On Fri, 2008-12-26 at 01:04 +1100, David Timms wrote: > William Case wrote: > > On Wed, 2008-12-24 at 08:30 +1030, Tim wrote: > >> On Tue, 2008-12-23 at 10:55 -0500, William Case wrote: > ... > > 1. Gives me a whole range of adjustments for different channels. (I > > assume channels means different sources e.g. Master, Headphone, PCM etc.). > Just to clarify, a source device generates an audio signal( line in, PCM > (pulse code modulation = wav uncompressed audio), that might then get > processed (eg volume control, master, headphone), and then sent to a > destination (often an output device like a speaker connector or > headphone jack). > > Because a typical soundcard has an internal hardware mixer, it can > usually mix together various inputs (sources) like CD input, mic input > and recorded audio signals, and produce a single output signal (mostly > in stereo=2 channels). When mixing together externally received signals, > no main CPU processing power is used, unless you are trying to record to > hard disk etc. > > > 2. gives me two choices and > > 3., 4., 5. gives me only Master. > > Which should I choose and why? > capture means recording - usually from a line in (eg from a vcr or mp3 > player etc), or from a microphone. So exclude those from your choice. > > > If I should be using HDA NVida (Alsa-mixer), why do I have PulseAudio > > options? > Consider pulse audio to be a real-time digital mixer and volume control, > where the audio calculations are performed inside your main CPU. In the > default setup, once pulseaudio has done it's processing, it passes the > result to the alsa driver which outputs the audio data to the soundcard. > The soundcard turns the digital audio data into analog audio signals for > use with amplifier, speakers, or headphones. > > Pulseaudio also has enhanced capabilities like remembering that when you > playback with xmms that you like to output via your amplifiers and > speakers, but when you are viewing a flash video, to playback into your > headphones instead, at a different level. Another capability let's you > choose the destination playback device while the material is actually > being played. A third capability let's the output go to an audio device > on another machine. Obviously, this is a bit trickier to set up. > > ... > >> These individual mixer input controls should normally be left off if you > >> never use them, as they can each introduce noise (hiss, beeps and > >> burbles, etc.) to the system. > > I will turn them off except for Master and Front. I will experiment > > with PC Speaker. Of course these are only available to me if I use the > > default alsa mixer setting. > It's not a one or other setting, both parts will still be involved; > pulseaudio will process, mix, and attenuate sound signals, whereas alsa > will drive the physical hardware. The setting you are seeing lets you > decide whether to control the physical driver volume levels or the > software generated pulseaudio volume controls. If you mute or turn the > alsa master way down, it wont matter how high you turn the pulseaudio > mixer, since the alsamixer comes after the pulseaudio one in the audio > chain. (also true for the reverse). > > If you play back a loud audio file, and turn both the pulseaudio source > and master up full. Then change to the alsa setting. You can then use > the also setting to set up an absolute maximum level that you would want > to hear, by adjusting the master. Then you could go back to the > pulseaudio setting to adjust the playback to a comfortable setting, and > from then on only use the pulseaudio setting. > > ... > >>> * How is sound related to video ? > > > >> Sound is the sound, video is the picture... The question is too vague > >> to be answerable. > In digital format, sound and vision are both represented with digital > 1's and 0's. With all video and audio file types, there is a packing > together of the audio and video information into the one file. The > multiplexed file provides information about when to playback each frame > of video in relation to the audio in the file. For example, an mpeg2 > (dvd) file might have two frames of video, then 2 of audio, then 1 of > video, two audio in an order to achieve a consistent throughput of audio > and video data. > > >>> * Why are there so many files associated with producing sound? > In digital audio, the most basic file type is waveform (.wav), where > each momentary value of audio is stored, on a 1 for 1 basis. Experiments > and calculations can show us that for something we store as quality > musical recording we need to sample that momentary value at 44kHz (times > per second) or higher so as not to disrupt our digital recording with > audio aliases. Since we also seem to enjoy the spatial enhancement > produced by stereo or more channels, the file needs to store both left > and right information. Finally, we found that if we only store the > digital value using a small no of bits per sample, when played back we > hear a harsh, chunky sound, rather than the CD like quality of using 16 > (or more) bits per sample. The catch with all that is it takes up a lot > space. > > To solve space issues (less a problem now that storage space costs a lot > less), compression schemes were developed. Most take advantage of > reducing the number of channels eg to mono, reducing the sampling rate, > or the number quantizing levels (bits/sample); but this is done in > context of the type of audio being compressed - eg human voices are > typically of lower frequency, and can sampled at a slower rate, and with > less levels. > > The biggest jump in compression was with psychoacoustic modelling, where > it was found that in a complex sound, a listener does not notice that > certain frequency (pitch) sounds become inaudible (or masked) by other > sounds. > > The reason there is so many formats, is because developers were > essentially competing to produce more highly compressed audio files, > without noticeable change in quality, when using a certain type of > audio, over a certain communication medium. Eg: when the fastest home > internet connections were slow modems, compression made it possible to > transmit voice signals over your internet connection. If you tried to > transmit music of higher quality that voice, you would have large audio > distortions that made it difficult to hear the original material. > > You might like to play with the audio editor program audacity (perhaps > from rpmfusion if you want to be able to import and save in certain > compressed formats (mp3)). It shows you a graphical representation of > the audio file, and eg lets you choose a zoom, start and stop position, > and just play back small parts of a file, so that you can work out what > the sound "looks" like to a computer. > > Hope that helps a bit more ;-) > DaveT. > Hi, Dave, I work in the IC Test industry, and that is the clearest non-math explanation I have ever read. I also teach applied DSP (fourier analysis, time series analysis and uses of IFTs.) I have endeavored to explain to many and varied audiences these effects, but never came up with such clarity. May I quote you? Thank you, Les Howell -- fedora-list mailing list fedora-list@xxxxxxxxxx To unsubscribe: https://www.redhat.com/mailman/listinfo/fedora-list Guidelines: http://fedoraproject.org/wiki/Communicate/MailingListGuidelines
