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Without -avoid-version, libtool creates a libwebrtc-util.so.0 and
libwebrtc-util.so.0.0.0 which are not cleaned up by make uninstall,
which in turn causes make distcheck to fail.
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This moves out the webrtc bits into a small helper library to shield the
rest of module-echo-cancel from being linked with a C++ linker. This is
required because automake will _always_ link module-echo-cancel in C++
mode if any of its deps (even conditional ones) are in C++.
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I forgot half of the front headphone patch, i e, to hide the
speaker output when the front headphone is connected. Thanks to
Shih-Yuan Lee for noticing.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Conflicts:
src/modules/alsa/mixer/paths/analog-output-speaker.conf
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Many desktops have headphone on the front and line outs on the back.
Sometimes this means that the headphone is labelled "Front Headphone Jack",
but the volume controls are only "Headphone Playback Volume", i e,
without the "Front" prefix.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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Some devices have inverted right channel, so when you add left and right,
the result is silence, or very faint sound. In recent kernels (3.5,
perhaps also 3.4) these are starting to be marked with a special
"Inverted Internal Mic" capture switch.
While we might want to add some reverse summing mechanism in the
future, for now, we just turn the thing off to avoid the problem of
recording silence.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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internal mic
Several laptops have speaker ports, and/or internal mic ports, but we have
no way of detecting that. So we make the port(s) always show up for these
devices.
BugLink: https://bugs.launchpad.net/bugs/946232
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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Credit to David for finding this one.
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Compilation with -DDEBUG_TIMING fails due to a missing header:
modules/alsa/alsa-source.c: In function 'check_left_to_record':
modules/alsa/alsa-source.c:426:9: warning: implicit declaration of function 'raise' [-Wimplicit-function-declaration]
modules/alsa/alsa-source.c:426:9: error: 'SIGTRAP' undeclared (first use in this function)
Signed-off-by: Eero Nurkkala <eero.nurkkala@offcode.fi>
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Compilation with -DDEBUG_TIMING fails due to a missing header:
modules/alsa/alsa-sink.c: In function 'check_left_to_play':
modules/alsa/alsa-sink.c:453:9: warning: implicit declaration of function 'raise' [-Wimplicit-function-declaration]
modules/alsa/alsa-sink.c:453:9: error: 'SIGTRAP' undeclared (first use in this function)
Signed-off-by: Eero Nurkkala <eero.nurkkala@offcode.fi>
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Based on a patch by Marko Ollonen.
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[These symbols were removed in libudev.so.1.0.0. Replace them with
hardcoded strings. -- heftig]
Signed-off-by: Marc-Antoine Perennou <Marc-Antoine@Perennou.com>
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There was a race in the existing code that could cause the pa_once code
to be run twice, see:
http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-April/013354.html
Therefore the existing implementation was rewritten to instead look like
the reference implementation here:
http://www.hpl.hp.com/research/linux/atomic_ops/example.php4
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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When a Bluetooth headset is connected only to HFP profile (not connected
to A2DP) and host streams to it, a crash occurs if host disconnects.
When HFP disconnects, audio thread will fail on POLLHUP then generate
a message to set PA profile to Off before ending.
If this message is managed before PA unload bluetooth device module,
all works fine.
But, if this message is managed during module unload, this finish by
re-entrance in release code (stop_thread) and a crash.
This fix prevents to process profile change when module is unloading.
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function
Yikes!
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This makes sure we only call sink/source update_rate() if the stream
being woken up has a sample spec that doesn't match the corresponding
device'.
https://bugs.freedesktop.org/show_bug.cgi?id=49875
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• s,Pulseaudio,PulseAudio,
• One last occurence of this “error” is in the French translation.
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pa_bool_t and dbus_bool_t cannot be used interchangably since their size
might (and do) vary. This caused a crash on some systems which was
reported and root caused by: Aidan Thornton <makosoft@googlemail.com>.
Ref: https://bugs.gentoo.org/show_bug.cgi?id=398097
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Changes in v2:
- Call the mapping a generic 4-channel input mapping
instead of a 4-channel mic array mapping. The mapping
will be used also by sound cards that have two stereo
input jacks, so in those cases talking about mic arrays
is wrong.
