Re: [pulseaudio-discuss] Channel Mode mixer element: 6ch vs. 8ch
'Twas brillig, and gee at 09/12/10 04:55 did gyre and gimble: I have not encountered anything since then that would lead me to believe it was wrong to do so. Even though forcing this might not be the best. Maybe give an option to the user in pavucontrol (instead of alsamixer) or something like this? Daniel (who deals a lot with this kind of thing) seems to think that there would indeed be some consequences so we'll have to think about this a bit more. Aside from the fact that it's rather complicated (PA abstracts ALSA from PA clients like pavucontrol, gnome-volume-control, kmix etc., so if we were to expose this, there would have to be some method of passing this enumeration from the server to the client and some way for the client to tell the server to change the value etc. so this would have to be designed in. Even with the hassle factor ignored, it's really not a route we want to go down anyway. We want the user to simply pick what setup they want and for it to work. Alsamixer is just a bizarre mess of sliders for 99% of users and we really do just want to do everything we can to keep that mess from creeping into PA. The best solution would be to deal with it in an intelligent and contextual way. I'm sure we'll find it :) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
[pulseaudio-discuss] Pulseaudio for two sessions at the same time
Hello all: I've a computer at home which uses to have two sessions opened at the same time: mine and my wife's. We use to leave both sessions opened instead of opening and closing sessions with each seat change. But the problem is that the first openes session gets sound and the other one does not: it is absolutely mute. How can we get session working in both sessions at the same time? Thanks Noel T. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Pulseaudio for two sessions at the same time
Hi, 'Twas brillig, and Noel David Torres Taño at 09/12/10 09:35 did gyre and gimble: I've a computer at home which uses to have two sessions opened at the same time: mine and my wife's. We use to leave both sessions opened instead of opening and closing sessions with each seat change. But the problem is that the first openes session gets sound and the other one does not: it is absolutely mute. This should not generally happen. PulseAudio is designed to work at a session level and ensure that when you switch between sessions, control of the audio device is handed gracefully over to the other user. All of this is actually handled at a lower level than PA. It's dealt with by a system called Console Kit. Console Kit maintains a record of who the active user is on a system (you can check via ck-list-sessions command in a terminal). Console Kit will instruct udev to write ACLs (access control lists) on various bits of hardware (including the sound card) so that only the active user has permission to use them at any one time (otherwise there could be security considerations - e.g. spying on voip calls etc. etc.) PA simply honours this lower level system. Now what is happening in your case is one of three things (the last is the most likely): 1. Console Kit is not working properly. To test this, open a session for both users and open a terminal and type ck-list-sessions. As you switch between the sessions, the active user should change. 2. Console Kit is not writing the ACLs properly. Use getfacl /dev/snd/* in each session to ensure that the relevant user appears in the ACLs for the sound. 3. One or both of your users is in the audio group. This bypasses all the nice ACL and session switching logic, but only really works if your sound hardware supports hardware mixing or you have specific reason to do something non-standard (see below). Just type groups in a terminal to see if you are in the audio group and if so, use the appropriate tools to remove this and then reboot (make sure you do this for both users) and you should get smooth switching of users. How can we get session working in both sessions at the same time? Well that's the important question. Do you *really* want it to work at the same time, or do you want it to hand over gracefully when you switch sessions. Most systems (including OSX and Windows etc. - although I've had odd experience with Windows...) do the latter but some users want the former. If you fix/debug the above mentioned issues, then you'll get a nice handover, but if you really do want both at the same time output, then the simplest way (if you are generally always logged in) is as follows: 1. Add your user to the audio group, but not your wife. 2. Login as you. 3. Start paprefs and tick the Enable Network Access box. 4. Copy the file ~/.pulse-cookie to your wife's home directory (so that you both have the same cookie file). 5. Edit/create the file ~/.pulse/client.conf in your wife's home directory and put the line default-server = localhost This will mean you run the PA daemon and your wife connects to your daemon. You can also use a system-wide daemon but this is probably easier and at least means one user can benefit from SHM IPC whereas with system-wide no users can. HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
Hi, 'Twas brillig, and Robert Gabriel at 09/12/10 11:10 did gyre and gimble: Hello, Im trying to implement pulseaudio on a Slackware based distribution which Im working on. I have also rebuild phonon so it will support properly pulseaudio and gstreamer package too. They really should compile support for this in by default. Basically it works, just there is no actual sound and I get this error in syslog: Dec 9 11:08:06 kongoni pulseaudio[898]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Dec 9 11:08:07 kongoni pulseaudio[898]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Dec 9 11:08:07 kongoni pulseaudio[898]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Dec 9 11:08:07 kongoni pulseaudio[898]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value min_avail. Dec 9 11:08:07 kongoni pulseaudio[898]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Dec 9 11:08:07 kongoni pulseaudio[898]: alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! Dec 9 11:08:07 kongoni pulseaudio[898]: alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Dec 9 11:08:07 kongoni pulseaudio[898]: alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value min_avail. This is generally a non-fatal error. It doens't usually result in a total lack of sound. Can you supply the output of pacmd list please? Also have you confirmed that your VM sound output is actually connected to something? e.g. can you get sound from alsa directly? e.g. doing type the following in a shell inside your VM: pasuspender bash speaker-test -D hw:0 -c 2 CTRL+C (after the test is run) exit Does it produce sound? [r...@kongoni ~]$ cat /etc/pulse/default.pa http://default.pa |grep hw load-module module-alsa-sink device=hw:0 load-module module-alsa-source device=hw:0 Is there a reason you had to change the default.pa from the default? Normally udev-detect should do everything for you. It's also generally bad practice to use hw directly, better to use e.g. front:0 NOTE: Pulseaudio is build with --disable-hal if it matters. As we'd recommend seeing as hal is dead these days. Is udev support compiled in? It replaced hal these days. Also what version of PA? I'd recommend 0.9.22. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
