Also, we’ve switched to the WebRTC canceller as the default canceller (you can still choose the Speex canceller manually, though). Overall, the quality is pretty good. I’d do a demo, but it’s effectively had zero learning time in my tests, so we’re not too far from a stage where this is a feature that, if we’re doing it right you won’t notice it exists.

There lot’s of things, big and small that need to be added and tweaked, but this does go some way towards bringing a hassle-free VoIP experience on Linux closer to reality. Once again, kudos to the folks at Google for the great work and for opening up this code. Also a shout-out to fellow Collaboran Sjoerd Simons for bouncing ideas and giving me those much-needed respites from talking to myself. :)

• 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.

• Today (Tuesday), at 4 pm, I’ll be talking about recent developments in PulseAudio
• Tomorrow (Wednesday), at 11am, we’re continuing the Audio BoF that I had mentioned earlier (since we ran out of time on Sunday)

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

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!

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!

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!

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. :)

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).

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

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:

With AEC:

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.

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.

I’ll be presenting a talk titled DLNA in a GNOME 3 World, talking about Rygel and the work we’ve been doing on gupnp-dlna to make DLNA rock on GNOME.

If you’re in or around Bangalore and contribute to or are interested in contributing to GNOME, you really have no excuse to not attend (heck, entry’s free). This applies doubly to students who are looking for cool stuff to do for the Google Summer of Code this year. So, do drop by and say hello! :)

So Nirbheek Chauhan and I designed a couple of quick mockups that I think would work well. This doesn’t address customising behaviour with an external monitor, but I don’t feel nearly as strongly about that being hidden in dconf-editor as I do about the rest.

My mockup

Nirbheek's mockup

While Nirbheek’s version looks decidedly prettier, I think the meaning of the icons is not absolutely obvious. This might be solvable by some explanatory text above and mouse-overs.

While doing all this, though, it’s clear that it is really hard to design a UI that you think will please enough people, and really easy to make assumptions about what “people” want and how they use their computers. So kudos to the GNOME3 UI designers for taking up this difficult job and I hope they take all the feedback flying around in a positive spirit (even if the messages are often not quite positive-sounding ;) )

For those that don’t know, the DLNA is a consortium that aims to standardise how various media devices around your house communicate with each other (that is, your home theater, TV, laptop, phone, tablet, …). One piece of this problem is having a standard way of identifying the type of a file, and communicating this between devices. For example, say your laptop (MediaServer in DLNA parlance) is sharing the movies you’ve got with your TV (MediaPlayer), and your TV can play only upto 720p H.264-encoded video. When the MediaServer is sharing files, it needs to provide sufficient information about the file so that the MediaPlayer knows whether it can play it or not, so that it can be intelligent about what files show up in its UI.

How the DLNA specification achieves this is by using “profiles”. For each media format supported by the DLNA specification, a number of profiles are defined, that identify the audio/video codec used, the container, and (in a sense) the complexity of decoding the file. (for multimedia geeks, that translates to things like the codec profile, resolution, framerate/samplerate, bitrate, etc.)

For example, if a file is indicated to be of a DLNA profile named AAC_ISO_320, this indicates that this is an audio file encoded with the AAC codec, contained in an MP4 container (that’s “ISO”), with a bitrate of at most 320 kbps. Similarly, a file with profile AVC_MP4_MP_SD_MPEG1_L3 represents a file with H.264 (a.k.a. AVC) video coded in the H.264 Main Profile at specific resolutions upto 720×576, MP3 audio, in an MP4 container (there are more restrictions, but I don’t want to swamp you with details).

So now we have a problem statement – given a media file, we need to get the corresponding DLNA profile. It’s easiest to break this problem into 3 pieces:

1. Discovery: First we need to get all the metadata that the DLNA specification requires us to check. Using GStreamer and Edward’s gst-convenience library, getting the metadata we needed was reasonably simple. Where the metadata wasn’t available (mostly codec profiles and bitrate), I’ve tried to expose the required data from the corresponding GStreamer plugin.

2. DLNA Profiles: I won’t rant much about the DLNA specification, because that’s a whole series of blog posts in itself, but the spec is sometimes overly restrictive and doesn’t support a number of popular formats (Matroska, AVI, DivX, OGG, Theora). With this in mind, we decided that it would be nice to have a generic way to store the constraints specified by the DLNA specification and use them in our library. We chose to store the profile constraints in XML files. This allows non-programmers to tweak the profile data when their devices resort to non-standard methods to work around the limitations of the DLNA spec.

3. Matching: With 1. and 2. above in place, we just need some glue code to take the metadata from discovery and match it with the profiles loaded from disk. For the GStreamer hackers in the audience, the profile storage format we chose looks suspiciously like serialized GstCaps, so matching allows us to reuse some GStreamer code. Another advantage of this will be revealed soon.

So there you have it folks, this covers the essence of what GUPnP DLNA does. So what’s next?

