On 03/25/2012 12:59 PM, Maarten Lankhorst wrote:
250 ms default fixed latency won't work well for some applications like wine.
Thanks for the patch. It seems reasonable to set the sink's latency to
match jack's latency, but jack_port_get_total_latency is deprecated and
therefore causes a comp
On 03/23/2012 05:02 PM, Rémi Denis-Courmont wrote:
...and this brokenness is the rule, not the exception. (And in the
example above one can wonder why a translation was made in the first
place.) In some cases these do not even fit their format strings. Btw, I
have seen some crashes in pa_log_leve
Hey David,
Op 26-03-12 11:30, David Henningsson schreef:
> On 03/25/2012 12:59 PM, Maarten Lankhorst wrote:
>> 250 ms default fixed latency won't work well for some applications like wine.
>
> Thanks for the patch. It seems reasonable to set the sink's latency to match
> jack's latency, but jack_
While looking at bug 44397,
https://bugs.freedesktop.org/show_bug.cgi?id=44397
Gnome's volume control somehow wanted to create a filter sink cycle. I'm
attaching a patch to prevent this, which seems to have worked (at least
there is no crash!), but before I push it:
Is there any reason, any
On 03/26/2012 11:54 AM, Maarten Lankhorst wrote:
Hey David,
Op 26-03-12 11:30, David Henningsson schreef:
On 03/25/2012 12:59 PM, Maarten Lankhorst wrote:
250 ms default fixed latency won't work well for some applications like wine.
Thanks for the patch. It seems reasonable to set the sink's
Valid channel id range is from 0 to SND_MIXER_SCHN_LAST,
inclusive, so the array size has to be
SND_MIXER_SCHN_LAST + 1.
---
src/modules/alsa/alsa-mixer.h |2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/src/modules/alsa/alsa-mixer.h b/src/modules/alsa/alsa-mixer.h
index c
So, how can I get a release candidate from Git? What URL should I probe?
There is no branch named "2.0" or like. Sorry, I am not familiar with
your terminology "*happen off* the master branch."
On 19.02.12 18:59, Arun Raghavan wrote:
Hey folks,
In order to make things a bit more regular, we've
On 03/26/2012 01:35 PM, Tanu Kaskinen wrote:
Valid channel id range is from 0 to SND_MIXER_SCHN_LAST,
inclusive, so the array size has to be
SND_MIXER_SCHN_LAST + 1.
This looks correct. A quick grep for SND_MIXER_SCHN_LAST shows similar
arrays in alsa-sink and alsa-source, and also, that my su
Coverity thinks that sample can be NULL when it's
dereferenced after this line. Adding an assertion doesn't
hurt here (in my opinion), and that should get rid of the
warning.
---
src/modules/dbus/iface-core.c |2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/src/modules/dbu
On Mon, 2012-03-26 at 13:44 +0200, David Henningsson wrote:
> On 03/26/2012 01:35 PM, Tanu Kaskinen wrote:
> > Valid channel id range is from 0 to SND_MIXER_SCHN_LAST,
> > inclusive, so the array size has to be
> > SND_MIXER_SCHN_LAST + 1.
>
> This looks correct. A quick grep for SND_MIXER_SCHN_LA
On Mon, 2012-03-26 at 14:42 +0300, Roman Beslik wrote:
> So, how can I get a release candidate from Git? What URL should I probe?
We've uploaded one RC already. Details at:
http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-March/012986.html
> There is no branch named "2.0" or like. S
On Mon, 2012-03-26 at 14:42 +0300, Roman Beslik wrote:
> So, how can I get a release candidate from Git? What URL should I probe?
> There is no branch named "2.0" or like. Sorry, I am not familiar with
> your terminology "*happen off* the master branch."
The phrase means that there won't be a se
'Twas brillig, and Tanu Kaskinen at 26/03/12 13:32 did gyre and gimble:
> On Mon, 2012-03-26 at 14:42 +0300, Roman Beslik wrote:
>> So, how can I get a release candidate from Git? What URL should I probe?
>> There is no branch named "2.0" or like. Sorry, I am not familiar with
>> your terminology
Hi,
I'm trying to install the Terratec 6Fire USB audio device. It's an
external audio device with several analog in & output channels.
The alsa-driver is running and I can play analog stereo music or movies
on pulseaudio based players.
I can output digital sound (ac3 or dts) on spdif via op
On Sun, 2012-03-25 at 16:45 +0200, Matěj Laitl wrote:
> Hi Tanu and pulseaudio-discuss,
> I'd like to participate as a student in GSoC 2012 working on PulseAudio.
> Among
> suggested ideas I've chosen Configurable maximum volume for sinks and
> sources,
> below is a very draft of my proposal.
Changes since v1:
Use max value of jack_port_get_latency_range to calculate the latency
and squash compiler warnings cased by using jack_port_get_total_latency
Modifying latency only works inside a callback, and for hardware the
latency is generally fixed on jack, so just take the max value.
Sign
I use tvtime (an open source TV application) to watch TV on my Fedora 16
system. As some of you may know forwarding audio from the sound card
built into the TV tuner card to the primary sound card is not done by
tvtime. There are various ways of forwarding the audio including SoX,
module-loopback
Hello,
I'm trying to clarify myself the memory requirements of pulseaudio and how
it should be done on embedded system. Actually I have arm based board (TI
dm365) with 128MB of RAM. I'm using the most of this memory as buffers for
video application so only 54MB are available for the linux.
On 03/26/2012 06:20 PM, Tanu Kaskinen wrote:
On Sun, 2012-03-25 at 16:45 +0200, Matěj Laitl wrote:
Hi Tanu and pulseaudio-discuss,
I'd like to participate as a student in GSoC 2012 working on PulseAudio. Among
suggested ideas I've chosen Configurable maximum volume for sinks and sources,
below i
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