PA version is 0.9.22, udev support is build, hal support not. Also now Im not in front of the test system, so I can't know if it's producing sound, but I get this: [kong...@kongoni ~]$ pasuspender bash [kong...@kongoni ~]$ speaker-test -D hw:0 -c 2 speaker-test 1.0.23 Playback device is hw:0 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 8 to 16384 Period size range from 8 to 16384 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 0 - Front Left 1 - Front Right Time per period = 5.444608 0 - Front Left 1 - Front Right Time per period = 6.004962 0 - Front Left 1 - Front Right pacmd list: [kong...@kongoni ~]$ pacmd list Welcome to PulseAudio! Use help for usage information. Memory blocks currently allocated: 1, size: 64.0 KiB. Memory blocks allocated during the whole lifetime: 1156, size: 3.8 MiB. Memory blocks imported from other processes: 0, size: 0 B. Memory blocks exported to other processes: 0, size: 0 B. Total sample cache size: 0 B. Default sample spec: s16le 2ch 48000Hz Default channel map: front-left,front-right Default sink name: alsa_output.hw_0 Default source name: alsa_input.hw_0 Memory blocks of type POOL: 1 allocated/1 accumulated. Memory blocks of type POOL_EXTERNAL: 0 allocated/0 accumulated. Memory blocks of type APPENDED: 0 allocated/0 accumulated. Memory blocks of type USER: 0 allocated/0 accumulated. Memory blocks of type FIXED: 0 allocated/1155 accumulated. Memory blocks of type IMPORTED: 0 allocated/0 accumulated. 20 module(s) loaded. index: 0 name: module-device-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute state of devices module.version = 0.9.22 index: 1 name: module-stream-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore the volume/mute/device state of streams module.version = 0.9.22 index: 2 name: module-card-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore profile of cards module.version = 0.9.22 index: 3 name: module-augment-properties argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Augment the property sets of streams with additional static information module.version = 0.9.22 index: 4 name: module-alsa-sink argument: device=hw:0 used: 0 load once: no properties: module.author = Lennart Poettering module.description = ALSA Sink module.version = 0.9.22 index: 5 name: module-alsa-source argument: device=hw:0 used: 0 load once: no properties: module.author = Lennart Poettering module.description = ALSA Source module.version = 0.9.22 index: 6 name: module-udev-detect argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Detect available audio hardware and load matching drivers module.version = 0.9.22 index: 7 name: module-bluetooth-discover argument: used: -1 load once: yes properties: module.author = Joao Paulo Rechi Vita module.description = Detect available bluetooth audio devices and load bluetooth audio drivers module.version = 0.9.22 index: 8 name: module-esound-protocol-unix argument: used: -1 load once: no properties: module.author = Lennart Poettering module.description = ESOUND protocol (UNIX sockets) module.version = 0.9.22 index: 9 name: module-native-protocol-unix argument: used: -1 load once: no properties: module.author = Lennart Poettering module.description = Native protocol (UNIX sockets) module.version = 0.9.22 index: 10 name: module-gconf argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = GConf Adapter module.version = 0.9.22 index: 11 name: module-default-device-restore argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = Automatically restore the default sink and source module.version = 0.9.22 index: 12 name: module-rescue-streams argument: used: -1 load once: yes properties: module.author = Lennart Poettering module.description = When a sink/source is removed, try to move their streams to the default sink/source module.version = 0.9.22 index: 13 name: module-always-sink argument: used: -1 load once: yes properties: module.author = Colin Guthrie module.description = Always keeps at least one sink loaded even if it's a null one module.version = 0.9.22 Can't paste everything... too big :) On Thu, Dec 9, 2010 at 12:43, Colin Guthrie gm...@colin.guthr.ie wrote: Hi, 'Twas brillig, and Robert Gabriel at 09/12/10 11:10 did gyre and gimble: Hello, Im trying to implement pulseaudio on a Slackware based distribution which Im working on. I have also rebuild phonon so it will
[pulseaudio-discuss] [PATCH 3/3] Fighting rewinds: Make sure there is some headroom after an underrun
If the amount of data in the implementor buffer is very tiny,
i e even less than what we will likely be asked for, don't ask
for a rewind as that would lead to another underrun.
--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
From 3c0bf348c3395b3cff0d77fd52a2e1e725c6e4cd Mon Sep 17 00:00:00 2001
From: David Henningsson david.hennings...@canonical.com
Date: Thu, 9 Dec 2010 14:25:58 +0100
Subject: [PATCH 3/3] Fighting rewinds: Make sure there is some headroom after an underrun
If the amount of data in the implementor buffer is very tiny,
i e even less than what we will likely be asked for, don't ask
for a rewind as that would lead to another underrun.
Signed-off-by: David Henningsson david.hennings...@canonical.com
---
src/pulsecore/protocol-native.c | 15 +--
1 files changed, 13 insertions(+), 2 deletions(-)
diff --git a/src/pulsecore/protocol-native.c b/src/pulsecore/protocol-native.c
index 5dab80e..378a401 100644
--- a/src/pulsecore/protocol-native.c
+++ b/src/pulsecore/protocol-native.c
@@ -1321,9 +1321,18 @@ static void handle_seek(playback_stream *s, int64_t indexw) {
/* pa_log(%lu vs. %lu, (unsigned long) pa_memblockq_get_length(s-memblockq), (unsigned long) pa_memblockq_get_prebuf(s-memblockq)); */
if (pa_memblockq_is_readable(s-memblockq)) {
+if (s-sink_input-sink) {
+pa_usec_t usec = pa_sink_get_latency_within_thread(s-sink_input-sink);
+int latency = pa_usec_to_bytes(usec, s-sink_input-sample_spec);
+if (latency pa_memblockq_get_length(s-memblockq)) {
+pa_log_debug(Skipping rewind, need at least %d bytes., latency);
+playback_stream_request_bytes(s);
+return;
+}
+}
-/* We just ended an underrun, let's ask the sink
- * for a complete rewind rewrite */
+/* We ended an underrun and got some head start,
+ * let's ask the sink for a complete rewind rewrite */
pa_log_debug(Requesting rewind due to end of underrun.);
pa_sink_input_request_rewind(s-sink_input,
@@ -1524,6 +1533,8 @@ static int sink_input_pop_cb(pa_sink_input *i, size_t nbytes, pa_memchunk *chunk
s-is_underrun = TRUE;
playback_stream_request_bytes(s);
