13

In Audacity, you can approach this in at least two ways, depending on the nature of the bit you want to silence and the surrounding material. The simple way is to highlight the section of the track / channel and select "Generate... Silence" from the menu. The suggested duration will be the length of your selection. This is a harsh edit, and very likely ...


13

Play-at-Speed Audacity has a "Play-at-Speed" feature located at Transcription Toolbar. Move the slider to choose a slower or faster speed than normal, then click the green triangle button to left of the slider to play at that speed. The benefits of changing speed with Transcription Toobar are that you don't have to change the audio data in the ...


5

If the amplification was linear, and didn't result in clipping, then comparing any given peak value (in Audacity) between the two files will give you the information.


5

Better is subjective, and taste differs, but a chain of effects might include the following: High pass filter, maybe set at 20Hz. Depends on the material, but if you have a recording done with microphones the very lowest frequencies often are aeroplanes or lorrys or air condition and not the signal you want. When the signal is spoken word, go higher in ...


4

You're quite right that it's some sort of nonlinear operation – hard digital clipping in fact. And honestly it's somewhat obvious why this happens: you're superimposing two sine waves, each of which is already peak-normalised. The result of such a combination will in general exceed 0 dBFS, and without special handling (limiter etc.) that means the ...


4

No. You have a systemic delay. When your recording software sends samples to the soundcard, they are buffered in the driver, sent to the card, buffered in the card, digitally filtered for oversampling (which also incurs a delay), buffered in the D/A converter and finally sent out. Then you hear this already delayed signal, add your new voice to it, it ...


4

ReplayGain tags aren't standard in WAV files, so you have to alter the PCM data with the required gain. As per my reading of the Replaygain specs, a correctly implemented Replaygain scanner will print out the gain required to attain 89 dB SPL (as defined in the specs). FFmpeg has a filter to detect replaygain. You can run ffmpeg -i in.wav -af replaygain -...


3

Try using a crossfade between the two tracks. Crossfading is slightly different from a normal fade in/out. Instead of a linear transition, crossfading uses a logarithmic curve to keep the volume constant all the way through. Audacity has built-in crossfading effects if you want to give it a try.


3

There are Cursor Short Jump Left, Cursor Short Jump Right, Cursor Long Jump Left and Cursor Long Jump Right commands, which are assigned to comma (,) and period (.) keys by default. They can be changed in Preferences/Keyboard. Comma: 1s left Period: 1s right Shift + Comma: 15s left Shift + Period 15s right Shortcut info found in Audacity 2.1.0 Guide.


3

Yes, there are ways to try to reduce echo, but they also negatively impact the sound with artifacts. You can use features like gates to try to cut off when someone stops talking directly, but those are probably best applied after recording. There is nothing that you can apply, in software, at the time of recording that you can't apply later. You don't ...


3

This question is a bit out of context on a sound design q&a site, but easy to answer. Whenever you re-encode an already encoded soundfile (mp3>export to mp3) you lose more 'sound' information. Look up 'lossy' and 'lossless encoding' on wikipedia and you'll understand it better. Oh and in general it's easier for us to answer if you include more details. ...


3

MountainX, I don't know if you ever got the answer you needed but I just found out how to do it. If you have a stereo recording in Audacity you click on "Audio Track" at the left of the stereo channel you're looking for. You'll see Split Stereo Track in the drop down menu and then you can edit a specific segment of the audio in that channel alone


3

There are enough patches in the recording with the hum and without the voice to use a noise removal tool -- Audacity has one built in and the LADSPA plugins provide this kind of thing. I'd be shocked if the likes of Protools, Logic etc didn't have such a thing. As the comment mentions below, the noise is sampled and then its audio spectrum removed from the ...


3

Add a label track, and find the start of each leitmotif. Place your cursor at the beginning of the area you want, and press ctrl+alt+v or Edit>Paste Text To New Label. This will create a new label, and if there is any text in your clipboard it will be added. However, this must be saved as a .aup (audacity project), since labels serve no purpose to audio ...


3

If I would approach it, I would go and try the following: First of all, FM radio broadcasting has an upper limit of 15 KHz, and lower limit of 30 Hz. So cut all frequencies outside that scope to start with a very steep filter. Then a "little desktop / clock radio" would use a very small speaker which would not reproduce any sound lower than 200 Hz, so roll ...


