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Again, no CamelCase in the directory name.
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Added configuration parameter "protocol" which lets the user choose
from 3 shout protocols. This adds support for real shoutcast servers.
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Case insensitivity isn't helpful, and comparing only the first 3 bytes
of a configured value may encourage users to supply wrong or
misleading values.
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The last bit of CamelCase in audio_format.h. Additionally, rename a
bunch of local variables.
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"bits" and "channels" cannot be negative.
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Having an array with disabled entries sucks. Removed that
DISABLED_SHOUT_ENCODER_PLUGIN macro, and fill the plugin list only
with plugins which are actually enabled. This should be done for all
plugin types.
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When there are standardized headers, use these instead of the bloated
os_compat.h.
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pause() puts the audio output into pause mode: if supported, it may
perform a special action, which keeps the device open, but does not
play anything. Output plugins like "shout" might want to play silence
during pause, so their clients won't be disconnected. Plugins which
do not support pausing will simply be closed, and have to be reopened
when unpaused.
This pach includes an implementation for the shout plugin, which
sends silence chunks.
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The old struct initializers are error prone and don't allow moving
elements around. Since we are going to overhaul some of the APIs
soon, it's easier to have all implementations use C99 initializers.
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We have eliminated direct accesses to the audio_output struct from
the all output plugins. Make it opaque for them, and move its real
declaration to output_internal.h, similar to decoder_internal.h.
Pass the opaque structure to plugin.init() only, which will return the
plugin's data pointer on success, and NULL on failure. This data
pointer will be passed to all other methods instead of the
audio_output struct.
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Since the output plugin returns a value indicating success or error,
we can have the output core code assign the "open" flag.
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Pass the globally configured audio_format as a const pointer to
plugin.init(). plugin.open() gets a writable pointer which contains
the audio_format requested by the plugin. Its initial value is either
the configured audio_format or the input file's audio_format.
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Add a check to write_page() which checks if there is actually data.
Don't bother to call shout_send() if there is not.
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The function is trivial, without a benefit. Also don't initialize
buf.data[0], this is not a null terminated string.
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Since the buffer size is known at compile time, we can save an
indirection by declaring it as a char array instead of a pointer.
That saves an extra allocation, and we can calculate with the
compile-time constant sizeof(data) instead of the attribute "max_len".
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Declare both shout plugins "const", since they will never change, once
initialized at compile time.
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Shout encoder plugins are known at compile time. There is no reason
to use a complex data structure as "List" to manage them at runtime -
just put the pointers into a static array.
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Don't typedef the structs at all. It is easier to forward-declare
this way.
Don't typedef methods. They are used exactly once, a few lines below.
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[mk: moved this patch after "Refactor and cleanup of shout Ogg and MP3
audio outputs". The original commit message follows, although it is
outdated:]
Creation of shout_mp3 audio output plugin. Basically I just copied the
existing shout plugin and replaced ogg with lame. Uses lame for mp3
encoding. Next step is to pull common functionality out of each shout
plugin and share it between them.
Configuration options for "shout_mp3" are the same as for "shout".
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I've perhaps gone a bit overboard, but here's the current rundown:
Both Ogg and MP3 use the "shout" audio output plugin. The shout audio
output plugin itself has two new plugins, one for the Ogg encoder,
and another for the MP3 (LAME) encoder.
Configuration for an Ogg stream doesn't change. For an MP3 stream,
configuration is the same as Ogg, with two exceptions. First, you must
specify the optional "encoding" parameter, which should be set to "mp3".
See mpd.conf(5) for more details. Second, the "quality" parameter is
reversed for LAME, such that 1 is high quality for LAME, whereas 10 is
high quality for Ogg.
I've decomposed the code so that all libshout related operations
are done in audioOutput_shout.c, all Ogg specific functions are in
audioOutput_shout_ogg.c, and of course then all LAME specific functions
are handled in audioOutput_shout_mp3.c.
To develop encoder plugins for the shout audio output plugin, I basically
just mimicked the plugin system used for audio outputs. This might be
overkill, but hopefully if anyone ever wants to support some other sort
of stream, like maybe AAC, FLAC, or WMA (hey it could happen), they will
hopefully be all set.
The Ogg encoder is slightly less optimal under this configuration.
It used to send shout data directly out of its ogg_page structures. Now,
in the interest of encapsulation, it copies the data from its ogg_page
structures into a buffer provided by the shout audio output plugin (see
audioOutput_shout_ogg.c, line 77.) I suspect the performance impact
is negligible.
As for metadata, I'm pretty sure they'll both work. I wrote up a test
scaffold that would create a fake tag, and tell the plugin to send it
out to the stream every few seconds. It seemed to work fine. Of course,
if something does break, I'll be glad to fix it.
Lastly, I've renamed lots of things into snake_case, in keeping with
normalperson's wishes in that regard.
