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.
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...
Some time one device announces multiple addresses (e.g. IPv4 one
and IPv6 one). Or some user may own multiple RAOP devices with
the same model name.
This patch adds device port to device description so that users
can distinguish appropriate RAOP sink by its address.
This patch switch the packet-buffer to use core memory pool instead of
manually allocating the room required for storing TCP/UDP packets. Packets
are now stored using pa_memchunk instead of internal struct. Quite a few
malloc saved compare to previous design.
ALAC encoding is to be prefered simply because ALAC audio packet reverse-
engineering and implementation is in better shape than raw PCM. Sending ALAC
audio does not mean compressing audio and thus linking an external library to
do so. ALAC packets has the ability to carry uncompressed PCM frames, and
that's what is implemented at the time.
TCP and UDP implementation are following two diffrent code path while code
logic is quite the same. This patch merges both code path into a unique one
and, thus, leads to a big refactoring. Major changes include:
- moving sink implementation to a separate file (raop-sink.c)
- move raop-sink.c protocol specific code to raop-client.c
- modernise RTSP session handling in TCP mode
- reduce code duplications between TCP and UDP modes
- introduce authentication support
- TCP mode does not constantly send silent audio anymore
About authentication: OPTIONS is now issued when the sink is preliminary
loaded. Client authentication appends at that time and credential is kept
for the whole sink lifetime. Later RTSP connection will thus look like this:
ANNOUNCE > 200 OK > SETUP > 200 OK > RECORD > 200 OK (no more OPTIONS). This
behaviour is similar to iTunes one.
Also this patch includes file name changes to match Pulseaudio naming
rules, as most of pulseaudio source code files seem to be using '-'
instead of '_' as a word separator.
RAOP authentication is using standard HTTP challenge-response authentication
scheme. This patch adds two helper functions that generate the proper hash
(for both techniques) given a username, a password and session related tokens.
MD5 hashing will be needed during the authentication process.
Original patch by Martin Blanchard. Patch splitted by
Hajime Fujita <crisp.fujita@nifty.com>.
Base64 implementation is now in a common file called raop_util.c.
Old Base64 files are removed but copyright is preserved.
Original patch by Martin Blanchard, patch splitted by
Hajime Fujita <crisp.fujita@nifty.com>.
When playback stops, a FLUSH command is send to the server and the sink
goes to IDLE. If playback resumes quickly, sink goes back to RUNNING
(without being SUSPENDED) and the sink should just start streaming again.
This patch implements this behaviour.
This patch adds an RTP audio packet retransmission support and a
circular buffer implementation for it.
This patch was originally written by Matthias Wabersich [1] and
later debugged and integrated into the latest tree by Hajime Fujita
[1]: https://bugs.freedesktop.org/show_bug.cgi?id=42804#c44
During the discovery phase, raop servers send their capabilities
(supported encryption, audio codec...). These should be passed to the
raop sink via module's arguments.
Original patch written by Martin Blanchard, then modified by Hajime
Fujita <crisp.fujita@nifty.com> based on review comments by
Anton Lundin <glance@acc.umu.se>.
Now resolver_cb always dtrdup()s string blocks given by Avahi,
to make the code easier to maintain.
There are two versions in the RAOP protocol; one uses TCP and the
other uses UDP. Current raop implementation only supports TCP
version.
This patch adds an initial UDP protocol support for RAOP.
It is based on Martin Blanchard's work
(http://repo.or.cz/w/pulseaudio-raopUDP.git/shortlog/refs/heads/raop)
which is inspired by Christophe Fergeau's work
(https://github.com/zx2c4/pulseaudio-raop2).
Matrin's modifications were edited by Hajime Fujita, so that it
would support both TCP and UDP protocol in a single module.
Also this patch includes a fix that was found thanks to Matthias,
who reported that his ALAC
codec support fixed the issue.
https://bugs.freedesktop.org/show_bug.cgi?id=42804#c30
In alsa-lib, snd_pcm_hw_params() internally calls snd_pcm_prepare(), thus
user space applications have no need to call snd_pcm_prepare() after calls
of snd_pcm_hw_params(). An explicit calls of snd_pcm_prepare() is expected
in a case to recover PCM substreams.
