients, but such settings sound unnatural for less percussive audio. Release: Determines how quickly compression stops after audio drops below the threshold. Possible values range from 0 milliseconds to 5000 milliseconds. The default, 100 milliseconds, works well for a wide range of audio. Try faster settings for audio with fast transients, and slower settings for less percussive audio.

Mode: Specifies the type of hardware emulation, determining equalization and distortion characteristics. Tape and Tube reflect the sonic character of vintage delay units, while Analog reflects later electronic delay lines. Dry Out: Determines the level of original, unprocessed audio.

Wet Out: Determines the level of delayed, processed audio. Delay: Specifies the delay length in milliseconds. Feedback: Creates repeating echoes by resending delayed audio through the delay line. For example, a setting of 20% sends delayed audio at one-fifth of its original volume, creating echoes that gently fade away. A setting of 200% sends delayed audio at double its original volume, creating echoes that quickly grow in intensity. Trash: Increases distortion and boosts low frequencies, adding warmth.

Spread: Determines the stereo width of the delayed signal. Scale: Determines how frequencies are arranged along the horizontal x‑axis:. For finer control over low frequencies, select Logarithmic.

A logarithmic scale more closely resembles how people hear sound. For detailed, high‑frequency work with evenly spaced intervals in frequency, select Linear. Spline Curves: Creates smoother, curved transitions between control points, rather than more abrupt, linear transitions. For more information, see. Reset: Reverts the graph to the default state, removing filtering. Advanced: Click the triangle to access these settings:.

FFT Size: Specifies the Fast Fourier Transform size, determining the tradeoff between frequency and time accuracy. For steep, precise frequency filters, choose higher values.

For reduced transient artifacts in percussive audio, choose lower values. Values from 1024 through 8192 work well for most material. Window: Determines the Fast Fourier Transform shape, with each option resulting in a different frequency response curve. These functions are listed in order from narrowest to widest. Narrower functions include fewer surrounding, or sidelobe, frequencies but less precisely reflect center frequencies.

Wider functions include more surrounding frequencies but more precisely reflect center frequencies. The Hamming and Blackman options provide excellent overall results. Gain sliders: Sets the exact boost or attenuation (measured in decibels) for the chosen band. Range: Defines the range of the slider controls. Enter any value between 1.5 dB and 120 dB.

(By comparison, standard hardware equalizers have a range of about 12 dB to 30 dB.). Accuracy: Sets the accuracy level for equalization. Higher accuracy levels give better frequency response in the lower ranges, but they require more processing time. If you equalize only higher frequencies, you can use lower accuracy levels. Master Gain: Compensates for an overall volume level that is too soft or too loud after the EQ settings are adjusted.

The default value of 0 dB represents no master gain adjustment. Frequency: Specifies the center frequency for each notch. Gain: Specifies the amplitude for each notch. Enable: Enable the button to pass without processing. Notch Width: Determines frequency range for all notches. The three options range from Narrow to Super Narrow. Narrow is for a second order filter, which removes some adjacent frequencies.

Super Narrow is for a sixth order filter, which is specific. Ultra Quiet: Virtually eliminates noise and artifacts, but requires more processing. This option is audible only on high-end headphones and monitoring systems.

Fix Gain to: Determines if notches have equal or individual gain levels. Frequency: Sets the center frequency for bands 1-5, and the corner frequencies for the band-pass and shelving filters. Gain: Sets the boost or attenuation for frequency bands, and the per-octave slope of the band-pass filters. Q/ Width: Controls the width of the affected frequency band. Low Q values affect a larger range of frequencies.

High Q values (close to 100) affect a narrow band and are ideal for notch filters removing particular frequencies, like 60-Hz hum. Band: Enables up to five intermediate bands, and high-pass, low-pass, and shelving filters, giving you fine control over the equalization curve. To activate the corresponding settings, click the band button.

The low and high shelving filters provide slope buttons that adjust the low and high shelves by 12 dB per octave, rather than the default 6 dB per octave. Constant: Describes a frequency band’s width as either a Q value (which is a ratio of width to center frequency) or an absolute width value in Hz.

