libwrap is an obscure artefact from a past long ago, when source IP
address meant something.
And its API is "interesting"; it requires the application to expose
two global variables `allow_severity` and `deny_severity`. This led
to bug #437. I don't want to declare those variables; instead, I'd
like to remove libwrap support.
Closes#437
This is similar to b177bffa6a, in that it fixes the Windows issue of connecting to the open socket. Also, the listen_socket is set to AcceptNonBlock after the connection anyways.
This function is sparsely documented and a look at the bluez-alsa
source code shows that implementations make undocumented assumptions
on the `struct pollfd` array parameter which can lead to strange
effects.
Since we switched from autotools to Meson in commit
94592c1406, we don't need to include
`config.h` early to properly enable large file support. Meson passes
the required macros on the compiler command line instead of defining
them in `config.h`.
This means we can include `config.h` at any time, whenever we want to
check its macros, and there are no ordering constraints.
Works around a problem where MPD goes into a busy loop because
snd_pcm_drain() always returns `-EAGAIN` without making any progress
(fixes#425).
This problem was triggered by snd_pcm_drain() after snd_pcm_cancel()
and snd_pcm_prepare(), but without submitting any data with
snd_pcm_writei().
I believe this is a kernel bug: in non-blocking mode, the kernel's
snd_pcm_drain() function returns early. In this mode, it only checks
whether snd_pcm_drain_done() has been called already, but
snd_pcm_drain_done() is never called if no data was submitted.
In blocking mode, the following `for` loop detects this condition, so
snd_pcm_drain_done() is not necessary, but without this extra check,
we get `-EAGAIN` forever.
This fixes a problem which caused a failure with snd_pcm_writei()
because snd_pcm_drain() had already been called in the previous
iteration. This commit makes sure that snd_pcm_drain() is only called
after the final snd_pcm_writei() call.
This fixes discarded samples at the end of playback.
MPD's default is 100ms, which is too long for the real-time I/O
thread. The OutputThread has 100us, but the real-time I/O thread
might have tighter deadlines.
This change has currently no effect (I believe), because nobody uses
timers on the RTIO thread.
If our `ring_buffer` is smaller than the ALSA-PCM buffer (if the
latter has more than the 4 periods we allocate), it can happen that
the start threshold is crossed and ALSA switches to
`SND_PCM_STATE_RUNNING`, but the `ring_buffer` is empty. In this
case, MPDD will generate silence, even though the ALSA-PCM buffer has
enough data. This causes stuttering (#420).
This commit amends an older workaround for a similar problem (commit
e08598e7e2) by adding a snd_pcm_avail()
check, and only generate silence if there is less than one period of
data in the ALSA-PCM buffer.
Fixes#420
The method Cancel() assumes that the `period_buffer` must be empty
when `active==false`, but that is not the case when Play() fails.
Of course the assertion in Cancel() is not 100% correct, but I decided
to rather fix this in LockCaughtError() because the `period_buffer`
should only be accessed from within the RTIO thread, and this is the
only code path where `active` can be set to `false` with a non-empty
`period_buffer`.
Fixes#423
This check was added 9 years ago in commit
4dc25d3908 to work around a dmix bug
which I assume has been fixed long ago.
Removing this fixes another corner case: if draining is requested
before the start threshold is reached, the PCM is still in
SND_PCM_STATE_PREPARED but not yet SND_PCM_STATE_RUNNING, which means
the submitted data will never be played. This corner case is
realistic when playing songs shorter than the ALSA buffer (if the
buffer is very large).
This fixes a corner case which has probably never occurred and
probably never will: if Cancel() is called, and then Play() followed
by Drain(), the plugin should really play that data. However
currently, this never happens, because snd_pcm_prepare() is never
called.
When `metadata_sent` is `false`, the plugin assumes there is metadata
which must be sent, even if no metadata page was passed to the plugin.
Initializing it to `true` avoids dereferencing this `nullptr`.
Fixes#412
If the output is already open, the `current_chunk` pointer may be
bogus and out of sync with `SharedPipeConsumer::chunk`, leading to an
assertion failure in `SharedPipeConsumer::Consume()`.
