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New Features Coming Up For PulseAudio 3.0

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  • #16
    How about PulseAudio recognize the on-board chipset features and actually have it configured for your system, instead of always haven't to bang on it to get it to recognize anything that is Intel Sound via RealTek 889 never mind 892. Each time some modification arrives in Debian my 7.1 turns into a 2.1 sound configuration. It's embarrassing that Audio in 2012 is still a joke in Linux.

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    • #17
      Originally posted by crazycheese View Post
      PulseAudio is meant for efficient audio mixing and broadcasting. For mixing of individual streams there is Jack.
      To record audio from youtube clip, download the clip using any plugin for your browser (at least 3 working), then drag-n-drop the clip onto Audacity 2.0+ instance.
      I think you missed the point. It's within PulseAudio's scope and any Windowing System(KDE, Gnome, etc) to allow PA to mix channels.

      Jack is really a hardware based(realtime access) layer that happens to have mixing capabilities.

      Also the example I put could also be a live broadcast and you want to record as the audio comes available, so downloading a video is not applicable in this scenario and is a round about way.
      Last edited by e8hffff; 12-13-2012, 02:21 AM.

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      • #18
        Originally posted by frantaylor View Post
        Perhaps your English is not so good! Can you tell the difference between "making an observation" and "volunteering for work"?
        You said PulseAudio devlopers have a "DUMB mind-set" since they don't support "PulseAudio everywhere" - definitely not just "making an observation". One of the obstacles to supporting everything is that some has to actually do the support part. I was just asking if you were ready to put your money where your mouth is or you were just shouting orders.

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        • #19
          Originally posted by Larian View Post
          Would you mind telling me what sort of circumstances have conspired to make you want to be able to do that kind of audio ninja kung-fu? I'm serious here, not trolling. It sounds like a neat feature, but why would someone want to do this? I've never understood who this functionality is marketed to.
          The Yate softphone is one of the few SIP clients that can work with my employer's goofy Avaya setup. However, it has no built-in functionality for manipulating audio inputs and outputs -- meaning there's no UI for configuring it to work with my USB headset. So without some method of altering the default, Yate sends output to my laptop's speakers and takes input from my laptop's built-in mic.

          Enter Veromix and PulseAudio. After installing Yate and during the first phone call, I can use controls in Veromix to change Yate's audio routing. I can send output to and take input from the headset. Fortunately, I only need to do this once after installing, as the configuration is saved. Furthermore, if I want to switch to speakerphone-like behavior, I can route the output and input back to the laptop's hardware.

          Agreed, this is suboptimal. Better would be for Yate to incorporate some audio configuration, like just about every other softphone I've tried. But, alas, I can't get any of those to properly perform (or hold) a SIP registration. The problem lies somewhere, I think, in Avaya's SIP implementation. But I'm no SIP expert, so I'm kind of guessing here.

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