Tag: pulseaudio (page 1 of 4)

Update from the PipeWire hackfest

As the third and final day of the PipeWire hackfest draws to a close, I thought I’d summarise some of my thoughts on the goings-on and the future.


Before I get into the details, I want to send out a big thank you to:

  • Christian Schaller for all the hard work of organising the event and Wim Taymans for the work on PipeWire so far (and in the future)
  • The GNOME Foundation, for sponsoring the event as a whole
  • Qualcomm, who are funding my presence at the event
  • Collabora, for sponsoring dinner on Monday
  • Everybody who attended and participate for their time and thoughtful comments


For those of you who are not familiar with it, PipeWire (previously Pinos, previously PulseVideo) was Wim’s effort at providing secure, multi-program access to video devices (like webcams, or the desktop for screen capture). As he went down that rabbit hole, he wrote SPA, a lightweight general-purpose framework for representing a streaming graph, and this led to the idea of expanding the project to include support for low latency audio.

The Linux userspace audio story has, for the longest time, consisted of two top-level components: PulseAudio which handles consumer audio (power efficiency, wide range of arbitrary hardware), and JACK which deals with pro audio (low latency, high performance). Consolidating this into a good out-of-the-box experience for all use-cases has been a long-standing goal for myself and others in the community that I have spoken to.

An Opportunity

From a PulseAudio perspective, it has been hard to achieve the 1-to-few millisecond latency numbers that would be absolutely necessary for professional audio use-cases. A lot of work has gone into improving this situation, most recently with David Henningsson’s shared-ringbuffer channels that made client/server communication more efficient.

At the same time, as application sandboxing frameworks such as Flatpak have added security requirements of us that were not accounted for when PulseAudio was written. Examples including choosing which devices an application has access to (or can even know of) or which applications can act as control entities (set routing etc., enable/disable devices). Some work has gone into this — Ahmed Darwish did some key work to get memfd support in PulseAudio, and Wim has prototyped an access-control mechanism module to enable a Flatpak portal for sound.

All this said, there are still fundamental limitations in architectural decisions in PulseAudio that would require significant plumbing to address. With Wim’s work on PipeWire and his extensive background with GStreamer and PulseAudio itself, I think we have an opportunity to revisit some of those decisions with the benefit of a decade’s worth of learning deploying PulseAudio in various domains starting from desktops/laptops to phones, cars, robots, home audio, telephony systems and a lot more.

Key Ideas

There are some core ideas of PipeWire that I am quite excited about.

The first of these is the graph. Like JACK, the entities that participate in the data flow are represented by PipeWire as nodes in a graph, and routing between nodes is very flexible — you can route applications to playback devices and capture devices to applications, but you can also route applications to other applications, and this is notionally the same thing.

The second idea is a bit more radical — PipeWire itself only “runs” the graph. The actual connections between nodes are created and managed by a “session manager”. This allows us to completely separate the data flow from policy, which means we could write completely separate policy for desktop use cases vs. specific embedded use cases. I’m particularly excited to see this be scriptable in a higher-level language, which is something Bastien has already started work on!

A powerful idea in PulseAudio was rewinding — the ability to send out huge buffers to the device, but the flexibility to rewind that data when things changed (a new stream got added, or the stream moved, or the volume changed). While this is great for power saving, it is a significant amount of complexity in the code. In addition, with some filters in the data path, rewinding can break the algorithm by introducing non-linearity. PipeWire doesn’t support rewinds, and we will need to find a good way to manage latencies to account for low power use cases. One example is that we could have the session manager bump up the device latency when we know latency doesn’t matter (Android does this when the screen is off).

There are a bunch of other things that are in the process of being fleshed out, like being able to represent the hardware as a graph as well, to have a clearer idea of what is going on within a node. More updates as these things are more concrete.

The Way Forward

There is a good summary by Christian about our discussion about what is missing and how we can go about trying to make a smooth transition for PulseAudio users. There is, of course, a lot to do, and my ideal outcome is that we one day flip a switch and nobody knows that we have done so.

In practice, we’ll need to figure out how to make this transition seamless for most people, while folks with custom setup will need to be given a long runway and clear documentation to know what to do. It’s way to early to talk about this in more specifics, however.


One key thing that PulseAudio does right (I know there are people who disagree!) is having a custom configuration that automagically works on a lot of Intel HDA-based systems. We’ve been wondering how to deal with this in PipeWire, and the path we think makes sense is to transition to ALSA UCM configuration. This is not as flexible as we need it to be, but I’d like to extend it for that purpose if possible. This would ideally also help consolidate the various methods of configuration being used by the various Linux userspaces.

