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.
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:
- The buffer size should be as large as possible to minimize the
risk of drop-outs.
- The buffer size should be as small as possible to guarantee
minimal latencies.
- The fragment size should be as large as possible to minimize the
number of interrupts, and thus the required CPU time used, to maximize
the time the CPU can sleep for between interrupts and thus the battery
lifetime (i.e. the fewer interrupts are generated the lower your audio
app will show up in powertop, and that's what all is about,
right?)
- The fragment size should be as small as possible to give the
application as much time as possible to fill up the playback buffer,
to minimize drop-outs.
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:
- The buffering metrics are highly dependant on what the sound hardware
can provide. Portable software needs to be able to deal with hardware
that can only provide a very limited set of buffer and fragment
sizes.
- The buffer metrics are configured only once, when the device is
opened, they usually cannot be reconfigured during playback without
major discontinuities in audio. This is problematic if more than one
application wants to output audio at the same time via a sound server
(or dmix) and they have different requirements on
latency. For these sound servers/dmix the fragment metrics are
configured statically in a configuration file, and are the same during
the whole lifetime. If a client connects that needs lower latencies,
it basically lost. If a client connects that doesn't need as low
latencies, we will continouisly burn more CPU/battery than
necessary.
- It is practically impossible to choose the buffer metrics optimal
for your application -- there are too many variables in the equation:
you can't know anything about the IRQ/scheduling latencies of the
OS/machine your software will be running on; you cannot know how much
time it will actually take to produce the audio data that shall be
pushed to the audio device (unless you start counting cycles, which is
a good way to make your code unportable); the scheduling latencies are
hugely dependant on the system load on most current OSes (unless you
have an RT system, which we generally do not have). As said, for sound
servers/dmix it is impossible to know in advance what the requirements
on latency are that the applications that might eventually connect
will have.
- Since the number of fragments is integral and at least 2
on almost all existing hardware we will generate at least two interrupts
on each buffer iteration. If we fix the buffer size to 2s then we will
generate an interrupt at least every 1s. We'd then have 1s to fill up
the buffer again -- on all modern systems this is far more than we'd
ever need. It would be much better if we could fix the fragment size
to 1.9s, which still gives us 100ms to fill up the playback buffer
again, still more than necessary on most systems.
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:
- We minimize the overall number of interrupts, down to what the
latency requirements of the connected clients allow us. i.e. we save power,
don't show up in powertop anymore for normal music playback.
- We maximize drop-out safety, because we buffer up to 2s in the
usual cases. Only with operating systems which have scheduling
latencies > 2s we can still get drop-outs. Thankfully no
operating system is that bad.
- In the event of an underrun we don't get stuck in it, but instead
are able to recover quickly and can make sure it doesn't happen again.
- We provide "zero-latency". Each client can rewrite its playback
buffer at any time, and this is forwarded to the hardware, even if
this means that the sample currently being played needs to be
rewritten. This means much quicker reaction to user input, a more
responsive user experience.
- We become much less dependant on what the sound hardware provides
us with. We can configure wakeup times that are independant from the
fragment settings that the hardware actually supports.
- We can provide almost any latency a client might request,
dynamically without reconfiguration, without discontinuities in
audio.
Of course, this scheme also comes with major complications:
- System timers and sound card timers deviate. On many sound cards
by quite a bit. Also, not all sound cards allow the user to query the
playback frame index at any time, but only shortly after each IRQ. To
compensate for this deviation PA contains a non-trivial algorithm
which tries to estimate and follow the deviation over time. If this
doesn't work properly it might happen that an underrun happens much
earlier than we expected.
- System timers on Unix are not very high precision. On traditional
Linux with HZ=100 sleep times for timers are rounded up to multiples
of 10ms. Only very recent Linux kernels with hrtimers can
provide something better, but only on x86 and x86-64 until now. This
makes the whole scheme unusable for low latency setups unless you run
the very latest Linux. Also, hrtimers are not (yet) exposed in
poll()/select(). It requires major jumping through loops to
work around this limitation.
- We need to keep a history of sample data for each stream around, thus increasing the memory
footprint and potentially increased cache pressure. PA tries to work
against the increased memory footprint and cache pressure this might cause by doing
zero-copy memory management.
- We're still dependant on the maximum playback buffer size the
sound hardware supports. Many sound cards don't even support 2s, but only
300ms or suchlike.
- The rewriting of the client buffers causing rewriting of the
hardware buffer complicates the resampling/converting step
immensly. In general the code to implement this model is more complex
than for the traditional model. Also, ALSA has not really been
designed with this design in mind, which makes some things very hard
to get right and suboptimal.
- Generally, this works reliably only on newest ALSA, newest kernel,
newest everything. It has pretty steep requirements on software and
sometimes even on hardware. To stay comptible with systems that don't
fulfill these requirements we need to carry around code for the
traditional playback model as well, increasing the code base by far.
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.
