http://summit.ubuntu.com/lpc-2012/ Audio

Wednesday 11:40 - 12:25 PDT
Not Attending Expose Routing && DSPs in ASoC
=== Exposing Routing to the Application Layer === In order to allow nicer configuration UIs and easier diagnostics it would be good if we could expose audio routing information to the application layer. === Representing DSPs in ASoC === As embedded systems continue to evolve, host-based processing is often offloaded to a co-processor dedicated for specific tasks in order to improve the performance and power utilization. In this the audio subsystem seems to be no exception and more vendors are trying to offload tasks to DSPs, be they embedded in SoCs or audio codecs. This evolution requires changes in ASoC to represent all the routing and processing capabilities of these DSPs. The current efforts are based on two methods being propagated. The first approach developed by Liam Girdwoodand TI, typically referred to as soc-dsp aka dynamic PCM, introduces the notion of dynamic PCM nodes, where audio front-ends (FEs-classic PCM device visible to user) and audio back-ends (BEs-soc-hardware interfaces) represent audio links. The second approach being discussed is referred as CODEC<-->CODEC model as proposed by Mark Brown, where the DSP is represented as another codec in the system and links to the real codec in the system through the machine map. This talk takes a look at both the approaches and latest evolutions, future plans on these methods and discusses the common infrastructure needs which need to considered for making this representation an effective one for SoC and codec vendors Topic Lead: Vinod Koul <email address hidden> Topic Lead: Pierre-louis Bossart <email address hidden>

Participants:
attending broonie (Mark Brown)
attending diwic (David Henningsson)
attending srwarren (Stephen Warren)
attending tiwai (Takashi Iwai)
attending vinod-koul (Vinod Koul)

Tracks:
  • Audio
Nautilus 2
Wednesday 14:50 - 15:35 PDT
Not Attending HD-audio cooking && QA/Testing
=== HD-Audio Cooking Recipe === This session will cover the debugging practices for typical problems seen with HD-audio, which is the standard component on all modern PCs, using the sysfs kernel interface and user-space helper programs. Also, we will discuss open problems, jack retasking, HDMI stream assignment, missing speaker/mic streams, and better interaction / organization with user-space applications like PulseAudio. === Bug Tracking and QA === Bug tracking in the sound subsystem is far away from the optimal situation. We'd like to discuss how to improve the situation, looking through the experiences from the current ALSA and distro bug trackers. Also, we're going to show the automated testing of HD-audio driver using hda-emu emulator for detecting the errors and regression coverage. Topic Lead: Takashi Iwai (<email address hidden>) Working as a member of hardware-enablement team in SUSE Labs at SUSE Linux Products GmbH in Nuremberg, Germany, while playing a role as a gatekeeper of the Linux sound subsystem tree over years. Topic Lead: David Henningsson (<email address hidden>) Working for Canonical with audio hardware enablement, fixing audio bugs and maintaining the audio stack, and also part of PulseAudio's current development team.

Participants:
attending broonie (Mark Brown)
attending diwic (David Henningsson)
attending srwarren (Stephen Warren)
attending tiwai (Takashi Iwai)

Tracks:
  • Audio
Nautilus 2
Thursday 09:30 - 10:15 PDT
Not Attending Simplify volume setting at startup/shutdown
Currently, on a normal desktop session, volume is set four times on startup - initally by the kernel, then by alsactl, then by PulseAudio in the DM session, then by PulseAudio in the logged in session. When shutting down, both PulseAudio and alsactl saves volumes to restore them later. And then we also have suspend and hibernate to consider, and that cards can be plugged in at any time. First, isn't this quite complex for something as simple as setting volumes? Second, can we facilitate new features, such as 1) having a "set this volume as default, for all users, on startup" button in the volume control, or 2) "allow the DM user to introspect different users' volumes"? Topic Lead: David Henningsson (<email address hidden>) Working for Canonical with audio hardware enablement, fixing audio bugs and maintaining the audio stack, and also part of PulseAudio's current development team.

Participants:
attending broonie (Mark Brown)
attending tiwai (Takashi Iwai)

Tracks:
  • Audio
Nautilus 2
Thursday 16:30 - 17:15 PDT
Not Attending Time Alignment && PulseAudio on Android
=== Time alignment in the Linux Audio Stack === The Linux Audio stack provides very little support for precise timing, despite the availability of hardware audio wall clocks and the adoption of new protocols such as IEEE1588 and Ethernet AVB, which align networked devices several orders of magnitude more precisely than NTP. In this presentation, we show how providing user-space applications with access to the audio wall clock can improve audio rendering/capture for local and networked devices. In the first case, the resolution of the wall clock can help PulseAudio track with more precision the drift between system time and audio time. Likewise for networked devices, the differences in audio wall clocks can help a server adjust asynchronous sample-rate conversions without large and frequent variations of the sample-rate ratio. We will present some ideas on modifications of the audio stack and data structures and gather feedback from the open-source community. Topic Lead: Pierre Bossart === PulseAudio on Android === As part of our efforts to make 'standard' Linux components available in the Android world, we are working on porting PulseAudio to Android. In this session, we talk about challenges in the initial porting effort, the approach we are taking to make PulseAudio an out-of-the-box replacement for the native system, and what advantages we hope to be able to provide with this work. Topic Lead: Arun Raghavan <email address hidden> Arun Raghavan is a long-time open source supporter and mainly hacks on the PulseAudio audio server at Collabora. He contributes to the GStremaer multimedia framework, and secretly is a developer on the Gentoo Linux distribution as well.

Participants:
attending broonie (Mark Brown)
attending tiwai (Takashi Iwai)

Tracks:
  • Audio
Nautilus 2
Thursday 17:25 - 18:10 PDT
Not Attending ALSA channel-mapping API
The functionality to query and/or set the PCM channel-mapping is a long-standing missing feature in ALSA. The session will cover the requirement by the actual hardware and discuss the pros and cons of proposed implementations. * REQUIRED AUDIENCE ALSA devs, PulseAudio devs, gstreamer devs Topic Lead: Takashi Iwai <email address hidden>: Working as a member of hardware-enablement team in SUSE Labs at SUSE Linux Products GmbH in Nuremberg, Germany, while playing a role as a gatekeepr of the Linux sound subsystem tree over years.

Participants:
attending broonie (Mark Brown)
attending diwic (David Henningsson)
attending srwarren (Stephen Warren)
attending tiwai (Takashi Iwai)

Tracks:
  • Audio
Nautilus 2

PLEASE NOTE The Linux Plumbers Conference 2012 schedule is still in a draft format and is subject to changes at any time.