Tag: gstreamer

PulseConf 2012: Report

For those of you who missed my previous updates, we recently organised a PulseAudio miniconference in Copenhagen, Denmark last week. The organisation of all this was spearheaded by ALSA and PulseAudio hacker, David Henningsson. The good folks organising the Ubuntu Developer Summit / Linaro Connect were kind enough to allow us to colocate this event. A big thanks to both of them for making this possible!

The room where the first PulseAudio conference took place

The room where the first PulseAudio conference took place

The conference was attended by the four current active PulseAudio developers: Colin Guthrie, Tanu Kaskinen, David Henningsson, and myself. We were joined by long-time contributors Janos Kovacs and Jaska Uimonen from Intel, Luke Yelavich, Conor Curran and Michał Sawicz.

We started the conference at around 9:30 am on November 2nd, and actually managed to keep to the final schedule(!), so I’m going to break this report down into sub-topics for each item which will hopefully make for easier reading than an essay. I’ve also put up some photos from the conference on the Google+ event.

Mission and Vision

We started off with a broad topic — what each of our personal visions/goals for the project are. Interestingly, two main themes emerged: having the most seamless desktop user experience possible, and making sure we are well-suited to the embedded world.

Most of us expressed interest in making sure that users of various desktops had a smooth, hassle-free audio experience. In the ideal case, they would never need to find out what PulseAudio is!

Orthogonally, a number of us are also very interested in making PulseAudio a strong contender in the embedded space (mobile phones, tablets, set top boxes, cars, and so forth). While we already find PulseAudio being used in some of these, there are areas where we can do better (more in later topics).

There was some reservation expressed about other, less-used features such as network playback being ignored because of this focus. The conclusion after some discussion was that this would not be the case, as a number of embedded use-cases do make use of these and other “fringe” features.

Increasing patch bandwidth

Contributors to PulseAudio will be aware that our patch queue has been growing for the last few months due to lack of developer time. We discussed several ways to deal with this problem, the most promising of which was a periodic triage meeting.

We will be setting up a rotating schedule where each of us will organise a meeting every 2 weeks (the period might change as we implement things) where we can go over outstanding patches and hopefully clear backlog. Colin has agreed to set up the first of these.

Routing infrastructure

Next on the agenda was a presentation by Janos Kovacs about the work they’ve been doing at Intel with enhancing the PulseAudio’s routing infrastructure. These are being built from the perspective of IVI systems (i.e., cars) which typically have fairly complex use cases involving multiple concurrent devices and users. The slides for the talk will be put up here shortly (edit: slides are now available).

The talk was mingled with a Q&A type discussion with Janos and Jaska. The first item of discussion was consolidating Colin’s priority-based routing ideas into the proposed infrastructure. The broad thinking was that the ideas were broadly compatible and should be implementable in the new model.

There was also some discussion on merging the module-combine-sink functionality into PulseAudio’s core, in order to make 1:N routing easier. Some alternatives using te module-filter-* were proposed. Further discussion will likely be required before this is resolved.

The next steps for this work are for Jaska and Janos to break up the code into smaller logical bits so that we can start to review the concepts and code in detail and work towards eventually merging as much as makes sense upstream.

Low latency

This session was taken up against the background of improving latency for games on the desktop (although it does have other applications). The indicated required latency for games was given as 16 ms (corresponding to a frame rate of 60 fps). A number of ideas to deal with the problem were brought up.

Firstly, it was suggested that the maxlength buffer attribute when setting up streams could be used to signal a hard limit on stream latency — the client signals that it will prefer an underrun, over a latency above maxlength.

Another long-standing item was to investigate the cause of underruns as we lower latency on the stream — David has already begun taking this up on the LKML.

Finally, another long-standing issue is the buffer attribute adjustment done during stream setup. This is not very well-suited to low-latency applications. David and I will be looking at this in coming days.

Merging per-user and system modes

Tanu led the topic of finding a way to deal with use-cases such as mpd or multi-user systems, where access to the PulseAudio daemon of the active user by another user might be desired. Multiple suggestions were put forward, though a definite conclusion was not reached, as further thought is required.

Tanu’s suggestion was a split between a per-user daemon to manage tasks such as per-user configuration, and a system-wide daemon to manage the actual audio resources. The rationale being that the hardware itself is a common resource and could be handled by a non-user-specific daemon instance. This approach has the advantage of having a single entity in charge of the hardware, which keeps a part of the implementation simpler. The disadvantage is that we will either sacrifice security (arbitrary users can “eavesdrop” using the machine’s mic), or security infrastructure will need to be added to decide what users are allowed what access.

