CODEWITHSUNDEEP
c++audioencodingcodecopus
rightaway717
rightaway717

Reputation: 2831

Are there any constraints to encode a audio signal?

I capture a pcm sound at some sampling rate, e.g. 24 kHz. I need to encode it using some codec (I use Opus for that) to send over network. I noticed that at some sampling rate I use for encoding with Opus, I often hear some extra "cracking" noise at the receiving end. At other rates, it sounds ok. That might be an implementation bug, but I though there might be some constraints also that I don't know.

I also noticed that if I use another sampling rate while decoding Opus-encoded audio stream, I get a lower or higher pitch of sound, which seems logical to me. So I've read, that I need to resample on the other end, if receiving side doesn't support the original PCM sampling rate.

So I have 2 questions regarding all this:

  1. Are there any constraints on sampling rate (or other parameters) of audio encoding? (Like I have a 24kHz pcm sound - maybe there are certain sample rates to use with it?)
  2. Are there any common techniques to provide the same sound quality at both sides when sending audio stream over network?

Upvotes: 1

Views: 577

Answers (2)

rightaway717
rightaway717

Reputation: 2831

The problem was solved by buffering packets at receiving end and writing them to the soundcard buffer as soon as some amount has been reached. The 'crackling' noise was then most likely due to the gaps between subsequent frames that were sent to the soundcard buffer

Upvotes: 1

mvd
mvd

Reputation: 256

The crackling noises are most likely a bug, since there is no limitations to the samplerate that would result in this kind of noise (there are other kinds of signal changes that come with sample rate conversion, especially when downsampling to a lower samplerate; but definitely not crackling).

A wild guess would be, that there is something wrong with the input buffer. Crackling often occurs if samples are omitted or duplicated, oftentimes the result of the boundaries of subsequent buffers not being correct.

Sending audio data over network in realtime will require compression, no matter what. The required data rate is simply too high. There are codecs which provide lossless audio compression (e.g. FLAC), but their compression ratio is comparatively low compared to e.g. Opus.

Upvotes: 1

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