Maarten Bosmans [Thu, 13 Oct 2011 18:54:17 +0000 (20:54 +0200)]
rtpoll: Update comment
to reflect changes made in 32e2cd6d3216f780c4cffed0f8eb3c30f2c8d732
core: get rid of rt sig/timer handling since modern Linux' ppoll() is finally fixed for granularity
Antti-Ville Jansson [Fri, 11 Nov 2011 14:22:24 +0000 (16:22 +0200)]
stream: Fix upload samples' cleanup
In pa_create_stream_callback, a stream is inserted into
s->context->record_streams only if it's a record stream. Otherwise it's
inserted into s->context->playback_streams. However, in stream_unlink the
stream is removed from s->context->playback_streams only if it's a playback
stream and otherwise it's removed from s->context->record_streams.
Thus, if the stream is an upload stream, we first insert it into
s->context->playback_streams in pa_create_stream_callback and then try to
remove it unsuccessfully from s->context->record_streams in stream_unlink. This
means that we are leaking hashmap entries until the context is freed,
constantly consuming more memory with applications that upload and unload a
large number of samples through one context.
Of course, this begs the question whether upload streams even belong in either
of those hashmaps. I don't want to mess around with the code too much at this
point though, so this patch should be a sufficient improvement.
Arun Raghavan [Fri, 4 Nov 2011 10:47:26 +0000 (16:17 +0530)]
echo-cancel: Add infrastructure for cancellers to do AGC
This adds some infrastructure for canceller implementations to also
perform acoustic gain control. Cancellers now have a couple of new API
calls that allow them to get/set capture volume.
This is made slightly complex by the fact that cancellation happens in
thread context while most volume mangling needs to be done in main
context. To deal with this, while getting the volume we save source
volume updates as they are propagated to thread context and use this
cached value for queries. To set the volume, we send an async message to
main context and let that set the source volume.
Arun Raghavan [Mon, 7 Nov 2011 09:31:25 +0000 (15:01 +0530)]
sink,source: Allow sample rate switching with corked streams
This updates corked streams' resamplers when switching sample rates on a
sink/source, which means the restriction of allowing sample rate updates
only when no streams are attached to a sink/source is now relaxed to
preventing updates only when there is a running stream attached.
Arun Raghavan [Thu, 18 Aug 2011 06:26:26 +0000 (11:56 +0530)]
cli: Add a dump-volumes command
The purpose of this command is to print all the internal volume
variables for sinks/sources and all corresponding
sink-inputs/source-outputs to make debugging and reasoning about
volume-related issues easier.
Lars R. Damerow [Thu, 3 Nov 2011 20:14:45 +0000 (21:14 +0100)]
alsa: support fixed latency range in alsa modules
This adds a boolean module parameter to disable automatic dynamic
latency readjustments on underruns, but leaves automatic dynamic
watermark readjustments untouched.
Frédéric Dalleau [Tue, 4 Oct 2011 07:37:25 +0000 (09:37 +0200)]
bluetooth: Set hfgw profile when HandsfreeGateway is playing
Allow module-bluetooth-device to listens to HandsfreeGateway state
changes using DBUS signals. When an handsfree connects, module-bluetooth-device
is loaded and goes to playing state. When the handsfree disconnect audio,
the card profile is set to "off". If the headset connects audio again after
that, the card profile should switch to "hfgw" again to match state of audio
connection.
Frédéric Dalleau [Tue, 4 Oct 2011 07:37:23 +0000 (09:37 +0200)]
bluetooth: Release MediaEnpoint if card profile is set to Off
If card profile is set to "off", the audio stream should be released.
Current implementation releases the stream when the card profile
is changed to "hsp" or "hfgw" again and immediatly reconnects after that.
Frédéric Dalleau [Tue, 4 Oct 2011 07:37:22 +0000 (09:37 +0200)]
bluetooth: Do not unload module-bluetooth-device on ERR or HUP
This happens in the following scenario :
An HandsfreeGateway connects RFCOMM and then SCO. A card appears in
PA and can be used. If for some reason, SCO is disconnected,
module-bluetooth-device is unloaded. The card will disappear, even
if RFCOMM is still connected. After that, it is not possible to
connect SCO again from PA.
Frédéric Dalleau [Tue, 4 Oct 2011 07:37:21 +0000 (09:37 +0200)]
bluetooth: Fix Media Endpoint for HandsfreeGateway
This patch will add the necessary quirks so that pulseaudio can register
an endpoint on the /MediaEndpoint/HFPHS path. This endpoint is to be
used for HFP Handsfree profile.