To that end, I’ve started trying to get a UCM setup on my desktop that PulseAudio can use, and be functionally equivalent to what we do with our existing configuration. There are missing bits and bobs, and I’m currently focusing on the ones related to hardware volume control. I’ll write about this in the future as the effort expands out to other hardware.

Onwards and upwards

The transition to PipeWire is unlikely to be quick or completely-painless or free of contention. For those who are worried about the future, know that any switch is still a long way away. In the mean time, however, constructive feedback and comments are welcome.

A Late GUADEC 2017 Post

It’s been a little over a month since I got back from Manchester, and this post should’ve come out earlier but I’ve been swamped.

The conference was absolutely lovely, the organisation was a 110% on point (serious kudos, I know first hand how hard that is). Others on Planet GNOME have written extensively about the talks, the social events, and everything in between that made it a great experience. What I would like to write about is about why this year’s GUADEC was special to me.

GNOME turning 20 years old is obviously a large milestone, and one of the main reasons I wanted to make sure I was at Manchester this year. There were many occasions to take stock of how far we had come, where we are, and most importantly, to reaffirm who we are, and why we do what we do.

And all of this made me think of my own history with GNOME. In 2002/2003, Nat and Miguel came down to Bangalore to talk about some of the work they were doing. I know I wasn’t the only one who found their energy infectious, and at Linux Bangalore 2003, they got on stage, just sat down, and started hacking up a GtkMozEmbed-based browser. The idea itself was fun, but what I took away — and I know I wasn’t the only one — is the sheer inclusive joy they shared in creating something and sharing that with their audience.

For all of us working on GNOME in whatever way we choose to contribute, there is the immediate gratification of shaping this project, as well as the larger ideological underpinning of making everyone’s experience talking to their computers better and free-er.

But I think it is also important to remember that all our efforts to make our community an inviting and inclusive space have a deep impact across the world. So much so that complete strangers from around the world are able to feel a sense of belonging to something much larger than themselves.

I am excited about everything we will achieve in the next 20 years.

(thanks go out to the GNOME Foundation for helping me attend GUADEC this year)

Sponsored by GNOME!

Beamforming in PulseAudio

In case you missed it — we got PulseAudio 9.0 out the door, with the echo cancellation improvements that I wrote about. Now is probably a good time for me to make good on my promise to expand upon the subject of beamforming.

As with the last post, I’d like to shout out to the wonderful folks at Aldebaran Robotics who made this work possible!


Beamforming as a concept is used in various aspects of signal processing including radio waves, but I’m going to be talking about it only as applied to audio. The basic idea is that if you have a number of microphones (a mic array) in some known arrangement, it is possible to “point” or steer the array in a particular direction, so sounds coming from that direction are made louder, while sounds from other directions are rendered softer (attenuated).

Practically speaking, it should be easy to see the value of this on a laptop, for example, where you might want to focus a mic array to point in front of the laptop, where the user probably is, and suppress sounds that might be coming from other locations. You can see an example of this in the webcam below. Notice the grilles on either side of the camera — there is a microphone behind each of these.

Webcam with 2 mics

Webcam with 2 mics

This raises the question of how this effect is achieved. The simplest approach is called “delay-sum beamforming”. The key idea in this approach is that if we have an array of microphones that we want to steer the array at a particular angle, the sound we want to steer at will reach each microphone at a different time. This is illustrated below. The image is taken from this great article describing the principles and math in a lot more detail.

Delay-sum beamforming

Delay-sum beamforming

In this figure, you can see that the sound from the source we want to listen to reaches the top-most microphone slightly before the next one, which in turn captures the audio slightly before the bottom-most microphone. If we know the distance between the microphones and the angle to which we want to steer the array, we can calculate the additional distance the sound has to travel to each microphone.

The speed of sound in air is roughly 340 m/s, and thus we can also calculate how much of a delay occurs between the same sound reaching each microphone. The signal at the first two microphones is delayed using this information, so that we can line up the signal from all three. Then we take the sum of the signal from all three (actually the average, but that’s not too important).

The signal from the direction we’re pointing in is going to be strongly correlated, so it will turn out loud and clear. Signals from other directions will end up being attenuated because they will only occur in one of the mics at a given point in time when we’re summing the signals — look at the noise wavefront in the illustration above as an example.


(this section is a bit more technical than the rest of the article, feel free to skim through or skip ahead to the next section if it’s not your cup of tea!)

The devil is, of course, in the details. Given the microphone geometry and steering direction, calculating the expected delays is relatively easy. We capture audio at a fixed sample rate — let’s assume this is 32000 samples per second, or 32 kHz. That translates to one sample every 31.25 µs. So if we want to delay our signal by 125µs, we can just add a buffer of 4 samples (4 × 31.25 = 125). Sound travels about 4.25 cm in that time, so this is not an unrealistic example.

