ALSA is at the centre and amixer, alsamixer and kMix all use the same
API and so stream control is more like
Code:
--------------------
kMixer
|
V
streamtuner2------->arecord ( streaming ALSA) ------->SBS------->SB
^
|
alsamixer
--------------------
> + There is a huge latency issue. It takes a stream over seven minutes to
> achieve 100% buffering.
When using arecord to a file - it will set very big buffers which means
you are using all PC memory as a buffer so it will always takes minutes
to buffer. You will see the "overrun" message when application cannot
read audio and put into a buffer in time.
When streaming to a pipe as setup in WaveInput buffers are still large
by default. The secret to minimising latency is to use smaller buffer
and double buffering - you should try using ecasound. Look at
custom-convert.conf.eca for a example in using ecasound but settings
may need to be tweaked to suit your situation.
> + Most critical: sound is getting to the Capture channel only as it is
> recorded from the PC Speakers by the PC microphone! If you mute the
> speakers the music stops but you still hear room noise (conversation,
> keyboard, barking dog, etc.!
Normally after pressing mute - one of the controls will go to 0% -
check it is normally the master. I think there is something odd (e.g.
electrical fault, signal loop) in your hardware/audio setup that mic
input bleeds into audio. This seems to be confirmed by the fact that
the Mic and Mic boost levels do not affect the audio you hear.
--
bpa
------------------------------------------------------------------------
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