| Commit message (Collapse) | Author | Files | Lines |
|---|
|
Don't use audio_format.reverse_endian.
|
|
This format is unused since the DSDIFF decoder plugin now reverses the
bit order.
|
|
|
|
Make sure we get a UTF-8 encoded string.
|
|
Basic support for Direct Stream Digital. No conversion yet, and no
decoder/output plugin support.
|
|
Support for conversion from float to 16, 24 and 32 bit integer
samples.
|
|
Pass audio_output objects around instead of void pointers. This will
give some more control to the plugin, and prepares for non-blocking
audio outputs.
|
|
Move the "extern" declarations from output_list.c, for more type
safety.
|
|
|
|
See code comment.
|
|
According to the Solaris dsp manpage, AFMT_S24_PACKED is
little-endian:
http://download.oracle.com/docs/cd/E19963-01/821-1475/6nmf5baot/index.html
The Minix soundcard.h header says the same.
|
|
Remove trailing whitespace found by this command:
find -name '*.[ch]' | xargs grep "[[:space:]]$"
|
|
|
|
MPD doesn't have child processes anymore, and thus we're not expecting
to receive SIGCHLD very often. Since hard disk access isn't
interrupted by signals anyway, we don't need those excessive checks.
|
|
Added #ifdefs on the macros AFMT_S24_NE and other OSS4 extensions.
|
|
Removed the "supported" / "unsupported" arrays, and auto-detect on
every open() call, just like the ALSA output plugin.
|
|
Removed the macro AFMT_S16_MPD.
|
|
|
|
This patch prepares support for floating point samples (and probably
other formats). It changes the meaning of the "bits" attribute from a
bit count to a symbolic value.
|
|
After we've been hit by Large File Support problems several times in
the past week (which only occur on 32 bit platforms, which I don't
have), this is yet another attempt to fix the issue.
|
|
Add a "mode" argument to open_cloexec() instead.
|
|
Added the "fd_util" library, which attempts to use the new thread-safe
Linux system calls pipe2(), accept4() and the options O_CLOEXEC,
SOCK_CLOEXEC. Without these, it falls back to FD_CLOEXEC, which is
not thread safe.
This is particularly important for the "pipe" output plugin (and
others, such as JACK/PulseAudio), because we were heavily leaking file
descriptors to child processes.
|
|
|
|
The mixer core library is now responsible for creating and managing
the mixer object. This removes duplicated code from the output
plugins.
|
|
This patch allows the output plugins to import only mixer_list.h,
instead of the full mixer_api.h (which would expose internal
structures).
|
|
mixer_control.h should provide the functions needed to manipulate a
mixer, without exposing the internal mixer API (which is provided by
mixer_api.h).
|
|
This updates the copyright header to all be the same, which is
pretty much an update of where to mail request for a copy of the GPL
and the years of the MPD project. This also puts all committers under
'The Music Player Project' umbrella. These entries should go
individually in the AUTHORS file, for consistancy.
|
|
The MPD core logs the audio format of all audio outputs. Remove the
duplicate message from the plugins.
|
|
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.
|
|
Eliminate one label and a bunch of gotos.
|
|
Use C instead of CPP.
|
|
Return type of oss_find_supported_param(), oss_can_convert() and
oss_find_unsupported_param() should be bool instead of int.
|
|
Convert the num_supported and num_unsupported variables from signed to
unsigned.
|
|
Renamed types, functions and variables.
|
|
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.
|
|
The meaning of the chunk depends on the audio format; don't suggest a
specific format by declaring the pointer as "char*", pass "void*"
instead.
|
|
The old API required an output plugin to not return until all data
passed to the play() method is consumed. Some output plugins have to
loop to fulfill that requirement, and may block during that. Simplify
these, by letting them consume only part of the buffer: make play()
return the length of the consumed data.
|
|
The output plugin shouldn't know any specifics of the mixer API. Make
it return the mixer object, and let the caller deal with it.
|
|
On some platforms, g_free() must be used for memory allocated by
GLib. This patch intends to correct a lot of occurrences, but is
probably not complete.
|
|
Allocate the mixer object when it is configured.
Merged mixer_configure() into mixer_new(). mixer_new() was quite
useless anyway.
|
|
Don't use statically allocated mixer objects.
|
|
Return the default value in the conf_get_block_*() functions when
param==NULL was passed.
This simplifies a lot of code, because all initialization can be done
in one code path, regardless whether configuration is present.
|
|
All config_get_block_*() functions should accept constant config_param
pointers.
|
|
This replaces lots of getBlockParam() invocations.
|
|
Renamed functions, types, variables.
|
|
It returns bool, not int.
|
|
This patch tryes to introduce pluggable mixer (struct mixer_plugin) along with some basic infrastructure (mixer_* functions). Instance of mixer (struct mixer) is used in
alsa and oss output plugin
|
|
"#ifdef G_BYTE_ORDER == G_BIG_ENDIAN" cannot work, of course.
|
|
|
|
|