I suggested that since these are broadly fringe use-cases, we should document how users can configure the system by hand for these purposes, the crux of the argument being that our architecture should be dictated by the main use-cases, and not the ancillary ones. The disadvantage of this approach is, of course, that configuration is harder for the minority that wishes multi-user access to the hardware.

Colin suggested a mechanism for users to be able to request access from an “active” PulseAudio daemon, which could trigger approval by the corresponding “active” user. The communication mechanism could be the D-Bus system bus between user daemons, and Ștefan Săftescu’s Google Summer of Code work to allow desktop notifications to be triggered from PulseAudio could be used to get to request authorisation.

David suggested that we could use the per-user/system-wide split, modified somewhat to introduce the concept of a “system-wide” card. This would be a device that is configured as being available to the whole system, and thus explicitly marked as not having any privacy guarantees.

In both the above cases, discussion continued about deciding how the access control would be handled, and this remains open.

We will be continuing to look at this problem until consensus emerges.

Improving (laptop) surround sound

The next topic dealt with being able to deal with laptops with a built-in 2.1 channel set up. The background of this is that there are a number of laptops with stereo speakers and a subwoofer. These are usually used as stereo devices with the subwoofer implicitly being fed data by the audio controller in some hardware-dependent way.

The possibility of exposing this hardware more accurately was discussed. Some investigation is required to see how things are currently exposed for various hardware (my MacBook Pro exposes the subwoofer as a surround control, for example). We need to deal with correctly exposing the hardware at the ALSA layer, and then using that correctly in PulseAudio profiles.

This led to a discussion of how we could handle profiles for these. Ideally, we would have a stereo profile with the hardware dealing with upmixing, and a 2.1 profile that would be automatically triggered when a stream with an LFE channel was presented. This is a general problem while dealing with surround output on HDMI as well, and needs further thought as it complicates routing.

Testing

I gave a rousing speech about writing more tests using some of the new improvements to our testing framework. Much cheering and acknowledgement ensued.

Ed.: some literary liberties might have been taken in this section

Unified cross-distribution ALSA configuration

I missed a large part of this unfortunately, but the crux if the discussion was around unifying cross-distribution sound configuration for those who wish to disable PulseAudio.

Base volumes

The next topic we took up was base volumes, and whether they are useful to most end users. For those unfamiliar with the concept, we sometimes see sinks/sources where which support volume controls going to > 0dB (which is the no=attenuation point). We provide the maximum allowed gain in ALSA as the maximum volume, and suggest that UIs show a marker for the base volume.

It was felt that this concept was irrelevant, and probably confusing to most end users, and that we suggest that UIs do not show this information any more.

Relatedly, it was decided that having a per-port maximum volume configuration would be useful, so as to allow users to deal with hardware where the output might get too loud.

Devices with dynamic capabilities (HDMI)

Our next topic of discussion was finding a way to deal with devices such as those HDMI ports where the capabilities of the device could change at run time (for example, when you plug out a monitor and plug in a home theater receiver).

A few ideas to deal with this were discussed, and the best one seemed to be David’s proposal to always have a separate card for each HDMI device. The addition of dynamic profiles could then be exploited to only make profiles available when an actual device is plugged in (and conversely removed when the device is plugged out).

Splitting of configuration

It was suggested that we could split our current configuration files into three categories: core, policy and hardware adaptation. This was met with approval all-around, and the pre-existing ability to read configuration from subdirectories could be reused.

Another feature that was desired was the ability to ship multiple configurations for different hardware adaptations with a single package and have the correct one selected based on the hardware being run on. We did not know of a standard, architecture-independent way to determine hardware adaptation, so it was felt that the first step toward solving this problem would be to find or create such a mechanism. This could either then be used to set up configuration correctly in early boot, or by PulseAudio for do runtime configuration selection.

Relatedly, moving all distributed configuration to /usr/share/..., with overrides in /etc/pulse/... and $HOME were suggested.

Better drain/underrun reporting

David volunteered to implement a per-sink-input timer for accurately determining when drain was completed, rather than waiting for the period of the entire buffer as we currently do. Unsurprisingly, no objections were raised to this solution to the long-standing issue.

In a similar vein, redefining the underflow event to mean a real device underflow (rather than the client-side buffer running empty) was suggested. After some discussion, we agreed that a separate event for device underruns would likely be better.

Beer

We called it a day at this point and dispersed beer-wards.