[mk: moved the MP3 patch after this one. Splitted this patch into
several parts; the others were already applied before this one. Fixed
a bunch GCC warnings and wrong whitespace modifications. Made it
compile with mpd-mk by adapting to its prototypes]
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Support sending metadata to a shout server using shout_metadata_new()
and shout_metadata_add(). The Ogg Vorbis encoder does not support
this currently.
[mk: this patch was separated from Eric's patch "Refactor and cleanup
of shout Ogg and MP3 audio outputs", I added a description]
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The Ogg encoder is slightly less optimal under this configuration. It
used to send shout data directly out of its ogg_page structures. Now,
in the interest of encapsulation, it copies the data from its ogg_page
structures into a buffer provided by the shout audio output plugin
(see audioOutput_shout_ogg.c, line 77.) I suspect the performance
impact is negligible.
[mk: this patch and its description was separated from Eric's patch
"Refactor and cleanup of shout Ogg and MP3 audio outputs"]
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Begin dividing audioOutput_shout.c: move everything OGG Vorbis related
to audioOutput_shout_ogg.c. The header audioOutput_shout.h has to
keep its dependency on vorbis/vorbisenc.h, because it needs the vorbis
encoder types.
For this patch, we have to export several internal functions with
generic names to the ABI; these will be removed later when the encoder
plugin patches are merged.
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Prepare the split of the shout plugin into multiple sources: move
all important declarations to audioOutput_shout.h.
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Remove unused code which is in comments. Remove that comment about
"stolen code", since the plugin has changed much, and it isn't obvious
which parts are derived.
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In the plugin's init() function, outAudioFormat is simply a copy of
reqAudioFormat. Use reqAudioFormat instead of outAudioFormat here.
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Storing pointers to immutable audio_format structs isn't worth it,
because the struct itself isn't much larger than the pointer. Since
the shout plugin requires the user to configure a fixed audio format,
we can simply copy it in myShout_initDriver().
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The audio_format argument to timer_new() should be constant, because
it is not modified. The same is true for ShoutData.audioFormat.
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Since the plugin struct is never modified, we should store it in
constant locations.
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Also rename AudioOutputPlugin to struct audio_output_plugin, and use
forward declarations to reduce include dependencies.
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Just like decoder_api.h, output_api.h provides the audio output API
which is used by the plugins.
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Get rid of CamelCase, and don't use a typedef, so we can
forward-declare it, and unclutter the include dependencies.
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The previous patch enabled these warnings. In Eric's branch, they
were worked around with a generic deconst_ptr() function. There are
several places where we can add "const" to pointers, and in others,
libraries want non-const strings. In the latter, convert string
literals to "static char[]" variables - this takes the same space, and
seems safer than deconsting a string literal.
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During the tag library refactoring, the shout plugin was disabled, and
I forgot about adapting it to the new API. Apply the same fixes to
the oggflac decoder plugin.
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Getting rid of CamelCase; not having typedefs also allows us to
forward-declare the structures.
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Don't use CPP macros when you can use C enum... this also allows
better type checking.
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The audio output plugins should get a constant pointer, because they
must not modify the buffer. Since the size is a non-negative buffer
size in bytes, we should change its type to size_t.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7293 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Local variables which are never read before the first assignment don't
need initialization. Saves a few bytes of text. Also don't reset
variables which are never read until function return.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7199 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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When we expect an integer as result, why would we use the double
precision floating point parser? strtol() is a better match, although
we should probably check for overflows...
git-svn-id: https://svn.musicpd.org/mpd/trunk@7198 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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[ew: cleaned up the dirty union hack a bit]
Signed-off-by: Eric Wong <normalperson@yhbt.net>
git-svn-id: https://svn.musicpd.org/mpd/trunk@7180 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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Signed-off-by: Eric Wong <normalperson@yhbt.net>
git-svn-id: https://svn.musicpd.org/mpd/trunk@7146 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This will make refactoring features easier, especially now that
pthreads support and larger refactorings are on the horizon.
Hopefully, this will make porting to other platforms (even
non-UNIX-like ones for masochists) easier, too.
os_compat.h will house all the #includes for system headers
considered to be the "core" of MPD. Headers for optional
features will be left to individual source files.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7130 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@6858 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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This attribute was set in log.c, but not exported to other
modules in log.h
This allows us to remove some unneccessary variable
initializations that were added in r6277. I did
audioOutput_shout.c a bit differently, to avoid some
jumps.
before:
$ size src/mpd
text data bss dec hex filename
225546 4040 14600 244186 3b9da src/mpd
after:
$ size src/mpd
text data bss dec hex filename
224698 4040 14600 243338 3b68a src/mpd
git-svn-id: https://svn.musicpd.org/mpd/trunk@6821 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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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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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git-svn-id: https://svn.musicpd.org/mpd/trunk@6556 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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