When given an explicit device.description in card_properties, prefer
this information over other default prefixes (e.g. 'Built-in Audio')
when constructing sink/source descriptions.
For example, if I manually configure the card description to be
"FooBar", I then expect that the sinks and created by the card also
have "FooBar" in their description instead of generic "Built-in
Audio".
In some cases, "Analog Input" could show up as well as
"Headset Mic" (or "Headphone Mic"), because I forgot to add the
relevant "required-absent" lines when I added the headset mic path.
As a result, both "Analog Input" and "Headset Mic" showed up on the
Logitech USB 530 Headset.
Reported-by: Steve Magoun <steve.magoun@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
CC modules/module_tunnel_sink_la-module-tunnel.lo
modules/module-tunnel-source-new.c: In function 'read_new_samples':
modules/module-tunnel-source-new.c:145:16: warning: declaration of 'read' shadows a global declaration [-Wshadow]
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
If mixer_handle is not NULL, then hctl_handle won't be NULL either.
The redundant check was confusing, because it looked like we would
leak the mixer_handle if mixer_handle is non-NULL and hctl_handle is
NULL.
Currently the latency information is being updated based on the encoded
SBC data instead of the decoded PCM data. Fixing this required moving
the timing update to be after the packet has been decoded.
The Nokia E7 running Symbian Belle Refresh seems to generate invalid SBC
packets every few minutes. This causes pulseaudio to disconnect the
stream and log "SBC decoding error (-3)".
If a single packet is bad, pulseaudio should keep playing the stream.
Some people want module-rtp-send to send silence when the sink that is
monitored goes idle, and some people want module-rtp-send to pause the
RTP stream to avoid unnecessary bandwidth consumption.
If a stream is started corked and remains corked, the sink/source
remained idle without being properly suspended. This patch fixes
that issue.
BugLink: https://bugs.launchpad.net/bugs/1284415
Tested-by: Ricardo Salveti <ricardo.salveti@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This fixes an assertion crash:
[pulseaudio] source.c: Assertion 'PA_SOURCE_IS_LINKED(s->state)' failed at pulsecore/source.c:734, function pa_source_update_status(). Aborting.
The crash happened when a Bluetooth headset profile was changed from
a2dp to hsp. During the profile change three devices are created:
a sink, a monitor source for the sink, and a regular source. First
pa_sink/source_new() are called for each device, and that puts the
devices to u->core->sinks/sources. Then, pa_sink_put() is called for
the sink, and that in turn calls pa_source_put() for the source. At
that point module-device-manager decides to reroute all source
outputs. The non-monitor source that the Bluetooth card created hasn't
been linked yet at this stage, because it will only be linked after
the sink and the monitor source have been linked. So,
module-device-manager should take into account during the rerouting
that not all sinks and sources are necessarily linked. This patch does
that.
Reported-By: Iskren Hadzhinedev <i.hadzhinedev@gmail.com>
A crash was observed that was caused by pa_memblockq_peek() returning
a NULL memblock in sink_input_pop_cb(). The scenario where this was
happening was
1. Delete 2 rtp-recv's connected to a ladspa-sink
2. Delete ladspa-sink
3. Delete alsa-sink
4. Create alsa-sink
5. Create ladspa-sink
6. Create 2 rtp-recv's connected to the ladspa-sink
The crash was probably caused by a rewind that made the read index go
negative while the write index was at least zero, causing there to be
a gap in the memblockq. The problematic rewind might have been caused
by adding the rtp-recv stream to the ladspa-sink. That has not been
proven, but this looks very similar to the bug that was fixed in
module-virtual-sink in commit 6bd34156b1.
The modargs are in both cases (a succesfull as well as a failed module
initialization) freed already in pa__done().
To avoid leaking modargs memory before they are assigned to u->modargs, the
code is reorganized to first allocate userdata, and then allocate the modargs.
Local variable ma is not needed anymore.
discussion here
http://lists.freedesktop.org/archives/pulseaudio-discuss/2013-December/019661.html
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Reported-by: poljar (Damir Jelić) <poljarinho@gmail.com>
I think this makes the code a bit nicer to read and write. This also
reduces the chances of off-by-one errors when checking the bounds of
channel count values.
I think this makes the code a bit nicer to read and write. This also
reduces the chances of off-by-one errors when checking the bounds of
sample rate values.
I think this makes the code a bit nicer to read and write. This also
reduces the chances of off-by-one errors when checking the bounds of
the sample format value.
PCM Devices which have the BATCH flag set update the PCM pointer only with
period size granularity. Using timer based scheduling does not have any
advantage in this mode. For one devices which have that flag set usually update
the position pointer in software after getting the period interrupt. So
disabling the period interrupt is not possible for this kind of devices.
Furthermore writing to or reading from the buffer slice for the current period
is not possible since the position inside the buffer is not known. On the other
hand the tsched algorithm seems to get easily confused for this kind of
hardware, which results in garbled audio output. This typically means that timer
based scheduling needs to be manually disabled on systems with such devices.
Auto disabling tsched in this case allows these systems to run with the default
configuration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
the main intent is to make testing different sample rate resampling
implementations easier; so far there is only global control via
resample-method (command line argument and /etc/pulse/daemon.conf)
module-remap-*'s only purpose is resampling (comprising format conversion,
channel remapping, sample rate adjustment), it can easily be introduced
into any audio pipeline
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
