[speex-dev] Speex settings and jitter

[Allen Drennan]

In my experience most of the DirectSoundCapture interfaces actually just map
over to standard Wave Multimedia recording so you are right in that
DSBCAPS_CTRLPOSITIONNOTIFY is usually not implemented.

The fastest way to record audio on Windows for the purposes of minimizing
latency with something like Speex is to use a thread which uses GetMessage
to extract buffers from the recording device.  As you noted though, most of
your latency is created when recording as Windows sound recording doesn't
like really small buffers.

-----Original Message-----
From: owner-speex-dev at xiph.org [mailto:owner-speex-dev at xiph.org] On Behalf
Of John Hayes
Sent: Tuesday, May 20, 2003 11:33 AM
To: speex-dev at xiph.org
Subject: RE: [speex-dev] Speex settings and jitter

Right - and I deal with that on the receiver end based on an approximation
of sender's and receiver's responsiveness - the minimum latency I've been
able to get into the system is about 150 ms. Of that, jitter buffering is
about 40-100ms. I'd love to figure out how to get that down without killing
myself on thread switching or Win32 kernel calls, but ms has to actually
implement the DSBCAPS_CTRLPOSITIONNOTIFY capability in direct sound capture
...

John

> -----Original Message-----
> From: owner-speex-dev at xiph.org [mailto:owner-speex-dev at xiph.org]On
> Behalf Of Allen Drennan
> Sent: Tuesday, May 20, 2003 11:18 AM
> To: speex-dev at xiph.org
> Subject: RE: [speex-dev] Speex settings and jitter
>
>
> In my experience most of the jitter related issues are because people are
> using too small of audio buffer sizes that match the framing size
> of Speex -
> particularly in Windows.  This isn't a problem with Speex, but as a
> programmer you should collect and append a few frames to match the size of
> your output audio frame buffer before attempting to play the sound.
>
> -----Original Message-----
> From: owner-speex-dev at xiph.org [mailto:owner-speex-dev at xiph.org] On Behalf
> Of John Hayes
> Sent: Tuesday, May 20, 2003 11:12 AM
> To: speex-dev at xiph.org
> Subject: RE: [speex-dev] Speex settings and jitter
>
> The audio frame speex generates sounds pretty terrible most of
> the time, and
> I don't use it for jitter correction instead I just use it for dropped
> packets - so I usually drop the late packet. It sounds acceptable
> as long as
> I drop less than 5% of traffic (dropping 2 in a row makes a bad
> robot noise,
> so I reset the stream in that case). The good news is that on an
> unsaturated
> DSL line jitter and packet loss are basically 0.[1]
>
> In my experience dropping a packet without decoding NULL or
> without playing
> the gap packet make a really bad tick.
>
> John
>
> [1] The bad news is on a saturated DSL connection - it's
> basically unusable
> without a second of jitter buffering.
>
> > -----Original Message-----
> > From: owner-speex-dev at xiph.org [mailto:owner-speex-dev at xiph.org]On
> > Behalf Of Chris Flerackers
> > Sent: Tuesday, May 20, 2003 9:05 AM
> > To: speex-dev at xiph.org
> > Subject: [speex-dev] Speex settings and jitter
> >
> >
> > Hi,
> >
> > Is there are document where the interaction between all the
> configuration
> > options of speex is explained?
> > Speex seems to have CBR, VBR and ABR. You can also use SPEEX_SET_QUALITY
> > (SPEEX_SET_VBR_QUALITY) and SPEEX_SET_BITRATE which I suppose
> can't be set
> > at the same time. Is there a list of possible combinations somewhere?
> >
> > I also have another question related to jitter. To minimize jitter, you
> > usually need to make the audio
> > shorter and longer and preferably while keeping it sound smooth.
> > What is the
> > best way to do this
> > with speex? I saw that you can pass NULL to the decode function
> > to generate
> > an audio frame which
> > might be used to make the audio longer. Will speex always generate a
> > "fluent" audio stream (without ticks, ...)
> > when inserting these NULLs. And is dropping a packet a viable
> way to make
> > the audio shorter?
> >
> > Best regards,
> > Chris
>
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