PulseConf Hackers

Our valiant attendees after a day of plotting the future of PulseAudio

User experience

David very kindly invited us to spend a day after the conference hacking at his house in Lund, Sweden, just a short hop away from Copenhagen. We spent a short while in the morning talking about one last item on the agenda — helping to build a more seamless user experience. The idea was to figure out some tools to help users with problems quickly converge on what problem they might be facing (or help developers do the same). We looked at the Ubuntu apport audio debugging tool that David has written, and will try to adopt it for more general use across distributions.

Hacking

The rest of the day was spent in more discussions on topics from the previous day, poring over code for some specific problems, and rolling out the first release candidate for the upcoming 3.0 release.

And cut!

I am very happy that this conference happened, and am looking forward to being able to do it again next year. As you can see from the length of this post, there are lot of things happening in this part of the stack, and lots more yet to come. It was excellent meeting all the fellow PulseAudio hackers, and my thanks to all of them for making it.

Finally, I wouldn’t be sitting here writing this report without support from Collabora, who sponsored my travel to the conference, so it’s fitting that I end this with a shout-out to them. :)

PulseConf Schedule

David has now published a tentative schedule for the PulseAudio Mini-conference (I’m just going to call it PulseConf — so much easier on the tongue).

For the lazy, these are some of the topics we’ll be covering:

  • Vision and mission — where we are and where we want to be
  • Improving our patch review process
  • Routing infrastructure
  • Improving low latency behaviour
  • Revisiting system- and user-modes
  • Devices with dynamic capabilities
  • Improving surround sound behaviour
  • Separating configuration for hardware adaptation
  • Better drain/underrun reporting behaviour

Phew — and there are more topics that we probably will not have time to deal with!

For those of you who cannot attend, the Linaro Connect folks (who are graciously hosting us) are planning on running Google+ Hangouts for their sessions. Hopefully we should be able to do the same for our proceedings. Watch this space for details!

p.s.: A big thank you to my employer Collabora for sponsoring my travel to the conference.

PulseConf!

For those of you who missed it, your friendly neighbourhood PulseAudio hackers are converging on Copenhagen in a month to discuss, plan and hack on the future of PulseAudio.

We’re doing this for the first time, so I’m super-excited! David has posted details so if this is of interest to you, you should definitely join us!

Picking your battles

Most of you have no doubt already seen that Mozilla will be changing their position on H.264 support for HTML5 video in future releases. This is an extremely important decision that I’ve been hoping to see for a while now, and I am really glad this is being done.

There is no doubt that we need patent-unencumbered standards for web codecs (or as much as is possible given the dismal patent ecology today), and while much giddy anticipation followed Google/On2’s release of VP8 into the open, I don’t believe it ever made sense to expect the codec landscape to change drastically in the short timespan everyone expected. There’s a lot of the hardware and software out there that needs to change (see any SoCs with VP8 support yet?), not to mention the interests of the MPEG-LA mafconsortium.

I love Firefox, both as a product and what it means for an open web (for those of you that know me, this might be hard to believe given all my ranting, but it’s true!). I’m glad Mozilla chose to live to fight another day rather than go out in a blaze of glory and (or a flicker of irrelevance).

p.s.: these are my views and do not necessarily represent those of my employer

p.p.s.: Alessandro’s been doing some great work to get the GStreamer multimedia backend going again (this makes so much more sense than going the NIH route!)

Talk video from GstConf 2011

For those of you who were interested but couldn’t make it to the GStreamer Conference this year, the cool folks at Ubicast have got the talk videos up (can be streamed or downloaded).

Among these is my talk about recent developments in the PulseAudio world.

Notes from the Prague Audio BoFs

As I’d blogged about last week, we had a couple of Audio BoF sessions last week. Here’s a summary of what was discussed. I’ve collected items in relevance order rather than chronological order to make for easier reading. I think I have everything covered, I’ll update this post if one of the attendees points out something I missed or got wrong.

  • Surround: There were a number of topics that came up with regards to multichannel/surround support:

    • There seems to be some lack of uniformity of channel maps, particularly in non-HDA hardware. While it was agreed that this should be fixed, we need some directed testing and bug reports to actually be able to fix this.

    • Multichannel mixers, while theoretically supported, are not actually exposed by any current drivers. It was suggested that these could be exposed without breaking existing applications by having new MC mixers exposed with device names corresponding to the surround PCM device (like “surround51”).

    • We need a way to query channel maps on a given PCM. This will be implemented as a new ALSA API which could be called after the PCM is opened. (AI: Takashi)

    • It would be good to have a way to configure the channel map as well (if supported by the hardware?). The suggestion was to do this as was done in PulseAudio, where an arbitrary channel map could be specified. (NB: is there hardware that supports multiple/arbitrary channel maps? If yes, how do we handle this?)

