document behaviour of pa_shared_remove() in case name does not exist
Coverity ID: #1380672
thanks to Georg Chini for suggesting to swap patch title and commit message
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
The "profile->card != c->card" check always evaluated to false, so the
CardProfileUpdated signal was never sent. The reason: call_data was
assigned to a pa_card_profile pointer, but the correct type is a pa_card
pointer.
readdir_r() was supposed to be a thread-safe version of readdir(), but
the interface turned out to be problematic. Due to the problems and the
fact that readdir() is safe enough on modern libc implementations, glibc
deprecated readdir_r() in version 2.24.
The man page contains more information about what's wrong with
readdir_r(): http://man7.org/linux/man-pages/man3/readdir_r.3.html
On systems with constrained CPUs, we might run into a situation where
the master source/sink is configured to have too high a latency.
On the source side, this would cause us to wake up with a large chunk of
data to process, which might cause us to exhust our RT limit and thus be
killed.
So it makes sense to limit the overall latency that we request from the
source (and correspondingly, the sink, so we don't starve for playback
data on the source side).
The 10 blocks maximum is somewhat arbitrary (I'm assuming the system has
enough headroom to process 10 chunks through the canceller without
getting close to the RT limit). This might make sense to make tunable in
the future.
If the ALSA device supports granular pointer reporting, we end up in a
situation where we write out a bunch of data, iterate, and then find a
small amount of data available in the buffer (consumed while we were
writing data into the available buffer space). We do this 10 times
before quitting the write loop.
This is inefficient in itself, but can also have wider consequences. For
example, with module-combine-sink, this will end up pushing the same
small chunks to all other devices too.
Given both of these, it just makes sense to not try to write out data
unless a minimum threshold is available. This could potentially be a
fragment, but it's likely most robust to just work with a fraction of
the total available buffer size.
don't ignore server port parsing errors as suggested by Hajime Fujita
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
Cc: Hajime Fujita <crisp.fujita@nifty.com>
wath may be NULL, as suggested by Hajime Fujita
Coverity ID: #1398156
setting val = NULL is not needed
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
Cc: Hajime Fujita <crisp.fujita@nifty.com>
for example, in case HAVE_MEMFD is #undef, checking with #if HAVE_MEMFD
gives a warning (gcc 5.4.1, Ubuntu)
pulsecore/shm.c: In function 'sharedmem_create':
pulsecore/shm.c:208:5: warning: "HAVE_MEMFD" is not defined [-Wundef]
#if HAVE_MEMFD
use #ifdef or #if defined() to check for presence of a #define
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
Both input and output device were chosen with the same device number.
This is problematic as those numbers don't have to correspond.
Additionally the input device was named after the output device. This
commit adresses both issues by providing specific parameters for each
type.
This is a rebase of Wim Taymans patch to support the HSP headset role that has
somehow been forgotten. Original patch can be found at
https://lists.freedesktop.org/archives/pulseaudio-discuss/2015-February/023242.html
Rebase and minor changes by Georg Chini.
In addition to the HSP Audio Gateway, also add support for the headset
role in the native bluetooth backend. In this role, pulseaudio is used as
headset.
In the headset role, we create source and sink to receive and send the samples
from the gateway, respectively. Module-bluetooth-policy will automatically load
loopback modules to link these to a sink and source for playback. Because this
makes the source the speaker and the sink the microphone, we need to reverse the
roles of source and sink compared to the gateway role.
In the gateway role, adjusting the sink volume generates a +VGS command to set
the volume on the headset. Likewise, receiving AT+VGS updates the sink volume.
In the headset role, receiving a +VGS should set the source volume and any
source volume changes should be reported back to the gateway with AT+VGS.
