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.
This serves to explicitly document the various cases we deal with in
pa_sink_update_rate()/pa_source_update_rate() rather than have some of
them hidden behind the initialisation of desired_rate.
This adds an "avoid-resampling" option to daemon.conf that makes the
daemon try to use the stream sample rate if possible (the device needs
to support it, which currently only ALSA does), and there should not be
any other stream connected).
This should enable some of the "audiophile" use-cases where users wish
to play high sample rate audio files without resampling.
We still will do conversion if sample formats don't match, though. This
means that if you want to play 96 kHz/24 bit audio without any
modification the default format will need to be set to be 24-bit as
well. This will force all streams to be upconverted, which, other than
the wasted resources, should be relatively harmless.
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
This macro compares if the given two strings, with the maximum length
of n, are equal. Useful for strings that are not NULL-terminated.
Reviewed-by: Anton Lundin <glance@acc.umu.se>
Constants should be declared simply with "const". With struct members,
"static" means that all struct instances share the same variable, i.e.
all instances always see the same value. That's of course already
implied in the concept of "constant". Newer Vala versions don't allow
mixing "const" and "static".
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
