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Tue, 08 Apr 2008

What's Cooking in PulseAudio's glitch-free Branch

A while ago I started development of special branch of PulseAudio which is called glitch-free. In a few days I will merge it back to PulseAudio trunk, and eventually release it as 0.9.11. I think it's time to explain a little what all this "glitch-freeness" is about, what made it so tricky to implement, and why this is totally awesome technology. So, here we go:

Traditional Playback Model

Traditionally on most operating systems audio is scheduled via sound card interrupts (IRQs). When an application opens a sound card for playback it configures it for a fixed size playback buffer. Then it fills this buffer with digital PCM sample data. And after that it tells the hardware to start playback. Then, the hardware reads the samples from the buffer, one at a time, and passes it on to the DAC so that eventually it reaches the speakers.

After a certain number of samples played the sound hardware generates an interrupt. This interrupt is forwarded to the application. On Linux/Unix this is done via poll()/select(), which the application uses to sleep on the sound card file descriptor. When the application is notified via this interrupt it overwrites the samples that were just played by the hardware with new data and goes to sleep again. When the next interrupt arrives the next block of samples is overwritten, and so on and so on. When the hardware reaches the end of the hardware buffer it starts from its beginning again, in a true ring buffer fashion. This goes on and on and on.

The number of samples after which an interrupt is generated is usually called a fragment (ALSA likes to call the same thing a period for some reason). The number of fragments the entire playback buffer is split into is usually integral and usually a power of two, 2 and 4 being the most frequently used values.

Schematic overview
Image 1: Schematic overview of the playback buffer in the traditional playback model, in the best way the author can visualize this with his limited drawing abilities.

If the application is not quick enough to fill up the hardware buffer again after an interrupt we get a buffer underrun ("drop-out"). An underrun is clearly hearable by the user as a discontinuity in audio which is something we clearly don't want. We thus have to carefully make sure that the buffer and fragment sizes are chosen in a way that the software has enough time to calculate the data that needs to be played, and the OS has enough time to forward the interrupt from the hardware to the userspace software and the write request back to the hardware.

Depending on the requirements of the application the size of the playback buffer is chosen. It can be as small as 4ms for low-latency applications (such as music synthesizers), or as long as 2s for applications where latency doesn't matter (such as music players). The hardware buffer size directly translates to the latency that the playback adds to the system. The smaller the fragment sizes the application configures, the more time the application has to fill up the playback buffer again.

Let's formalize this a bit: Let BUF_SIZE be the size of the hardware playback buffer in samples, FRAG_SIZE the size of one fragment in samples, and NFRAGS the number of fragments the buffer is split into (equivalent to BUF_SIZE divided by FRAG_SIZE), RATE the sampling rate in samples per second. Then, the overall latency is identical to BUF_SIZE/RATE. An interrupt is generated every FRAG_SIZE/RATE. Every time one of those interrupts is generated the application should fill up one fragment again, if it missed one interrupt this might become more than one. If it doesn't miss any interrupt it has (NFRAGS-1)*FRAG_SIZE/RATE time to fulfill the request. If it needs more time than this we'll get an underrun. The fill level of the playback buffer should thus usually oscillate between BUF_SIZE and (NFRAGS-1)*FRAG_SIZE. In case of missed interrupts it might however fall considerably lower, in the worst case to 0 which is, again, an underrun.

It is difficult to choose the buffer and fragment sizes in an optimal way for an application:

As you can easily see it is impossible to choose buffering metrics in a way that they are optimal on all four requirements.

This traditional model has major drawbacks:

Due to the limitations of this model most current (Linux/Unix) software uses buffer metrics that turned out to "work most of the time", very often they are chosen without much thinking, by copying other people's code, or totally at random.