+/* Don't return a block here - it confuses the underrun handling in sink-input later */
+return -1;
}
/* This call will not fail with prebuf=0, hence we check for
--
1.7.1
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Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
'Twas brillig, and Robert Gabriel at 09/12/10 14:19 did gyre and gimble: No, the sound works in the VM. Works perfectly in VirtualBox when using just alsa. I also remove load-module alsa-sink and still the same. OK, then thanks for confirming. NB: Make sure you have not SSH'ed in to the machine with X11 forwarding enabled from a Machine running PulseAudio. In this scenario PA will try to play sound on your (e.g. the source) machine, not inside the VM. So what I'd like to see (all commands to be run as a regular user, *not* root, I'll prefix the command with sudo if it should be run by root). 1. Full pacmd list output 2. ck-list-sessions 3. getfacl /dev/snd/* 4. sudo lsof | grep /dev/snd 5. PULSE_LOG=99 paplay -v /path/to/a/wav.wav That'll probably be enough to get going on with :D I'll try and advise as best I can once that debug info is available. Take care Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
Apologies, but your HTML formatting makes this reply hard to read. As you had the manually added sinks, they actually technically conflicted with the built in (i.e. udev detected) sinks. They both referred to the same alsa device, and thus during our probing phase, we were unable to open the alsa devices (as we were essentially conflicting with ourselves). See annotation below. 'Twas brillig, and Robert Gabriel at 09/12/10 11:57 did gyre and gimble: 2 sink(s) available. * index: 0 name: alsa_output.hw_0 driver: module-alsa-sink.c flags: HARDWARE DECIBEL_VOLUME LATENCY state: SUSPENDED suspend cause: IDLE priority: 9950 volume: 0: 100% 1: 100% 0: 0.00 dB 1: 0.00 dB balance 0.00 base volume: 100% 0.00 dB volume steps: 65537 muted: no current latency: 0.00 ms max request: 0 KiB max rewind: 0 KiB monitor source: 0 sample spec: s16le 2ch 48000Hz channel map: front-left,front-right Stereo used by: 0 linked by: 0 fixed latency: 80.00 ms module: 4 properties: alsa.resolution_bits = 16 device.api = alsa device.class = sound alsa.class = generic alsa.subclass = generic-mix alsa.name http://alsa.name = Intel 82801AA-ICH alsa.id http://alsa.id = Intel ICH alsa.subdevice = 0 alsa.subdevice_name = subdevice #0 alsa.device = 0 alsa.card = 0 alsa.card_name = Intel 82801AA-ICH alsa.long_card_name = Intel 82801AA-ICH with STAC9700,83,84 at irq 21 alsa.driver_name = snd_intel8x0 device.bus_path = pci-:00:05.0 sysfs.path = /devices/pci:00/:00:05.0/sound/card0 device.bus = pci device.vendor.id http://device.vendor.id = 8086 device.vendor.name http://device.vendor.name = Intel Corporation device.product.id http://device.product.id = 2415 device.product.name http://device.product.name = 82801AA AC'97 Audio Controller device.form_factor = internal device.string = hw:0 device.buffering.buffer_size = 15360 device.buffering.fragment_size = 1920 device.access_mode = mmap device.description = Internal Audio device.icon_name = audio-card-pci The above is from your manually added sink. I'd remove it from your default.pa and retart PA. index: 1 name: alsa_output.pci-_00_05.0.analog-stereo driver: module-alsa-card.c flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY state: SUSPENDED suspend cause: IDLE priority: 9959 volume: 0: 28% 1: 28% 0: -33.00 dB 1: -33.00 dB balance 0.00 base volume: 63% -12.00 dB volume steps: 65537 muted: no current latency: 0.00 ms max request: 0 KiB max rewind: 0 KiB monitor source: 2 sample spec: s16le 2ch 48000Hz channel map: front-left,front-right Stereo used by: 0 linked by: 0 fixed latency: 80.00 ms card: 0 alsa_card.pci-_00_05.0 module: 18 properties: alsa.resolution_bits = 16 device.api = alsa device.class = sound alsa.class = generic alsa.subclass = generic-mix alsa.name http://alsa.name = Intel 82801AA-ICH alsa.id http://alsa.id = Intel ICH alsa.subdevice = 0 alsa.subdevice_name = subdevice #0 alsa.device = 0 alsa.card = 0 alsa.card_name = Intel 82801AA-ICH alsa.long_card_name = Intel 82801AA-ICH with STAC9700,83,84 at irq 21 alsa.driver_name = snd_intel8x0 device.bus_path = pci-:00:05.0 sysfs.path = /devices/pci:00/:00:05.0/sound/card0 device.bus = pci device.vendor.id http://device.vendor.id = 8086 device.vendor.name http://device.vendor.name = Intel Corporation device.product.id http://device.product.id = 2415 device.product.name http://device.product.name = 82801AA AC'97 Audio Controller device.form_factor = internal device.string = front:0 device.buffering.buffer_size = 15360 device.buffering.fragment_size = 1920 device.access_mode = mmap device.profile.name http://device.profile.name = analog-stereo device.profile.description = Analog Stereo device.description = Internal Audio Analog Stereo alsa.mixer_name = SigmaTel STAC9700,83,84 alsa.components = AC97a:83847600 module-udev-detect.discovered = 1 device.icon_name = audio-card-pci ports: analog-output;output-amplifier-on: Analog Output / Amplifier (priority 9910) analog-output;output-amplifier-off: Analog Output / No Amplifier (priority 9900) analog-output-mono;output-amplifier-on: Analog Mono Output / Amplifier (priority 5010) analog-output-mono;output-amplifier-off: Analog Mono Output / No Amplifier (priority 5000) analog-output-lfe-on-mono;output-amplifier-on: Analog Output (LFE) / Amplifier (priority 4010) analog-output-lfe-on-mono;output-amplifier-off: Analog Output (LFE) / No Amplifier (priority 4000) active port: analog-output;output-amplifier-on This is the interesting bit. You'll see that you have various potential ports available to use. The active port is actually the one with highest priority, so it's likely the best choice, but just to be sure, you can try other ports (see the Output Devices tab in pavucontrol) If possible please try playing some sound and then
Re: [pulseaudio-discuss] Mic input volume controls
'Twas brillig, and Kulikov, Vitaliy at 01/12/10 17:34 did gyre and gimble: Hello everybody, I would like to add more details for issue #1. Just changing order of the capture and MIC volume controls in the path is not enough if those volumes have gain only ranges and the reason is that, AFAIK, currently PA adjusts requested volume to the higher HW volume step. But if we change it to lower HW step it should work just fine. Here is the example how that works. Let's say we have 22.5 dB gain only range with the 1.5 dB steps for capture HW control and 40 dB gain only range with the 10 dB steps for MIC boost control. Now, for all volumes between 0 and 10 dB PA will keep MIC boost level at 0 dB (when it is first in the path) and set capture level to 1.5, 3, 4.5 etc levels. And when level reaches 10 dB, then MIC boost will be set at 10 dB and capture at 0 dB. For the requested levels between 10 and 20 db, MIC boost will be kept at 10 dB and capture level will take the difference and so on. What you describe above is how PA currently works with regards to volume changes. We use the alsa API with a +1 dir argument to various functions e.g. http://www.alsa-project.org/alsa-doc/alsa-lib/group___simple_mixer.html#gef9c6ce9deb46de7b5727dc9982dc6d6 So we will use accurate or first above this means that less attenuation will be done if the accuracy cannot be performed exactly. So no need to worry about this. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] [PATCH 0/3] Fighting rewinds
