The state_changed callbacks fulfill multiple roles, which is both a problem
regarding separation of concerns and regarding code clarity. De facto,
these callbacks cover error reporting, opening connections, and closing
connection, all in one, depending on a state that is arguably an internal
stream detail. The code in these callbacks tie these internal states to
assumptions that opening/closing callbacks is directly tied to specific
state changes in a common way, which is not always true. For example,
stopping the stream may not _actually_ stop it if a background send timer
is still running.
The notion of a "state_changed" callback is also problematic because the
pw_streams that are used in rtp-sink and rtp-source also have a callback
for state changes, causing confusion.
Solve this by replacing state_changed with three new callbacks:
1. report_error : Used for reporting nonrecoverable errors to the caller.
Note that currently, no one does such error reporting, but the feature
does exist, so this callback is introduced to preserve said feature.
2. open_connection : Used for opening a connection. Its optional return
value informs about success or failure.
3. close_connection : Used for opening a connection. Its optional return
value informs about success or failure.
Importantly, these callbacks do not export any internal stream state. This
improves encapsulation, and also makes it possible to invoke these
callbacks in situations that may not neatly map to a state change. One
example could be to close the connection as part of a stream_start call
to close any connection(s) left over from a previous run. (Followup commits
will in fact introduce such measures.)
config.h needs to be consistently included before any standard headers
if we ever want to set feature test macros (like _GNU_SOURCE or whatever)
inside. It can lead to hard-to-debug issues without that.
It can also be problematic just for our own HAVE_* that it may define
if it's not consistently made available before our own headers. Just
always include it first, before everything.
We already did this in many files, just not consistently.
Add support for FairPlay SAP v2.5 (encryption type 5) type devices such as Apple Home Pod Minis.
Apparently only these devices require the `POST /feedback` heartbeat, so fix that.
Initialize the byte array with bytes instead of a string because the 0
byte at the end of the string does not fit in the array and causes a
compiler warning.
Intercept the Output Latency paran and parse it for later.
Use the computed latency as the ProcessLatency and expose this
as the ProcessLatency param and the updated Input latency.
Accept updates to ProcessLatency to modify the latency, which then also
updates the Input Latency param.
See #4270
32 bits are enough, and additionally this also fixes an incorrect
format string, which caused the default `audio.rate` to be
incorrectly set on some platforms, such as 32-bit arm ones.
Fixes#4080
Since `spa/utils/cleanup.h` is not a private header anymore, there is
no need for a separate `pipewire/cleanup.h` since the definitions of
the cleanup routines can now be moved into the respective headers.
As part of LibreSSL's cleanup of removed ENGINE support, <openssl/x509.h> is no longer included in <openssl/engine.h>. <openssl/evp.h> was transiently included in <openssl/x509.h>, compared to <openssl/rand.h> for OpenSSL.
We first need to do the setup, which we start when setting the Format
param.
This causes the module to unload when starting pavucontrol because it
tries to start the record in the wrong state.
See #3778
The Pro Link 1 replies with Audio-latency=0, patch that up to
1500ms to make it work again.
Previously it configured 1500ms as the default latency but that seems
unnecessary in the usual case.
Fixes#3698
The RTSP FLUSH request does not seem to be required.
This change also fixes an issue where another RECORD request is
erroneously sent when the stream switches back from paused to streaming
by only setting `impl->streaming = false` in `rtsp_do_teardown()`.
Use both the volume and mute to decide what volume to send.
Don't let the mute state overwrite the volume. Also never mute the
stream.
Pressing mute and unmute restores the previous volume this way.
The volume interval that RAOP devices understand is [-30,0],
where -30.0 equals min vol, and 0.0 equals max. vol.
The local system volume is represented as a cubic (volumetric)
value in the [0,1] interval.
So cube root system volume value, scale by 30 and
translate -30 to map to target output range.
The special value -144 denotes volume mute. Send a corresponding RTSP
message when mute is not already toggled on.
This reverts commit 21d16b1ad5.
The change causes the sound of videos to be way out of sync when streaming from Fedora to Sonos.
It seems the issue is a device-specific quirk, and the change cannot be applied universally to all devices,
thus reverting until a better solution is found that does not affect other devices.
The function already returns `ssize_t`, so do not use `errno`
to communicate the reason for failure, instead, return the
negative errno.
`pw_getrandom()` was inconsistent in this regard because
sometimes it simply returned a negative errno without
setting `errno`. This change fixes that as well.