Maarten Bosmans [Tue, 4 Oct 2011 12:12:20 +0000 (14:12 +0200)]
tests: Revisit which tests to run with make check
Some tests (remix-test, sig2str) only display information, so they are not
useful for automated testing. Others (interpol-test, once-test, thread-test)
do return an error on failure, so should be included in TESTS.
Maarten Bosmans [Tue, 4 Oct 2011 12:01:03 +0000 (14:01 +0200)]
tests: More useful output of make check
Instead of spilling thousands of lines of output, make check now runs the
test-suite in about 100 lines or so. If running under make check, the output of
tests is reduced. The MAKE_CHECK environment variable is used for this, so that
when running the test manually, the full output is still shown. Furthermore,
pa_log is used consistently instead of printf, so that all test output goes to
stderr by default. Colored output from make check goes to stdout.
Maarten Bosmans [Fri, 28 Oct 2011 14:30:05 +0000 (16:30 +0200)]
tests: Make sure tests assert on failures and return error status
When a test program exits with a nonzero return value (or an assert is hit),
the test is regarded as a FAIL.
This makes `make check` a little more useful.
Arun Raghavan [Fri, 7 Oct 2011 10:58:11 +0000 (16:28 +0530)]
echo-cancel: Plug in WebRTC drift compensation
This adds the ability for echo cancellers to provide their own drift
compensation, and hooks in the appropriate bits to implement this in the
WebRTC canceller.
We do this by introducing an alternative model for the canceller. So
far, the core engine just provided a run() method which was given
blocksize-sized chunks of playback and record samples. The new model has
the engine provide play() and record() methods that can (in theory) be
called by the playback and capture threads. The latter would actually do
the processing required.
In addition to this a set_drift() method may be provided by the
implementation. PA will provide periodic samples of the drift to the
engine. These values need to be aggregated and processed over some time,
since the point values vary quite a bit (but generally fit a linear
regression reasonably accurately). At some point of time, we might move
the actual drift calculation into PA and change the semantics of this
function.
NOTE: This needs further testing before being deemed ready for wider use.
Siarhei Siamashka [Thu, 20 Oct 2011 12:47:48 +0000 (15:47 +0300)]
bluetooth: sbc: overflow bugfix and audio decoding quality improvement
The "(((audio_sample << 1) | 1) << frame->scale_factor[ch][sb])"
part of expression
"frame->sb_sample[blk][ch][sb] =
(((audio_sample << 1) | 1) << frame->scale_factor[ch][sb]) /
levels[ch][sb] - (1 << frame->scale_factor[ch][sb])"
in "sbc_unpack_frame" function can sometimes overflow 32-bit signed int.
This problem can be reproduced by first using bitpool 128 and encoding
some random noise data, and then feeding it to sbc decoder. The obvious
thing to do would be to change "audio_sample" variable type to uint32_t.
However the problem is a little bit more complicated. According
to the section "12.6.2 Scale Factors" of A2DP spec:
scalefactor[ch][sb] = pow(2.0, (scale_factor[ch][sb] + 1))
And according to "12.6.4 Reconstruction of the Subband Samples":
sb_sample[blk][ch][sb] = scalefactor[ch][sb] *
((audio_sample[blk][ch][sb]*2.0+1.0) / levels[ch][sb]-1.0);
Hence the current code for calculating "sb_sample[blk][ch][sb]" is
not quite correct, because it loses one least significant bit of
sample data and passes twice smaller sample values to the synthesis
filter (the filter also deviates from the spec to compensate this).
This all has quite a noticeable impact on audio quality. Moreover,
it makes sense to keep a few extra bits of precision here in order
to minimize rounding errors. So the proposed patch introduces a new
SBCDEC_FIXED_EXTRA_BITS constant and uses uint64_t data type
for intermediate calculations in order to safeguard against
overflows. This patch intentionally addresses only the quality
issue, but performance can be also improved later (like replacing
division with multiplication by reciprocal).