Now, instead, assume the signal needs to be delayed by 80 µs. This translates to 2.56 samples. We’re working in the digital domain — the mic has already converted the analog vibrations in the air into digital samples that have been provided to the CPU. This means that our buffer delay can either be 2 samples or 3, not 2.56. We need another way to add a fractional delay (else we’ll end up with errors in the sum).

There is a fair amount of academic work describing methods to perform filtering on a sample to provide a fractional delay. One common way is to apply an FIR filter. However, to keep things simple, the method I chose was the Thiran approximation — the literature suggests that it performs the task reasonably well, and has the advantage of not having to spend a whole lot of CPU cycles first transforming to the frequency domain (which an FIR filter requires)(edit: converting to the frequency domain isn’t necessary, thanks to the folks who pointed this out).

I’ve implemented all of this as a separate module in PulseAudio as a beamformer filter module.

Now it’s time for a confession. I’m a plumber, not a DSP ninja. My delay-sum beamformer doesn’t do a very good job. I suspect part of it is the limitation of the delay-sum approach, partly the use of an IIR filter (which the Thiran approximation is), and it’s also entirely possible there is a bug in my fractional delay implementation. Reviews and suggestions are welcome!

A Better Implementation

The astute reader has, by now, realised that we are already doing a bunch of processing on incoming audio during voice calls — I’ve written in the previous article about how the webrtc-audio-processing engine provides echo cancellation, acoustic gain control, voice activity detection, and a bunch of other features.

Another feature that the library provides is — you guessed it — beamforming. The engineers at Google (who clearly are DSP ninjas) have a pretty good beamformer implementation, and this is also available via module-echo-cancel. You do need to configure the microphone geometry yourself (which means you have to manually load the module at the moment). Details are on our wiki (thanks to Tanu for that!).

How well does this work? Let me show you. The image below is me talking to my laptop, which has two microphones about 4cm apart, on either side of the webcam, above the screen. First I move to the right of the laptop (about 60°, assuming straight ahead is 0°). Then I move to the left by about the same amount (the second speech spike). And finally I speak from the center (a couple of times, since I get distracted by my phone).

The upper section represents the microphone input — you’ll see two channels, one corresponding to each mic. The bottom part is the processed version, with echo cancellation, gain control, noise suppression, etc. and beamforming.

WebRTC beamforming

WebRTC beamforming

You can also listen to the actual recordings …

… and the processed output.

Feels like black magic, doesn’t it?

Finishing thoughts

The webrtc-audio-processing-based beamforming is already available for you to use. The downside is that you need to load the module manually, rather than have this automatically plugged in when needed (because we don’t have a way to store and retrieve the mic geometry). At some point, I would really like to implement a configuration framework within PulseAudio to allow users to set configuration from some external UI and have that be picked up as needed.

Nicolas Dufresne has done some work to wrap the webrtc-audio-processing library functionality in a GStreamer element (and this is in master now). Adding support for beamforming to the element would also be good to have.

The module-beamformer bits should be a good starting point for folks who want to wrap their own beamforming library and have it used in PulseAudio. Feel free to get in touch with me if you need help with that.

Audio Devices and Configuration

This one’s going to be a bit of a long post. You might want to grab a cup of coffee before you jump in!

Over the last few years, I’ve spent some time getting PulseAudio up and running on a few Android-based phones. There was the initial Galaxy Nexus port, a proof-of-concept port of Firefox OS (git) to use PulseAudio instead of AudioFlinger on a Nexus 4, and most recently, a port of Firefox OS to use PulseAudio on the first gen Moto G and last year’s Sony Xperia Z3 Compact (git).

The process so far has been largely manual and painstaking, and I’ve been trying to make that easier. But before I talk about the how of that, let’s see how all this works in the first place.

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A Quick Update

Happy 2016 everyone!

While I did mention a while back (almost two years ago, wow) that I was taking a break, I realised recently that I hadn’t posted an update from when I started again.

For the last year and a half, I’ve been providing freelance consulting around PulseAudio, GStreamer, and various other directly and tangentially related projects. There’s a brief list of the kind of work I’ve been involved in.

If you’re looking for help with PulseAudio, GStreamer, multimedia middleware or anything else you might’ve come across on this blog, do get in touch!

PulseAudio 7.1 is out

We just rolled out a minor bugfix release. Quick changelog:

  • Fix a crasher when using srbchannel
  • Fix a build system typo that caused symlinks to turn up in /
  • Make Xonar cards work better
  • Other minor bug fixes and improvements

More details on the mailing list.

Thanks to everyone who contributed with bug reports and testing. What isn’t generally visible is that a lot of this happens behind the scenes downstream on distribution bug trackers, IRC, and so forth.

