Distortion in ULAW

[mdgarrett]

Getting distortion when playing back a ulaw encoded wav file. I initially did this using a websocket connection to a server in order to make use of being able to stream and begin playback relatively quickly before having a complete buffer and playing back using AV.Player.fromBuffer. This is a telephony recording and you can hear everything being said, but there is artifacting/distortion that can be heard that is not in original audio (slight popping, etc). I then switched to do AV.Player.fromURL and it is also present. Given ulaw is a very simple codec, I do not believe this is related to tickets I've seen dealing with FLAC/CPU usage, so was curious if this was an issue in general?

[mdgarrett]

For what it is worth, tested ulaw decoding and piped to xaudiojs and it sounds fine, none of the distortions mentioned above are present.

[crackofdusk]

@mdgarrett Which browser are you using? Is the sound distorted in other browsers?

[mdgarrett]

Was using Chrome, didn't test in FF. Will do so, but likely won't be till tomorrow. That said, it was brought to my attention by a colleague for code we were collaborating on, and he is more FF. So know it was happening on different PCs and believe browsers.

[ikorchynskyi]

Confirmed, I've faced with this issue too. I've created a sample build to demonstrate an issue: https://github.com/ikorchynskyi/aurora-test/

ULAW file from /prompts played correctly in vlc, but in browser sound is distorted. Tested in:

$ google-chrome --version
Google Chrome 43.0.2357.132 
$ firefox --version
Mozilla Firefox 38.0

Would you please give some advice? Maybe, some custom filter can helps?

[ikorchynskyi]

After a deep investigation:

1. Decoded by aurora buffer played by xaudiojs w/o noise.
2. AudioContext.createScriptProcessor in aurora called with bufferSize = 4096, in xaudiojs with bufferSize = 2048.

bufferSize
This value controls how frequently the audioprocess event is dispatched and how many sample-frames need to be processed each call. Lower values for bufferSize will result in a lower (better) latency. Higher values will be necessary to avoid audio breakup and glitches. It is recommended for authors to not specify this buffer size and allow the implementation to pick a good buffer size to balance between latency and audio quality.

• https://developer.mozilla.org/en-US/docs/Web/API/AudioContext/createScriptProcessor

Increasing bufferSize to 8192 or decreasing it to 2048 (sic!) completely removes clucks from output.