Subject: [AVT] Decrease rate in AC-3 audio with minimal loss of quality

From: Alejandro Cabrera Obed <[email-masked]>
Date: Fri, 4 Nov 2005 10:43:03 -0300

Hi people,
 I'll transmit AC-3 audio over IP, and suppose I monitor the audio buffer at
the receiver in order to detect the "underflow" state and then I decrease
the audio rate to make some type of congestion control.
 I ask you: what is the best manner to decrease the AC-3 audio rate with
minimal loss of quality: decreasing the sampling rate, decreasing the bits
per sample, decreasing the words per syncframe or what ????????
  Thanks a lot,
 Alejandro

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

---

From: "Link, Brian" <[email-masked]>
Date: Wed, 9 Nov 2005 14:53:48 -0800

Hi Alejandro,
 
The directly parameterized way to change the  data rate of AC-3 is to
change the words per sync frame. Typically, encoder implementations
offer a choice of data rates to a user, and these data rates are
translated by the encoder into frame sizes.
 
Lowering the sampling rate does have the effect of reducing the AC-3
data rate, but is not recommended. First, it does not offer many
choices. Also, it introduces continuity challenges on the playback side
when transitioning from audio of one sampling rate to that of another.
There are likely to be audible artifacts at transition points. Note that
altering the sampling rate of the audio requires additional processing
of the audio prior to encoding.
 
Reducing the bits per sample of the audio data does not by itself alter
the AC-3 data rate and it adds undesirable quantization distortion, so
it is also not recommended.

Depending on the configuration used in your application, you may need to
be concerned about tandem coding losses. It's not clear if this is true
from your question. If the audio is already encoded in the AC-3 format
before it enters your system, it is necessary to decode it and re-encode
it at the new data rate. Each generation of re-encoding (tandem coding)
introduces some loss of quality. Also, if your application will use
SPDIF (a common interface to hardware decoders), be aware that there are
added complexities in making sure frames with differing frame sizes are
aligned correctly over SPDIF.

Regards,
Brian 

________________________________

	From: [email-masked] [mailto:[email-masked]] On
Behalf Of Alejandro Cabrera Obed
	Sent: Friday, November 04, 2005 5:43 AM
	To: AVT Lista
	Subject: [AVT] Decrease rate in AC-3 audio with minimal loss of
quality

	Hi people,
	 
	I'll transmit AC-3 audio over IP, and suppose I monitor the
audio buffer at the receiver in order to detect the "underflow" state
and then I decrease the audio rate to make some type of congestion
control. 
	 
	I ask you: what is the best manner to decrease the AC-3 audio
rate with minimal loss of quality: decreasing the sampling rate,
decreasing the bits per sample, decreasing the words per syncframe or
what ????????

	Thanks a lot,
	 
	Alejandro

	-- 
	Alejandro Cabrera Obed
	[email-masked] 

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