[vdr] Re: mpeg1 layer 2 (mp2) -> iec958 ?

["Dr. Werner Fink" <werner@suse.de>]

On Mon, Oct 13, 2003 at 11:40:29PM +0200, Sven Goethel wrote:
> On Monday 13 October 2003 23:30, Sven Goethel wrote:
> > are the most digital receivers able to
> > receive mp2 via iec958 ?
> >
> > is this a stdandard ?
> >
> > if so .. may we can pass this digital data with bitstreamout
> > as well ?
> >
> > werner ?
> >
> 
> well in werner's linked page:
> 	http://www.epanorama.net/documents/audio/spdif.html
> they say:
> 
> 
> +++
> 
> Characteristics
> 
> Standard IEC958 "Digital audio interface" from EBU (European Broadcasting Union) details:
> 
>     * Audio format : linear 16 bit default, up to 24 bit expandable
>     * Allowed sampling frequencies (Fs) of the audio:
>           o 44.1kHz from CD
>           o 48 kHz from DAT
>           o 32 kHz from DSR 
> 
> +++
> 
> Multi channel audio and S/PDIF
> 
> IEC958 was named IEC60958 at 1998. IEC60958 (The S/PDIF) can carry normal audio and IEC61937 datastreams. 
> IEC61937 datastreams can contain multichannel sound 
> like MPEG2, AC3 or DTS. When IEC61937 datastrams are transferred, 
> the bits which normally carry audio samples are replaced with the databits from the datastream 
> and the headers of the S/PDIF signal. 
> Channel-status information contains one bit (but 1) which tells if the data in S/PDIF frame is 
> digital audio or some other data (DTS, AC3, MPEG audio etc.). 
> This bit will tell normal digital audio equipments that they don't try to play back this data 
> as they were audio samples. (would sound really horrible if this happens for some reason).
> 
> The equipments which can handle both normal audio and IEC61937 
> just look at those header bits to determine what to do with the received data.
> 
> +++
> 
> so we might can add at least to more formats to bitstreamout:
> 
> - encoded samples of 48 kHz, e.g. of mp2
> - mp2 itself encapsulated like todays ac3/dts ..
> 
> is this true ?

Yep.

> 
> if so .. there would be no more need to 
> switch to analog audio for mp2 ...

Hmm .. never tried that because I've connected the S/P-DIF out
of the DVB card to the S/P-DIF in of the sound card and loop
the data through the sound card in the case of normal mpeg
audio.  The advantage is simply that in that case there are
no problems with the quart synchronization of the quart of
the transmitting station (e.g. Pro7) and the quart of the
sound card.  For an example, my sound card is about 100ppm
faster than the time normal of Pro7 in Live.  This leads
all 15 minutes to a short shutter due to the workaround
within the bitstreamout plugin (doubling all fourth AC3
frame in the risk of getting an underrun).

Nevertheless it should be possible to embed mp2 audio
frames within an PCM frames each representing the length
the duration of the included mp2 frame.  This for all
DVB cards which do not have a S/P-DIF out ... or sound cards
which lost the S/P-DIF in pin due to the vendors `kindness'.
And the question is: Are all A/V receivers and amplifiers
out there able to detect mp2 and switch on _automatically_
the mp2 decoder.


          Werner


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