Fixes#411
Meson always enables large file support on the compiler command line,
thus config.h doesn't need to be included anymore. We'll remove the
whole `check.h` header soon.
Closes#409
`AVCodecParameters` contains values from the codec detected by
avformat_find_stream_info(), but after avcodec_open2(), a different
codec might be selected with a different `AVSampleFormat`. This leads
to misinterpretation of data returned from FFmpeg, leading to random
noise or silence.
This was observed with FFmpeg 4.0.2 and a TS container file containing
MP2. A mp3-float codec was detected returning `AV_SAMPLE_FMT_FLTP`,
but finally the `mpegaudiodec_fixed.c` was used, returning
`AV_SAMPLE_FMT_S16`.
By using the audio format from `AVCodecContext`, we ensure that MPD
and FFmpeg always agree on the actual audio format in the buffer.
This removes the FFmpeg bug workaround from commit e1b032cbad which I
assume is obsolete after 7 years.
Fixes#380
ensure that valid mixer values are set also when the ALSA driver
does not report a valid dB range ('set_raw' fallback)
correct a bug in which volume is assumed to lie in [0..100]
instead of [0..1]
The protocol documentation says that the difference between `find` and
`search` is that `search` is case insensitive, but that's only half
the truth: `search` also searches for sub strings instead of matching
the whole string. This part is undocumented and unfortunate, but at
this point, we can't change it.
However leaking this surprising behavior to the new filter expressions
was a bad idea; the "==" operator should never match substrings. For
people who need that, we should add a new operator.
Thanks to C++14, we can declare and fill variables inside `constexpr`
functions. This means me can stop make assumptions on the `struct`
layouts without losing `constexpr`.
Closes#393
Bugs in libroar which broke the MPD build have been annoying me for
quite some time, and the newest bug has now hit my main build machine:
https://github.com/MusicPlayerDaemon/MPD/issues/377
Problem is the usage of the typedef `_IO_off64_t` in libroar's
`vio_stdio.h`:
int roar_vio_to_stdio_lseek (void *__cookie, _IO_off64_t *__pos, int __w);
This `_IO_off64_t` is an internal implementation detail of glibc and
was removed in version 2.28. Nobody must ever use it. Why the ****
did the RoarAudio developers use it? Not using internal typedefs
isn't exactly rocket science.
This annoys me enough to finally remove the plugin. Anyway, I've
never heard of anybody using RoarAudio, so my best guess is that
nobody will notice.
The compile-time calculation for `factor` overflows because `1<<31`
cannot be represented by `int`. By casting to `uintmax_t` first, we
can avoid this overflow.
Closes#380
Grouping in the "list" command was completely broken from the start,
unlike "count group". I have no idea what I have been thinking when I
wrote commit ae178c77bd, but it didn't
make any sense.
This commit is a rewrite of the feature.
For clients to be able to detect this feature, this commit also
increments the protocol version.
So long, autotools! This is my last MPD related project to migrate
away from it. It has its strengths, but also very obvious weaknesses
and weirdnesses. Today, many of its quirks are not needed anymore,
and are cumbersome and slow. Now welcome our new Meson overlords!
Previously, there was the setting `buffered_before_play` which
specified a percentage of the audio buffer, defaulting to `10%`. That
was working well enough for quite some time, until high-quality audio
formats became common.
At 44.1 kHz, 16 bit stereo, MPD collected 2.3 seconds worth of data in
the buffer before starting playback. With the same default settings
and 192 kHz, 24 bit stereo, that was only 0.27 seconds.
Making this depend on the byte size only leads to high latency at low
quality, and too little data at high quality. The natural choice
would be to use a duration instead of a byte size, which should give
the same good experience with all audio formats.
Since the `buffered_before_play` configuration setting was not
understood well by users and caused more harm than good, this commit
deprecates it. It has now no effect.
Simplify the formula, and I guess this makes the formula more
reliable. Imagine somebody configured `buffered_before_play` larger
than 25%; then the decoder would be woken up all the time. This
doesn't seem logical. On the other hand, it's easy to understand that
the decoder should be woken up below 75% buffer fill.
