On Tue, Dec 14, 2010 at 1:31 PM, Marco Ballesio Hopefully there's more than just source-code to describe the policies and enforcement. Is there a design document stating the "organizational" or "legal" or "human" reasons for said policies and enforcements? And, as a developer, where is the documentation for the appropriate layer to use, the appropriate API, etc -- e.g. for the original question that started this thread -- how to switch on/off speakers, FM-radio transmitter, etc on handset. ( cf my original reply http://lists.meego.com/pipermail/meego-handset/2010-December/000066.html ) Something high-level like the following Maemo document would be very helpful -- but I have been unsuccessful finding such documentation for Meego: http://wiki.maemo.org/Documentation/Maemo_5_Developer_Guide/Architecture/Mu… Also, why not use ALSA use-case-manager: http://opensource.wolfsonmicro.com/node/14 UCM appears to be scheduled for other distros "handset" work: https://wiki.ubuntu.com/Specs/N/ARM/AlsaScenarioManager Potential to blow out my handset speakers duly noted (the n900 goes http://en.wikipedia.org/wiki/Up_to_eleven !), however, I wasn't advocating playing around with the ALSA layer indiscriminately, in fact, I specifically stated: ................................ Regarding my old nemesis, pulseaudio ( http://tinyurl.com/2976vu6 == http://old.nabble.com/uninstall-pulseaudio-to-increase-audio-app-stability-… ), I think the only "elegant" thing about pulseaudio is that it's bluetooth handling works for those that care about bluetooth; given the number of bug-reports and incompatibilities pulseaudio generates in different distros, I'm not sure "elegant" is the right word.... to go through pulseaudio sounds like an all-around bad idea, and to have "enforcement" or "compliance" attached makes it sound even worse. Tell me it ain't so! There is a growing class of applications that do not want or need pulseaudio around -- those using http://jackaudio.org/ . When the jack audio server launches, the first thing it does it use dbus to disable pulseaudio. Is that also non compliant? It seems inappropriate to preclude an entire class of application -- real-time "pro" audio and video apps that utilize the Jack audio server to provide low-latency, tightly synchronized audio -- as needed for modern multimedia creation and playback. Perhaps such applications are a stretch for an OMAP3-class device, but given the many audio/media apps listed in http://omappedia.org/wiki/PEAP_Projects , clearly OMAP4 and beyond might not be, even on a puny "handset." Of course, those making such audio apps might sidestep pulseaudio compliance/latency/inefficiency issues by using http://opensoundcontrol.org/ and an external DAC ( http://gregsurges.com/tag/dac/ ). Finally, it seems odd that in a "handset" environment, pulseaudio is an absolute requirement. To me, it is just a waste of batteries, and a terrible source of unnecessary context switching and interrupts during audio playback. It's sort of like being forced to drive around in a gas-guzzling, oversized sport-utility vehicle with 10 foot tires and 5 feet of ground clearance -- just to drive to the market or work on a well-paved freeway on a summer's day -- even when one might prefer a bicycle, motorcycle, sports-car, subway, or whatever tool is best for the job. Given that's the argument against Java/Android on the handset (it's the SUV of languages/environments) it's unfortunate that lighter-weight and less monolithic and more "unixy" solutions aren't being pursued for audio on the Meego handset. Take for example HD audio/video playback -- something where you need a "sportscar" to get the latency and sample rate associated with the audio stream while also performing HD decoding (e.g. 16bit/96K audio is supported on omap3430 per http://and-developers.com/device_information ). Pulseaudio typically operates at a fixed rate and forces resampling to that rate, causing an oft perceptible loss in fidelity. So in order to allow "digital mixing" for notification signals or phone calls while watching an HD movie, either pulseaudio will be upmixing those signals to 16/96, which is inefficient for audio that doesn't need the higher bitrate; the alternative, which is what we get with pulseaudio, is that the DAC is running at 44 or 48k, and even though we might be listening to material at a higher sample-rate, it'll be resampled and sonic artefacts may be introduced in order to work with the 'lowest common denominator'. IMHO what is needed is not a "digital mixer" and resampler and extra user-space processing before going into the kernel and out to the sound hardware via ALSA drivers. We certainly need "use case