However, the problem is quite complex and there does not seem to be one perfect fix, it's more of an optimisation problem. GStreamer in particular sends out many small data packages, and PulseAudio does not handle that very well. That's the default behavior, but you can cut the traffic by using the latency-time property in pulsesink. This makes sure you send bigger buffers, up to the 64k limit that PulseAudio has internally. -Pierre ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
OK, here it is: pacmd list: http://pastebin.com/raw.php?i=Kn7tZBH4 [kong...@kongoni ~]$ ck-list-sessions Session1: unix-user = '1000' realname = 'Kongoni' seat = 'Seat1' session-type = '' active = TRUE x11-display = ':0' x11-display-device = '/dev/tty7' display-device = '' remote-host-name = '' is-local = TRUE on-since = '2010-12-09T17:02:12.843441Z' login-session-id = '' [kong...@kongoni ~]$ getfacl /dev/snd/* getfacl: Removing leading '/' from absolute path names # file: dev/snd/by-path # owner: root # group: root user::rwx group::r-x other::r-x # file: dev/snd/controlC0 # owner: root # group: audio user::rw- user:kongoni:rw- group::rw- mask::rw- other::--- # file: dev/snd/pcmC0D0c # owner: root # group: audio user::rw- user:kongoni:rw- group::rw- mask::rw- other::--- # file: dev/snd/pcmC0D0p # owner: root # group: audio user::rw- user:kongoni:rw- group::rw- mask::rw- other::--- # file: dev/snd/pcmC0D1c # owner: root # group: audio user::rw- user:kongoni:rw- group::rw- mask::rw- other::--- # file: dev/snd/timer # owner: root # group: audio user::rw- user:kongoni:rw- group::rw- mask::rw- other::--- [kong...@kongoni ~]$ sudo lsof |grep /dev/snd kmix 1114kongoni 12u CHR 116,6 0t0 2372 /dev/snd/controlC0 pulseaudi 1121kongoni 21u CHR 116,6 0t0 2372 /dev/snd/controlC0 pulseaudi 1121kongoni 28u CHR 116,6 0t0 2372 /dev/snd/controlC0 PULSE_LOG=99 paplay -v /path/to/a/wav.wav - running the sound work properly! But even if the sound work properly when loggin into KDE the error is still there and the login sound didn't work, but now KDE didn't throw an error and Falling back to . The pop-up error from the KDE notifier or whatever is called. On Thu, Dec 9, 2010 at 16:51, Colin Guthrie gm...@colin.guthr.ie wrote: Apologies, but your HTML formatting makes this reply hard to read. As you had the manually added sinks, they actually technically conflicted with the built in (i.e. udev detected) sinks. They both referred to the same alsa device, and thus during our probing phase, we were unable to open the alsa devices (as we were essentially conflicting with ourselves). See annotation below. 'Twas brillig, and Robert Gabriel at 09/12/10 11:57 did gyre and gimble: 2 sink(s) available. * index: 0 name: alsa_output.hw_0 driver: module-alsa-sink.c flags: HARDWARE DECIBEL_VOLUME LATENCY state: SUSPENDED suspend cause: IDLE priority: 9950 volume: 0: 100% 1: 100% 0: 0.00 dB 1: 0.00 dB balance 0.00 base volume: 100% 0.00 dB volume steps: 65537 muted: no current latency: 0.00 ms max request: 0 KiB max rewind: 0 KiB monitor source: 0 sample spec: s16le 2ch 48000Hz channel map: front-left,front-right Stereo used by: 0 linked by: 0 fixed latency: 80.00 ms module: 4 properties: alsa.resolution_bits = 16 device.api = alsa device.class = sound alsa.class = generic alsa.subclass = generic-mix alsa.name http://alsa.name = Intel 82801AA-ICH alsa.id http://alsa.id = Intel ICH alsa.subdevice = 0 alsa.subdevice_name = subdevice #0 alsa.device = 0 alsa.card = 0 alsa.card_name = Intel 82801AA-ICH alsa.long_card_name = Intel 82801AA-ICH with STAC9700,83,84 at irq 21 alsa.driver_name = snd_intel8x0 device.bus_path = pci-:00:05.0 sysfs.path = /devices/pci:00/:00:05.0/sound/card0 device.bus = pci device.vendor.id http://device.vendor.id = 8086 device.vendor.name http://device.vendor.name = Intel Corporation device.product.id http://device.product.id = 2415 device.product.name http://device.product.name = 82801AA AC'97 Audio Controller device.form_factor = internal device.string = hw:0 device.buffering.buffer_size = 15360 device.buffering.fragment_size = 1920 device.access_mode = mmap device.description = Internal Audio device.icon_name = audio-card-pci The above is from your manually added sink. I'd remove it from your default.pa and retart PA. index: 1 name: alsa_output.pci-_00_05.0.analog-stereo driver: module-alsa-card.c flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY state: SUSPENDED suspend cause: IDLE priority: 9959 volume: 0: 28% 1: 28% 0: -33.00 dB 1: -33.00 dB balance 0.00 base volume: 63% -12.00 dB volume steps: 65537 muted: no current latency: 0.00 ms max request: 0 KiB max rewind: 0 KiB monitor source: 2 sample spec: s16le 2ch 48000Hz channel map: front-left,front-right Stereo used by: 0 linked by: 0 fixed latency: 80.00 ms card: 0 alsa_card.pci-_00_05.0 module: 18 properties: alsa.resolution_bits = 16 device.api = alsa device.class = sound alsa.class = generic alsa.subclass = generic-mix alsa.name http://alsa.name = Intel 82801AA-ICH alsa.id http://alsa.id = Intel ICH alsa.subdevice = 0 alsa.subdevice_name = subdevice #0
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
[ Just reiterating my previous request to not top post (it's bad etiquette on 99% of mailing lists, and also to trim the quoted sections appropriately rather than include them verbatim at the end - that's of no use to anybody. Thanks ] 'Twas brillig, and Robert Gabriel at 09/12/10 17:09 did gyre and gimble: OK, here it is: pacmd list: http://pastebin.com/raw.php?i=Kn7tZBH4 Cool. All looks good now that you've removed the manual sink you added. [kong...@kongoni ~]$ ck-list-sessions Session1: unix-user = '1000' realname = 'Kongoni' seat = 'Seat1' session-type = '' active = TRUE ^ This is the interesting bit. It's set correctly :) [kong...@kongoni ~]$ getfacl /dev/snd/* # file: dev/snd/pcmC0D0c # owner: root # group: audio user::rw- user:kongoni:rw- ^^^ Your user appearing in the ACL is what I wanted to check here. All good. [kong...