Current implementation of PulseAudio modules for ALSA playbacking/capturing
results in double calls of snd_pcm_prepare(). The second call for hw plugin
of alsa-lib executes ioctl(2) with SNDRV_PCM_IOCTL_PREPARE command in state
of SNDRV_PCM_STATE_PREPARED for the PCM substream. This has no effects to
the PCM substream as long as corresponding drivers are implemented
correctly.
This commit removes the second call for the reason.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Issue: When HFP/HSP profile is used with certain BT chipsets, the
audio sounds heavily distorted, with very slow playback full of noise.
During recording, the samples are dropped and it distorts the recorded
audio samples.
The root cause of both the issues are related to the fixed MTU sizes
in the PA stack, which is 48 bytes. Here, the BT chipset CC256x had
180 bytes MTU and it was being under-utilized and the rate at which
the samples were being accepted where not matching the expected rate,
and hence the distortion.
Solution: The appropriate solution to this problem is by reading the
MTU size of the SCO socket using getsockopts dynamically.
BugLink: http://bit.ly/2gDpGPv
BugLink: http://bit.ly/2hQsARK
X11 has its own bell volume setting, controlled with the "xset b"
command. If we use that volume, then the "System Sounds" slider in
pavucontrol doesn't affect the x11-bell sample volume, which in my
opinion is a bad thing. Ignoring the volume suggestion from X11 allows
module-stream-restore to apply the "event" role volume.
Not all VOIP applications (specially those which use alsa) set media.role to
phone. This means we need some heuristic to determinate if we want to switch
from a2dp to hsp profile based on number and types of source output (recording)
streams.
And also some people want to use their bluetooth headset (with microphone) as
their default recording device but some do not want to because of low quality.
This patch implements optional heuristic which is disabled by default. It is
disabled by default to not break experience of current pulseaudio users because
heuristic cannot be optimal. Heuristic is implemented in module-bluetooth-policy
module and decide if pulseaudio should switch to a hsp profile or not. It checks
if there is some source output with pass all these conditions:
* does not have set media.role
* does not use peak resample method (which is used by desktop volume programs)
* has assigned client/application (non virtual stream)
* does not record from monitor of sink
And if yes it switch to hsp profile.
By default this heuristic is disabled and can be enabled when loading module
module-bluetooth-policy with specifying parameter auto_switch=2
Because it is disabled by default nobody will be affected by this change unless
manually change auto_switch parameter.
Signed-off-by: Pali Rohár <pali.rohar@gmail.com>
In the current RTSP implementation, there is a vulnerable window
between the RTSP object creation and the URL initialization.
If any RTSP command is issued during this period, it will lead to
crash by assertion violation.
This patch introduces pa_rtsp_exec_ready(), which returns if it is
safe to issue RTSP commands.
Reviewed-by: Anton Lundin <glance@acc.umu.se>
Add a function performing a call to the OPTIONS request; also,
in some special cases, tuning transport parameters is required (default:
"RTP/AVP/TCP;unicast;interleaved=0-1;mode=record") ! The RAOP client for
example needs to overwrite them.
Reviewed-by: Anton Lundin <glance@acc.umu.se>
pa_ioline_close does not free the ioline structure itself, so we
have to unref the structure if we want to free it.
Reviewed-by: Anton Lundin <glance@acc.umu.se>
In the "unlink post" hook it's not guaranteed that the stream's old
device exists any more, so let's use the "unlink" hook that is safer.
For example, module-bluetooth-policy does a card profile change in the
source-output "unlink post" hook, which invalidates the source-output's
source pointer.
When the "unlink" hook is fired, the stream is still linked to its
device, which affects the return values of the check_suspend()
functions. The unlinked streams should be ignored by the check_suspend()
functions, so I had to add extra parameters to those functions.
This fixes a crash that happens if the bluetooth headset is the only
non-monitor source in the system and the last "phone" stream dies.
