I'm in the process of beta testing a stream for internet radio broadcast. We are looking to stream in mp3 as it's still the most universally accepted format for opening from a URL or hyperlink, at least until we develop our standalone app and flash player on the site.
That said, we were toying with 128kbps 44k lame and like how it sounds, but have concerns for mobile users and data usage. So we'd like to go with a 64kbps stream instead and lower the overhead for everyone. Right now, running the lame converter at 64kbps 44k joint stereo just doesn't sound good. Lots of high end artifacts as expected. Cutting the sampling frequency down to 22k helps but I'm losing a lot of high end.I've listened to other services using this bit rate, and they sound a lot better. I listen to Radio.com quite a bit and their stations are encoded 64kbps AAC or MP3 by default. Even when switching the player to MP3 in settings, it sounds really clean. Honestly nothing like a 64kbps lame encoded stream that I'm used to.
Sorry if I left you with more questions than answers.
Quote from: Rescator on 14 January, 2013, 02:40:05 PMSorry if I left you with more questions than answers.Hi Rescator. Thanks for the advice on coding HTML 5 and WebRTC. Yeah, I'm kinda bewildered. The only thing i can think is that the app saying its playing 64kbps is really 64kbps mono / 128 stereo.. which would be misleading for all the mobile data crunchers out there.I tend to think it's not a processing thing, unless they are using a ton of pre-emphasis before 22khz sample rate, which I think would only work marginally well. You have my curiousity on mp3Pro...
=======================================================================LOW BITRATES=======================================================================At lower bitrates, (like 24 kbps per channel), it is recommended thatyou use a 16 kHz sampling rate combined with lowpass filtering. LAME,as well as commercial encoders (FhG, Xing) will do this automatically.However, if you feel there is too much (or not enough) lowpassfiltering, you may need to try different values of the lowpass cutoffand passband width (--resample, --lowpass and --lowpass-width options).
We may end up implementing HE-AAC from the get go..