Test for the difference of sbc encoding/decoding roundtrip vs.
the original audio file for joint stereo, bitpool 128, 8 subbands
and http://media.xiph.org/sintel/sintel-master-st.flac sample
demonstrates some quality improvement:
=== before ===
--- comparing original / sbc_encoder.exe + sbcdec ---
stddev: 4.64 PSNR: 82.97 bytes:170495708/170496000
=== after ===
--- comparing original / sbc_encoder.exe + sbcdec ---
stddev: 1.95 PSNR: 90.50 bytes:170495708/170496000
Marcel Holtmann [Fri, 26 Aug 2011 18:18:54 +0000 (11:18 -0700)]
bluetooth: audio: Update license for shared header files
The header files with constants and structures for audio specific
interaction with Pulseaudio are suppose to be under LGPL license.
For some odd reason a2dp-codecs.h ended up being under GPL license
which is against the intention of this being shared and re-used by
non-GPL programs. Fix this now to avoid any future confusion.
Arun Raghavan [Thu, 27 Oct 2011 10:49:18 +0000 (12:49 +0200)]
core: Add a string list membership check function
This adds a pa_str_in_list() to check for a given string in a
space-separated list of strings. For now, this is merely present to
avoid duplication of role matching code (intended roles can be a
space-separate list) across modules.
The documentation says we expect a comma-separate list of intended
roles, but the code splits the string on whitespaces, so this corrects
the documentation to match the implementation.
Colin Guthrie [Mon, 24 Oct 2011 21:35:38 +0000 (23:35 +0200)]
role-cork: Make module-role-cork more generic.
Operate on a list of 'trigger roles' and 'cork roles'. i.e.
react to any stream with a role in the trigger list and apply a
cork to any stream with the a role in the cork list.
The trigger roles default to 'phone' and the cork roles default
to both 'music' and 'video' thus achieving the same functionality
as currently when called without any arguments.
Colin Guthrie [Thu, 20 Oct 2011 13:11:53 +0000 (14:11 +0100)]
conf: Use .nofail when loading module-jackdbus-detect
When starting via a console login, PA will likely not have a session DBus
to play with. As there is no X11 environment, libdbus will be unable
to launch a session DBus for us and thus the module will fail to load
which in turn prevents PA from loading.
If the user subsequently logs into X11 this it will still not be possible
to load the module as the server will be ignorant of the X11 and DBus
environment variables so a longer term solution for handling this should
be found.
Colin Guthrie [Thu, 20 Oct 2011 09:04:49 +0000 (10:04 +0100)]
Update LICENSE.
Some of the license wording was less than clear. Try to clarify the
different GPL 'downgrade' scenarios but also be generic to ensure that
those packagers where GPL is a problem check thoroughly before they ship.
Inspired by comments from Brian Cameron @ Oracle via fdo#41822
Colin Guthrie [Tue, 11 Oct 2011 19:42:42 +0000 (20:42 +0100)]
build-sys: Provide a simple CMake Config setup (similar to pkgconfig)
I'd rather not have to do this, as I don't really see the point in
duplicating what is done in pkgconfig, but this is likely the
easiest way to avoid nasty hacks.
Arun Raghavan [Tue, 18 Oct 2011 04:23:20 +0000 (09:53 +0530)]
sink,source: Account for corked streams in update_rate()
pa_sink/source_used_by() ignores corked/monitor streams, but we need to
make sure there aren't any of these while updating rate (at least for
now -- this is a restriction that would be nice to get rid of).
Arun Raghavan [Mon, 10 Oct 2011 19:17:56 +0000 (00:47 +0530)]
source: Bring rate update code in sync with sink code
Basically adds code to handle passthrough sources. This isn't a tested
path at the moment, but in the future, when we do wish to support these,
it'll save us the trouble of having to sync all the code again.
Pierre-Louis Bossart [Tue, 2 Aug 2011 23:37:29 +0000 (18:37 -0500)]
alsa: support for alternate sampling rate
This is where the actual changes happen.
Some additional checks would be required to make sure the
rate is actually supported
Tested with both PCM and passthrough streams
This adds the WebRTC echo canceller as another module-echo-cancel
backend. We're exposing both the full echo canceller as well as the
mobile echo control version as modargs.
Pending items:
1. The mobile canceller doesn't seem to work at the moment.
2. We still need to add bits to hook in drift compensation (to support
sink and source from different devices).
The most controversial part of this patch would probably be the
mandatory build-time dependency on a C++ compiler. If the optional
--enable-webrtc-aec is set, then there's also a dependency on libstdc++.