  • Routing: Unsurprisingly, we came back to the problem of building a simplified mixer graph for PulseAudio.

    • The current status is that ALSA builds a simplified mixer for use by userspace, and PulseAudio further simplifies this by means of some name-based guessing.

    • PulseAudio would ideally like a simplified version of the original mixer graph, but something more complete than what we get now

    • However, since PulseAudio has fairly unique requirements of what information it wants, it probably makes sense to have ALSA provide the entire graph and leave the simplification task to PulseAudio (discussion on this approach continues)

    • There was no consensus on who would do this or how this should be done (creating a new serialisation format, exposing what HDA provides, adding node metadata to ALSA mixer controls, or something else altogether)

    • As an interim step, it was agreed that it would be possible to provide ordering in the simplified ALSA mixer (that is, add metadata to the control to signal what control comes “before” it and what comes “after” it). This should go some way in making the PA mixer simplification logic simpler and more robust. (AI: Takashi)

  • HDMI: A couple of things came up in discussion about the status of HDMI.

    • There was a question about the reliability of ELD information as this will be used more in future versions of PulseAudio. There did not appear to be conclusive evidence in either direction, so we will probably assume that it is reliable and deal with reliability problems as they arise.

    • It was mentioned that it might be desirable to only expose the ALSA device if a receiver is plugged in. This had been mooted earlier as something to do in PulseAudio as an alternative. One thing to consider with this approach is dealing with a device switch on the receiver side. Doing this without a notification to userspace was generally agreed to be a bad idea.

  • Jack detection: The long-standing debate on exposing jacks as input devices or ALSA controls came to a conclusion, with the resolution being that jacks would be exposed as ALSA controls. This requires a change in the kernel (and potentially alsa-lib?) which should not be too complex. Actual buttons (such as volume/mute control) will continue to be input devices. Once this is done, the pending jack detection patches will be adapted to use the new interface. (AI: Takashi (patches are in a branch already!), David)

  • UCM: Another long-standing issue was the merging of the ALSA UCM patches into PulseAudio. Most of the problems thus far had been due to an incomplete understanding of how ALSA and PA concepts mapped to each other. Some consensus was arrived at in this regard after a lengthy discussion:

    • As is the case now, every ALSA PCM maps to a PA sink

    • Each UCM verb maps to a PA card profile

    • Each combination of UCM devices that can be used concurrently maps to a PA port

    • Each UCM modifier is mapped to an intended role on the corresponding sink

    The code should (as is in the patches currently submitted) be part of the PA ALSA module, and there will be changes required to use the UCM-specified mixer list instead of PA’s guessing mechanism. (AI: ???)

    (NB: It was mentioned that PulseAudio needs to support multiple intended roles for a sink/source. This is actually already supported — the intended roles property is a whitespace-separated list of roles)

    (NB2: There was further discussion with the Linaro folks this week about the UCM bits, and there’s likely going to be an IRC/phone gathering to clarify things further in the near future)

  • GStreamer latency settings: We currently do not actually use PulseAudio’s power saving features from GStreamer for several reasons. Suggestions to over come this were mooted. While no definite agreement was reached, one suggestion was to add a “powersave” profile to pulsesink that chose higher latency/buffer-time values. Players would need to set this if they are not using features that break when these values are raised.

  • Corking: The statelessness of current the corking mechanism was discussed in one session, and between the PulseAudio developers later. The problem is that we need to be able to track cork/uncork reasons more closely. This would give us more metadata that is needed to make policy decisions without breaking streams. Particularly, for example, if PA corks a music stream due to an incoming call, then the user plays, then pauses music, and then the call ends, we must not uncork the music stream. We intend to deal with this with two changes:

    • We need to add a per-cause cork/uncork request count

    • We need to associate a “generation” with cork/uncork requests, so certain conditions (such as user intervention) can bump the generation counter, and uncork requests corresponding to old cork requests will be ignored

    This will make it possible to track the various bits of state we need to do the right thing for cases like the one mentioned before.

So that’s that — lots of things discussed, lots of things to do! Thanks to everyone who came for participating.

PragueAudio

For those who are in Prague for GstConf, LinuxCon, ELCE, etc. — don’t forget we’ve a couple of interesting audio-related things happening:

If you’re around and interested, do drop in!

More conferences than you can shake a stick at

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!

1.w00t!

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. :)

Hello … hello … hello!

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.