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Windows has multiple levels of volume controls. How to set properly?

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  • Windows has multiple levels of volume controls. How to set properly?

    My W2K carputer uses an Audigy 4 board to send audio to my I-Bus head unit and thus pretend to be a cd changer.

    The thing is, there are multiple levels of volume controls on Windows to adjust the line out volume:
    • WAV level (I'm playing wav files)
    • Master volume control
    • WinAmp volume control

    What combination of setting these will result in the best output sound quality?

    I will be using a 1K Hz (-3db) test tone to set the input level to the head unit so that the level out of my amp matches the original level amp input level I set using the cdchanger playing the test tone.

    I will be using the head unit volume control to adjust volume in actual use, I hope to get this to be a one time setup on the carputer side. Should I max the WAV volume, and maybe the master volume and use the winamp volume to adjust the line out? Should no control ever be maxed out, but always limited to 80%? What is the best strategy?

  • #2
    I personally would never run anything at Max, in order to avoid any possible distortion. To advance your idea start with setting everything at 50%.

    As the final sound is subjective, increase each setting slightly until you find a quality/volume you are happy with.

    At the end of the day it will be the quality of the speakers which will decide.


    • #3
      In All the Recording and Pro Audio I do I always Run My Volume At 75%...
      Microsoft Is a Evil Empire that has corrupted a generation!

      Thanks - Eric Scott
      Area 3 Productions


      • #4
        Use winamp set at 100% because using less than that actually causes winamp to render the audio at a lower level than it is meant to be. Winamp then uses the Wav output to send the audio to the sound card. Keep this as 100% as well. Use your Master to control your output volume. Adjust it to your liking.
        Christmas Car Project 2008

        ArrowDash Skin Worklog


        • #5
          Thanks. That's the info I hoped to learn.


          • #6
            lots of wrong info here.

            The Windows Kmixer set anything below 100% degrades the signal by reducing the bit resolution. A good quality soundcard in the computer should never distort at full volume unless the original signal/recording is distorted. I cannot say the same for cheaper quality cards and the way they setup the outputs of them, but I believe onboard sound should not distort.

            Same thing with WAV since it is part of windows kmixer.

            However, one of the developers of winamp specifically stated that the volume control implemented in winamp does not reduce bit resolution IIRC. I'd have to dig around to find the link.

            Windows Vista was suppsoed to redesign the way the mixer works and it was supposed to eliminate the bit resolution issue.
            System always under construction


            • #7
              Well, i have to agree with Durwood, Digital volume controls are bad, decreases bit resolution.. Keep them at 100% and use the Winamp volume control only as it is supposedly designed better.

              I cannot get the link to work, but here is some tech info on Digital Volume.