management" and the notion of a digital patch bay, and some way of smoothly mixing between sounds, glitch free switching of sample rates at the ALSA level, and then choosing the direct path to hardware that'll best handle the "main audio task" we're doing -- e.g. playing music, watching a movie, making a phone call, or just using the screen UI. What isn't needed is "desktop networked audio" capabilities of pulseaudio, or any extra user-space streaming/mixing/resampling of signals. The inefficiencies introduced by pulseaudio is evidenced by the n900/meego-1.1 handset, which cannot maintain audio synchronization during audio or video playback if even the slightest trace of something else is going on, including a wireless network scan, or even just cursoring around in a networked terminal (which goes through the wireless stack in my setup). On Maemo/n900 note what happens when playing back an Mp3 -- pulseaudio consumes twice the resources at "high priority" of the decoding process: Mem: 239968K used, 5572K free, 0K shrd, 1856K buff, 68204K cached CPU: 31.5% usr 10.3% sys 0.0% nice 53.9% idle 4.1% io 0.0% irq 0.0% softirq Load average: 0.88 0.56 0.27 PID PPID USER STAT RSS %MEM %CPU COMMAND 781 1 pulse S < 3832 1.5 22.0 /usr/bin/pulseaudio --system --high-priority 1317 705 user S < 6320 2.5 12.2 /usr/bin/mafw-dbus-wrapper mafw-gst-renderer 971 1 user S < 1868 0.7 1.3 /usr/bin/dbus-daemon --fork --print-pid 5 --print-address 7 --session On Meego/n900, here's what playing track one in "Music player" looks like -- where even cursoring around on the command-line in bash in the terminal is enough to "desync" the audio stream for a few seconds -- and that's with pulseaudio running at nice=-11 and high priority! Note pulseaudio consumes 31.6% CPU while the decompression, presumably happening in bognor-regis "Media daemon and play queue manager" == 34.5% and finally the media player app itself at 6.7%.... all while consuming 21.3% memory just to play back an MP3.... top - 17:56:15 up 20 min, 1 user, load average: 3.31, 3.12, 2.07 Tasks: 106 total, 2 running, 103 sleeping, 0 stopped, 1 zombie Cpu(s): 18.1%us, 26.6%sy, 17.2%ni, 36.5%id, 1.2%wa, 0.0%hi, 0.3%si, 0.0%st Mem: 226260k total, 215948k used, 10312k free, 84k buffers Swap: 786428k total, 5944k used, 780484k free, 81596k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1063 meego 20 0 127m 9m 7720 S 34.5 4 .5 1:35.09 bognor-regis-da 892 meego 9 -11 191m 4588 3256 S 31.6 2.0 2:45.14 pulseaudio 1104 meego 20 0 2364 960 744 R 10.5 0.4 0:00.22 top 1059 meego 25 5 91628 32m 29m R 6.7 14.8 0:29.26 meegomusic 803 root 20 0 38452 28m 17m S 3.8 12.8 0:38.55 Xorg 876 root 20 0 0 0 0 S 1.9 0.0 0:13.20 ipolldevd Here's *just* watching a video (big buck bunny 240p) on the meego handset -- and no audio is even playing back (constant triggering of desync bug seen when playing back audio stream?) but pulseaudio is working hard anyways, consuming almost 50% of CPU needed to decompress the video and audio stream: top - 17:46:48 up 11 min, 1 user, load average: 4.47, 2.20, 1.03 Tasks: 106 total, 2 running, 103 sleeping, 0 stopped, 1 zombie Cpu(s): 12.8%us, 31.9%sy, 55.2%ni, 0.0%id, 0.0%wa, 0.0%hi, 0.0%si, 0.0%st Mem: 226260k total, 222564k used, 3696k free, 100k buffers Swap: 786428k total, 1224k used, 785204k free, 85424k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1007 meego 25 5 269m 41m 32m R 59.3 18.9 1:27.18 meegovideo 892 meego 9 -11 191m 5272 3948 S 26.1 2.3 0:40.00 pulseaudio 803 root 20 0 33092 23m 13m S 4.1 10.6 0:10.83 Xorg I'm sure a simple experiment could determine exactly the "effect" of pulseaudio: Play the same playlist until the battery wears out using the same software player outputting first to pulseaudio (pref not through ALSA's pulseaudio driver because that wouldn't be "fair" to go through the kernel twice) then play the same through ALSA "dmix" interface, just to emulate pulseaudio's mixing/resampling functionality. I would imagine the pure-ALSA solution would pass the "energizer bunny' test for more hours and far fewer kernel/userspace context switches. Although a realtime userspace process like pulseaudio can help deliver stable audio in a multicore environment -- it may end starving out other processes on a slower uniprocessor. Which is why I believe a pulse-audio-free solution should be available and still be "compliant" on low-end systems. The one place where pulseaudio is currently helpful is in hooking up bluetooth devices. But ALSA has it's own bluetooth layer as well, and other architectures for audio shouldn't be precluded: http://bluetooth-alsa.sourceforge.net/future.html or Phonon (http://pulseaudio.org/wiki/KDE ). -- Niels http://nielsmayer.com