@kongoni ~]$ sudo lsof |grep /dev/snd kmix 1114kongoni 12u CHR 116,6 0t0 2372 /dev/snd/controlC0 pulseaudi 1121kongoni 21u CHR 116,6 0t0 2372 /dev/snd/controlC0 pulseaudi 1121kongoni 28u CHR 116,6 0t0 2372 /dev/snd/controlC0 Cool. No application hogging things. One thing I would point out is that kmix is accessing /dev/snd/controlC0 directly. If kmix is build with pulseaudio support, it should not do that. I'll just go via PulseAudio. This shouldn't be a problem, but you will see confusing things by looking directly at the alsa mixer while pulseaudio is used, primary due to it's Flat Volume logic. A Kmix with PA support is likely the better option for you. PULSE_LOG=99 paplay -v /path/to/a/wav.wav - running the sound work properly! Great! So it seems simply that it's a KDE problem. But even if the sound work properly when loggin into KDE the error is still there and the login sound didn't work, but now KDE didn't throw an error and Falling back to . The pop-up error from the KDE notifier or whatever is called. This shouldn't happen with a fairly up to date Phonon build. What version are you using? 4.4.3 was released just the other day. I'd recommend that. Is PulseAudio auto-spawning enabled? (it is by default - you can check if you have autospawn=no in /etc/pulse/client.conf or ~/.pulse/client.conf. If you have not specifically disabled the autospawning, it will work automatically (as it's the default). This page may have some useful information for you (incl. screen shots as to how things are supposed to look and work etc.): http://pulseaudio.org/wiki/KDE Either way this is more of a KDE problem than a PA one. Although it'll likely be myself that helps you either way (as I wrote all the KDE integration). HTHs Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
On Thu, Dec 9, 2010 at 18:54, Colin Guthrie gm...@colin.guthr.ie wrote:
[ Just reiterating my previous request to not top post (it's bad
etiquette on 99% of mailing lists, and also to trim the quoted sections
appropriately rather than include them verbatim at the end - that's of
no use to anybody. Thanks ]
'Twas brillig, and Robert Gabriel at 09/12/10 17:09 did gyre and gimble:
OK, here it is:
pacmd list: http://pastebin.com/raw.php?i=Kn7tZBH4
Cool. All looks good now that you've removed the manual sink you added.
[kong...@kongoni ~]$ ck-list-sessions
Session1:
unix-user = '1000'
realname = 'Kongoni'
seat = 'Seat1'
session-type = ''
active = TRUE
^
This is the interesting bit. It's set correctly :)
[kong...@kongoni ~]$ getfacl /dev/snd/*
# file: dev/snd/pcmC0D0c
# owner: root
# group: audio
user::rw-
user:kongoni:rw-
^^^
Your user appearing in the ACL is what I wanted to check here. All good.
[kong...@kongoni ~]$ sudo lsof |grep /dev/snd
kmix 1114 kongoni 12u CHR 116,6 0t0
2372 /dev/snd/controlC0
pulseaudi 1121 kongoni 21u CHR 116,6 0t0
2372 /dev/snd/controlC0
pulseaudi 1121 kongoni 28u CHR 116,6 0t0
2372 /dev/snd/controlC0
Cool. No application hogging things. One thing I would point out is that
kmix is accessing /dev/snd/controlC0 directly. If kmix is build with
pulseaudio support, it should not do that. I'll just go via PulseAudio.
This shouldn't be a problem, but you will see confusing things by
looking directly at the alsa mixer while pulseaudio is used, primary due
to it's Flat Volume logic. A Kmix with PA support is likely the better
option for you.
PULSE_LOG=99 paplay -v /path/to/a/wav.wav - running the sound work
properly!
Great!
So it seems simply that it's a KDE problem.
But even if the sound work properly when loggin into KDE the error is
still there and the login sound didn't work,
but now KDE didn't throw an error and Falling back to . The pop-up
error from the KDE notifier or whatever is called.
This shouldn't happen with a fairly up to date Phonon build. What
version are you using? 4.4.3 was released just the other day. I'd
recommend that.
I just wrote before that phonon was just build with pulseaudio support
and of course the version is 4.4.3.
Rebuild kmix... isn't in my list, as it most probably need to rebuild
KDE as this is how Slackware doesn't things.
So for me should be a different fix.
Now in Phonon config there is Default Audio Device which is set in
/etc/asound.conf, but
this is not the default and would like to make default globaly, just
not sure how.
pcm.!default {
type pulse
hint.description Default Audio Device
}
Is PulseAudio auto-spawning enabled? (it is by default - you can check
if you have autospawn=no in /etc/pulse/client.conf or
~/.pulse/client.conf. If you have not specifically disabled the
autospawning, it will work automatically (as it's the default).
This page may have some useful information for you (incl. screen shots
as to how things are supposed to look and work etc.):
http://pulseaudio.org/wiki/KDE
Either way this is more of a KDE problem than a PA one. Although it'll
likely be myself that helps you either way (as I wrote all the KDE
integration).
HTHs
Col
--
Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/
Day Job:
Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
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Re: [pulseaudio-discuss] Forcing a mapping from a stream to a device
Hi Matt, Sorry for the late reply. The message was stuck in the moderation queue and I hadn't checked it for a while :( I've a funny feeling we discussed this on IRC but perhaps not. 'Twas brillig, and Matt Feifarek at 19/11/10 21:59 did gyre and gimble: Thanks for PulseAudio; I'm a big fan. I have two different sound devices in my system; a sound card and a usb telephony device. I want to force certain streams (by software source) to use the telephony device but leave music and other things going to the standard sound card. I can't seem to make this work; I can make either device default and then manually move streams from one sink to another, but PA doesn't seem to remember correctly. Each phone call requires that I move the stream again. Normally the device will be remembered and moving it once should be enough. Not sure why this wouldn't work (module-stream-restore takes care of this for you). Can you provide output pacmd list-sink-inputs output when one of these calls is in progress. The interesting bit will be the stream restore id in the proplist. Specifically, I want to use the new Gmail dial phones feature in a browser, in this case Chrome, with a ClearOne Chat 50... but I want everything else to go through my desktop speakers. I suppose if I could get it to work, I'd try Skype next. This will be more complicated as the browser is a general purpose system, and yet it's trying to do use-case specific things. Skype should work fine. Skype even identifies it's streams properly - phone for it call audio and event for the ringing sound etc. Generally speaking Chrome will just identify itself as a generic stream and thus if you use it for phone calls but also e.g. soundcloud.com or last.fm etc. then these are all handled as the same type of stream and will be saved and restored accordingly. There is no easy way to identify them separately :( Is there some invocation I can add to my pulse config files to force this? I've tried using Earcandy to do this, but it's not very reliable. Generally speaking, pavucontrol is the app you want. I have no idea what earcandy is. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
'Twas brillig, and Robert Gabriel at 09/12/10 18:09 did gyre and gimble:
I just wrote before that phonon was just build with pulseaudio support
and of course the version is 4.4.3.
