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Use GLib's GError library for reporting output device failures.
Note that some init() methods don't clean up properly after a failure,
but that's ok for now, because the MPD core will abort anyway.
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Return true/false instead of 1/0.
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audio_output_get_name() has been removed, which was the only function
left in output_api.h. The output plugin doesn't need the audio_output
object at all, remove the parameter from the init() method.
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Similar to the decoder plugin API: added wrapper functions to increase
code readability.
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Added audio_format_parse() in a separate library, with a modern
interface: return a GError instead of logging errors. This allows the
caller to deal with the error.
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Renamed audio_output struct members.
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All config_get_block_*() functions should accept constant config_param
pointers.
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Renamed functions, types, variables.
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Removed yet another superfluous buffer layer: return the PCM buffer
from pcm_convert() instead of copying PCM data into the
caller-supplied buffer.
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When one of several output devices failed, MPD tried to reopen it
quite often, wasting a lot of resources. This patch adds a delay:
wait 10 seconds before retrying. This might be changed to exponential
delays later, but for now, it makes the problem go away.
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Don't return 0/-1 on success/error, but true/false. Instead of int,
use bool for storing flags.
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Instead of manually calling memset(0) on the pcm_convert_state struct,
client code should use a library function from pcm_utils.c. This way,
we can change the semantics of the struct easily.
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No CamelCase, and a struct instead of a typedef.
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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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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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To keep I/O nastiness and latencies away from the core, move the audio
output code to a separate thread, one per output. The thread is
created on demand, and currently runs until mpd exits.
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As long as the device isn't open, both attributes are not used. Since
they will both be initialized in audio_output_open(), we do not need
the initialization in audio_output_init().
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Eliminate sameInAndOutFormats and check with audio_format_equals()
each time it this information is needed. Another 4 bytes saved.
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Instead of checking convertAudioFormat, we can simply check if
reqAudioFormat is defined. This saves 4 bytes in the struct.
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The "!src" check in copyAudioFormat() used to hide bugs - one should
never pass NULL to it. There is one caller which might pass NULL, add
a check in this caller.
Instead of doing mempcy(), we can simply assign the structures, which
looks more natural.
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Getting rid of CamcelCase, again.
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Similar to decoder_control.c, output_control.c will provide functions
for controlling the output thread (which will be implemented later).
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No CamelCase. Also don't declare typedefs for the methods.
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Instead of having to register each output plugin, store them
statically in an array. This eliminates the need for the List library
here, and saves some small allocations during startup.
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Instead of copying all that stuff from the audio output plugin to the
audio output structure, store a pointer to the plugin.
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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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Getting rid of CamelCase; not having typedefs also allows us to
forward-declare the structures.
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And again, convert arguments to const.
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We want to expose the AudioFormat structure to plugins; remove some
clutter by moving its declaration to a separate header file.
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7345 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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When growing the audioOutput->convBuffer, we can use free()+malloc()
instead of realloc(), which saves a memcpy().
git-svn-id: https://svn.musicpd.org/mpd/trunk@7295 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@7294 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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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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In my previous patch set, I forgot to change the
pcm_sizeOfConvBuffer() invocation in convertAudioFormat() to also use
size_t.
git-svn-id: https://svn.musicpd.org/mpd/trunk@7292 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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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@6527 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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one now, and trying to call NULL was causing a segfault at exit.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6398 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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because lsr may return less than the input buffer size, and the rest of the
audio code needs to know the new size. This fixes the clicking that was
introduced with recent changes to the lsr code. A huge thanks to remiss
for figuring this out.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6273 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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audio at once, so it won't work for us. The old full API code was still
heavily broken, as each call to pcm_convertSampleRate() used the same
state, even if it was processing two streams of audio. The new code keeps
a separate state for each audio stream that's being converted.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6255 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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and samplerate conversion. This makes the code much easier to read, and
fixes a few bugs that were previously there.
git-svn-id: https://svn.musicpd.org/mpd/trunk@6224 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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git-svn-id: https://svn.musicpd.org/mpd/trunk@5834 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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- don't close and reopen an audioOutput when it has a fixed output format, and closing and reopening the device is unneccessary when the input audio format changes
git-svn-id: https://svn.musicpd.org/mpd/trunk@4908 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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