[Flac-dev] Synchronizing a streaming client to the server Was: Idea to possibly improve flac?

Brian Willoughby brianw at sounds.wa.com
Sat Jan 8 02:48:55 PST 2011

On Jan 7, 2011, at 20:44, Ben Allison wrote:
> The main problem is in the Ogg layer, in my opinion.

I think you are right.

What is the reason that the Ogg layer was chosen for streaming  
broadcast?  My admittedly naive assumption was that Ogg is a file  
format, not a streaming format.

> Imagine this extreme use-case with __completely made up__ numbers.   
> This
> is a scenario where the server is encoding to FLAC on-the-fly from  
> a raw
> PCM input, either from disk or a live stream.
> Let's say the FLAC block size is 1024 samples, or 23ms at 44100 Hz.
> Let's say each silent block compresses to 1 byte.  Let's also say  
> that the
> Ogg packeting layer wants 4096 bytes before creating a page.  Again -
> these numbers are completely made up, but illustrate the point.  In  
> this
> example, it would take 95 seconds of digital silence before Ogg  
> decided to
> send another page out over the network.

I realize that these are made-up numbers, and I wouldn't expect a  
great deal of precision, but it would seem worthwhile to at least  
look at realistic estimates.

The minimum FRAME blocking in FLAC would be 12 bytes for stereo, not  
just 1 byte.  That's with 40-bit minimum FRAME_HEADER, 8-bit  
SUBFRAME_HEADER, 2 silent CONSTANT 16-bit audio samples, and 16-bit  

This means the 4096-byte Ogg packeting layer would only represent 7.9  
seconds of latency.  I think it's perfectly reasonable for a client  
tuning in to a lossless broadcast to wait just under 8 seconds for  
pre-buffering of the stream.

Better yet, ditch the Ogg packeting layer entirely, and use raw FLAC  
streaming.  That should mean that any size packet is possible.  TCP/ 
IP should allow shorter packets than the maximum, although it may be  
more efficient for broadcast traffic to have packets closer to the  
normal size for file transfers, which would be more like 3.4  
seconds.  I admit that I don't know for sure what restrictions there  
are on internet broadcast packets, but it can't be much worse.

3.4 seconds is a long time to go without a packet, but buffering of  
6.7 seconds or more should easily take care of that.

> If the input audio on the server is coming from a live-source (such as
> simulcast of an FM station) or if the disk I/O is very slow, this  
> can be
> extremely problematic.
> Ben Allison
> Principal Software Engineer
> Nullsoft, Inc.

I don't see why live-source audio would be a problem.  The system  
would simply buffer the required amount of uncompressed audio data  
before calling the FLAC encoder.  The only cost would be latency, and  
all digital broadcast systems that involve codecs involve a serious  
amount of latency.  I can hear a distinct delay between my HDTV set  
tuner and computer USB tuner, since each has a different amount of  
buffering in its pipeline, exacerbated by the digital audio system.

Brian Willoughby
Sound Consulting

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