Constant Q is the most common setting. Ultra-Quiet: Virtually eliminates noise and artifacts, but requires more processing. This option is audible only on high-end headphones and monitoring systems. Range: Sets the graph to a 30-dB range for more precise adjustments, or a 96-dB range for more extreme adjustments. Types: Specifies the type of scientific filter. The available options are as follows. Bessel: Provides accurate phase response with no ringing or overshoot.

However, the pass band slopes at its edges, where rejection of the stop band is the poorest of all filter types. These qualities make Bessel a good choice for percussive, pulse-like signals. For other filtering tasks, use Butterworth.

Butterworth: Provides a flat pass band with minimal phase shift, ringing, and overshoot. This filter type also rejects the stop band much better than Bessel and only slightly worse than Chebychev 1 or 2. These overall qualities make Butterworth the best choice for most filtering tasks. Chebychev: Provides the best stop band rejection but the worst phase response, ringing, and overshoot in the pass band. Use this filter type only if rejecting the stop band is more important than maintaining an accurate pass band.

Elliptical: Provides a sharp cut-off and narrow transition width. It can also notch out frequencies, unlike the Butterworth and Chebychev filters. It can introduce ripples in both the stop band and the pass band. Modes: Specify a mode for the filter.

The available options are as follows. LowPass: Passes the low frequencies and removes high frequencies. Specify the cutoff point at which the frequencies are removed. HighPass: Passes high frequencies and removes low frequencies. Specify the cutoff point at which the frequencies are removed. BandPass: Preserves a band, a range of frequencies, while removing all other frequencies.

Specify two cutoff points to define the edges of the band. BandStop: Rejects any frequencies within the specified range. Also known as a notch filter, Band Stop is the opposite of Band Pass. Specify two cutoff points to define the edges of the band. Master Gain: Compensates for an overall volume level that might be too loud or too soft after you adjust the filter settings.

Cutoff: Defines the frequency that serves as a border between passed and removed frequencies. At this point the filter switches from passing to attenuating, or conversely. In filters requiring a range (Band Pass and Band Stop), Cutoff defines the low frequency border, while High Cutoff defines the high frequency border. High Cutoff: Defines the high frequency border in filters that require a range (Band Pass and Band Stop). Order: Determines the filter’s precision. The higher the order, the more precise the filter (with steeper slopes at the cutoff points, and so on).

However, high orders can also have high levels of phase distortion. Transition Bandwidth: (Butterworth and Chebychev only) Sets the width of the transition band. (Lower values have steeper slopes.) If you specify a transition bandwidth, the Order setting is filled in automatically, and conversely. Flanging is an audio effect caused by mixing a varying, short delay in roughly equal proportion to the original signal. It was originally achieved by sending an identical audio signal to two reel‑to‑reel tape recorders, and then pressing the flange of one reel to slow it down.

Combining the two resulting recordings produced a phase‑shifted, time‑delay effect, characteristic of psychedelic music of the 1960s and 1970s. The Flanger effect lets you create a similar result by slightly delaying and phasing a signal at specific or random intervals.

Initial Delay Time: Sets the point in milliseconds at which flanging starts behind the original signal. The flanging effect occurs by cycling over time from an initial delay setting to a second (or final) delay setting. Final Delay Time: Sets the point in milliseconds at which flanging ends behind the original signal. Stereo Phasing: Sets the left and right delays at separate values, measured in degrees. For example, 180° sets the initial delay of the right channel to occur at the same time as the final delay of the left channel. You can set this option to reverse the initial/final delay settings for the left and right channels, creating a circular, psychedelic effect.

Feedback: Determines the percentage of the flanged signal that is fed back into the flanger. With no feedback, the effect uses only the original signal. With feedback added, the effect uses a percentage of the affected signal from before the current point of playback. Modulation Rate: Determines how quickly the delay cycles from the initial to final delay times, measured either in cycles per second (Hz) or beats per minute (beats).