- Added a comment about using the "hw" device name.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=45813
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libsamplerate_resample() assumed that src_process() would
always consume the whole input buffer. That was an invalid
assumption leading to crashes.
This patch adds a leftover memchunk for storing any
non-consumed input. When pa_resampler_run() is called next
time, the leftover is prepended to the new input.
Changes in v3:
- Make the calculations in pa_resampler_result() and
pa_resampler_max_block_size() more readable and more
correct.
- Rework the leftover storing: instead of using a dedicated
buffer for it, store it in the beginning of remap_buf.
This can avoid some memory copying. (The idea was
suggested by Wang Xingchao.)
- Use a generic save_leftover() function instead of doing
the leftover copying in the resampler implementation.
- Use the leftover logic also with the speex and ffmpeg
resamplers.
[ed: dropped the speex bit since the API guarantees that
it will consume everything -- Arun]
Changes in v2:
- If add_leftover() is called with zero-length input while
the leftover length is non-zero, we don't try to acquire
the input memblock.
- Instead of taking a reference to the original input in
libsamplerate_resample(), we copy the leftover data to a
new memblock. This is done, because otherwise, if the
input is one of the internal buffers, the data can get
overwritten before reading it in add_leftover().
- Store add_leftover_buf size in bytes instead of samples
(more convenient, but less consistent with other code).
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=47156
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All utilities should have the --help and --version command line options.
These two were added to pacmd, also the goto label was changed from fail
to quit like in the other utilities.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=45030
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In addition to changing "plen > u->blocksize" to "plen >=
u->blocksize", I also removed one of the duplicated ifs.
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State which thread calls set_port in which scenario.
Reported-by: Jyri Sasha <oku@iki.fi>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=48376
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
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This is needed for 0b421f on Intel/NVidia hardware.
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Log in as user A, fast user switch to user B, let user B change
port, volume or mute status, then switch back to user A.
At this point we must make sure that the ALSA and PA volumes are
synchronised by writing to the ALSA mixer when the ALSA device
becomes available.
BugLink: https://bugs.launchpad.net/bugs/915035
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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This should have been done a long time ago but it brings symmetry to the API
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This just documents the cli interface syntax. Mostly a lift from
http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/CLI
with some minor editing and bringing up to date.
Also document a few undocumented commands.
Shuffle around the order in 'pacmd help' output to match the order
in the new docs for consistency.
https://bugs.freedesktop.org/show_bug.cgi?id=45029
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If deferred volumes were activated, set_volume does not really set
the volume, and is probably only meant to be called from the main
thread.
As we're currently really setting the port and the mute here (i e
modifying ALSA), we should really modify the volume as well.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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Make sure we can't be called into by remaining references to
sink-inputs and source-outputs after we have unloaded, as
that will likely lead to segfaults.
Thanks to Tanu for providing valuable input on this patch.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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If an error during pa__init() causes a jump to fail: u->asyncmsgq is not
initialized.
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Coverity thinks that expected_method_sig can be NULL when
it's dereferenced by pa_streq(). Adding assertions doesn't
hurt here (in my opinion), and that should get rid of the
warnings.
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Coverity thinks that device_name can be NULL when it's
dereferenced by strcmp. Adding an assertion doesn't hurt
here (in my opinion), and that should get rid of the
warning.
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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.
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Valid channel id range is from 0 to SND_MIXER_SCHN_LAST,
inclusive, so the size of the masks array in pa_alsa_element
has to be SND_MIXER_SCHN_LAST + 1. Similar "too small"
arrays were also in alsa-sink's and alsa-source's userdata,
but actually those arrays were not used at all so they were
removed.
element_is_subset() in alsa-mixer.c skipped the last channel
id when iterating the element masks array; that's now fixed
as well.
Thanks to David Henningsson for spotting the too small
arrays in alsa-sink and alsa-source and the
element_is_subset() problem.
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Add also an assertion for the sample spec validity. The
existing code already does crash in case of an invalid
sample spec, but the error would not be as obvious: the
crash would happen due to a divide-by-zero operation in
pa_frame_aligned().
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