10 votes
Accepted

WAV file recorded with wrong header

The most direct, if somewhat "hands-dirty", way of doing this is to manipulate the header with a hex editor. There are two entries in the RIFF header that need to be modified: the sample rate itself, ...
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6 votes

WAV file recorded with wrong header

Very easy. Just use SoX: sox -r 44.1k infile.wav outfile.wav The -r 44.1k overrides the header information in infile.wav and will not perform any effects on the samples themselves. More info: sox....
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  • 111
5 votes

48kHz vs 44.1 kHz

is ABSOLUTELY necessary to record at 48 kHz that's not for film where the final audio will be bounced to 44.1 format anyway? We know that at 44.1kHz we can accurately record and playback the ...
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  • 703
5 votes

WAV file recorded with wrong header

After some more searching, I have found that ffmpeg is able to do this by converting the WAV to raw PCM format, and then converting back to WAV with the correct sample rate in the header. For example:...
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  • 243
5 votes
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WAV files sound high pitched - how do I bulk fix the sample rate in the metadata?

Nevermind, I found something. The command line tool sox seems to do the trick: find . -name '*.wav' | xargs -I {} basename {} .wav | xargs -I {} sox -r 44100 {}.wav {}.edited.wav find . -name '*....
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  • 403
4 votes
Accepted

48kHz vs 44.1 kHz

As a possible side bar to the excellent answer provided by JCPedroza above, If human ears are the only listeners (i.e., signals between 20 and 20,000 Hz) then it is never* absolutely necessary to ...
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  • 175
4 votes

Is there any tradeoff to get the best quality/file size ratio for mp3 conversion?

To get the best audio quality in your final lossy-eccoded audio file output at the smallest possible file size, you have to do 3 steps: sample rate convert from 96kHz to 44.1kHz, because the lossy ...
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4 votes
Accepted

Converting from float 32 PCM to signed 16 PCM. Does it matter if I resample before or after the bit depth conversion?

I think the way you currently have the pipeline set-up is the right way to do it. Conversion should be the last block in the chain for sure as this will ensure that any processing artifacts are ...
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  • 8,474
3 votes
Accepted

Is it possible to record higher frequencies using multiple microphones sampling in lockstep?

frequencies above what microphone allows Microphones have a frequency response curve. This is a random example: The exact shape of the diagram is different for each microphone, but they all have ...
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  • 1,558
3 votes
Accepted

What is the real time from sample number?

OK you are getting confused between frames as in "SMPTE" frames and audio buffers. The frames you are referring to are actually buffers. They are not 'vision' frames. 25fps refers to vision frames. ...
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  • 8,474
3 votes

WAV file recorded with wrong header

One way i do this, using Adobe Audition, is to "Open as" raw PCM data (one has to type an * in the name bar of the opening window to show all files), and then define the sample rate and mono/stereo ...
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  • 51
3 votes
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What sample rates can an MP3 file have?

If this wikipedia page is to be believed, what you call mp3 and is actually a MPEG-1 audio layer 3 encoded audio file only supports three sample rates : 32 kHz 44.1 kHz 48 kHz EDIT As @PkP has ...
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  • 2,733
3 votes
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Verification of upsampled audio file

It seem to me that your code does not any interpolation with further addition of samples between each pair of existing ones (in that case the size of the file would double, but pitch and speed should ...
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3 votes

Why do mp3 have sample rate?

A sample rate is the rate at which samples are taking from the source sound. It says nothing about how much information is stored in those samples. Whether an audio codec is considered lossy or ...
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3 votes

Mixing audio with different sample rates

You will only get benefit from sampling to the next highest sampling rate you are able to mix at. For instance, if you have a 32kHz stream and a 48kHz stream, resample the 32kHz stream to 48kHz. You ...
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  • 8,474
3 votes
Accepted

Bit depth calculation doesn't seem to add up

It's an MP3 file. You can't deduce anything from that other than the fact that it's probably got a bit depth of 16 bits. Try the same exercise but using a WAV file. You will have better luck. MP3 is a ...
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  • 8,474
3 votes
Accepted

44100 Hz encoded in 24bit vs 16bit

It seems that as defined by the CDDA Red Book, CD quality is 16 bit with a maximum sample rate of 44.1 kHz. I understand 44.1kHz was chosen due to human hearing limitations so I assume 16 bit was ...
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  • 1,558
2 votes

Why do mp3 have sample rate?

Your understanding of lossy compression is close, but a little bit off. An MP3 (or any lossy compression) doesn't actually reduce the number of samples per second, but rather, it alters the values of ...
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  • 8,091
2 votes

How to convert a stereo MP3 to mono MP3

On Linux systems like Ubuntu, you can use ffmpeg: ffmpeg -i input.mp3 -c:a libmp3lame -q:a 2 -ac 1 output.mp3 where: -c:a libmp3lame: The audio codec to use -q:a 2: The audio quality (bitrate), see ...
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2 votes

Pitch-Shifting makes sample sound out of tune

I am no expert, but could your guitar have been tuned to Just temperament, instead of Equal temperament? Correct me if I am wrong, but if one tunes the guitar by the ear, it does not necessarily ...
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2 votes

Reasamplomatic5000 Change Pitch Without Changing Tempo

Firstly it's nice to see someone making use of RS5000! Sadly, though, it works in the same way as most samplers in that it achieves the pitch changes simply by playing back the sample data at ...
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2 votes
Accepted

How to get number of Frames(or Samples) per sec or ms in a audio (.wav or .mp3) file?

The bit depth in your case is 16. It refers to how large each sample is, so if you have a 16 bit value for each sample, that is your bit depth. Bit rate is a measure of data per second (as I think ...
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  • 283
2 votes

What is Tidal's 'Master' Quality sample rate and bit depth?

Fi: 44.1khz/16bit (1411kbps) Master: 96khz/24bit (4608kbps) Pitchfork Article Note that only ~30,000 tracks are MASTER quality right now You can tell that Master is available by the M symbol and ...
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  • 292
2 votes

MP3 and bitrate

Your assumption that bitrate = bitdepth * channels * samplerate is true for raw PCM encoded audio. It is not true for bitrate reduction encodings, be it lossless (as flac) or lossy (as mp3). The ...
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  • 2,733
2 votes

Is 96kHz sampling rate becoming standard for high quality audio?

There are several reasons for sampling with 96kHz: this allows for more natural results when doing non-linear processing (all kinds of distortion, modulation, pitch shifting) since that causes ...
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2 votes
Accepted

What's the lowest analog audio sound that a CD/44.1kHz audio system can make?

You can encode any frequency down to 0 Hertz (DC signal) in the digital signal. Most real-world recording and playback systems will not go this low unless you modify them in some way. The frequency ...
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  • 1,264
1 vote

Why does increasing the sample rate with SoundForge make the pitch go way up?

All you are doing here is changing the project sampling rate. Remember, all the audio files you are using will remain at their original sampling rate. It's not clear exactly what the original sampling ...
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  • 8,474
1 vote
Accepted

Is it possible to fake decimation with FM?

(This question intrigued me, so I though I'd try it out for myself and write a full explanative answer) Phase modulation is often used as a step in the process of frequency modulation, so they are ...
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  • 2,176
1 vote

Audio recording at wrong speed

You can use ffmpeg or audacity to do this. Both are free. In Audacity, use Effect > Change Tempo... or Effect > Sliding Time Scale / Pitch Shift
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  • 5,845
1 vote

How can I sync audio and video from a different source?

If you didn't use a clapper when you filmed, then you need to find something in the filmed portion where you can see something happen that causes a percussive sound that was recorded. A drummer ...
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  • 2,040

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