PulseAudio <= 0.9.10 uses a fragment size of 25ms by default, with four fragments. That means that right now, unless you reconfigure your PulseAudio manually clients will not get latencies lower than 100ms whatever you try, and as long as music is playing you will get 40 interrupts/s. (The relevant configuration options for PulseAudio are default-fragments= and default-fragment-size-msec= in daemon.conf)

dmix uses 16 fragments by default with a size of 21 ms each (on my system at least -- this varies, depending on your hardware). You can't get less than 47 interrupts/s. (You can change the parameters in .asoundrc)

So much about the traditional model and its limitations. Now, we'll have a peek on how the new glitch-free branch of PulseAudio does its things. The technology is not really new. It's inspired by what Vista does these days and what Apple CoreAudio has already been doing for quite a while. However, on Linux this technology is new, we have been lagging behind quite a bit. Also I claim that what PA does now goes beyond what Vista/MacOS does in many ways, though of course, they provide much more than we provide in many other ways. The name glitch-free is inspired by the term Microsoft uses to call this model, however I must admit that I am not sure that my definition of this term and theirs actually is the same.

Glitch-Free Playback Model

The first basic idea of the glitch-free playback model (a better, less marketingy name is probably timer-based audio scheduling which is the term I internally use in the PA codebase) is to no longer depend on sound card interrupts to schedule audio but use system timers instead. System timers are far more flexible then the fragment-based sound card timers. They can be reconfigured at any time, and have a granularity that is independant from any buffer metrics of the sound card. The second basic idea is to use playback buffers that are as large as possible, up to a limit of 2s or 5s. The third basic idea is to allow rewriting of the hardware buffer at any time. This allows instant reaction on user-input (i.e. pause/seek requests in your music player, or instant event sounds) although the huge latency imposed by the hardware playback buffer would suggest otherwise.

PA configures the audio hardware to the largest playback buffer size possible, up to 2s. The sound card interrupts are disabled as far as possible (most of the time this means to simply lower NFRAGS to the minimal value supported by the hardware. It would be great if ALSA would allow us to disable sound card interrupts entirely). Then, PA constantly determines what the minimal latency requirement of all connected clients is. If no client specified any requirements we fill up the whole buffer all the time, i.e. have an actual latency of 2s. However, if some applications specified requirements, we take the lowest one and only use as much of the configured hardware buffer as this value allows us. In practice, this means we only partially fill the buffer each time we wake up. Then, we configure a system timer to wake us up 10ms before the buffer would run empty and fill it up again then. If the overall latency is configured to less than 10ms we wakeup after half the latency requested.

If the sleep time turns out to be too long (i.e. it took more than 10ms to fill up the hardware buffer) we will get an underrun. If this happens we can double the time we wake up before the buffer would run empty, to 20ms, and so on. If we notice that we only used much less than the time we estimated, we can halve this value again. This adaptive scheme makes sure that in the unlikely event of a buffer underrun it will happen most likely only once and never again.

When a new client connects or an existing client disconnects, or when a client wants to rewrite what it already wrote, or the user wants to change the volume of one of the streams, then PA will resample its data passed by the client, convert it to the proper hardware sample type, and remix it with the data of the other clients. This of course makes it necessary to keep a "history" of data of all clients around so that if one client requests a rewrite we have the necessary data around to remix what already was mixed before.

The benefits of this model are manyfold:

Of course, this scheme also comes with major complications:

The advantages of the scheme clearly outweigh the complexities it causes. Especially the power-saving features of glitch-free PA should be enough reason for the embedded Linux people to adopt it quickly. Make PA disappear from powertop even if you play music!

The code in the glitch-free is still rough and sometimes incomplete. I will merge it shortly into trunk and then upload a snapshot to Rawhide.

I hope this text also explains to the few remaining PA haters a little better why PA is a good thing, and why everyone should have it on his Linux desktop. Of course these changes are not visible on the surface, my hope with this blog story is to explain a bit better why infrastructure matters, and counter misconceptions what PA actually is and what it gives you on top of ALSA.

posted at: 19:54 | path: /projects | permanent link to this entry | comments


It should be obvious but in case it isn't: the opinions reflected here are my own. They are not the views of my employer, or Ronald McDonald, or anyone else.

Please note that I take the liberty to delete any comments posted here that I deem inappropriate, off-topic, or insulting. And I excercise this liberty quite agressively. So yes, if you comment here, I might censor you. If you don't want to be censored you are welcome to comment on your own blog instead.


Lennart Poettering <mzoybt (at) 0pointer (dot) net>
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