Small setting adjustments produce widely varying effects. Mode: Provides three ways of flanging:. Inverted: Inverts the delayed signal, canceling out audio periodically instead of reinforcing the signal.

If the Original ‑ Expanded mix settings are set at 50/50, the waves cancel out to silence whenever the delay is at zero. Special Effects: Mixes the normal and inverted flanging effects.

The delayed signal is added to the effect while the leading signal is subtracted. Sinusoidal: Makes the transition from initial delay to final delay and back follow a sine curve. Otherwise, the transition is linear, and the delays from the initial setting to the final setting are at a constant rate. If Sinusoidal is selected, the signal is at the initial and final delays more often than it is between delays. Mix: Adjusts the mix of original (Dry) and flanged (Wet) signal. You need some of both signals to achieve the characteristic cancellation and reinforcement that occurs during flanging.

With Original at 100%, no flanging occurs at all. With Delayed at 100%, the result is a wavering sound, like a bad tape player. Stages: Specifies the number of phase-shifting filters.

A higher setting produces denser phasing effects. Intensity: Determines the amount of phase‑shifting applied to the signal. Depth: Determines how far the filters travel below the upper frequency. Larger settings produce a wider tremolo effect; 100% sweeps from the upper frequency to zero Hz. Mod Rate: Modulation rate controls how fast the filters travel to and from the upper frequency. Specify a value in Hz (cycles per second).

Phase Diff: Determines the phase difference between stereo channels. Positive values start phase shifts in the left channel, negative values in the right. The maximum values of +180° and -180° produce a complete difference and are sonically identical. Upper Freq: Sets the upper-most frequency from which the filters sweep. To produce the most dramatic results, select a frequency near the middle of the selected audio’s range. Feedback: Feeds a percentage of the phaser output back to the input, intensifying the effect. Negative values invert phase before feeding audio back.

Mix: Controls the ratio of original to processed audio. Output Gain: Adjusts the output level after processing. Frequency: Sets the root frequency of the hum. If you’re unsure of the precise frequency, drag this setting back and forth while previewing audio.

Q: Sets the width of the root frequency and harmonics above. Higher values affect a narrower range of frequencies, and lower values affect a wider range. Gain: Determines the amount of hum attenuation. Number of Harmonics: Specifies how many harmonic frequencies to affect.

Harmonic Slope: Changes the attenuation ratio for harmonic frequencies. Output Hum Only: Lets you preview removed hum to determine if it contains any desirable audio. Impulse: Specifies a file that simulates an acoustic space. Click Load to add a custom impulse file in WAV or AIFF format.

Mix: Controls the ratio of original to reverberant sound. Room Size: Specifies a percentage of the full room defined by the impulse file. The larger the percentage, the longer the reverb. Damping LF: Reduces low-frequency, bass-heavy components in reverb, avoiding muddiness and producing a clearer, more articulate sound. Damping HF: Reduces high-frequency, transient components in reverb, avoiding harshness and producing a warmer, lusher sound. Pre-Delay: Determines how many milliseconds the reverb takes to build to maximum amplitude.

To produce the most natural sound, specify a short pre-delay of 0–10 milliseconds. To produce interesting special effects, specify a long pre-delay of 50 milliseconds or more. Width: Controls the stereo spread.

A setting of 0 produces a mono reverb signal. Gain: Boosts or attenuates amplitude after processing. Characteristics. Room Size: Sets the room size. Decay: Adjusts the amount of reverberation decay in milliseconds. Early Refections: Controls the percentage of echoes that first reach the ear, giving a sense of the overall room size.

Too high a value can result in an artificial sound, while too low a value can lose the audio cues for the room’s size. Half the volume of the original signal is a good starting point.

Width: Controls the spread across the stereo channels. 0% produces a mono reverb signal; 100% produces maximum stereo separation. High Frequency Cut: Specifies the highest frequency at which reverb can occur.

Low Frequency Cut: Specifies the lowest frequency at which reverb can occur. Damping: Adjusts the amount of attenuation applied to the high frequencies of the reverb signal over time. Higher percentages create more damping for a warmer reverb tone.

Diffusion: Simulates the absorption of the reverberated signal as it is reflected off surfaces, such as carpeting and drapes. Lower settings create more echoes, while higher settings produce a smoother reverberation with fewer echoes.Output Level. Dry: Sets the percentage of source audio to output with the effect. Wet: Sets the percentage of reverb to output.

Input Center: Determines the percentage of the center channel included in the processed signal. Input LFE: Determines the percentage of the Low Frequency Enhancement channel used to excite reverb for other channels. The LFE signal itself is not reverberated. Reverb Settings.

Impulse: Specifies a file that simulates an acoustic space. Click Load to add a custom, 6-channel impulse file in WAV or AIFF format. Room Size: Specifies a percentage of the full room defined by the impulse file. The larger the percentage, the longer the reverb. Damping LF: Reduces low-frequency, bass-heavy components in reverb, avoiding muddiness and producing a clearer, more articulate sound. Damping HF: Reduces high-frequency, transient components in reverb, avoiding harshness and producing a warmer, lusher sound. Pre-Delay: Determines how many milliseconds the reverb takes to build to maximum amplitude.

To produce the most natural sound, specify a short pre-delay of 0–10 milliseconds. To produce interesting special effects, specify a long pre-delay of 50 milliseconds or more. Front Width: Controls the stereo spread across the front three channels.A width setting of 0 produces a mono reverb signal. Surround Width: Controls the stereo spread across the rear surround channels (Ls and Rs). Output.

C Wet Level: Controls the amount of reverb added to the Center channel. (Because this channel usually contains dialog, reverb should typically be lower.). L/R Bal: Controls left-right balance for front and rear speakers. 100 outputs reverb to only the left, -100 to only the right. F/B Bal: Controls front-back balance for left and right speakers. 100 outputs reverb to only the front, -100 to only the back. Mix: Controls the ratio of original to reverberant sound.

A setting of 100 outputs only reverb. Gain: Boosts or attenuates amplitude after processing. Positive and Negative graphs: Specify separate distortion curves for positive and negative sample values. The horizontal ruler (x‑axis) indicates input level in decibels; the vertical ruler (y‑axis) indicates output level. The default diagonal line depicts an undistorted signal, with a one‑to‑one relationship between input and output values. Click-and-drag to create and adjust points on the graphs.

Drag points off a graph to remove them. Reset: Returns a graph to its default, undistorted state. Curve Smoothening: Creates curved transitions between control points, sometimes producing a more natural distortion than the default linear transitions. Time Smoothing: Determines how quickly distortion reacts to changes in input levels.

Denoiser Premiere Pro

Level measurements are based on low-frequency content, creating softer, more musical distortion. dB Range: Changes the amplitude range of the graphs, limiting distortion to that range. Linear Scale: Changes the amplitude scales of the graphs from logarithmic decibels to normalized values. Compressor: Reduces dynamic range to maintain consistent amplitude and help guitar tracks stand out in a mix. Filter: Choose an option from this menu, and then set options below:. Filter: Simulates guitar filters ranging from resonators to talk boxes. Type: Determines which frequencies are filtered.

Specify Lowpass to filter high frequencies, Highpass to filter low frequencies, or Bandpass to filter frequencies above and below a center frequency. Frequency: Determines the cutoff frequency for Lowpass and Highpass filtering, or the center frequency for Bandpass filtering. Resonance: Feeds back frequencies near the cutoff frequency, adding crispness with low settings and whistling harmonics with high settings. Distort: Adds a sonic edge often heard in guitar solos. To change the distortion character, choose an option from the Type menu. Amplifier: Simulates various amplifier and speaker combinations that guitarists use to create unique tones. Mix: Controls the ratio of original to processed audio.

Equalizer: Adjusts the overall tonal balance. Graph: Shows frequency along the horizontal ruler (x‑axis) and amplitude along the vertical ruler (y‑axis), with the curve representing the amplitude change at specific frequencies. Frequencies in the graph range from lowest to highest in a logarithmic fashion (evenly spaced by octaves).

Low Shelf Enable: Activate shelving filters at the low end of the frequency spectrum. Peaking Enable: Activates a peaking filter in the center of the frequency spectrum. High Shelf Enable: Activate shelving filters at the high end of the frequency spectrum.

Hz: Indicates the center frequency of each frequency band. dB: Indicates the level of each frequency band. Q: Controls the width of the affected frequency band. Low Q values (up to 3) affect a larger range of frequencies and are best for overall audio enhancement. High Q values (6–12) affect a narrow band and are ideal for removing a particular, problematic frequency, like 60-Hz hum. Reverb: Adds ambience. Drag the Amount slider to change the ratio of original to reverberant sound.

Exciter: Exaggerates high-frequency harmonics, adding crispness and clarity. Retro: Adjusts light distortion.

Tape: Adjusts bright tone. Tube: Adjusts quick, dynamic response. Amount: Adjust the level of processing. Widener: Adjusts the stereo image (disabled for mono audio). Drag the Width slider to the left to narrow the image and increase central focus.

Drag the slider to the right to expand the image and enhance spatial placement of individual sounds. Loudness Maximiser: Applies a limiter that reduces dynamic range, boosting perceived levels.

A setting of 0% reflects original levels; 100% applies maximum limiting. Output Gain: Determines output levels after processing. For example, to compensate for EQ adjustments that reduce overall level, boost the output gain. Pitch Transpose: Contains options that adjust pitch. Semi-tones: Transposes pitch in semi-tone increments, which equalmusical half-notes (for example, the note C# is one semi-tone higher than C). A setting of 0 reflects the original pitch; +12 semi-tones are an octave higher; -12 semi-tones are an octave lower. Cents: Adjusts pitch in fractions of semi-tones.

Possible values range from -100 (one semi-tone lower) to +100 (one semi-tone higher). Ratio: Determines the relationship between shifted and original frequency.

Possible values range from 0.5 (an octave lower) to 2.0 (an octave higher). Precision: Determines sound quality. Low Precison: Use the Low setting for 8 bit or low-quality audio. Medium Precison: Use the medium setting for medium quality audio. High Precison: High setting taking longest to process. Use the High setting for professionally recorded audio. Pitch Settings: Control how audio is processed.

Splicing Frequency: Determines the size of each chunk of audio data. (The Pitch Shifter effect divides audio into small chunks for processing.) The higher the value, the more precise the placement of stretched audio over time. However, artifacts become more noticeable as values go up. At higher Precision settings, a lower Splicing Frequency may add stutter or echo. If the frequency is too high, sound becomes tinny and voices have a tunnel-like quality. Overlapping: Determines how much each chunk of audio data overlaps with the previous and next ones. If stretching produces a chorus effect, lower the Overlapping percentage.

If doing so produces a choppy sound, adjust the percentage to strike a balance between choppiness and chorusing. Values range from 0% to 50%. Use appropriate default settings: Applies good default values for Splicing Frequency and Overlapping. Reduce Noise By: Determines the level of noise reduction.

Values between 6 dB and 30 dB work well. To reduce bubbly background effects, enter lower values.

Noisiness: Indicates the percentage of original audio that contains noise. Fine Tune Noise Floor: Manually adjusts the noise floor above or below the automatically calculated floor. Signal Threshold: Manually adjusts the threshold of desirable audio above or below the automatically calculated threshold. Spectral Decay Rate: Determines how quickly noise processing drops by 60 dB. Fine‑tuning this setting allows greater noise reduction with fewer artifacts. Values that are too short, create bubbly sounds.

Values that are too long, create a reverb effect. Broadband Preservation: Retains desirable audio in specified frequency bands between found artifacts. A setting of 100 Hz, for example, ensures that no audio is removed 100 Hz above or below found artifacts.

Lower settings remove more noise but may introduce audible processing. FFT Size: Determines how many individual frequency bands are analyzed. To increase frequency resolution, choose a high setting. To increase time resolution, choose a low setting. High settings work well for artifacts of long duration (like squeaks or power-line hum), while low settings better address transient artifacts (like clicks and pops). LFO (Low Frequency Oscillator) Type: Specifies wave type of Low Frequency Oscillator: Sin(e), Rect(angle), or Tri(angle).

Rate: Determines the maximum rate at which amplitude changes occur. With low values, the resulting voice slowly gets louder and quieter, like singers that cannot keep their breath steady. With high settings, the result can be jittery and unnatural. Depth: Determines the maximum variation in amplitude that occurs. For example, you can alter the amplitude of a chorused voice so that it is 5 dB louder or quieter than the original. At low settings (less than 1 dB), the depth could be unnoticeable unless the Modulation Rate is set high.

At high settings, however, the sound could cut in and out, creating an objectionable warble. Natural vibratos occur around 2 dB to 5 dB. This setting is a maximum only; the vibrato volume would not always go as low as the setting indicates. This limitation is intentional, as it creates a more natural sound. Delay: Specifies the maximum amount of delay allowed. An important component of chorusing is the introduction of short delays (often in the 15-35 millisecond range) that vary in duration over time. If the setting is small, all the voices start merging into the original, and an unnatural flanging effect could occur.

If the setting is high, a warbled effect could occur, like a cassette deck eating a tape. Feedback: Adds a percentage of processed voices back into the effect input.

Feedback can give a waveform an extra echo or reverb effect. A little feedback (less than 10%) can provide extra richness, depending on the delay and vibrato settings. Higher settings produce more traditional feedback, a loud ringing that can get loud enough to clip the signal. Sometimes this clipping is a desired effect, as in the Flying Saucers preset, which generates the warbled sounds of UFOs whizzing around your head. Mix: Determines the ratio of Dry and Effects signal. A setting of 100% corresponds to a ratio of 1/1 while a setting of 0 defeats the effect signal.

Threshold: Determines the threshold for the detection and thus determines how much of the signal gets affected. This control ranges from 0% to 100%. DePlop: Determines the extent of the reduction of low frequency clicks.

It sometimes sounds more like a plop than a click. This control ranges from 0% to 100%. Audition: When selected, this control lets you hear only the sounds that get removed. When the actual contents of the audio can be heard in audition mode, it is a strong indication that the threshold is set too low. If the threshold is left unadjusted, the audio signal gets harmed.

Efficiency meter: This meter indicates the efficiency of the DeCrackler. The Threshold dial should be tweaked to get the maximum value. The maximum is also reached when the threshold is low, but at this point the fundamental audio signal gets harmed. Threshold: Determines the detection level for the crackles. This control ranges from 0% to 100%.

Reduction: Determines the amount by which the crackles get reduced. This control ranges from 0% to 100%. Audition: When selected, this control lets you hear only the sounds that get removed. When the actual contents of the audio can be heard in audition mode, it is a strong indication that the threshold is set too low.

If the threshold is left unadjusted, the audio signal gets harmed. Filter: Specifies the number of filters to use to remove the hum.

Hum is included not only of the fundamental frequencies of 50 Hz or 60 Hz, but also contain harmonics with frequencies that are multiples of the fundamental (100/110 Hz, 150/160 Hz, and such). Higher values cause greater CPU usage. Adjusting this value determines the number of harmonic frequencies to filter. Reduction: Specifies the amount of reduction to apply to the hum. High values could also cut necessary audio information in the low end.

Frequency: Specifies the center frequency of the hum. Usually this frequency is 50 Hz in Europe and Japan, and 60 Hz in the US and Canada. Often the frequency of the hum is not static, but varies by +/– 5 Hz.

To set the respective frequency, click the 50 Hz or 60 Hz buttons. Freeze: Stops the noise floor estimation at the current value. Use this control to locate noise that drops in and out of a clip.

