[Vorbis-dev] Fwd: Object Audio - Misusing Vorbis?

[Richard Furse]

Thanks for this - made my life MUCH easier!

I took a proper look at Opus which we hadn't actually considered before. It
proved wonderfully easy to integrate and I keep having to play the results
as I'm not really believing the file sizes! So far the only things that seem
to be missing are a 44.1kHz sample rate and multichannel coupling, but I
reckon we can live without those...

Thanks again,

--Richard

> -----Original Message-----
> From: vorbis-dev-bounces at xiph.org [mailto:vorbis-dev-bounces at xiph.org]
> On Behalf Of xiphmont at xiph.org
> Sent: 11 November 2014 10:58
> To: vorbis-dev at xiph.org
> Subject: [Vorbis-dev] Fwd: Object Audio - Misusing Vorbis?
> 
> Oops, had meant to reply on-list as well.
> 
> Monty
> 
> ---------- Forwarded message ----------
> From:  <xiphmont at xiph.org>
> Date: Tue, Nov 11, 2014 at 12:57 AM
> Subject: Re: [Vorbis-dev] Object Audio - Misusing Vorbis?
> To: Richard Furse <richard at muse440.com>
> 
> 
> Hello Richard,
> 
> On Fri, Oct 31, 2014 at 3:50 AM, Richard Furse <richard at muse440.com>
> wrote:
> > Hi there
> >
> > I'm trying to use Vorbis in a slightly unusual way, and I'm a little
unclear
> > if I'm going to be able to do what I want to do nicely, or at all. Many
> > apologies if I'm on the wrong list, or if this is a FAQ, or off-topic.
I'd
> > be very happy to be re-routed!
> 
> If Vorbis can't do what you need, perhaps Opus can. In any case, let's
> get down to it...
> 
> > I'm doing a little experimental work on a block-based bitstream format
for
> > object audio streaming. This format allows any number of audio objects
to
> > start or stop on block boundaries, and each audio object might be mono
or
> > multichannel. Blocks may vary in length within the stream, from 1 to
65535
> > samples, and each block is encoded as a single "chunk" of data
containing
> > all audio for all audio objects for that block of time. There's a bunch
of
> > other data in the chunk, for instance spatalisation information. The
> > bitstream format is already (experimentally) operational using a very
> simple
> > codec that is lossless for all this data, including the audio.
> 
> ...yup, already thinking Opus might be a better fit because it can
> transition without new headers...
> 
> > What I was hoping to do next is a lossy version, and thought Vorbis
might
> be
> > suitable for the underlying audio coding, but I'm struggling a little to
> > understand exactly what I could/should be using. I thought initially I
could
> > use just the vorbis/vorbisenc calls to encode the audio for each audio
> > object each as an embedded bitstream, and then write the bytes returned
> in
> > ogg_packet structures from vorbis_analysis_headerout() and
> vorbis_analysis
> > () within my packets, but I'm not sure if this is supported behaviour -
and
> > for decoding, I'm not sure how I should populate the reconstructed
> > ogg_packet structures correctly.
> 
> What you would likely do in this case is write self-contained Ogg
> bitstreams.  The code will be happy to do it.  The same would be true
> of Opus, though it's easier to omit the high-level container in the
> case of Opus, as you can predict packet size much more easily.
> 
> 
> > I could use the whole Ogg binding, which would clearly take me back into
> > standard API space. Then, for each block of time, my bitstream chunk
> would
> > contain, for each audio object, a number of Ogg packets, which would in
> turn
> > contain Vorbis data. There's clearly quite a bit of redundancy here.
> 
> Right.  Nothing to add, you understand this well already.
> 
> > One of the things I'm unclear about is whether or not I'm guaranteed to
get
> > the right data into my chunk if I do this. If I submit, say, 4800
samples of
> > audio, and call all the relevant vorbis/ogg calls to fully flush the
> > resulting bytes, am I guaranteed to get exactly the ogg/vorbis packets
that
> > are required to resynthesize those 4800 samples of audio?
> 
> Yes. You could do it without 'complete' bitstream construction, though
> that would be the most strightforward way.
> 
> > During
> > resynthesis, no ogg/vorbis packets from the previous or next block/chunk
> > will be available (except any internal state the vorbis decoders might
be
> > keeping).
> 
> Vorbis packets are lapped, however the encoder also goes to some
> trouble to minimize dicontinuities at the boundaries between
> self-contained streams, as individual streams are not lapped, but you
> often want seamless playback boundaries regardless.  The seamlessness
> is not perfect, but it's pretty good.
> 
> > Another thing I'm not too sure about is how I should handle lost chunks.
> > Because the ogg/vorbis streams for new audio objects can start up at any
> > point in time, it's possible that the new ogg/vorbis header packets are
> > within the chunk that is lost. Is there then a way I can recover this
audio
> > object?
> 
> So long as you have the header matching a chunk, you can begin
> playback anywhere.  If the header requirement is an issue, Opus will
> get around it (as it does not have Vorbis-style headers and can
> transition between many playback states without restarting).
> 
> >I periodically send "sync" packets which are designed to allow
> > reconstruction of all state, so some more header data could go in there,
but
> > I'm not too sure what should be used. Maybe all the ogg packets relating
to
> > vorbis_analysis_headerout? Can I reset all the decoders with this header
> > data and then start feeding the (much later) analysis packets?
> 
> Yes.
> 
> > Is there a
> > lighter way to do this (I don't really want to be resending all the Ogg
> > headers every second)?
> 
> I don't blame you :-)
> 
> For any given Vorbis setup (number of channels, rate, quality), the
> headers for that stream will be the same for our encoder, and you can
> count on that.  If you're always using the same encoder, you can just
> assume the given headers.
> Or you could stick the headres that will be used somewhere once if you
> want to proof against encoder updates.
> 
> And, again, Opus sidesteps the issue by not having the heavyweight
> headers to begin with.  But it sounds like what you want can indeed be
> done with Vorbis.
> 
> > PS, some audio objects will be carrying HOA data, so will have
potentially
> > LOTS of channels (64?). How many channels can Vorbis handle in coupled
> mode
> > these days?
> 
> Specialized coupling goes to 5/7 channels.  You can do pairwise
> coupling up to 254 channels and independent up to 255.  You could also
> design a custom coupling if the project warrants it; that's in the
> headers.  That's not quite as scary as it sounds.
> 
> Monty
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