9

Converting a from wma file to wav is technically un-compressing it. You will not be able to regain the quality lost from the original compression, but you shouldn't lose any additional quality as long as the algorithm of the conversion software is decent. ffmpeg should work well for your needs. Edit: (Thanks to @evilsoup for the command line given in the ...


5

Mixing audio in a computer is very analogous to how sound mixes in the real world. In the real world, sound is simply pressure waves that cause vibrations in our eardrum that get converted to what we hear by our brain. In the computer, that level of pressure of the wave is represented as a series of samples that describe how strong the pressure wave was. ...


5

You can use the amerge and pan filters in ffmpeg to combine two mono streams into one stereo output: ffmpeg -i input -filter_complex "[0:3] [0:4] amerge,pan=stereo:c0=c0:c1=c1" -c:v copy -c:a pcm_s16le output or using -ac instead of pan: ffmpeg -i input -filter_complex "[0:3] [0:4] amerge" -c:v copy -c:a pcm_s16le -ac 2 output [0:3] and [0:4] refer to ...


5

You didn't supply very much information, such as your OS and your desired output format, so I can not give you a detailed answer. CAF is a container format that can support several audio formats, but Apple Lossless (ALAC) is probably the most common. ffmpeg can decode ALAC in CAF: ffmpeg -i input.caf output.wav This can be turned into a "batch" command ...


4

I just came up with a rule of thumb. I don't know if it is is really any good or not, but maybe worth considering. If you are taking things away (cleanup) better to do it in mono. If you are adding things, (sweetening such as reverb) better to do it in stereo.


4

Audio sampled at 44.1 kHz (like normal audio CDs) can in theory contain content up to the Nyquist limit of 22.05kHz. However, you need a filter to remove all content above that limit, otherwise it folds back into the hearable range: you hear this as aliasing. A perfect, theoretical filter would remove all content above 22.05kHz and leave everything else ...


3

Dither, Noise Shaping, and Bit Quantization. These are the reasons for the separate mastering passes. All of which do not need to be considered when mixing/bouncing the audio in the native digital format it was converted to (24bit/48khz). Also, it sounds like he will provide you with individual "HiRes" native files OR Will downsample for you and create a ...


3

This will depend a bit on exactly what is in the files. They may be just audio, but they may have midi bits and pieces in there as well. In your .band file, you should have a media subfolder, and the audio tracks should be in there (as .aiff files if I recall). Any audio app should let you import them and let you save as wav, mp3 or whatever. I'd use ...


3

I'm not sure of any way to easily do that in the program. However, since the project files are simply text it should be easy enough to write a program to parse the file and create markers to match the items. Each item starts with the position property. <ITEM POSITION 234.73069114906761 When adding a marker through the UI, you would see a dialog like ...


3

Yes, any analog transition or transmission has quality loss. How noticeable that loss is depends on a variety of factors including the quality of the DAC (digital to analog converter), the quality of connections, the quality of cables, the quality of any amplifiers, etc. That said, it isn't likely to practically matter particularly much. All digital audio ...


3

Use the Marantz as your pre-amp. Route the audio out of one of its 'Recorder' outputs. This should give you consumer-level output, nominally -20dB. Input to the Behringer on a switchable input mic/line/instrument & flick between the options to see which gives you the best impedance/level match - probably line or instrument. Watch levels as you do this, ...


2

I believe this article sums it up far better than I ever could. And I like it because it doesn't seem to lean to either side of the "sample rate war". I recommend everyone read it of you have the time. http://www.trustmeimascientist.com/2013/02/04/the-science-of-sample-rates-when-higher-is-better-and-when-it-isnt/ From my personal experience, record at ...


2

WAV is an uncompressed format. Unless you are merging tracks, reducing the sample rate, or lowering the bit depth (all unlikely to happen by accident), a plain conversion to WAV will always be lossless. Of course, this will not remove noise introduced by the lossy compression on the source material. Basically, any program you find will suffice.


2

It is impossible to upsample the bitrate. Your files have already lost the info from being compressed to 192. It looks to me as if the AC3 is going to give you a similar if not slightly more consistant result to the MP3. Seeing as this option is also easier to do using the program you mentioned... Go with the AC3


2

Yes, there will be a difference. Mp3 is a lossy codec that deteriorates with every generation. It's bad practice to use mp3 (or lossy compression in general) at any stage where the audio may yet be edited or reencoded. Some programs are smart enough to simply copy the original audio stream without reencoding if possible, but I can't say whether this applies ...


2

Taking the 2 MSBs should be just what you're after. The way I think of it is that 1 bit (1 or 0) simply tells it the signal is above or below the middle value - and since we're talking audio, that would be the zero line. So, 2 bits would do the same, but subdivide the range into 4 'zones'. 3 bits into 4 'zones' and generally, x bits = 2^x 'zones'.


2

First convert dbm to Watts using the following: Pw = 10^(Pdbm/10) / 1000 Pw = 10^(-10/10) / 1000 = 0.0001 Watts So the max output power is 0.0001 Watts, and the output load impedance is 10 kOhm. We can now use Ohm's Law to calculate the Voltage: V = I * R P = I * V which makes V = SQRT(P * R) V = SQRT(0.0001 * 10000) = 1 Volt now you can go from Volt ...


2

Assuming that the preamp outputs are line level, then yes it is perfectly safe to do this. You will wire the connection up as follows: Ground (RCA) to Pin 1 and 3 (XLR) Signal (RCA) to Pin 2 (XLR) A converter plug should be fine and it should have the same wiring as this.


2

There is no formula you can use to make this conversion. It is unlikely your webcam would be suitable as a sound sensor due to the processing involved in detecting and multiplexing the audio into the visual data stream. Also, the 'dB' values you are quoting are meaningless. In order to be able to detect something useful, you would need a raw feed from the ...


2

FLAC is lossless compression. During compression, you can set a compression option: setting this to a higher level creates a smaller file that is still lossless by taking more time to process the file. This compression level is included in the file, so the decoder will know how to decompress it. So two FLACs created from the same source using different ...


1

The decisive step for answering this question is the "converting to stereo" step. You are adding another dimension not present in the original after the step. If this step is done with a specialized convert-to-stereo filter (like there are special ways to add color to black-and-white films), your preprocessing in mono should be focused on improving the ...


1

Page 23 of the following document contains the circuit diagram for the PT40 transmitter. http://cloud.akg.com/9784/wms40_service.pdf The Schematic indicates that the connection is balanced however the schematic does not indicate a balanced circuit. Pin 3 appears to be a bias voltage supply. Therefore I think that shorting pins 2 and 3 would be inadvisable....


1

There is a limit to the amount of compression you can achieve with a "Lossless" compression algorithm - at some point all the recognizable patterns are removed and your file essentially becomes random data. The bitrate you are referring to is simply the number of bits of data that have to pass through the decoder to reproduce a unit second of uncompressed ...


1

You would rather turn to a program like Matlab or its open source equivalent Octave. Open your mono-channel audio data as a vector (using wavread for instance), and turn each sample value to its square value. Then compute the mean every N samples, N depending on the smoothness/time-precision you need.


1

Off the top of my head.. Sum - Adds Signals; any sounds present in both signals are reinforced by a varying amount, depending on how in-phase they are. Subtract - Subtracts one signal from the other(same as inverting the phase of one, then adding it to the other); any sounds present in both signals are attenuated or cancelled out completely, depending on ...


1

Thanks to Tetsujin for formula The basic formula is (Bar X Beats per Bar X Tick Resolution) + (Beats X Tick Resolution) + Remaining FL Ticks = Ticks


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