Hi Tanu,
I think you weren't rebased on master upstream with this patch or I have a
git problem :)
I can't find this negation of pa_bytes_to_usec "extra =
-pa_bytes_to_usec(p->read_index, pa_stream_get_sample_spec(p->stream));"
Last commit which touch this was made by Tanu Kaskinen - Sun Jul 13
1
On Fri, 7 Jul 2017, at 08:35 PM, Tanu Kaskinen wrote:
> On Fri, 2017-07-07 at 20:33 +0800, Qu Wenruo wrote:
> > After a quick glance into the code (without much knowledge about
> > pulseaudio), I found that pulseaudio is just using sbc library to do the
> > encode.
> >
> > From a2dp_process_ren
On Wed, 2017-07-12 at 09:57 +0200, Grzegorz Kołodziejczyk wrote:
> Hi Tanu,
>
> I think you weren't rebased on master upstream with this patch or I have a
> git problem :)
> I can't find this negation of pa_bytes_to_usec "extra =
> -pa_bytes_to_usec(p->read_index, pa_stream_get_sample_spec(p->stre
On Tue, 2017-07-11 at 23:21 +0300, Denis Shulyaka wrote:
> Hi list,
>
> I am making a small project, a home sound server. It will have a 5.1 analog
> output and S/PDIF optical input, besides others. The server has to mix all
> inputs and send them to the output.
> My initial proof-of-concept with
12 июля 2017 г. 7:42 PM пользователь "Tanu Kaskinen" написал:
On Tue, 2017-07-11 at 23:21 +0300, Denis Shulyaka wrote:
> Hi list,
>
> I am making a small project, a home sound server. It will have a 5.1
analog
> output and S/PDIF optical input, besides others. The server has to mix all
> inputs a
If the description is not updated when moving, the old automatically
generated description will refer to the old master sink after the move,
which is not nice.
---
src/modules/module-equalizer-sink.c | 26 ++
1 file changed, 22 insertions(+), 4 deletions(-)
diff --git a/sr
Asus Xonar DX requires toggling the Front Panel switch when changing
output between the front and back panel.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=99479
---
src/modules/alsa/mixer/paths/analog-output-headphones.conf | 7 +++
src/modules/alsa/mixer/paths/analog-output-lineout.
On Wed, 2017-07-12 at 22:29 +0300, Denis Shulyaka wrote:
> Well, that was more or less expected. Is such support planned?
I haven't heard of anyone planning to implement the feature.
> It means I need to write my own decoding application. But that leads me to
> another question:
> When I open the
12 июля 2017 г. 23:58 пользователь "Tanu Kaskinen" написал:
I don't know how the endpoints are supposed to agree on the format, and
what kind of interface alsa provides for that (if any). The alsa-devel
list might be more appropriate for questions about spdif details.
Ok, I will ask there. I've
On Sat, Jun 3, 2017 at 8:53 PM, David Michael wrote:
> Can this be applied to support building the Solaris module for GNU Hurd?
> The change itself is basic portability (including the POSIX for
> POLLIN etc. and optionally using the Solaris ), so maybe it
> also helps some other systems. I also
Hi all,
I was able to successfully connect using HFP AG profile(git
PulseAudio) with a handsfree device.
If I switch the profile from sound setting to HSP/HFP I can see the
sounds played on the laptop are transferred using SCO channel. Also, i
can record the sound from the handsfree device.
Inste
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