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6

Your sine has come out a couple of octaves too low, for some reason. Indeed it's infrasound, 6.6 Hz to be precise, which rather explains why you don't hear anything. A couple of ways you could have found this out: Actually the sound is still "audible": over a decent subwoofer, you'll hear (or rather feel) a throbbing similar to a ship engine, and things in ...


6

Usually, an application that plays encoded files (be it audio or video) will need to decode that file to a format that the target interface (be it a video or audio interface) can output on a standard port (like analog or spdif for audio, hdmi or vga for video). Most audio interfaces are waiting for PCM datas. Depending on the OS host of the platform, the ...


5

Lower bit depths are not easily available simply because they are not looked for commercially, not because of some intrinsic technical difficulty. Audio interfaces or other capturing devices have their ADCs (Analog do Digital converters) based on integrated chips optimized for the features that are most requested commercially. In principle it would not be ...


5

If you want to be technically accurate, a sound file does not have a volume, it has a level. Volume can only exist when there is an actual speaker producing sound as that has a fixed SPL(sound pressure level). A signal can be played back at any volume provided the speaker is able to produce it at that volume. Instead, it is referred to as signal level ...


3

0dBFS 1kHz sine defines a peak value. Alignment of audio levels is usually done at an RMS level of -18dBFS or -20dBFS depending on the standard you are applying. The line level alignment level that the amplifier will use depends entirely on the manufacturer. For the sake of argument, let us consider that the alignment level is -18dBFS. -18 dBFS = 0 dBu = ...


2

ffmpeg -i original-file.wav -f u8 -acodec pcm_u8 8bit-file.pcm See also here. It's not comma-separated but binary, but that's definitely the more sensible choice – especially for a board with very limited bandwidth.


1

These spikes are high frequency components contained in the resulting signal decode. There are two possible reasons for this. Method one uses a low pass filter which removes these components. Method two has a decoding fault. Both of these suggestions are speculative as there is no reference signal to compare.


1

According to this spectral output you are not dealing with 48kHz sampled audio. For some reason the audio has been decimated and the actual sampling rate is 24kHz. Consequently the 'nyquist' frequency is 12kHz. One of the key things to remember about digital audio, is that during the process of sampling analogue audio the spectral content is mirrored ...


1

Do this in two steps. First generate the RAW file (in audacity, there is an option to export to headerless RAW audio file under Export -> Export Audio). Make sure you export to 8-bit signed or unsigned values. Then, using a linux tool called xxd, you convert the binary file into a c header file(I suppose you could do this with any hex viewer but I haven't ...


1

Nowadays it can be edited with programs like Tascam Hi-Res Editor: http://tascam.com/product/hi-res_editor/ From their site: TASCAM Hi-Res Editor is a new, free application that plays and edits up to 11MHz DSD or 384kHz WAV files. The software allows playback and export of DSD files without converting to PCM audio. This app is available now as a free ...


1

This is a complex question =-) there's only one software that allows for true DSD editing, Sonoma Systems. This software however relies on sound card dependencies. DSD cannot be edited because there's only one bit, therefore it cannot be manipulated. At the point of edit Sonoma converts the audio to multi-bit DSD, and then goes back to 1bit after the edit. ...


1

First off I think it's a big decision to buy a field recording rig. It's very good that you ask about these things, because it can prevent you from buying the wrong gear for the purpose. But you never mentioned your purpose of buying it in the first place. What do you plan to record? Quiet Ambiences or cars? My question is, is the combination Denecke PS-2 /...


1

PCM audio is the nowadays common method of encoding analog audio to a digital form (without data compression, i.e. we're talking about "lossless digital audio"). Practically PCM audio is contained in (or "is equal to") audio formats such WAV and AIFF. Usually WAV. Channels 3&4 might refer to the so called M&E (music & effects) track. I.e. it's a ...


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