Tanu Kaskinen [Thu, 6 Oct 2011 20:09:15 +0000 (23:09 +0300)]
alsa: New modarg "paths_dir" for module-alsa-card
The new module argument can be used to provide a custom
directory for loading alsa path configuration files. This is
useful for testing: no need to be root to create test
configuration files.
Arun Raghavan [Wed, 12 Oct 2011 12:14:30 +0000 (17:44 +0530)]
filter-apply: Move sink/source unlink callbacks before m-s-r
module-stream-restore and modile-filter-apply can get into an infinite
loop if m-s-r is called before m-f-a (m-s-r rescues a stream and
attaches it to a sink/source, which then triggers m-f-a to move it back
to the filter sink/source, and so on). The purpose of the m-f-a hooks is
to beat m-s-r, so moving them to be run first.
Arun Raghavan [Wed, 5 Oct 2011 06:59:10 +0000 (12:29 +0530)]
echo-cancel: Skip processing till there's enough data
This makes sure that we only perform any processing (resync or actual
cancellation) after the source provides enough data to actuall run the
canceller.
echo-cancel: Skip canceller when no source outputs are connected
When a source-output isn't connected to our virtual source, we skip echo
cancellation altogether. This makes sense in general, and makes sure
that we don't end up adjusting for delay/drift when nothing is
connected. This should make convergence faster when the canceller
actually starts being used.
Arun Raghavan [Wed, 5 Oct 2011 08:11:43 +0000 (13:41 +0530)]
echo-cancel: Increase threshold for resyncing, make it configurable
This increase the threshold for difference between the playback and
capture stream before samples are dropped from 1ms to 5ms (the
cancellers are generally robust to this much and higher). Also, we make
this a module parameter to allow easier experimentation with different
values.
Pierre-Louis Bossart [Fri, 7 Oct 2011 23:12:32 +0000 (18:12 -0500)]
alsa: reset watermark to initial values on resume
Watermark level and latency values are not restored when
resuming, the values used prior to suspending are reused.
This leads to side effects when underruns happen and buffer
sizes are updated, PulseAudio can never meet lower latency
requirements.
Solution: keep track of watermark and latency values on sink or
source creation, and reapply them on resume to start with
a clean slate.
David Henningsson [Wed, 5 Oct 2011 09:15:53 +0000 (11:15 +0200)]
source-output: Do not use unset channel map in pa_source_output_new
This problem was found when tracing down a crash coming from the
esound protocol, which does not set a channel map.
BugLink: http://bugs.launchpad.net/bugs/864071 Signed-off-by: David Henningsson <david.henningsson@canonical.com>
N.B.: As Colin notes, this is because commit 117c7145 was incomplete
("format: Fix channel map handling")
Arun Raghavan [Tue, 4 Oct 2011 19:28:52 +0000 (00:58 +0530)]
alsa: Make mixer error handling more robust still
Instead of relying on the snd_mixer_* functions failing, we check for
POLLERR and POLLNVAL first. After this, any errors in handling the mixer
events are deemed fatal (that is we cause the ALSA source/sink thread to
terminate).
The case where POLLERR is set but POLLNVAL is not does not actually
occur, but we're making this a soft failure (stop polling the mixer, but
don't kill the I/O thread). If other conditions where POLLERR occurs
turn up, we need to handle them explicitly.
Thanks to Linus Torvalds for helping get this right.
Arun Raghavan [Tue, 4 Oct 2011 08:36:26 +0000 (14:06 +0530)]
echo-cancel: Fail if loaded between a sink and its monitor
Loading between a sink and its monitor causes a deadlock (while sending
messages for latency snapshots). It isn't a case that has any real
conceivable use, so let's just disallow it.
Arun Raghavan [Tue, 4 Oct 2011 04:59:03 +0000 (10:29 +0530)]
alsa: Better error handling in mixer rtpoll callback
This improves the error handling in the mixer rtpoll callback. It avoids
a crash if an error occurs (the rtpoll_item is freed but still
referenced), and specifically makes sure we don't continue trying to
poll the device if the card is disconnected.
Arun Raghavan [Tue, 4 Oct 2011 05:35:59 +0000 (11:05 +0530)]
alsa: Give compressed formats preference over PCM
This makes set_formats() put PCM formats lower down the list than
compressed formats since we negotiate by picking the first format in
this list that is also in the client-provided list of possible formats
during sink input creation.
This will be incorrect if we ever decide to do encoding in PA (for
things like AC3/DTS encoding for multichannel output over S/PDIF).