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
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.
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 error message for snd_pcm_hw_params_set_period_wakeup was
printing "ret", but "ret" wasn't being set.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Allow a module argument to specify that we should act globally
rather than just within a given sink.
The default value is to not opporate globally thus retaining the
current behaviour.
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.
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
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
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++.
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.
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.
This makes sure that we only perform any processing (resync or actual
cancellation) after the source provides enough data to actuall run the
canceller.
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.
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.
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.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
If module-jackdbus-detect failed in the later part of initialization,
the ma variable was freed twice.
BugLink: http://bugs.launchpad.net/bugs/867444
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
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.
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.
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.
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).
The purpose of this patch is to make it possible to configure stream volumes
before pulseaudio is run for the first time. This is useful, for example, in
embedded products where the default volumes have to be sensible already in
the first boot.
These are not used for anything at this point, but this
makes it easy to add ad-hoc debug prints that show the
memblockq name and to convert between bytes and usecs.
If module-dbus-protocol fails to start, pa__done is still called,
which falsified the assumption that u->connections was always set.
BugLink: http://bugs.launchpad.net/bugs/855729
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Sometimes the ALSA mixer can be modified during a point at shutdown
which causes a race condition trying to update the volume of an
unlinked sink.
Includes typo fix by our Chief Typo Spotter, Colin, and a clarifying
comment by me.
BugLink: http://bugs.launchpad.net/bugs/841968
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Uses the shared volume infrastructure by default with an option to
fallback on the old pretend-volume-sharing-that-kind-of-works if someone
wants it that way.
Users who keep left != right (or any sort of unbalanced channel volumes)
will likely want to disable shared volumes since it will cause their
master sink/source volume to be balanced.
This really isn't a very pleasant scenario since users would need to
manually set up echo cancellation in their config for this (until we
have a way to store module configuration). That said, the majority case
benefits from the volume sharing, so let's not wait for the
configuration infrastructure to be ready to use this.
Uses the shared volume infrastructure by default with an option to
fallback on the old pretend-volume-sharing-that-kind-of-works if someone
wants it that way.
This just covers Lennart's concern over the terminology used.
The majority of this change is simply the following command:
grep -rli sync[-_]volume . | xargs sed -i 's/sync_volume/deferred_volume/g;s/PA_SINK_SYNC_VOLUME/PA_SINK_DEFERRED_VOLUME/g;s/PA_SOURCE_SYNC_VOLUME/PA_SOURCE_DEFERRED_VOLUME/g;s/sync-volume/deferred-volume/g'
Some minor tweaks were added on top to tidy up formatting and
a couple of phrases were clarified too.
The Kinect shows up as a UAC device after the firmware has been loaded,
but in order to be detected by pulseaudio a 4-channels input only
mapping is needed. Provide a new profile for that and set it with a udev
rule.
fdo#39664
pa_core_maybe_vacuum now vacuums if there are either no streams or all devices are suspended.
The mempool_vacuum argument to module-suspend-on-idle is gone and defaults to true now.
We used to support older DBus versions but 1.3.0 is two years old
now and by requiring it we cut down of deviated code paths at
runtime and thus have less support issues.
fdo#40635