OK, that's good to know. I appreciated you built you're own but you
never know which version people use - e.g. there may have been a dep
that required you use an older version.
Rebuild kmix... isn't in my list, as it most probably need to rebuild
KDE as this is how Slackware doesn't things.
So for me should be a different fix.
Now in Phonon config there is Default Audio Device which is set in
/etc/asound.conf, but
this is not the default and would like to make default globaly, just
not sure how.
pcm.!default {
type pulse
hint.description Default Audio Device
}
Hmm, this is very odd. The ALSA hint should *not* appear in Phonon
config if PulseAudio support is fully working. Can you attach/link a
screen shot just to be sure?
The device list should mirror your PulseAudio setup (e.g. the same list
of devices as is shown in the Output Devices tab of pavucontrol.
The page I linked earlier has screen shots of kcm_phonon config screens.
It appears you are seeing ALSA devices in that list due to the fact that
the ALSA hint is being disaplayed and this should not happen with a
fully integrated setup.
Can you run:
PHONON_PULSEAUDIO_DEBUG=4 kcmshell4 kcm_phonon
And post/link the debug output please?
Col
--
Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/
Day Job:
Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
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Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
OK here it is:
Screenshot of phonon: http://mirror.visualserver.org/phonon.png
Output of phonon: http://pastebin.com/Cikqx7QM
On Thu, Dec 9, 2010 at 20:34, Colin Guthrie gm...@colin.guthr.ie wrote:
'Twas brillig, and Robert Gabriel at 09/12/10 18:09 did gyre and gimble:
I just wrote before that phonon was just build with pulseaudio support
and of course the version is 4.4.3.
OK, that's good to know. I appreciated you built you're own but you
never know which version people use - e.g. there may have been a dep
that required you use an older version.
Rebuild kmix... isn't in my list, as it most probably need to rebuild
KDE as this is how Slackware doesn't things.
So for me should be a different fix.
Now in Phonon config there is Default Audio Device which is set in
/etc/asound.conf, but
this is not the default and would like to make default globaly, just
not sure how.
pcm.!default {
type pulse
hint.description Default Audio Device
}
Hmm, this is very odd. The ALSA hint should *not* appear in Phonon
config if PulseAudio support is fully working. Can you attach/link a
screen shot just to be sure?
The device list should mirror your PulseAudio setup (e.g. the same list
of devices as is shown in the Output Devices tab of pavucontrol.
The page I linked earlier has screen shots of kcm_phonon config screens.
It appears you are seeing ALSA devices in that list due to the fact that
the ALSA hint is being disaplayed and this should not happen with a
fully integrated setup.
Can you run:
PHONON_PULSEAUDIO_DEBUG=4 kcmshell4 kcm_phonon
And post/link the debug output please?
Col
--
Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/
Day Job:
Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
Mageia Contributor [http://www.mageia.org/]
PulseAudio Hacker [http://www.pulseaudio.org/]
Trac Hacker [http://trac.edgewall.org/]
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Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
'Twas brillig, and Robert Gabriel at 09/12/10 20:09 did gyre and gimble: OK here it is: Screenshot of phonon: http://mirror.visualserver.org/phonon.png OK, so as I'm sure you're already aware from reading the link I previously gave you, you do not have proper PulseAudio integration here. Something with your build is broken. Your System Settings window should look like that shown in the screenshots I attached to the link. Output of phonon: http://pastebin.com/Cikqx7QM Line 73 is the most interesting PulseSupport(2): Enabled Breakdown: mEnabled: No, s_pulseActive Yes mEnabled is No. This basically means that the backend you are using does not support PulseAudio. What Phonon Backend are you using? If you've not done so already, I'd rebuild this backend against your new Phonon build. If you installed your own build of phonon into e.g. /usr/local, you must ensure that the system installed phonon backend is not being used. Col. -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
Im not sure what you mean with phonon backend? What exactly can that
be, give me some examples :)
Phonon is build properly for the system where it should be
/usr/{bin,lib,share} and stuff like that:
On Thu, Dec 9, 2010 at 21:18, Colin Guthrie gm...@colin.guthr.ie wrote:
'Twas brillig, and Robert Gabriel at 09/12/10 20:09 did gyre and gimble:
OK here it is:
Screenshot of phonon: http://mirror.visualserver.org/phonon.png
OK, so as I'm sure you're already aware from reading the link I
previously gave you, you do not have proper PulseAudio integration here.
Something with your build is broken.
Your System Settings window should look like that shown in the
screenshots I attached to the link.
Output of phonon: http://pastebin.com/Cikqx7QM
Line 73 is the most interesting
PulseSupport(2): Enabled Breakdown: mEnabled: No, s_pulseActive Yes
mEnabled is No.
This basically means that the backend you are using does not support
PulseAudio.
What Phonon Backend are you using? If you've not done so already, I'd
rebuild this backend against your new Phonon build. If you installed
your own build of phonon into e.g. /usr/local, you must ensure that the
system installed phonon backend is not being used.
Col.
--
Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/
Day Job:
Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
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Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
'Twas brillig, and Robert Gabriel at 09/12/10 20:20 did gyre and gimble:
Im not sure what you mean with phonon backend? What exactly can that
be, give me some examples :)
Phonon is build properly for the system where it should be
/usr/{bin,lib,share} and stuff like that:
Phonon is just a library. It needs a backend to actually do the audio
decoding. Typical backends include GStreamer, Xine and VLC (only these
three support PulseAudio and even then Xine is rubbish and not recommended).
In the screenshot you previously posted you'll notice a Backend tab...
try clicking on it and then having a look.
