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The shout plugin will now feign playback until the connect timeout is hit,
preventing connection attempts from blocking playback on local outputs.
Note that this patch is very different from remiss' original one.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6738 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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except that it now uses a timer for throttling.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6621 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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if the clock ticks right after we get the start time and the timeout is
only one second, we'll still wait a full second instead of returning
immediately.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6557 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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pretending to play while we wait for the connection to timeout. This
removes the need for timers, and thus removes the now unnecessary
timer_get_runtime_* function(s) from the timer code.
The changes made compared to the pre-patch shout plugin are:
* Block while connecting, timing out after 2 seconds.
* Close the device, and not just the connection, if play returns -1.
* Remove sd->last_err (it's always assigned before use).
* Some minor cleanups.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6555 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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from blocking to non-blocking shout api
* Wait ten seconds before declearing the shout server unreachable
* Fix a state where it would never attempt to connect if it had previously failed
It isn't perfect yet, but I'd like some testing on it from other setups
git-svn-id: https://svn.musicpd.org/mpd/trunk@6523 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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state (when users press stop, previous snd_pcm_drop(), then
snd_pcm_drain() was called. this would lockup dmix)
git-svn-id: https://svn.musicpd.org/mpd/trunk@6517 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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used in other plugins (fifo, shout, etc.).
git-svn-id: https://svn.musicpd.org/mpd/trunk@6397 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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uninitialized variables and non-returning functions that return. Let's
tell it to stfu.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6277 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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call to FATAL().
git-svn-id: https://svn.musicpd.org/mpd/trunk@6276 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6232 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5909 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5897 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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or no" parameters.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5896 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5733 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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just casting to int because it's the simplest (%z is not
well-supported)
Noticed-by: avuton on a 64-bit machine
git-svn-id: https://svn.musicpd.org/mpd/trunk@5257 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Some compilers and linkers aren't smart enough to optimize this,
as global variables are implictly initialized to zero. As a
result, binaries are a bit smaller as more goes in the .bss and
less in the text section.
git-svn-id: https://svn.musicpd.org/mpd/trunk@5254 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5252 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5251 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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There's no reason they shouldn't be static. Additionally,
output_ports doesn't need to be initialized to NULLs; that is
(and has always been) implicit (for all global variables)
git-svn-id: https://svn.musicpd.org/mpd/trunk@5247 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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..and not the enum value that corresponds to that bitrate
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4912 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This patch should continue to allow mpd to play as well as
possible to icecast servers while avoiding stalls on local
devices. This has eliminated ALSA underrun errors
for me while streaming to a remote host while the network
connection was bad.
Of course, this makes opening a connection non-blocking, too,
so myShout_openShoutConn is a bit more complex.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4898 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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I'm checking for zero-size allocations and assert()-ing them,
so we can more easily get backtraces and debug problems, but we'll
also allow -DNDEBUG people to live on the edge if they wish.
We do not rely on errno when checking for OOM errors because
some implementations of malloc do not set it, and malloc
is commonly overridden by userspace wrappers.
I've spent some time looking through the source and didn't find any
obvious places where we would explicitly allocate 0 bytes, so we
shouldn't trip any of those assertions.
We also avoid allocating zero bytes because C libraries don't
handle this consistently (some return NULL, some not); and it's
dangerous either way.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4690 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@4689 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Unfortunately there doesn't seem to be an indent switch for this,
but we have find + perl:
find src -name '*.[ch]' | xargs perl -i -p -e \
's/^\s+(\w+):/$1:/g unless /^\s+default:/'
This is a followup to r4605, and there are no actual code
changes in this.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4661 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Passing a ref to snd_pcm_hw_params_set_{buffer,period}_time_near
can modify our internal {period,buffer}_time members inside the
AlsaData structure, making re-initializing the device across
sample/bit rate and channel changes non-idempotent.
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* less-commonly compiled things like ao/mvp outputs
* Adding -Wno-transparent-union to SPARSE_FLAGS makes it check
inside decode.c, directory.c, player.c, and sig_handlers.c
* remove unused variables leftover from the master process
in sig_handlers.c
git-svn-id: https://svn.musicpd.org/mpd/trunk@4598 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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making it more presentable isn't something I'm willing to do before 0.12. It will likely be added back after 0.12, along with some very experimental stuff to make it more usable.
git-svn-id: https://svn.musicpd.org/mpd/trunk@4472 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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We'll try setting an initial value of 50ms, and halve it each
time snd_pcm_hw_params fails with -EPIPE.
This way we'll can use a larger (50ms) period_size whenever a device
supports it, and automatically pick smaller ones if we can't set
larger ones.
This removes the calculation borrowed from libao (svn) as well.
Other minor things:
"Alsa" => "ALSA" in error messages
_US appended to *_TIME constants so we won't get confused
(shank's request)
git-svn-id: https://svn.musicpd.org/mpd/trunk@4438 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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to 0ms sounds like complete crap. 50ms is the default that xmms has used for years, so lets just stick with that.
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