Yep, if we keep this up, it could even become a habit!
PulseAudio 1.1 is out. It’s mostly a bunch of bug fixes on top of 1.0. Most important of these are fixes for: a libpulse dependency on libsamplerate (if enabled) which would make our LGPL license invalid, broken Skype audio capture (because we changed from a 3 number version to 2 numbers), broken startup without a DBus session bus running, and not going crazy on USB disconnects.
This should be a very safe upgrade, so grab it while it’s hot!
I’ve just pushed a bunch of patches by Pierre-Louis Bossart that can have a pretty decent CPU/power impact. These introduce the concept of an “alternate sample rate”.
Currently, PulseAudio runs all your devices at a default sample rate, which is set to 44.1 kHz on most systems (this can be configured). All streams running at different sample rates are resampled to this sample rate. Pierre’s patches add an alternate sample rate that we try to switch to under certain circumstances if it means that we can save on resampling cost. This would happen if the stream uses exactly the alternate sample rate, or some integral-or-so multiple of it.
The default value for the alternate sample rate is 48 kHz. So if you’re playing a movie off a DVD where the audio track is typically a 48 kHz stream, and your card supports it, we switch to 48 kHz and avoid resampling altogether. Similarly, while making voice calls, common sample rates are 8, 16, and 32 kHz. These can be resampled to 48 kHz much faster than to 44.1 kHz.
Now for the big caveat — this won’t work if there’s any other stream connected to your sink/source. So if your music player is playing (or even paused) when you get that voip call, we can’t update the rate. This situation can probably be improved by at least allowing corked streams have their sample rate change (so having some random stream connected but not playing — I’m looking at you, Flash! — won’t block rate updates altogether). Hopefully we’ll get this fixed before this feature is released in PulseAudio 2.0.
Thanks to Pierre for all his work on this, and to my company, Collabora, for giving me some time for upstream work!
Prague is an interesting place to be at this time of the year — next week it’s playing host to the Real Time Linux Workshop. The week after that, we have the Kernel Summit, GStreamer Conference, Embedded Linux Conference Europe and LinuxCon Europe. I’m going to be at the last 3, and there’s some great audio stuff happening!
On the evening of Oct 23rd, we’re having an Audio BoF to discuss pending issues that cut across the stack — ALSA, PulseAudio, GStreamer and any other similar bits that people have complaints about.
Then there’s GstConf, where there are going to be a bunch of awesome talks. Also included is a talk by yours truly about recent developments in the PulseAudio world.
At some point during that week, possibly Oct 26th morning, plans are afoot to have an ALSA BoF to discuss the state and future of the HDA driver.
There are also rumours of excellent beer that will need to be scrupulously verified. ;)
It’s going to be a pretty exciting week!
As Colin Guthrie reports, PulseAudio 1.0 is now out the door! There’s a lot of new things in the release, and we should be getting a much more regular release schedule going. Head over to the full release notes for more details.
A lot of people have contributed to this release and thanks to them all. Special props to Colin all the patch-herding, tireless help, and code ninjutsu!
p.s.: Gentoo packages are already available, of course. :)
In an unsurprising turn of events, Adobe completely fails to play well with modern Linux systems. Well done, guys. Well done, indeed.
p.s.: I was quite happy to see that the Google Talk plugin has proper PulseAudio support (thanks to the WebRTC née GIPS code, it looks like).
I’m going to be at the Linux Plumbers’ Conference next week, speaking about the things we’ve been doing to make passthrough audio on Linux kick ass.
If you’re around and interested, do drop by!
I have a secret to confess. I’ve spent a great deal of time over the last few months talking to myself. I can’t say I haven’t enjoyed it — it turns out my capacity to entertain myself is far greater than initially suspected. But I hear you ask … why?
Here at Collabora, I’ve been building on Wim’s previous work on adding echo cancellation to PulseAudio. Thanks go to Intel for supporting us in continuing this work. Before too long, all this work will be trickling down to your favourite Linux distribution and all your friends will stop hating you.
First, a quick recap on what acoustic echo cancellation (AEC) is. If you already know this, you might want to skip this paragraph and the next. Say you’re on your laptop, and you receive a voice call from your friend. You don’t have a pair of headphones lying around, so you’re just going to use your laptop’s built-in speakers and mic. When your friend speaks, what she says is played out the speakers, but is also captured by the microphone and she gets to hear herself speak, albeit a short while (a few hundred milliseconds or more) later. This is called acoustic echo, and can be frustrating enough to make conversation nigh impossible. There are other types of echo for phone systems, but that’s not interesting to us at the moment.
This problem is common on pretty much all devices that you use to make phone calls. Astute readers will ask why they don’t actually face this problem on their phone. That’s because your phone (or, if you have a cheap phone, your phone company) has special software hidden away that removes the echo before sending your signal along to the other end. On laptops, which are general-purpose hardware, the job of echo cancellation is left to either your operating system (Windows XP onwards, for example) or your chat client (Skype, for example) to provide.
On Linux, we implement echo cancellation as a PulseAudio module (code-ninja Wim Taymans wrote this last year). We use the Speex DSP library to perform the actual echo cancellation. The code’s quite modular, so it’s not very hard to plug in alternate echo cancellers (we even include an alternate implementation, which isn’t quite as effective as Speex).
