Subject: RE: [AVT] RE: <draft-ietf-avt-rtp-vmr-wb-03.txt>: sampling rate

From: Scribano Gino-QA1087 <[email-masked]>
Date: Tue, 21 Sep 2004 08:22:05 -0500

Hi Magnus,

I think that you have captured these issues perfectly. According to my understanding, your explanations and assumptions are correct; and your conclusions are logical. Thank you for taking the time and effort to research these issues; and for clearly explaining your reasoning.

Best regards,
Gino

-----Original Message-----
From: [email-masked] [mailto:[email-masked]] On Behalf Of Magnus Westerlund
Sent: Monday, September 13, 2004 9:42 AM
To: Xie Qiaobing-QXIE1
Cc: Colin Perkins; [email-masked]; [email-masked]
Subject: Re: [AVT] RE: <draft-ietf-avt-rtp-vmr-wb-03.txt>: sampling rate

Hi Sassan and Colin,

I think we have two issues:

A. Is there any benefit to indicate or request that the sampling 
frequency used at the sender.

B. Is it necessary to use the sampling frequency as RTP timestamp rate.

I will start with A that I think is easier to explain and also can 
provide some information for issue B. If you find any of my assumptions 
and statements are incorrect, please correct me.

To my understanding of the VMR-WB after a conversation with Jonas 
Svedberg is that the VMR-WB will provide a somewhat better encoding of 
8kHz material if it is indicated that the input is 8kHz. However there 
is no need due to compatibility or decoder operation to signal the case 
where the 8kHz is used as input into the encoder. These would then 
result that the only case needed to be signaled between encoder and 
decoder is cases where the decoder will use output at 8kHz. Because if 
the decoder can request that the encoder uses 8kHz input some 
improvement of the 8kHz material is achieved. In the other cases where 
the receiver is capable of 16kHz it doesn't matter for the receiver if 
the original audio was 8 or 16kHz from a decoding point of view.

Colin, if one looks at issue B. Is it really needed to use the RTP 
timestamp frequency equal to the sampling rate used? I would say NO to 
that question. My reasoning is the following.

- Many audio input is sampled from a source at a higher rate then the 
encoder may handle. Thus a resampling and pre-processing stage is 
employed based on the encoders input frequency rather then producing 
that rate initially from the hardware. Some of the reason is that the 
pre-processing may actually yield better results than what the hardware 
at given input rate can gain. Another reason may be that one like to 
avoid switching the hardware between rate if changing the encoding.

- The frame based decoders does not need to know the encoders input 
rate. The encoder may anyway resample this into other rates for internal 
processing and band limited signals. I would claim that VMR-WB, AMR-WB+ 
and AAC are all example of codecs that perform this kind of tricks. On 
the receiver side they produce a output signal that has any sampling 
frequency the receiver finds most useful. Either causing clipping of the 
higher frequencies, but more commonly to a higher clock rate, despite 
that no more information is provided simply for ease of use.

- The frame based codecs do only need a RTP timestamp that allows the 
receiver to correctly reconstruct the time line when the encoding is 
done with the most audio bandwidth. In the VMR-WB case this is 16kHz. 
AMR-WB+ is even more strange, as we have selected an RTP timestamp rate
that results in that all internal sampling frequencies will result in 
integer timestamp ticks. Thus actually allowing one to correctly 
calculate frame alignment when the internal sampling frequency changes. 
That the frequency also is possible to recalculate into several common 
sampling frequencies with few partial sample alignments was also 
considered.

Thus I would use this to argue that indicating the actual sampling 
frequency is not necessarily as long as the receiver is capable of 
correctly reconstruct the media stream with its timing information in 
full resolution.

In the VMR-WB case I would think that having only one timestamp rate of 
16kHz does not effect codec operation and would simplify the handling 
when one has some senders that do use 8kHz, especially when gateways 
need to encoded sometime 8kHz material from pre-recorded responses and 
in other cases WB channel data. This do avoid the need to perform RTP 
timestamp rate switches.

If desired to have this possibility to request by a receiver that the 
sender do use 8kHz input then one should introduce a MIME parameter for 
this. However I would like to avoid using the "rate" parameter as it 
results in unnecessary barriers in form of signalling and RTP timestamp 
rate switching.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: [email-masked]

_______________________________________________
Audio/Video Transport Working Group
[email-masked]
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
[email-masked]
https://www1.ietf.org/mailman/listinfo/avt

---

From: Colin Perkins <[email-masked]>
Date: Thu, 23 Sep 2004 09:59:32 +0200

Gino,

As I've just replied to Magnus, I disagree with these conclusions.
Colin

On 21 Sep 2004, at 15:22, Scribano Gino-QA1087 wrote:

> Hi Magnus,
>
> I think that you have captured these issues perfectly. According to my 
> understanding, your explanations and assumptions are correct; and your 
> conclusions are logical. Thank you for taking the time and effort to 
> research these issues; and for clearly explaining your reasoning.
>
> Best regards,
> Gino
>
-- 
Colin Perkins
http://csperkins.org/