20
votes
Accepted
What audio formats and codecs are used for YouTube videos?
Your best bet is to see what formats youtube-dl returns by running at a command-line,
youtube-dl -F URL
which will produce output looking like this
format code extension resolution note
249 ...
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 ...
3
votes
Why is Constant Bit Rate (CBR) not favorable?
Constant bit rate is just that - constant, whether it's needed or not.
It wastes data when it's not needed & it can't keep up when it is.
In areas of low complexity, therefore, CBR is just ...
3
votes
LAME: Why is a lowpass filter used with --preset insane
This is an anti-aliasing filter. The --preset insane settings must allow aliasing to become apparent.
Your iPhone can't reproduce the frequencies the filter is applied to anyway, and if it could, you ...
2
votes
Accepted
Is lossy encoding supposed to produce higher signal level peaks than those present in the original?
Yes, certain compression formats can result in higher peak levels than the same material pre-compression. -1dBTP ("true peak" level, relative to full scale) seems to be the accepted maximum safe level ...
2
votes
How to tell if several audio tracks were encoded from the same source?
You can see the difference between a very low-bitrate mp3 & an unsullied file with a spectrograph; but you can also hear it.
By the time you're up in the decent file-types, high-Q mp3 & AAC, I ...
2
votes
Split audio by frequency
Solution with sox:
HZ="2700"
sox inp.wav "$HZ"-.wav lowpass $HZ
sox inp.wav "$HZ"+.wav highpass $HZ
sox -M "$HZ"-.wav "$HZ"+.wav out.wav
Solution ...
2
votes
Opus vs MP3 (Size & Quality Comparison)
Assuming your ripping from lossless source to Opus/MP3, Opus is definitely superior sound quality as this is shown in listening tests from Hydrogenaudio website.
as a general rule... Opus/AAC(standard ...
1
vote
Audio equivalent to QR code
Dual-tone multi-frequency signaling (DTMF) seems to be the obvious answer, because it also has two dimensions (two sets of tones).
To make the tones more tolerable, you could move them to a lower ...
1
vote
Split audio by frequency
With Reaper (free evaluation licence, low cost - $60 - usage licence), you can :
Use a 4-band crossover splitter, assign each band to one of 2 outputs (and use the crossover as a 2-band crossover ...
1
vote
Possible to ffmpeg encode an MP3 media segment for HLS to follow existing MP3?
The Thing Is, you are not perfectly replicating the output of -f hls by doing what you are doing. You are creating discrete mp3's and expecting them to behave the same way as hls mp3's. HLS is ...
1
vote
Accepted
Is audio better quality through a DAC meant for microphone encoing? (U-Phoria UM2)
An external ADC/DAC chain might sound better than if you did it in the PC.
Is it really? Possibly, or it could be apophenia or other psychological improvement. You would need to do controlled ...
1
vote
44100 Hz encoded in 24bit vs 16bit
The theory around sampling contains quite a bit of mathematics. Leaving that aside, let us look att the practical aspects of your questions.
In practice, today, there are only two effects of sample ...
1
vote
Accepted
FLAC --bps setting real meaning
According to the man page, --bps specifies the bits per sample. As far as I can tell, this parameter controls the data that will be encoded using FLAC, and can have a lossy effect. For example, here's ...
1
vote
Accepted
What should be the bit-rate & sample-rate settings, if I were to convert a WebM audio file to MP3?
The audio is encoded using opus, which is a high quality codec. When decoding from Opus, you will likely want to go via PCM - 16 bit at 48000. The more significant parameter with mp3 encoding will be ...
1
vote
Accepted
What is the difference between changing mp3 file extension to .wav and using a converter instead?
The difference (in basic terms):
.mp3 is a compressed audio format: some of the information from the original signal is thrown away when you create an .mp3.
.wav is an uncompressed audio format. ...
1
vote
Accepted
Are BRSTM files lossless?
The Binary Revolution Stream format uses ADPCM to represent audio data. ADPCM uses a lossless compression algorithm.
ADPCM is an adaptation of DPCM, which is in turn, an adaptation of PCM.
...
1
vote
Are BRSTM files lossless?
From a quick read through several of the links above, I believe it's most appropriate to call BRSTM a lossless audio format. For a start, let's define "lossless audio":
For simplicity's sake, a ...
1
vote
Does converting mp3 to wav make sense with regard to timing accuracy?
It does not make sense. You get a lot bigger filesizes, precisely 620 MB per hour of audio (which translates to 20 3-minute-long songs). For thousands, you're looking at about 60 GB of music. A minute ...
1
vote
Accepted
Do I lose any quality when trimming an MP3 with Adobe Audition?
Yes, you ALWAYS lose quality when open a file created with lossy compression (i.e. virtually all flavors of MP3) It may not be very much, but it is unavoidable. That is why it is called "lossy".
...
1
vote
Which audio file format gives smallest size for silent tracks?
Just an idea (which needs further research): do you actually need the silence to be a file or could you just play dynamically created data? If the latter is the case, you could probably find a ...
1
vote
Encoding OGG vorbis as CBR?
I would highly recommend using CELT or Opus for low bitrate audio (48kbps-96kbps), it sounds far superior, even compared to high performance ACC-HE+. the later is an IETF standard, and already widely ...
1
vote
Accepted
LAME bitrate and lowpass filters
Source:
http://wiki.hydrogenaud.io/index.php?title=LAME#Recommended_settings_details
Deeper source(code):
https://github.com/rbrito/lame/blob/origin/libmp3lame/lame.c#L211
Update:
--preset insane ...
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