1. Frankie Says Relax: Since the DLNA spec can often be too strict about what media is supported, we’ve decided to introduce a soon-to-come “relaxed mode” which should make a lot more of your media match some profile.

2. I Can Haz Trancoding: While considering how to store the DLNA profiles loaded from the XML on disk, we chose to use GstEncodingProfiles from the gst-convenience library since the restrictions defined by the DLNA spec closely resemble the kind of restrictions you’d expect to set while encoding a file (codec, bitrate, resolution, etc. again). One nice fallout of this is that (in theory), it should be easy to reuse these to transcode media that doesn’t match any profile (the encodebin plugin from gst-convenience makes this a piece of cake). That is, if GStreamer can play your media, Rygel will be able to stream it.

Apart from this, we’ll be adding support for more profiles, extending the API as more uses arise, adding more automated tests, and on and on. If you’re interested in the code, check out (sic) the repository on Gitorious.

Collabora Multimedians, looking for a canal

Nevertheless, I had a great time meeting all the cool folks at Collabora Multimedia at our company meeting. Managed to trundle out for my Rygel + DLNA lightning talk (more updates on this in a subsequent post). Things did get better subsequently, and I had an amazing week-long vacation in Germany, and now I’m back at home with my ninja skillz fully recharged!

There’s a nice tutorial for generating Vala bindings that serves as a good starting point. The process basically involves running a tool called vapigen, which examines your headers and libraries, and generates a GIR file from them (it’s an XML file describing your GObject-based API). It then converts this GIR file into a “VAPI” file which describes the API in a format that Vala can understand. Sounds simple, doesn’t it?

Now if only it were that simple :). The introspected file is not perfect, which means you need to manually annotate some bits to make sure the generated VAPI accurately represents the C API. These annotations are specified in a metadata file. You need to include things like “the string returned by this function must be freed by the caller” (that’s a transfer_ownership), or, object type Foo is derived from object type FooDaddy (specified using the base_class directive). Not all these directives are documented, so you might need to grok around the sources (specifically, vapigen/valagidlparser.vala) and ask on IRC (#vala on irc.gnome.org).

All said and done, the process really is quite straightforward. The work is in [my gst-convenience repository][arun-gst-conv-ks.git] right now (should be merged with the main repository soon). I really must thank all the folks on #vala who helped me with all the questions and some of the bugs that I discovered. Saved me a lot of frustration!

I’ve already got Rygel using these bindings, though that’s not been integrated yet. More updates in days to come.

As Srini had announced, Dec 5th was GNOME Day at FOSS.IN this year. We kicked the day off with Shreyas giving a developer’s introduction to GNOME 3.0. This was followed by another well-received talk by Srini on the Mobiln2 UI and Clutter.

By the end of lunch, it turned out our already packed schedule had got some new additions from the other enthusiastic GNOME folks around! The afternoon session was kicked off by Arun ‘vimzard’ Chaganty introducing what newbies need to know to dive into GNOME development. Tobias Mueller followed with a talk about GNOME Bugsquadding. Sayamindu and Dimitris then took the stage for a short L10n talk. Next up was a talk about Anjal by Puthali. Olivier then gave a hackers’ introduction to Empathy/Telepathy, Srinidhi and Bharath did a quick introduction to using the OpenSUSE Build Service.

Wait, I’m not done yet. :) The final session on GNOME Performance was a 4-hit combo with me giving a quick introduction to Sysprof, Lennart introducing mutrace, Krishnan giving a pretty wow introduction to using DTrace to profile GNOME, and Dhaval giving a short introduction to how cgroups could help make GNOME more responsive.

Phew! That was a long and awesome day, with some icing on the cake in the form of stickers and T-shirts. The last were possible thanks to the GNOME Foundation, so a huge thanks to them!

Sponsored by GNOME!

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. :)

The GNOME Foundation has funded a very significant part of my expense for this trip (making it possible at all), so a huge thanks to Travel Committee for giving me this opportunity. :) To summarise …

Sponsored by GNOME!

Shreyas and I reached Gran Canaria early in the morning of Day 1, but were too tired to make it to the first 2 keynotes. We woke up, had breakfast by the beach (the apartment we were in was <100 steps from the beach, and the auditorium was a 20 minute walk down the same beach — photos soon).

We did make it to Richard Stallman’s talk. It was quite generic, not surprisingly about software freedom, and nothing new to most of us. Of note were the great vitriol towards C# and the heathens who use it to create new software and a rather terrible and inappropriate attempt at humour that has been blogged about to death.

I met a huge number of people subsequently, some who’ve been at FOSS.IN before, and many whom I only knew by their online presence. The second half of the day was devoted to a number of Lightning Talks. I was pleasantly surprised to see the amount of work happening on semantic-aware projects. Good stuff.

Way to sleepy to continue making sense. More details on subsequent days, photos and so forth to come soon.

Edit: In the name of avoiding further procrastination, here are the photos.