3

You could try working with an expander, which is the reverse of a compressor and increases the dynamic range.


3

Processing always has the negative side effect of either increasing noise or taking away from the quality. The secret to processing is you want to do as little as possible to achieve what you want, and in this case that means removing steps that essentially do the same thing. As for the order here's what I would do. Normalize - This will amplify it to ...


3

To do it one marker at a time, without messing up the ones you haven't yet done, start with the last marker & work towards the beginning. This assume that once you've done a cut, you no longer need the marker.


3

Having listened to mentioned in comments - the effect doesn't come in at the 49s mark, it's there right from the start. It is definitely a phase issue - but I'm not sure what's causing it. If you've eliminated the potential physical issues as described by Joel, than what remains is a routing issue. I'm going to take a ...


3

It really depends on the material; are these sounds that you want to remove truly peaking at a lower level than every bit of the speech? How about in loudness or 'energy'? You need to look into the kind of sounds you want to remove and see how they differ from the ones you want to keep. This is key to being able to separate them. Are they of a shorter ...


3

If you mean just the dialog, you'd do better starting from the 5.1 audio as a source. The centre channel should be dialog only. If you want to include the music, or if you only have the audio as stereo, then it's not possible. You cannot unbake a cake. Late Edit: There is now the very good yet very expensive Izotope RX which has a "Music Rebalance" plugin ...


3

I assume that you export sound in order to import it somewhere else. This process has a lot of details and used to be done in the step called "mastering" in the old recording days. I'm no expert, but I could try to give you some pointers. Your first choice is whether to compress or not. The best known example of compressed formats is probably MP3. ...


2

As others have said, this is the expected behavior. Your computer mic input is a stereo mic jack that takes signal from the mono mic on both channels by nature of the connector design. (On a mono 1/8" connector, the ring contact will touch the sleeve of a mono connector.) Your Focusrite on the other hand has two distinct inputs, not a L/R configuration. ...


2

No, it is not ideal. It isn't doing anything to the text file other than reading it incorrectly. RAW data is expecting binary data, not text. But text is an encoded as binary data, so the file actually is a binary file. The encoding of the text file impacts how the binary data is formed, but reading it as a RAW audio file is really completely inaccurate ...


2

8-bit 8K isn't going to be great quality in any case, but WAV is as good as you can do. Straight uncompressed MS-format (RIFF) WAV is the most compatible format for interchange. There are two senses of 'compressed' at work here. U (mu)-law and A-law refer to schemes for compressing dynamic range, not for reducing file size. Typically they would be 'expanded'...


2

See previous discussion here, SPL De-Verb is probably what you're looking for. You can watch a comparison between SPL, iZotope, Zynaptic in this YouTube video.


2

You choose when to stop, Original file (as downloaded from Vocaroo) First, a cut at 8 kHz... Audio after 8 kHz cut Next, a 70 Hz high-pass... Audio after 8 kHz cut and 70 Hz high-pass And just for fun a pass of iZotope's "Dialogue Denoiser" (auto setting)... Audio after 8 kHz cut, 70 Hz high-pass, and Dialogue Denoiser


2

Quality of recorded audio (in the digital realm) is measured by captured frequencies (sample rate) and dynamic accuracy (Bit Depth). Once recorded, changing those wont improve the signal, only increase the file size. If your noise floor is too loud when compared to your voice then you need to reduce the noise. There are expensive programs that do this quite ...


2

I'd be far more inclined to try fix it at source first. There are several plugins I've tried that attempt to de-verb, but none anywhere near as successfully as just not recording it in the first place. Simplest trick would be to hang a heavy blanket, duvet or similar on the wall behind you [or hang it over a spare boom stand etc, close up behind you], ...


2

A long-shot, but this might work if you have a similar setup to mine. It works equally well for DVD [I don't have a BluRay] TV, catchup, YouTube, Netflix, VLC, etc etc I use my PC's built-in 7.1 analog outs to my 5.1 amp [which saves double decoding & odd phase issues I get over S/PDIF, or HDMI.] My built-in sound is a Realtek HD device [I know, not ...


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