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.
Two bugs here led to a large number of interrupts being generated on the
sound card when ALSA output is being used. Because we specify no default
period_time, the sound card gives us 3000 interrupts/sec rather than a more
sane 20 or 30. This completes the revert of dd7711 already started by
4ca24f.
The larger bug was in the change to config_get_block_unsigned() and using 0
as the default value for both 'buffer_time' and 'period_time'. This means
any pre-setting of these options in newAlsaData() gets wiped out. Add a new
default for period_time, and ensure default values for buffer_time and
period_time are used if none are provided by the user.
Signed-off-by: Dan McGee <dan@archlinux.org>
[mk: set defaults in newAlsaData() to fix auto-configuration; renamed
"_MS" back to "_US" because ALSA expects microseconds, not milliseconds]
Signed-off-by: Max Kellermann <max@duempel.org>
The null plugin synchronizes the playback so it will happen in real
time. This patch adds a configuration option which disables this: the
playback will then be as fast as possible. This can be useful to
profile MPD.
I was having problems with shoutcast stream outputs before applying
the attached patch, which enlarges the shoutcast output
buffer. Ideally, this should be configurable, but this resolves the
issue for my needs.
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
JACK documentation states: "The caller is responsible for calling
free(3) any non-NULL returned value."
This does not seem to include the array elements. Duplicate them
after jack_get_ports(), and free only the array. Convert
JackData.output_ports to non-const.
There have been bug reports on MPD regarding 24 bit output via
libao/esd. The "ao" plugin does not attempt fall back to 16 bit
currently, and thus fails to play 24 bit audio (i.e. all mp3 files).
Make it always use 16 bit samples for now, until more bits are
well-tested.
The OS X output does not seem to support 24 bit audio in the way MPD
implements it currently. Fall back to 16 bit for now, and schedule
24 bit support on OS X for MPD 0.15.
Commit dd7711d8 removed MPD's default ALSA buffer_time. The result
was a buffer size which was way too small for playing streams on some
sound hardware, and caused skips and distorted sound. Revert the
default to 500 ms.
"float (*lamebuf)[2] = g_malloc()" does NOT allocate two float*
buffers. The formula is even wrong: it should be applied to LAME's
output buffer, not its input buffer.
Converted "lamebuf" to the two variables "left" and "right", and
allocate them independently with the exact buffer size. Set
right=left if mono output is configured.