This is turning into something that is really not relevant to the PA
mailing list. If you are able, please come on to IRC and join
#pulseaudio on Freenode. I'm coling there and I can likely help you
interactively for the next few hours.
Cheers
Col
--
Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/
Day Job:
Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
Mageia Contributor [http://www.mageia.org/]
PulseAudio Hacker [http://www.pulseaudio.org/]
Trac Hacker [http://trac.edgewall.org/]
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Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
OK I get it now, I'm just a brighter idiot now :)
Anyway, xine, at the moment is not build with pulseaudio support,
but gstreamer has
but still doesn't work, if I set gstreamer as preferred backend.
On Thu, Dec 9, 2010 at 21:28, Colin Guthrie gm...@colin.guthr.ie wrote:
'Twas brillig, and Robert Gabriel at 09/12/10 20:20 did gyre and gimble:
Im not sure what you mean with phonon backend? What exactly can that
be, give me some examples :)
Phonon is build properly for the system where it should be
/usr/{bin,lib,share} and stuff like that:
Phonon is just a library. It needs a backend to actually do the audio
decoding. Typical backends include GStreamer, Xine and VLC (only these
three support PulseAudio and even then Xine is rubbish and not recommended).
In the screenshot you previously posted you'll notice a Backend tab...
try clicking on it and then having a look.
This is turning into something that is really not relevant to the PA
mailing list. If you are able, please come on to IRC and join
#pulseaudio on Freenode. I'm coling there and I can likely help you
interactively for the next few hours.
Cheers
Col
--
Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/
Day Job:
Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
Mageia Contributor [http://www.mageia.org/]
PulseAudio Hacker [http://www.pulseaudio.org/]
Trac Hacker [http://trac.edgewall.org/]
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Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
'Twas brillig, and Robert Gabriel at 09/12/10 20:37 did gyre and gimble: OK I get it now, I'm just a brighter idiot now :) Anyway, xine, at the moment is not build with pulseaudio support, but gstreamer has but still doesn't work, if I set gstreamer as preferred backend. Can you resupply the kcmshell debug after switching backends? Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
'Twas brillig, and Robert Gabriel at 09/12/10 21:07 did gyre and gimble: Here it is... seems to not work so well :) http://pastebin.com/dmu0eWTf That all looks correct to me. What problems are you seeing? (try a full reboot now this is configured) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
OK, seems to work but daemon.conf needs some tweaks. As a stupid question, does xine support pulseaudio? On Thu, Dec 9, 2010 at 22:11, Colin Guthrie gm...@colin.guthr.ie wrote: 'Twas brillig, and Robert Gabriel at 09/12/10 21:07 did gyre and gimble: Here it is... seems to not work so well :) http://pastebin.com/dmu0eWTf That all looks correct to me. What problems are you seeing? (try a full reboot now this is configured) Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Mic input volume controls
'Twas brillig, and David Henningsson at 07/12/10 08:35 did gyre and gimble: On 2010-12-04 19:09, Colin Guthrie wrote: 'Twas brillig, and David Henningsson at 02/12/10 10:38 did gyre and Let's play with the idea that we added a configurable Direction key to these profiles that controlled whether we moved up or down to the nearest step. Wouldn't that help? Then we could have Mic Boost first in chain, then set it to move down rather than up. Or maybe that we should instead say that if we're above 0 dB, we move down, and if we're below 0 dB, we move up? Would that make sense? I'm not really sure here. A specific direction key may complicate what is already a rather complicated thing, so if we can possibly do things sensibly without it, then I'd suggest that would be more attractive (if it works reliably enough). We want to maximise quality while avoiding digital distortion, that's basically the problem in a nutshell. We're assuming (sometimes incorrectly; but that's our best guess) that this golden spot will be achieved with all sliders at 0 dB. I think my approach makes sense, unless I'm missing something: If we're aiming for something above 0 dB, let's round down to make sure we avoid distortion, and if we're below 0 dB, let's round up to make sure we get maximum quality. I see what you mean here. At present we always pass +1 to the various alsa functions that take it (e.g. snd_mixer_selem_set_playback_dB()), but after we adjust for base volume, we may be requesting alsa volumes of 0dB (e.g. if our base volume is at 30%, then 35% will be a +dB in alsa). So yeah, I think you are correct here, the rounding needs to factor this in. It's a pity there is not a fourth option to these calls in alsa (i.e. in addition to -1, 0, +1) that did appropriate rounding, but we can deal with this I guess. And then we always start with the control that's closest to the physical hardware and work our way in. I guess so. What's in and out here; that's currently determined by the order in which PulseAudio reads them in, right? That complicates matters a little given the include file structure and so I'm thinking maybe we should add a extremity=number key to the config file? (This is unrelated to the direction key suggested previously.) We would then start with the one having the highest extremity score. I guess that would allow for a clearer, more deterministic approach. Otherwise we would get in trouble if we have a profile with a volume control B (with extremity=2), which then includes a file with both control A (with extremity=1) and control C (with extremity=3) in the same include file. I think that would actually make things clearer. What do you think? While I've never completely grasped absolutely everything in the mixer profiles stuff, I think you speak sense :) Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
Well modified initially resample-method to resample-method=speex-float-1, and the sound was chopy, but modified now to resample-method=speex-float-0 and works fine. OK, I need to know what proper backend to use. This is a distribution and I suppose people want something that works out of the box. Can xine do it? Or I should go to VLC (note this is a GNU distro, so only freee :)) On Thu, Dec 9, 2010 at 22:29, Colin Guthrie gm...@colin.guthr.ie wrote: 'Twas brillig, and Robert Gabriel at 09/12/10 21:22 did gyre and gimble: OK, seems to work but daemon.conf needs some tweaks. Great! Out of curiosity, what did you need to tweak? As a stupid question, does xine support pulseaudio? Not a stupid question! It does have a pulseaudio output layer but in my experience it kinda sucks. Xine is not really actively developed any more and it's very much lost favour by the KDE community. Most hopes are being currently pinned on VLC. It's pulseaudio support is also not great at the moment but I'm hoping to find time to make it work nicely. GStreamer has some issues as a Phonon backend, but has very good pulseaudio integration. Nothing is ever perfect! Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Mic input volume controls