PSA: Breaking webrtc-audio-processing API

I know it’s been ages, but I am now working on updating the webrtc-audio-processing library. You might remember this as the code that we split off from the webrtc.org code to use in the PulseAudio echo cancellation module.

This is basically just the AudioProcessing module, bundled as a standalone library so that we can use the fantastic AEC, AGC, and noise suppression implementation from that code base. For packaging simplicity, I made a copy of the necessary code, and wrote an autotools-based build system around that.

Now since I last copied the code, the library API has changed a bit — nothing drastic, just a few minor cleanups and removed API. This wouldn’t normally be a big deal since this code isn’t actually published as external API — it’s mostly embedded in the Chromium and Firefox trees, probably other projects too.

Since we are exposing a copy of this code as a standalone library, this means that there are two options — we could (a) just break the API, and all dependent code needs to be updated to be able to use the new version, or (b) write a small wrapper to try to maintain backwards compatibility.

I’m inclined to just break API and release a new version of the library which is not backwards compatible. My rationale for this is that I’d like to keep the code as close to what is upstream as possible, and over time it could become painful to maintain a bunch of backwards-compatibility code.

A nicer solution would be to work with upstream to make it possible to build the AudioProcessing module as a standalone library. While the folks upstream seemed amenable to the idea when this came up a few years ago, nobody has stepped up to actually do the work for this. In the mean time, a number of interesting features have been added to the module, and it would be good to pull this in to use in PulseAudio and any other projects using this code (more about this in a follow-up post).

So if you’re using webrtc-audio-processing, be warned that the next release will probably break API, and you’ll need to update your code. I’ll try to publish a quick update guide when releasing the code, but if you want to look at the current API, take a look at the current audio_processing.h.

p.s.: If you do use webrtc-audio-processing as a dependency, I’d love to hear about it. As far as I know, PulseAudio is the only user of this library at the moment.


This one’s a bit late, for reasons that’ll be clear enough later in this post. I had the happy opportunity to go to GUADEC in Gothenburg this year (after missing the last two, unfortunately). It was a great, well-organised event, and I felt super-charged again, meeting all the people making GNOME better every day.

GUADEC picnic @ Gothenberg

GUADEC picnic @ Gothenberg

I presented a status update of what we’ve been up to in the PulseAudio world in the past few years. Amazingly, all the videos are up already, so you can catch up with anything that you might have missed here.

We also had a meeting of PulseAudio developers which and a number of interesting topics of discussion came up (I’ll try to summarise my notes in a separate post).

A bunch of other interesting discussions happened in the hallways, and I’ll write about that if my investigations take me some place interesting.

Now the downside — I ended up missing the BoF part of GUADEC, and all of the GStreamer hackfest in Montpellier after. As it happens, I contracted dengue and I’m still recovering from this. Fortunately it was the lesser (non-haemorrhagic) version without any complications, so now it’s just a matter of resting till I’ve recuperated completely.

Nevertheless, the first part of the trip was great, and I’d like to thank the GNOME Foundation for sponsoring my travel and stay, without which I would have missed out on all the GUADEC fun this year.

Sponsored by GNOME!

Sponsored by GNOME!

GNOME Asia 2015

I was in Depok, Indonesia last week to speak at GNOME Asia 2015. It was a great experience — the organisers did a fantastic job and as a bonus, the venue was incredibly pretty!

View from our room

View from our room

My talk was about the GNOME audio stack, and my original intention was to talk a bit about the APIs, how to use them, and how to choose which to use. After the first day, though, I felt like a more high-level view of the pieces would be more useful to the audience, so I adjusted the focus a bit. My slides are up here.

Nirbheek and I then spent a couple of days going down to Yogyakarta to cycle around, visit some temples, and sip some fine hipster coffee.

All in all, it was a week well spent. I’d like to thank the GNOME Foundation for helping me get to the conference!

Sponsored by GNOME!

Sponsored by GNOME!

Notes from the PulseAudio Mini Summit 2014

The third week of October was quite action-packed, with a whole bunch of conferences happening in Düsseldorf. The Linux audio developer community as well as the PulseAudio developers each had a whole day of discussions related to a wide range of topics. I’ll be summarising the events of the PulseAudio mini summit day here. The discussion was split into two parts, the first half of the day with just the current core developers and the latter half with members of the community participating as well.

I’d like to thank the Linux Foundation for sparing us a room to carry out these discussions — it’s fantastic that we are able to colocate such meetings with a bunch of other conferences, making it much easier than it would otherwise be for all of us to converge to a single place, hash out ideas, and generally have a good time in real life as well!

Incontrovertible proof that all our users are happy

Happy faces — incontrovertible proof that everyone loves PulseAudio!

With a whole day of discussions, this is clearly going to be a long post, so you might want to grab a coffee now. :)

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