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 ...


6

The following example will make a directory called outputdir and then re-encode all wav inputs in the current directory to mp3: mkdir outputdir for f in *.wav; do ffmpeg -i "$f" -c:a libmp3lame -q:a 4 outputdir/"${f%.wav}.mp3"; done You can use a slightly modified command to convert multiple formats to MP3: mkdir outputdir for f in *.{wav,aiff,flac,m4a}; ...


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 ...


5

Short answer: Because people still buy them, and it's not worth the cost to convert. Longer Answer: Those are the native settings of their existing files. It would cost them time (man hours) and storage space to convert all of those files to 48k for "convenience" on hard disk based delivery for the end user. Both mean more money spent. Note that these new ...


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. ...


4

Well, you're not really going to 30fps...you're going to either 29.97 (or maybe 23.98). I realize you may just be rounding, but who knows what newbies are out there looking at this page? Don't want them to get confused. ;) Just pitch shift your finished mix by the inverse percentage of the pull-down (i.e. here you're talking about a 4.1% pull-down, so pitch ...


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 ...


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.


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

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

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 ...


2

businesses respond to market demands. some customers still request 44.1 because that's "cd quality" and takes up the least hard drive space while still being full rez. If an sfx company were to only offer 192k sfx they'd almost certainly be costing themselves some sales on the lower ends of the market, and they'd be doing it for no reason.


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

dBPowerAmp (http://www.dbpoweramp.com/) It's the one and only, and not very expensive either.


2

Well, you will lose quality - there is no way to gain it back from 128kbps files. A simple filter can roll off your high pitched 'jangles' but because they have elements across multiple frequencies you aren't going to get rid of them all. So give a low pass filter a try or use a treble reducer (similar effect in this type of scenario) and you might get a ...


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

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 ...


2

Not all sound design is for picture. Plenty of audio-only projects are 44.1 - music, art, theatre etc.


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.


1

Audacity has a playback speed control. Make sure that it is set to 1x. Transcription Toolbar


1

Flac is just flac - it will accurately encode whatever is fed into it. You are right though, that if you feed in a poor quality audio file it will remain a poor quality file. The only likely way to do this would be through spectrum analysis: Analysis of the frequencies in the file may indicate compression or sampling artifacts, but automatically scanning ...


1

A DI-box would do the job, for example most of these would. Look in the specs if it supports speaker output (usually the maximum wattage is specified). Note that most of them are mono, so look for a stereo version or hook up two of them parallel. Also look at the voltage (if it is an active one), you need 12V for most cars. Most DI-box's have balanced ...


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