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

Ben Allison benski at winamp.com
Fri Jan 7 16:48:10 PST 2011

The issue is that silent frames compress to a very small size, and the Ogg
packeting layer can put more than one FLAC frame into a page.  So if you
have an extended period of silence with a live or rate-limited input
stream, the client buffers may exhaust themselves before a new page can be
put together and transmitted.

> This thread has raised several good topics.  It's surprising that the
> FLAC-Dev list has been silent for years, and now suddenly there are
> several good ideas to discuss.
> On Jan 7, 2011, at 15:04, David Richards wrote:
>> I am interested in streaming lossless audio, FLAC is probably the best
>> option for that. Currently the OggFLAC way of doing it mostly works
>> with a few hacks in libflac and my version of edcast. It might be that
>> the Ogg packaging layer is ill suited for this purpose, and an
>> alternative model developed.  I've seen that its possible to stream
>> native flac with netcat, but thats not really the solution I'm looking
>> for.
> I have not done much work with streaming.  I have written a lot of
> serious code that uses the FLAC library.  I remember that there used
> to be separate objects in the FLAC library for streams, and they were
> unique from the file objects because you can seek backwards in a
> file, but you cannot seek backwards in a stream.  For some reason, it
> seems that these objects have been removed in the latest versions of
> the FLAC library.
> Can anyone explain the issues with streaming pure FLAC?  What does
> OggFLAC add to make streaming possible, or even easier than pure
> FLAC?  I thought that OggFLAC was just a way to put FLAC blocks into
> the Ogg file format.  Apple's CAF specification would also allow FLAC
> blocks to be placed inside their file container, although this still
> would not force iTunes to play FLAC unless a decoder were installed
> in the system.
> What is it about netcat that you don't like?  Can you describe what
> you're looking for, and why the specific details are important?  I
> was always under the impression that the FLAC format was already
> designed for streaming, but I must admit that I've never studied the
> issue.
>> On Fri, Jan 7, 2011 at 5:58 PM, Tor-Einar Jarnbjo <tor-
>> einar at jarnbjo.name> wrote:
>>> Am 07.01.2011 23:38, schrieb David Richards:
>>>> I'm also interested in another concept of lossless streaming with
>>>> flac. Lets call it broadcast flac. A problem with streaming for long
>>>> periods of time is that the sending and receiving computers
>>>> clocks go
>>>> out of sync, for example even if I stream myself on localhost, with
>>> This is not a FLAC specific problem, but has to be handled in all
>>> situations
>>> where the streaming server is in control of the transmitting data
>>> rate. It's
>>> caused by a playback device, which actual sample rate is slightly
>>> higher
>>> than the sample rate actually requested or a streaming source,
>>> which system
>>> clock is running slowly. Since these parameters (at least an exact
>>> playback
>>> sample rate) is hard to achieve, this is a rather common problem.
>>> Or to
>>> shorten it: If the data has a sample rate of 44100 and your sound
>>> card
>>> consumes more than 44100 samples per "sender-time" second, your
>>> buffer will
>>> eventually exhaust. If it's the other way around, your buffer may
>>> overflow
>>> if the client does not handle these cases properly.
>> I am well aware its not flac specific, but such a standard way of
>> handling such a matter could be part of the packaging for streaming
>> flac.
> I think that this would be a good opportunity to design a solution
> that is specific to broadcast.  At the sending end, the server should
> have knowledge of when there are breaks in the content.  If the
> stream could send flags at these breaks, then the receiving client
> could go silent and reset the synchronization.  As you describe, the
> situation only becomes a problem after long periods of time, but I
> would guess that there are enough station breaks (or at least song
> breaks) in a long broadcast that there would be a chance for a reset.
> CoreAudio is a pull model, and the API provides a time line that can
> be used to find the audio samples for a specific time.  However,
> there are many cases where this time line gets reset.  Usually, each
> callback has a time stamp that occurs precisely after the previous
> callback.  Obviously, the audio should not glitch when the time line
> is contiguous, and thus the data must be sample-accurate.  However,
> CoreAudio code must also deal with situations where the time line
> starts over from 0, usually under control of the host application.
> CoreAudio also has a flag in the callback to indicate when the
> buffers are totally silent.  I'd like to borrow these ideas, or at
> least similarly-inspired ideas, and have FLAC streaming designed such
> that the stream can tell the playback software when to reset.
> The typical process to deal with synchronization of separate systems
> is sample rate conversion.  However, this introduces distortion into
> the audio, especially with real-time SRC.  The only way to avoid SRC
> is to have some way to reset the alignment without dropping or adding
> samples.  As I said above, if the broadcast server were to put flags
> in the stream to indicate silent breaks in the audio, then the
> playback client could drop silent samples or insert silent samples
> until the two time lines are resynchronized.  But, since this would
> only add or remove silence, there should be absolutely no audible
> glitch.  Perhaps the stream would need more than simple silent flags,
> or resync flags.  It might be necessary to transmit an actual running
> time line counter, with enough bits to count the longest stretch of
> contiguously-clocked audio blocks.  When the broadcast server sees a
> break in the content material, the time code could be reset to zero,
> and this would tell the client to start the sync over, thus avoiding
> dropped samples in the middle of real audio content.
>>>> Anyway what could happen is the client could do a little bit of
>>>> re-sampling here or there to ensure its in sync with the servers
>>>> clock.
>>> That is how streaming clients usually solve this problem, although
>>> is not
>>> really improving sound quality.
>> Its probably not a big deal if you don't resample all the time, just
>> when your off by X amount, all of this would just be client side
>> preferences. As long as the client side "knows" its off by X amount
>> you could handle it in any number of ways, I'd be fine if its just
>> crossfaded to the correct timing if was off by more than half a
>> second, then no resampling would ever happen, you would just get a
>> weird effect about once an hour, better than a buffer underrun or lag,
>> or perhaps the client could look for a half second of silence and just
>> cut it out.
> I don't think it's a good idea to resample just some of the time,
> although your idea to crossfade would work since it never resamples.
> I think that there are a number of PC-based digital audio playback
> systems, and perhaps even in the television broadcast industry, where
> this idea of intermittent resampling is done.  I hear a regular
> glitch in audio about once per second in many syndicated television
> shows, and my suspicion is that they are speeding up the show so that
> they can sell more commercial time.  Another place that I hear this
> glitching is in some of the PC audio software oriented for DJs which
> can play MP3 files at different speeds and mix them together.  I hear
> the same sound - one glitch per second - and it is very annoying.
> But, as you said, a crossfade once per hour would not be as bad.
> Also, the stream could be completely resynchronized even without a
> crossfade.  Some streaming servers are so bad that they can't run for
> hours without rebuffering, but I guess it's probably pretty lazy to
> design something that does that on purpose (the rebuffering, that
> is).  However, as I suggested, it might be better if the broadcast
> server gives hints so that the client player can do these crossfades
> during the silence between tracks.  Using my idea, you'd need to
> "crossfade" more than once per hour, because there probably isn't
> enough silence to handle it that seldom.  But a fraction of a second
> between tracks several times per hour would never be noticed, unless
> there is a continuous audio broadcast with absolutely no silence.
> Brian Willoughby
> Sound Consulting
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