'Twas brillig, and Colin Guthrie at 09/12/10 16:12 did gyre and gimble: 'Twas brillig, and Kulikov, Vitaliy at 01/12/10 17:34 did gyre and gimble: Hello everybody, I would like to add more details for issue #1. Just changing order of the capture and MIC volume controls in the path is not enough if those volumes have gain only ranges and the reason is that, AFAIK, currently PA adjusts requested volume to the higher HW volume step. But if we change it to lower HW step it should work just fine. Here is the example how that works. Let's say we have 22.5 dB gain only range with the 1.5 dB steps for capture HW control and 40 dB gain only range with the 10 dB steps for MIC boost control. Now, for all volumes between 0 and 10 dB PA will keep MIC boost level at 0 dB (when it is first in the path) and set capture level to 1.5, 3, 4.5 etc levels. And when level reaches 10 dB, then MIC boost will be set at 10 dB and capture at 0 dB. For the requested levels between 10 and 20 db, MIC boost will be kept at 10 dB and capture level will take the difference and so on. What you describe above is how PA currently works with regards to volume changes. We use the alsa API with a +1 dir argument to various functions e.g. http://www.alsa-project.org/alsa-doc/alsa-lib/group___simple_mixer.html#gef9c6ce9deb46de7b5727dc9982dc6d6 So we will use accurate or first above this means that less attenuation will be done if the accuracy cannot be performed exactly. So no need to worry about this. I think I perhaps misunderstood your point. Perhaps you were talking about volumes above 0dB, in which case the +1 direction is not really appropriate. I kinda missed that distinction in David's original messages but I saw sense in my latest reply! Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
'Twas brillig, and Robert Gabriel at 09/12/10 21:34 did gyre and gimble: Well modified initially resample-method to resample-method=speex-float-1, and the sound was chopy, but modified now to resample-method=speex-float-0 and works fine. We default to speex-float-0 in Mandriva too. OK, I need to know what proper backend to use. This is a distribution and I suppose people want something that works out of the box. Can xine do it? Or I should go to VLC (note this is a GNU distro, so only freee :)) I think your choice should be between GStreamer and VLC. I'd test both to see which works best for you. VLC is still quite young but it's getting a lot of development of late. You'll probably get differing opinions. Mandriva and I think Fedora and SuSE, use GStreamer by default but not 100% certain. If you do go with GStreamer, there is a patch related to PA output recently committed that you should add in: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=8ca094795add18733faeb2a1f335bb33f40f9894 Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Mic input volume controls
-Original Message- From: pulseaudio-discuss-boun...@mail.0pointer.de [mailto:pulseaudio-discuss-boun...@mail.0pointer.de] On Behalf Of Colin Guthrie Sent: Thursday, December 09, 2010 3:27 PM To: pulseaudio-discuss@mail.0pointer.de Subject: Re: [pulseaudio-discuss] Mic input volume controls 'Twas brillig, and Colin Guthrie at 09/12/10 16:12 did gyre and gimble: 'Twas brillig, and Kulikov, Vitaliy at 01/12/10 17:34 did gyre and gimble: Hello everybody, I would like to add more details for issue #1. Just changing order of the capture and MIC volume controls in the path is not enough if those volumes have gain only ranges and the reason is that, AFAIK, currently PA adjusts requested volume to the higher HW volume step. But if we change it to lower HW step it should work just fine. Here is the example how that works. Let's say we have 22.5 dB gain only range with the 1.5 dB steps for capture HW control and 40 dB gain only range with the 10 dB steps for MIC boost control. Now, for all volumes between 0 and 10 dB PA will keep MIC boost level at 0 dB (when it is first in the path) and set capture level to 1.5, 3, 4.5 etc levels. And when level reaches 10 dB, then MIC boost will be set at 10 dB and capture at 0 dB. For the requested levels between 10 and 20 db, MIC boost will be kept at 10 dB and capture level will take the difference and so on. What you describe above is how PA currently works with regards to volume changes. We use the alsa API with a +1 dir argument to various functions e.g. http://www.alsa-project.org/alsa-doc/alsa-lib/group___simple_mixer.htm l#gef9c6ce9deb46de7b5727dc9982dc6d6 So we will use accurate or first above this means that less attenuation will be done if the accuracy cannot be performed exactly. So no need to worry about this. I think I perhaps misunderstood your point. Perhaps you were talking about volumes above 0dB, in which case the +1 direction is not really appropriate. I kinda missed that distinction in David's original messages but I saw sense in my latest reply! Yes, that's what I meant and my point was just to support David's suggestions. And I was about to type that scenario above actually does not work with the current PA but then saw your new response to David. Another thing, if my memory does not fail me, I believe that somebody in that same thread stated that 20 dB gain should be good enough for all purposes but that is not what my experience tells me. It may be good enough when someone has its mic close to the mouth but may be not good enough for mic far on the desk. This is why all those mic boost controls with the ranges of 30 or 40 dB gain are in the audio codecs. Vitaliy Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Unable to make pulseaudio work on a Slackware based distribution (Kongoni)
For a proper gstreamer backend what do I need? gstreamer, gst-plugins-base and gst-plugins-good? On Thu, Dec 9, 2010 at 22:52, Colin Guthrie gm...@colin.guthr.ie wrote: 'Twas brillig, and Robert Gabriel at 09/12/10 21:34 did gyre and gimble: Well modified initially resample-method to resample-method=speex-float-1, and the sound was chopy, but modified now to resample-method=speex-float-0 and works fine. We default to speex-float-0 in Mandriva too. OK, I need to know what proper backend to use. This is a distribution and I suppose people want something that works out of the box. Can xine do it? Or I should go to VLC (note this is a GNU distro, so only freee :)) I think your choice should be between GStreamer and VLC. I'd test both to see which works best for you. VLC is still quite young but it's getting a lot of development of late. You'll probably get differing opinions. Mandriva and I think Fedora and SuSE, use GStreamer by default but not 100% certain. If you do go with GStreamer, there is a patch related to PA output recently committed that you should add in: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=8ca094795add18733faeb2a1f335bb33f40f9894 Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
Re: [pulseaudio-discuss] Forcing a mapping from a stream to a device
On Thu, 2010-12-09 at 18:32 +, Colin Guthrie wrote: 'Twas brillig, and Matt Feifarek at 19/11/10 21:59 did gyre and gimble: Is there some invocation I can add to my pulse config files to force this? I've tried using Earcandy to do this, but it's not very reliable. Generally speaking, pavucontrol is the app you want. I have no idea what earcandy is. I just checked it out based on his mail, its quite interesting, monitors the streams and adjusts/fades volume based on some user-defined rules (through a gtk GUI). Not really very flexible though, doesn't work very well in multi-output environments from the same card (like built-in Intel volume controlling both speaker and headphones). It does introduce an additional point of failure though. Matt, you should try pavucontrol instead. ___ pulseaudio-discuss mailing list pulseaudio-discuss@mail.0pointer.de https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss
