De-hum
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I've started this thread for the discussion of audio noise reduction/repair software and techniques. I've only just started using iZotope's RX2, but compared the other noise reducers I've tried it's damn near magical (disclaimer: I'm far from an expert in these things). IZotope, Inc. Is an audio technology company based in Cambridge, Massachusetts, United States.iZotope develops professional audio software for audio recording, mixing, broadcast, sound design, and mastering which can be used in wide range of Digital Audio Workstation (DAW) programs. Here are a few important tips for achieving the best results with dithering: Check our guides. Ozone's Mastering and Dithering guides contain useful information on the science and application of dithering. Work in a high-fidelity format. When mixing a project, applying effects, and mastering, it's always advantageous to work at the highest sample rates and bit-depths possible on your system. Oct 15, 2015 And for professionals who need to quickly deliver quality results, the RX 5 Advanced Audio Editor offers even more specialized post-production tools. In this review I will take a look at the new iZotope RX5 Advanced Audio Editor software.
De-hum is designed to remove low frequency buzz or hum from your audio file. Hum is often caused by lack of proper electrical ground. This tool includes a series of notch filters that can be set to remove both the base frequency of the hum (usually 50 or 60 Hz) as well as any harmonics that may have resulted. The De-hum module is effective for removing hum that has up to seven harmonics above its primary frequency. For hum that has many harmonics that extend into higher frequencies (often described as 'buzz'), try using the De-noise module. For tricky hum problems, De-noise features tonal noise reduction controls that can make short work of any extra-harmonic hum and buzz. Some very high frequency buzz can also be removed with the De-click module.
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Base Frequency
Sets the base frequency of the hum to be removed. The two most common base frequencies that cause hum are 50 Hz (Europe) and 60 Hz (U.S.). You can manually specify a base notch by choosing the Free option.
Note: When the De-hum module's 'Base Frequency' is set to 'Free,' you can use the Spectrum Analyzer (under View > Spectrum Analyzer) and its peak readout display to help find the exact peak frequency of any unwanted hum.
Manual/Adaptive
Adaptive mode will allow the De-hum module to adjust its noise profile based on changes over time in the incoming audio. In this mode, RX will analyze incoming audio for the specified learning time to determine what is hum and what is desired audio material. Adaptive mode can work better with sources that are constantly evolving.
In the Manual mode, the base hum frequency does not change over time.
Learn
The De-hum module can also automatically locate the root fundamental of any hum in your audio. Simply make a selection containing the trouble frequencies, and click the Learn button. This will automatically set the Base Frequency to the result of the Learn calculation. RX can analyze any audio with prominent hum, if you don’t have a passage of pure hum. This is, however, less reliable than learning from only hum.
The two most common base frequencies that cause hum are 50 Hz (Europe) and 60 Hz (U.S.). Under the Frequency Type field in the De-hum module, choose the appropriate frequency and then hit Preview to hear if this has an effect.
In some cases, you may need to choose the Free Frequency Type (e.g., when a recording made from analog tape is not precisely at its original recorded speed). Selecting this option unlocks the Base Frequency control and allows you to manually find the Hum's root note. With Preview engaged, move the slider up and down until you find the point where the hum lessens or disappears.
For even more precise settings, use RX's Peak Finding feature in the Spectrum Analyzer window. Simply single click to place RX's anchor sample on top of the hum you are trying to remove, and then drag your mouse over the peaks that appear in the Spectrum Analyzer window to view the exact frequencies of your audio. Djay pro windows free crack.
Filter Q
Controls the bandwidth of filters for base frequency and harmonics.
Linear Phase filters
Enables linear-phase FIR filters with a high FFT size. De-hum’s linear phase filters have a very accurate frequency response at the expense of latency and filter pre-ringing. When this is disabled, De-hum will use minimum-phase IIR filters, which are only susceptible to post-ringing (which is usually less noticeable than the pre-ringing introduced by FIR filters).
High/Low-pass filters
These filters allow high/low frequencies to pass while attenuating low/high frequencies respectively.
- Frequency (Hz): sets the cutoff frequency for the filter
- Q: sets the bandwidth of the high-pass filter
Number of harmonics
Because higher frequency harmonics often result from hum, RX's De-hum module has control for attenuating these overtones. Using the Number of Harmonics control, you can select up to 7 harmonics above the primary hum frequency. Again, the spectrogram display in many cases makes it easy to identify the number of hum harmonics in your project. After selecting number of harmonics, use the Harmonic Slope control to set how aggressively the higher harmonics are being cut. The Filter Q control adjusts the width of the hum filters.
60 Hz hum with harmonics
Link harmonics
Links the gain of all of the filters, none of the filters, or odd/even filters.
![Sample rate recording Sample rate recording](/uploads/1/2/6/0/126020915/319132956.gif)
Slope
When harmonics are linked, this controls the slope of the gain/suppression. As the harmonic order increases, the gain/suppression level resolves closer to 0 dB. When linking type is odd/even, an odd/even slope separate control appears that allows you to control the amount of gain/suppression for both odd and even harmonics.
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Gives numerical readout of gain settings in dB. You can also manually type in your gain settings for any of the harmonics.
Output hum only
By selecting the Output hum only checkbox, you can also hear the hum that is being removed. This is useful for fine-tuning your settings. Play through a section of your file where the hum is mixed with other material, select this mode, and hit Preview.
Now you can adjust parameters like the Filter Q (width) control and the Harmonic Slope control to maximize hum removal while minimizing the effect on the program material.
Resample
Resample allows you to convert an audio file from one sampling rate to another.
Sample Rate Conversion (SRC) is a necessary process when converting material from one sampling rate (such as studio-quality 96 kHz or 192 kHz) to another rate (such as 44.1 kHz for CD or 48 kHz for video).
It is common to record and edit in high sampling rates since higher rates allow higher frequencies to be represented. For example, a 192 kHz audio sample can represent frequencies up to 96 kHz whereas a 44.1 kHz audio sample can only represent frequencies up to 22.05 kHz. The highest frequency that can be represented accurately by a sampling rate is half of the sampling rate, and is known as the Nyquist frequency.
When reducing the sampling rate, or downsampling, it is crucial to remove the frequencies that cannot be represented at the lower sampling rate. Leaving frequencies above this point causes aliasing. Aliasing can be heard as the frequencies in an inaudible range are shifted into an audible range, causing distortion and noise. With iZotope SRC's steep low-pass filter, users can completely avoid the common aliasing artifacts while maintaining the maximum frequency content. A comparison of iZotope’s SRC process versus other sample rate convertors can be viewed at: http://src.infinitewave.ca/
You can also engage the Post-limiter option in order to limit the output levels of your signal to prevent any clipping from occurring.
Note: The Aliasing portion of the curve displayed in red shows the reflected frequencies during downsampling or imaged frequencies during upsampling — both due to aliasing.
New sampling rate
This setting chooses the sampling rate you want to convert to. Choose a sampling rate from the drop-down list, or click on the field to type in a custom sampling rate.
Change tag only
Changes the declared sampling rate of the file in the file’s properties without resampling the file, effectively changing the playback rate and pitch of the file.
This feature is useful if the sampling rate tag was damaged by a previous audio editing process and the file is playing back incorrectly.
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Filter steepness
This allows you to control the steepness of the SRC filter cutoff. The white line is representative of an ideal low-pass filter.
Higher filter steepness means better frequency performance of the filter: wider passband retains more useful signal, while stronger stopband attenuation provides better rejection of aliasing. At the same time, higher steepness of the frequency response requires a longer filter, which produces more ringing in time domain and energy smearing near the cutoff frequency.
Cutoff shift
SRC filter cutoff frequency shift (scaling multiplier).
Allows shifting the filter cutoff frequency up or down, to balance the width of a passband vs. amount of aliasing.
Pre-ringing
![View Sample Rate Izotope Rx View Sample Rate Izotope Rx](/uploads/1/2/6/0/126020915/618442693.jpg)
SRC filter pre-ringing amount in time domain (0 for minimum phase, 1 for linear phase, or anywhere in between).
Adjusts the phase response of the filter, which affects its time-domain ringing characteristic. The value of 0 produces a minimum-phase filter, which has no pre-ringing, but maximal post-ringing. The value of 1 produces a linear-phase filter with a symmetric impulse response: the amount of pre-ringing is equal to the amount of post-ringing. Intermediate values between 0 and 1 produce so-called intermediate-phase filters that balance pre- and post-ringing while maintaining linear-phase response across a possibly wider range of frequencies.
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Post-limiter
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Keeps true peak levels of the output signal below 0 dBTP to prevent any clipping from occurring.
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This option is important when resampling signals that are very close to 0 dB, because filtering during resampling can change peak levels of a signal.