This breaks a lot of headsets, so disabling by default. Can be
re-enabled in configuration for specific hardware where it is deemed
necessary.
Also added some debug logging to be able to examine what MTU size is
reported by the device.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102660
Some sound cards don't have any alsa-lib configuration, but they used to
work well enough up to PulseAudio 10. PulseAudio 11 stopped using "hw:0"
for the analog-stereo mapping, and instead defined it as a fallback
mapping without any mixer handling. As a result, switching between
headphones and speakers stopped working without changing the mixer
settings manually at least on Toshiba Chromebook 2. This patch adds the
mixer handling back to the fallback mapping.
I also renamed "unknown-stereo" to "stereo-fallback", because I like
that name more.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102560
We're supposed to prioritize USB sound cards over PCI sound cards, but
the priority bonus for the "internal" form factor prevents this from
happening. Not all (if any) USB sound cards have the form factor
property set, whereas at least on my laptop the on-board sound card has
the form factor set to "internal".
The order of the pa_sink_input_put() and pa_sink_put() calls in filter
modules was swapped in commit edc465da77 ("virtual sources and sinks:
Don't double attach a sink input or source output on filter load").
If flat volumes and volume sharing is enabled, the pa_sink_input_put()
call will update volumes of the whole tree of virtual sinks that are
connected to the root sink. The recursive updating procedure tried to
also update the volume of the new sink for which pa_sink_put() had not
yet been called, causing an assertion failure.
This patch tries to make sure that the volume of not-yet-linked sinks
is never changed. pa_sink_put() will set the sink volume correctly, so
it's fine to skip the not-yet-linked sinks during pa_sink_input_put().
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102549
When a phone is connected via bluetooth and switches to HFP, the sinks
and sources will have higher priority than the built-in devices.
Therefore they are chosen as default and module-bluetooth-policy will
incorrectly insert loopback modules that loop the phone back to itself.
This patch fixes the problem by lowering the priority of sink and source
if PulseAudio is in the headset role. The priority is also lowered if the
device is an a2dp source. In both cases it does not make sense to make the
source or sink default unless there is no other sound device available.
Sources should probably be added to pa_core.sources in pa_source_put(),
but currently they're added in pa_source_new(). A lot of stuff can
happen between the pa_source_new() and pa_source_put() calls, and
it has happened that the default source was updated during this time.
Therefore, pa_core_update_default_source() needs to take it into account
that not every source is necessarily linked.
Currently pulseaudio crashes with an assertion in pa_rtpoll_item_new_asyncmsgq_read()
or pa_rtpoll_item_new_asyncmsgq_write() if a loopback is applied to a tunnel-new
sink or source, because tunnel-{sink,source}-new do not set thread_info.rtpoll.
The same applies to module-combine-sink and module-rtp-recv.
This patch is not a complete fix for the problem but provides a temporary band-aid
by initializing thread_info.rtpoll properly. The rtpoll created is never run, but
loopback and combine-sink nevertheless work, see comments in the code.
This patch does not work for module-rtp-recv, but using rtp-recv with a remote
sink does not seem to make a lot of sense anyway.
Bug link: https://bugs.freedesktop.org/show_bug.cgi?id=73429
PA builds fine on MinGW except for the use of the scandir function in
pulsecore/conf-parser.c, so I provided a Win32 implementation. With this
patch the latest code builds on Win32 without problems.
If only part of the buffer is written into stdout by stdout_callback,
the buffer_index variable is increased by the number of written bytes,
buffer_length variable is decreased while the allocated buffer size
remains the same. That suggests that the current allocated size is
calculated as (buffer_index + buffer_length). However the current
stream_read_callback implementation writes new data to the start of the
buffer and allocates too little space, so that (buffer + buffer_index +
buffer_length - 1) could actully point outside of the allocated buffer.
pa_usec_t is an unsigned type, but there were calculations that used it
as if it were a signed type.
If the latency is negative, pa_simple_get_latency() now reports 0.
Added some comments too.
The on_the_fly_snapshot variable contains the amount of bytes that has
been sent from the source IO thread to the main thread, but not yet
pushed to the stream memblockq. The data is in the stream format, but
the bytes-to-usec conversion used the source format, which caused random
latency reporting errors.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=81075
This allows us to restore the default device properly when a
hotpluggable device (e.g. a USB sound card) is set as the default, but
unplugged temporarily. Previously we would forget that the unplugged
device was ever set as the default, because we had to set
configured_default_sink to NULL to avoid having a stale pa_sink pointer,
and also because module-default-device-restore couldn't resolve the name
of a currently-unplugged device to a pa_sink pointer.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=89934
This reverts commit 69c212f8c1.
Reasons:
The original reason for the patch was to work around some issue
regarding the profile not connecting immediately (sorry, I don't really
know the details), but that issue was fixed later by commit 998dfdf4cc,
so the original reason doesn't apply any more.
Automatically changing the profile when the transport state changes to
PLAYING has traditionally been handled by module-bluetooth-policy, and
as far as I can tell, there's no reason to change that.
The assertion is unsafe. It's not guaranteed that the profile change
will always succeed (at least pa_thread_mq_init() can fail due to
reaching the maximum file descriptor limit).
There are two reasons for this change:
1. If it is a Dell desktop machine with the realtek codec, and there
is no internal microphone on it, there is one physical audio jack
which can support headphone, headset and microphone, but this audio
jack does not have hardware capability to distinguish what is plugged
in, after users plug in a headphone and select headphone from UI
program, the headphone can't output any sound. There are many reasons
for this issue, one of them is the active_port of pa_source is set
to headphone-mic, that means the kernel audio driver will configure
this audio jack to be a microphone jack instead of headphone jack.