Noisefloor: Specifies the level (in decibels) of the noise floor as the clip plays. Reduction: Specifies the amount of noise to remove within a range of –20 dB to 0 dB. Offset: Sets an offset value between the automatically detected noise floor and the value defined by the user. The offset is limited to a range between –10 dB and +10 dB. Offset allows more controls when the automatic denoising is not sufficient. AutoGate: Cuts off a signal when the level falls below the specified threshold.

Use this control to remove unwanted background signals in recordings, such as a background signal in a voice-over. Set the gate to close whenever the speaker stops, thus removing all other sounds. The LED display colors indicate the gate’s mode: open (green), attack or release (yellow), and closed (red). Use the following controls for Gate:. Threshold: Specifies the level (between –60 dB and 0 dB) that the incoming signal must exceed to open the gate.

Flanging is an audio effect caused by mixing a varying, short delay in roughly equal proportion to the original signal. It was originally achieved by sending an identical audio signal to two reel-to-reel tape recorders, and then pressing the flange of one reel to slow it down. Combining the two resulting recordings produced a phase-shifted, time-delay effect, characteristic of psychedelic music of the 1960s and 1970s. The Flanger effect lets you create a similar result by slightly delaying and phasing a signal at specific or random intervals. LFO (Low Frequency Oscillator) Type: Specifies the wave type for the Low Frequency Oscillator: Sin(e), Rect(angle), or Tri(angle). Rate: Specifies the speed of the Low Frequency Oscillator.

Depth: Determines the gain level of the modulation waveform, thus controlling the depth of the effect. Delay: Sets the point in milliseconds at which flanging starts behind the original signal. The flanging effect occurs by cycling over time from an initial delay setting to a second (or final) delay setting. Feedback: Determines the percentage of the flanged signal that is fed back into the flanger. With no feedback, the effect uses only the original signal.

With feedback added, the effect uses a percentage of the affected signal from before the current point of playback. Mix: Adjusts the mix of original (Dry) and flanged (Wet) signal. You need some of both signals to achieve the characteristic cancellation and reinforcement that occurs during flanging. With Original at 100%, no flanging occurs at all. With Delayed at 100%, the result is a wavering sound, like one coming from a bad tape player. Use the following controls for each band:.

Threshold 1‑3: Specifies the level (between –60 dB and 0 dB) the incoming signal must exceed to start compression. Ratio 1‑3: Specifies the rate of compression, up to 8:1.

Attack 1‑3: Specifies the time (between 0.1 milliseconds and 10 milliseconds) the compressor takes to respond to a signal that exceeds the threshold. Release 1‑3: Specifies the time required for the gain to return to the original level when the signal falls below the threshold. LFO (Low Frequency Oscillator) Type: Selecting Sine, Rect, or Tri determines the waveform of the low-frequency oscillator used to modulate the phase shift. Rate: Determines the speed of the low frequency oscillator. Ranges from 0 to 10. Depth: Determines the gain level of the modulation waveform, and thus controls the depth of the effect. Ranges from 0% to 100%.

Delay: To achieve various possible effects, the phase-shifted signal gets delayed against the original signal. The Delay property sets the time for the delay.

Ranges from 0.1 ms to 4.0 ms. Feedback: Determines the amount of phase-shifted signal that gets mixed to the input signal. Using negative values inverts the phase again by 180°. Ranges from -50 to 50. Mix: Determines the ratio of Dry and Effects signal.

A setting of 100% corresponds to a ratio of 1/1 while a setting of 0 defeats the effect signal. It ranges from 0% to 100%.

Pre Delay: Specifies the time between the signal and the reverberation. This setting correlates to the distance a sound travels to the reflecting walls and back to the listener in a live setting. Absorption: Specifies the percentage in which the sound is absorbed. Size: Specifies the size of the room as a percentage.

Density: Specifies the density of the reverb “tail.” The Size value determines the range in which you can set Density. Lo Damp: Specifies the amount of dampening for low frequencies (in decibels). Dampening lower frequencies prevents the reverb from rumbling or sounding muddy. Hi Damp: Specifies the amount of dampening of high frequencies (in decibels). Low settings make the reverb sound softer. Mix: Controls the amount of reverb.