              Digital volume control with a PCM bitstream is actually very simple, you simply multiply
              each sample by a binary constant. This apparent simplicity is a trap for improperly
              implemented digital volume controls that can really mess up a digital signal. The result
              length in bits of any binary multiplication is equal to the sum of the word lengths of both
              input terms. For example a 16 bit pcm signal multiplied by a 16 bit volume control
              constant results in a 32 bit binary number. This conversion is completely loss-less, the
              original signal can be completely reconstructed from the 32 bit output data. However as
              there are no 32 bit DACs a 32 bit bitstream must truncated and possibly dithered or
              noise shaped to a manageable word length before it is converted into analog audio.
              Truncation is always a lossy process, once a signal is truncated no amount of processing
              may restore the original, it may be possible to to make any truncation artifacts inaudible
              with dither or noise shaping but the key to good digital volume control is to avoid any
              lossy translations. Instead of multiplying the pcm bitstream by a 16 bit constant you
              may multiply it by constant with an 8 bit or less word length, a 16 bit data stream and 8
              bit volume coefficient will result in a 24 bit word, which is attainable by many high
              quality DAC designs so no truncation is required. However the loss-less resolution of the
              result comes at the price of volume control steps that do not accurately track the ideal
              logarithmic attenuation curve, an 8 bit word my only attenuate a signal by a maximum
              of 48db and the last few steps will be very large making a reasonable attenuation limit
              of an 8 bit volume coefficient about 30db or so.
              The best digital volume controls are those that use a table that attempts to limit the
              amount of truncation needed. Such a volume control may use 4 bit constants for the
              first few attenuation levels, then gradually increase the word length as needed for
              greater amounts of attenuation. This table should include as many "magic" attenuation
              values as possible, -6.02db is only a 1 bit coefficient, -12.04db is only 2 bits ect... This
              approach maximizes the signal quality but also minimizes the linearity of the attenuation
              steps so that each volume step will be slightly different, however as volume control is more of a
              bulk attenuation and human hearing can easy adapt, a linear volume control is not really
              It should also be apparent that the least amount of attenuation possible should be used in a
              system with digital volume control, in an ideal system you should always be listening as near as
              possible full volume, this usually requires amplifiers with low gain.
              This can be a real problem since
              most amplifiers are designed with as much gain as possible to give the customer the illusion of
              "Power". Using amplifiers with too much gain requires too much digital attenuation at normal
              listening levels leading to very low resolution (too much truncation of the digital signal before it is
              converted in the DAC.) Digital volume control should also never be used with DACs of insufficient
              word length, a 16 bit DAC will require truncating a CD resolution digital input for any attenuation
              value at all so digital volume controls should only be used with DACs capable of converting word
              lengths of 20 bits or more. Of course better, higher resolution DACs result in better sound quality
              with digital volume control than lesser DACs. An acceptable DAC used without digital volume
              control may become unlistenable when used with digital volume control due to linearity errors that
              are normally masked when converting full level data, delta sigma DAC designs are perhaps the
              worst in this respect while sign-magnitude DAC designs are the best. A sign-magnitude DAC always
              increases its linearity with decreasing signal level making it ideal for digital volume control use. A delta-sigma DAC on the other hand has a noise level that increases with decreasing signal level
              making most of DACs of this type very poor choices to use with digital volume control.
              Ours is the age that is proud of machines that think and suspicious of men who try to.

              Current SYSTEM
              AMD64X2 2.5
              ZAPCO DSP-6
              ARCAUDIO SE Amps
              Rainbow Profi CS365 & Profi Vanadium 12"


              • #8
                Cool good info felix.

                Found my link

                System always under construction


                • #9
                  good info in here!

                  I always felt that using WinAmp for volume control was the best option, but never knew exactly why.

                  Jan Bennett
                  FS: VW MKIV Bezel for 8" Lilliput - 95% Finished

                  Please post on the forums! Chances are, someone else has or will have the same questions as you!


                  • #10
                    What about for folks using Centrafuse. To my knowledge the developer ties stragiht into DirectX for sound output. Doesn't use WMP or Winamp.


                    • #11
                      That is a good question that I don't have an answer to. Does centrafuse use the windows kmixer still to change volume or can volume adjustment be performed directly in DirectX? Maybe ask the developer or specifically in his forum. Or do some research on DirectX.
                      System always under construction


                      • #12
                        The Media Player is the exact reason i am not using Centrafuse, i really like the software though..

                        The way I understood Centrafuse is adjusting the windows master volume remotely...

                        That and the lack of support for FLAC files did it in for my frontend.
                        Ours is the age that is proud of machines that think and suspicious of men who try to.

                        Current SYSTEM
                        AMD64X2 2.5
                        ZAPCO DSP-6
                        ARCAUDIO SE Amps
                        Rainbow Profi CS365 & Profi Vanadium 12"


                        • #13
                          So leaving WAV and master volume at 100% and using winamp if needed to adjust output level is a good (simple) choice.

                          Bypassing kmixer with some type of kernel streaming plugin to winamp is a better sound quality choice (but more complicated install/configure/troubleshoot & may require resampling media files to 24-bit?).

                          Can someone expand on this a little?