If we make the priority of headset-mic a bit higher than headphone-mic,
the headset-mic will be the active_port of pa_source unless users
select the headphone-mic on purpose, then this issue will be fixed.
2. Nowadays, the headset is more popular than traditional microphone,
It is highly possible that users plug in a headset instead of
microphone, it makes sense to make the headset-mic's priority higher
than headphone-mic's.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
module-stream-restore primarily uses the role of a stream for restoring. The sink-inputs
and source-outputs of filters all have role "filter", therefore currently all filters are
treated equally and are restored to the same device and volume.
This patch lets module-stream-restore ignore the streams that connect the filter to the
master.
Bug link: https://bugs.freedesktop.org/show_bug.cgi?id=100065
These changes are from fedora.zanata.org. The authors are
Sam Friedmann <sfriedma@redhat.com>
Wim Taymans <wim.taymans@gmail.com>
Edouard Duliege <edouard.duliege@gmail.com>
When a filter sink is moving, it's not connected to any master sink, and
therefore it's not connected to any IO thread either. In this situation
trying to move a stream that is connected to the filter sink is likely
to result in crashing, because starting the move involves sending a
message to the IO thread. Sometimes this works by accident (the
asyncmsgq of the filter sink still points to the old master sink's
asyncmsgq), but we really should never attempt it. This patch blocks all
moves where the moving stream is connected to a filter sink that itself
is in the middle of a move.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=100277
In sink_put() and source_put(), pa_core_update_default_{sink,source}() was called
before the PA_CORE_HOOK_{SINK,SOURCE}_PUT hook. Therefore module-switch-on-connect
could not correctly determine the old default sink/source if no user default was
set and a sink/source with higher priority than any other sink/source turned up.
This patch corrects the problem by swapping the order of the hook call and the
pa_core_update_default_sink() call.
Additionally it corrects a problem in module-switch-on-connect. If, after the
change above, the new sink/source was the first sink/source to appear, pulseaudio
would crash because module-switch-on-connect assumed that the default sink/source
was not NULL. The patch checks if the default sink/source is NULL and only sets
the new default sink/source in that case.
When a filter is loaded and module-switch-on-connect is present, switch-on-connect
will make the filter the default sink or source and move streams from the old
default to the filter. This is done from the sink/source put hook, therefore streams
are moved to the filter before the module init function of the filter calls
sink_input_put() or source_output_put(). The move succeeds because the asyncmsq
already points to the queue of the master sink or source. When the master sink or
source is attached to the sink input or source output, the attach callback will call
pa_{sink,source}_attach_within_thread(). These functions assume that all streams
are detached. Because streams were already moved to the filter by switch-on-connect,
this assumption leads to an assertion in pa_{sink_input,source_output}_attach().
This patch fixes the problem by reverting the order of the pa_{sink,source}_put()
calls and the pa_{sink_input,source_output}_put calls and creating the sink input
or source output corked. The initial rewind that is done for the master sink is
moved to the sink message handler. The order of the unlink calls is swapped as well
to prevent that the filter appears to be moving during module unload.
The patch also seems to improve user experience, the move of a stream to the filter
sink is now done without any audible interruption on my system.
The patch is only tested for module-echo-cancel.
Bug-Link: https://bugs.freedesktop.org/show_bug.cgi?id=100065
When sinks are compared during the default sink selection, the active
port's availability is inspected. Therefore, the default sink should be
updated when the active port changes, because the new port may have
different availability status than the old port.
For example, let's say that a laptop has an analog sink with a speaker
and a headphone port, and headphones are initially plugged in, so both
ports can be used[1]. The headphone port is initially the active port.
There's also a null sink in the system. When the headphones are
unplugged, the headphone port becomes unavailable, and the null sink
becomes the new default sink. Then module-switch-on-port-available
changes the analog sink port to speakers. Now the default sink should
change back to the analog sink, but that doesn't happen without this
patch.
[1] Actually we currently mark speakers as unavailable when headphones
are plugged in, but that's not strictly necessary. My example relies on
both ports being available initially, so the bug can't be reproduced
with the current mixer configuration.
When the ofono backend released a tranport during suspend of sink or source, the
transport state was not changed to IDLE. Therefore pa_bluetooth_transport_set_state()
would return immediately when trying to resume. Even though the transport was acquired
correctly, setup_stream() would never be called and the resume failed.
This patch sets the transport state to IDLE when the transport is released. On resume,
the first call to transport_acquire() will be done from the message handler of the
*_SET_STATE message when source or sink are set to RUNNING. This call will only request
the setup of the connection, so setup_stream() cannot be called.
When the transport changes the state to PLAYING in hf_audio_agent_new_connection(),
handle_transport_state_change() is called. Because the sink or source state is already
RUNNING, the pa_{source,sink}_suspend() call will not lead to a state change message
and the I/O thread must be signaled explicitely to setup the stream.
The first setup of the device would also fail, which was only visible when the profile
was restored after connecting the headset. When trying to restore the headset_head_unit
profile, the profile was shortly set to off, so the headset always returned to a2dp.
This patch allows a delayed setup for the headset_head_unit profile, so that the profile
can successfully be restored.
When suspending due to idle timeout the transport will not change its
state, also in case the fd is closed due to POLLERR/POLLHUP events
the release shall check if the fd is still set otherwise it will fail
to be acquired again.
This means something went wrong, which in case of ofono backend it is
probably due to the profile not connecting immediately, but it can be
safely restored in that case the transport is playing which means the
profile has recovered connectivity.