Recently, we plugged in some more bits from the Speex library to do noise suppression and digital gain control (so you can quit twiddling with your mic volume for the other end to be able to hear you). We also added a bunch of fixes to reduce CPU consumption significantly — this should be good enough to run on a netbook and reasonably recent ARM platforms.
While all this sounds nice, I think a demo would sound (haha!) nicer …
Without AEC: /downloads/pulseaudio/aec/call-no-aec (or download ogg, aac)
With AEC: /downloads/pulseaudio/aec/call-with-aec (or download ogg, aac)
This is a recording of a call between my laptop and N900. The laptop is playing audio out the speakers and recording with the built-in mic. What you hear is the conversation as heard on the N900.
All this echo cancelling goodness will come to a Linux distribution near you in the upcoming 1.0 release of PulseAudio. The next version of the GNOME IM client, Empathy (3.2), will actually make use of this functionality. In due time, we intend to make it so that all voice applications will end up using this functionality (so if you’re writing a VoIP application and don’t want to use this functionality, you need to set a special stream property to disable this — filter.suppress="echo-cancel").
For the impatient among you, you can try all this out by getting recent testing versions of PulseAudio (I know packages are available for Ubuntu, Debian, Gentoo and Mageia at least). To force your phone streams to use echo cancellation, just run pactl load-module module-echo-cancel, and you’re done.
There’s still some work to be done, refining quality and using other AEC implementations (in the short-term, the WebRTC one looks promising). Things don’t work at all if you’re using different devices for playback and capture (e.g. laptop speakers and webcam mic). These are things that will be addressed in coming weeks and months.
I’m in Berlin at the Desktop Summit, so you can drop me a note and we can meet if you want to yell about PulseAudio things that annoy you (or even, y’know, things you like).
[tl;dr — if you’re using GNOME or a GStreamer-based player, not using the Rhythmbox crossfading backend, and want to try to save ~0.5 W of power, jump to end of the post]
Lennart pointed to another blog post about actually putting PulseAudio’s power-saving capabilities to use on your system. The latter provides a hack-ish way to increase buffering in PulseAudio to the maximum possible, reducing the number of wakeups. I’m going to talk about that a bit.
Summarising the basic idea, we want music players to decode a large chunk of data and give it to PA so that we can then fill up ALSA’s hardware buffer, sleep till it’s almost completely consumed, fill it again, sleep, repeat. More details in this post from Lennart.
The native GNOME audio/video players don’t talk to PulseAudio directly — they use GStreamer, which has a pulsesink element that actually talks to PulseAudio. We could configure things so that we send a large amount (say 2 seconds’ worth) to PulseAudio, sleep, and then wake up periodically to push out more. Now in the audio player (say Rhythmbox), the user hits next, prev, or pause. We need to effect this change immediately, even though we’ve already sent out 2 seconds of data (it would suck if you hit pause and the actual pause happened 2 seconds later, wouldn’t it?). PulseAudio already solves because it can internally “rewind” the buffer and overwrite it if required. GStreamer can and does take advantage of this by sending pause and other control messages out of band from the data.
This all works well for relatively simple GStreamer pipelines. However, if you want to do something more complicated, like Rhythmbox’ crossfading backend, things start to break. PulseAudio doesn’t offer an API to do fades, and since we don’t do rewinds in GStreamer, we need to apply effects such as fades with a latency equal to the amount of buffering we’re asking PulseAudio to do. This makes for unhappy users.
Well, all is not as bleak as it seems. There was some discussion on the PA mailing list, and the need for a proper fade API (really, a generic effects API) is clear. There have even been attempts to solve this in GStreamer.
But you want to save 0.5 W of power now! Okay, if you’re not using the Rhythmbox crossfading backend (or are okay with disabling it), this will make Rhythmbox, Banshee, pre-3.0 Totem (and really any GNOMEy player that uses gconfaudiosink, which will soon be replaced by gsettingsaudiosink, I guess), you can run this on the command line:
gconftool-2 --type string \
--set /system/gstreamer/0.10/default/musicaudiosink \
"pulsesink latency-time=100000 buffer-time=2000000"
On my machine, this brings down the number of wakeups per second because of alsa-sink to ~2.7 (corresponding nicely to the ~350ms of hardware buffer that I have). With Totem 3.0, this may or may not work, depending on whether your distribution gives gconfaudiosink a higher rank than pulseaudiosink.
This is clearly just a stop-gap till we can get things done the Right Way™ at the system level, so really, if things break, you get to keep the pieces. If you need to, you can undo this change by running the same command without the latency-time=… and buffer-time=… bits. That said, if something does break, do leave a comment below so I can add it to the list of things that we need to test the final solution with.
Yesterday was my last day at NVidia. I’ve worked with the Embedded Software team there for the last 15 months, specifically on the system software for a Linux based stack that you will see some time next year. I’ve had a great time there, learning new things, and doing everything from tweaking bit-banging I²C implementations with a CRO to tracking down alleged compiler bugs (I’m looking at you -fstrict-aliasing) by wading through ARM assembly.
As some of you might already know, my next step, which has had me bouncing off the walls for the last month, is to join the great folks at Collabora Multimedia working on the PulseAudio sound server. I’ll be working from home here, in Bangalore (in your face, 1.5-hour commute!). It is incredibly exciting for me to be working with a talented bunch of folks and actively contributing to open source software as part of my work!
More updates as they happen. :)