When the stream dies, the native protocol calls pa_source_output_unlink()
and would call pa_source_output_unref() next, but without this patch,
things happen during the unlinking, and the unreffing ends up being
performed on a stream that is already freed.
pa_source_output_unlink() fires the "unlink" hook before doing anything
else. module-bluetooth-policy then switches the headset profile from HSP
to A2DP within that hook. The HSP source gets removed, and at this point
the dying stream is still connected to it, and needs to be rescued.
Rescuing fails, because there are no other sources in the system, so the
stream gets killed. The native protocol has a kill callback, which again
calls pa_source_output_unlink() and pa_source_output_unref(). This is
the point where the native protocol drops its own reference to the
stream, but another unref call is waiting to be executed once we return
from the original unlink call.
I first tried to avoid the double unreffing by making it safe to do
unlinking recursively, but I found out that there's code that assumes
that once unlink() returns, unlinking has actually occurred (a
reasonable assumption), and at least with my implementation this was not
guaranteed. I now think that we must avoid situations where unlinking
happens recursively. It's just too hairy to deal with. This patch moves
the bluetooth profile switch to happen at a time when the dead stream
isn't any more connected to the source, so it doesn't have to be
rescued or killed.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=97906
Bug 96741 shows a case where an assertion is hit, because
pa_asyncq_new() failed due to running out of file descriptors.
pa_asyncq_new() is used in only one place (not counting the call in
asyncq-test): pa_asyncmsgq_new(). Now pa_asyncmsgq_new() can fail too,
which requires error handling in many places. One of those places is
pa_thread_mq_init(), which can now fail too, and that needs additional
error handling in many more places. Luckily there weren't any places
where adding better error handling wouldn't have been easy, so there are
many changes in this patch, but they are not complicated.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=96741
pa_memblockq_push() expects all memchunks to be aligned to PCM frame
boundaries, and that means both the index and length fields of
pa_memchunk. pa_rtp_recv(), however, used a memblock for storing both
the RTP packet metadata and the actual audio data. The metadata was
"removed" from the audio data by setting the memchunk index
appropriately, so the metadata stayed in the memblock, but it was not
played back. The metadata length is not necessarily divisible by the PCM
frame size, which caused pa_memblock_push() to crash in an assertion
with some sample specs, because the memchunk index was not properly
aligned. In my tests the metadata length was 12, so it was compatible
with many configurations, but eight-channel audio didn't work.
This patch adds a separate buffer for receiving the RTP packets. As a
result, an extra memcpy is needed for moving the audio data from the
receive buffer to the memblock buffer.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=96612
The module doesn't build any more[1], and when starting to investigate
the build failure, I asked the module author if he'd know something
about the breakage. He said that he didn't know about backward
compatibility problems with libxen, but more importantly, he said that
the module probably doesn't have any users[2]. It doesn't make sense to
keep maintaining a module that doesn't have users, so let's drop it.
[1] https://bugs.freedesktop.org/show_bug.cgi?id=98793
[2] https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-November/027172.html
SW: Pulseaudio 8.0 / BlueZ 5.39
Symptoms:
While disconnecting/reconnecting a paired bluetooth headset (LG HBS750)
audio fails roughly on every other connection.
On a failed connection "pactl list cards" shows the bluetooth device's
card but "Active Profile: off". Issuing "pacmd set-card-profile X
a2dp_sink" makes audio work immediately.
I realized that when this happened, the previous disconnection did not
remove the card, instead it was only configured for "Active Profile:
off" but otherwise left in place.
Upon looking at PA debug logs I saw that the transport for the a2dp_sink
was first set into disconnected state and then into idle state. In
"device_connection_changed_cb()" this causes the
"pa_bluetooth_device_any_transport_connected()" return true and the
module-bluez5-device is not unloaded.
Further investigation shows that this is caused by a race of
module-bluez5-device.c:thread_func() and
MediaPoint1::ClearConfiguration().
When the FD in thread_func() is closed (POLLHUP) an
BLUETOOTH_MESSAGE_STREAM_FD_HUP message is sent into the main thread.
The handler of this message unconditionally sets the transport into IDLE
state. This is a problem if it has already been set into DISCONNECTED
state.
