Can't set desired bitrate for ogg via cmdl defaults to 500k

[muammar]

Can't set desired bitrate for ogg via cmdl defaults to 500k

[muammar]

@ao-il thanks for your report because in that way I can keep improving mkchromecast. Can you tell me if it happens just with ogg codec, or also with mp3 and acc?.

Regarding latency, it is a very hard thing. I recently changed of ISP where I live, and with the new WIFI router I had an improvement from 8s to just 4s lag. I then decided to remove the -re flag from ffmpeg encoder backed and it improved to 3s lag in my case. I am thinking of ways to improve this (I am for instance implementing gstreamer pipes to capture audio and see if it gets any better). I have realised that when I using mkchromecast in macOS, the node server I am using to capture and encode is much more efficient in the sense that lag is below 2s. For sure this problem you are getting of not being able to change 500k bitrate will request more from your router!.

[ao-il]

I just tried mp3 and it works, but my files are ogg and would like to use the same format to stream. I realize that with audacity you can't set bitrate for ogg but instead you specify a quality number ranging from 1-10, where 9=320. Do you think I can use -q 9 instead? This is because ogg uses VBR. Regarding the sample rate, do you the cast device resamples? Cuz in the asoundrc, I saw you commented out 44100. I uncommented that because my files are not 48000 but the latter. Hope the device does not up resample to 48000?

[muammar]

@ao-il I have pushed a commit in the devel branch that should fix this problem. For the moments you have to always set bitrate with integer numbers with mkchromecast. I could work to implement a quality flag maybe.

By default the sample rate of the app is at 44100. You may modify it if you want, and as far as I can tell the google cast devices do not resample. That was a bug they fixed. Could you check that commit b010da5 fix your problem?.

Thanks.

[ao-il]

It now works to set 320 for ogg. Thanks alot!