This code was added in 21851c0673 but
looks completely broken:
- the status code is "206 OK" but "206" would be "Partial Content"
- the "Content-Length" header has a bogus value
- the "Content-RangeX" parameter has different bogus values (why
"Content-RangeX" anyway and not "Content-Range"?)
Apart from that, there are strange undocumented non-standard headers
which are probably there to work around bugs/expectations in one
broken proprietary client product. But these days, MPD doesn't bend
over to support broken clients. So let's kill this code.
Closes#304
For remote files (not streams), this downloads as quickly as possible
to a large buffer instead of throttling the stream during playback.
Throttling can make the server impatient and it may then disconnect.
This is what Qobuz and Tidal do, and this commit attempts to solve
this by not letting the Qobuz/Tidal server wait (closes#241).
Error message sent to client was "basic_string::_M_construct null not
valid" due to passing nullptr to the std::string constructor.
Regression caused by commit 386688b87a
When switching to another song manually, the player checks if the
decoder is already decoding that song; if so, it will attempt to reuse
it by seeking it to the new position. That however fails if the
decoder is not seekable (e.g. a radio stream) which leaves the user
unable to switch to that song with the bogus error message "Not
seekable".
Instead of stopping playback (due to seek time overflow), reject the
seek command. Closes#240
Relative negative values (with "seekcur") are still allowed, and MPD
will fix the resulting position if it turns out to be negative. But
the "seek" and "seekid" commands use an unsigned time stamp which must
not be negative.
With Grand Central Dispatch used in Main.cxx, debug builds on macOS
crash as the IsInside() assertion gets triggered in the event loop. As
a simple fix, usage of GCD is removed. Plugging and unplugging
headphones or changes of the default output device was tested without
issues. Whatever the original commit tried to fix by GCD probably does
not need fixing anymore.
From: Christian Kröner <ckroener@gmx.net>
This just copies the necessary bits and pieces from the ALSA plugin and applies them to OSXOutput based on dop config setting. It only changes the OSXOutput plugin as needed for DoP (further changes to support additionally e.g. integer mode or setting the physical device mode require rather a complete rewrite of the output plugin).
Fortunately the Core Audio API is by default bit perfect and supports DoP with minimal changes (setting the sampling rate accordingly after ensuring that the physical mode supports at least 24 bits per channel seems to be enough). This was tested on an Amanero Combo384 device hooked up to a ES9018 DAC.
USAGE (try only on DACs that support DoP):
- Add dop "yes" option to mpdconf
- Be sure to set at least 24bits per channel before playing some DSD file (using Audio-MIDI-Setup)
- Based on the dop setting, MPD will change the sample rate as required and output DoP signal to the DAC
- Hog mode is recommended to ensure that no other program will try to mix some output with the DoP stream (resulting in bad noise)
- Alternatively set the default output device to another device (e.g. the built-in output) to avoid having other audio interfere with DSD playback
support for chaining ogg opus streams to enable changing stream' metadata on the fly.
currently support is opt-in (enabled by additional option) because lots of clients can't handle this properly yet.
This addresses two problems:
1. the libFLAC write callback had to send an error status to its
caller when SubmitData() returned a command; this disrupted libFLAC
and the resulting command could not be used for anything;
2. the libFLAC function FLAC__stream_decoder_seek_absolute() also
calls the write callback, but its result cannot be used, because
seeking is still in progress, so we lose all data from one FLAC frame.
By moving the SubmitData() call until after CommandFinished(), we
avoid losing this data. This fixes another part of #113
Instead of passing whole chunks to the MusicPipe and checking the
end_time after each chunk, truncate the last chunk if it would exceed
the end_time. This requires keeping track of the absolute PCM frame
number.
This fixes a problem with gapless CUE song transitions: a small part
of the following song was always played twice.
Closes#113
Due to rounding errors, a slightly negative value can be passed to
set_normalized_volume(), which will make the log10() call fail.
Actually, volume 0 is already failing because log10(0) is illegal. So
let's fix this by implementing two corner cases: <=0 and >=100.
Closes#212
There is a POSIX definition for sched_setscheduler(), but Linux does
not implement that; instead of changing the process's scheduler, it
only affects one thread. This has caused some confusion among
application developers and C library developers.
While glibc implements Linux semantics, Musl has made their
sched_setscheduler() function an always-failing no-op, causing the
error message "sched_setscheduler failed: Function not implemented".
http://git.musl-libc.org/cgit/musl/commit/src/sched/sched_setscheduler.c?id=1e21e78bf7a5c24c217446d8760be7b7188711c2
Instead of relying on the C library which may be unreliable here, we
now roll our own system call wrapper.
Closes#218
This attribute shall be used only for IsInside() to make this safe
against a race condition described in #188:
> There is no requirement on the implementation that the ID of the
> created thread be available before the newly created thread starts
> executing.
http://pubs.opengroup.org/onlinepubs/009695399/functions/pthread_create.html):
This means that on some pthread implementations (e.g. Haiku), the
assert(thread.IsInside()) could fail.
Closes#188
This completes the bug fix commit
2065e3290452377b2931f3129b230c8cc536cbc8; if we clear "queued" then we
must clear "queued_song" as well, or another variant of the assertion
fails.
In current Android SDK releases, Ant support was removed. Move the
necessary build steps from the former Ant build system to our Makefile,
and call the required build tools from the Android SDK (aapt and dx),
Java SDK (javac) and Info-ZIP (zip) directly.
[mk: copied from Felix's commit
e52b906dba971a1173f9e8f83d32b52ee9f89af3 in the XCSoar project)
Supporting GCC 4.9 is becoming cumbersome as we refactor more code to
C++14.
This shouldn't be a problem for anybody, because Debian Stable
(Stretch) has version 6.3, and Ubuntu LTS (Xenial) has version 5.3.
Fixes deadlock because FileInputStream::Read() unlocks the mutex
(which was not locked) and then locks it, keeping it locked. This can
result in a deadlock. This happens because the archive and the file
mutex are different.
If a directory is a mount point, omit the "directory: " as well.
This bug is years old, but has become more visible now that mount
points are persistent in the state file.
These attributes are printed in the "outputs" response, and the new
command "outputset" allows setting new values.
No attributes are currently implemented.
If `SimpleDatabase::Visit` is called on a database that contains a mounted directry the URIs of the elements passed to the callbacks are not prefixed by the mountpoint path. This leads to lsinfo and add not working because they use the wrong URI. This pull request is using the `WalkMount` helper function to create prefixed versions of `VisitDirectory`, `VisitSong` and `VisitPlaylist` to add the correct prefix to the parameters of the callback functions.
[mk: the following text was copied from
https://github.com/MusicPlayerDaemon/MPD/pull/167]
For certain format (hi-res files) and normal buffer size hardware, The
hardware may at once consume most of the buffers. However, in Delay()
function, MPD is supposed to wait for 25 ms after the next try. it
will create a hiccup. The negative impact is much major than
increasing the latency.
I understand larger buffers come at a price. That's why in my earlier
commit last year I significantly reduced it. However, the buffer size
in CoreAudio is set according to the hardware, which is super small
latency. For instance, the system audio of 2015 generation of macbook
pro has maximum buffer size of 4096 samples, which is just 0.09s for
44.1k framerate, or 0.04s for 96k frames --- . compare to the 0.5 sec
latency alsa plugin has, even if we quadruple it, it's still super
tiny.