Clang didn't like the variable length array:
pulsecore/iochannel.c:358:17: error: fields must have a constant size:
'variable length array in structure' extension will never be supported
uint8_t data[CMSG_SPACE(sizeof(int) * nfd)];
^
Commit 451d1d6762 introduced the variable length array in order to have
the correct value in msg_controllen. This patch reverts that commit and
uses a different way to achieve the same goal.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=99458
do...while not reachable, loop should try different ports in case EADDRINUSE is returned
Coverity ID: #1398161
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
the modeling file help to avoid false positives and increase scanning
accuracy by explaining code Coverity can't see (out of tree libraries);
the model file must be uploaded by an admin to:
https://scan.coverity.com/projects/pulseaudio?tab=analysis_settings
the pa_assert_se() macro needs to be rewritten for Coverity so that
the assignment is not declared a side-effect
Signed-off-by: Peter Meerwald-Stadler <pmeerw@pmeerw.net>
The previous commit, "loopback: Initialize latency at startup and during
source/sink changes", was an old version of the patch that got
accidentally pushed instead of the last version. This commit does the
changes that were omitted when applying the old patch.
The current code does not make any attempt to initialize the end-to-end latency
to a value near the desired latency. This leads to underruns at startup because
the memblockq is initially empty and to very long adjustment times for long
latencies because the end-to-end latency at startup is significantly shorter
than the desired value.
This patch initializes the memblockq at startup and during source or sink changes
so that the end-to-end latency will be near the configured value. It also ensures
that there are no underruns if the source is slow to start and that the latency
does not grow too much when the sink is slow to start by adjusting the length of
the memblockq until the source has called push for the first time and the sink
has called pop for the second time. Waiting for the second pop is necessary
because the sink has not been started when the first pop is called.
For clarity, variables have been separated into input, output and main thread
variables.
FlatCarbon was the flattened Carbon version used in Mac OS Classic
(i.e., pre Mac OS X.)
It was shipped as legacy software until 10.8, then dropped completely.
Using CoreServices is good enough, manually including FlatCarbon headers
only lead to build failures for users who had old versions of Xcode
lingering around their machines.
v2: don't accidentally drop the OS X semaphore implementation.
A recent patch changed the MTU size from the default value of 48 to the value
returned by getsockopt(). This breaks HSP for some setups. To circumvent the
problem, this patch introduces a boolean parameter "autodetect_mtu" for
module-bluetooth-discover, module-bluez5-discover and module-bluez5-device to
make this use of getsockopt() configurable.
This serves to explicitly document the various cases we deal with in
pa_sink_update_rate()/pa_source_update_rate() rather than have some of
them hidden behind the initialisation of desired_rate.
This adds an "avoid-resampling" option to daemon.conf that makes the
daemon try to use the stream sample rate if possible (the device needs
to support it, which currently only ALSA does), and there should not be
any other stream connected).
This should enable some of the "audiophile" use-cases where users wish
to play high sample rate audio files without resampling.
We still will do conversion if sample formats don't match, though. This
means that if you want to play 96 kHz/24 bit audio without any
modification the default format will need to be set to be 24-bit as
well. This will force all streams to be upconverted, which, other than
the wasted resources, should be relatively harmless.
The RTSP client is not waiting anymore a new header after the
previous one (which can never occurs if RAOP is disconnected)
but after sending a command.
This patch fixes Issue #36.
https://github.com/hfujita/pulseaudio-raop2/issues/36
This patch is based on a similar idea as the previous one -- disabling
the flag right after the session is getting closed, rather than waiting
for a response from the server.
This patch fixes the issue #31.
https://github.com/hfujita/pulseaudio-raop2/issues/31
This patch sets c->is_recording = false when the RTSP FLUSH command
is issued. This avoids a race between the server response and
the record activation in some cases.
Regression introduced in commit 8c6407f:
raop: Merge TCP and UDP code paths + refactoring
Anyway, we need to determine if initial volume has to be setup before
sending RECORD or after:
- Setting it up *before* shouldn't be a problem: sink.c waits for
CONNECT state, set the volume and client.c triggers RECORD only once
he's got the SET_PARAMETER reply from server.
- Setting it up *after* seems to be more difficult if we try not to
send any audio before receiving the SET_PARAMETER reply form server. A
solution may be to send SET_PARAMETER just after the RECORD server
response is received and hope that it get processed by server during the
2sec latency/buffering time...
Attached patch implement that last solution. Works for me, but I cannot
guaranty it will with your hardware...
