| Commit message (Collapse) | Author | Age | Files | Lines |
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Don't modify the configured string.
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Determine the audio output name with audio_output_get_name() instead
of obtaining the name from the configuration again.
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No "force" parameter, pass a default value instead.
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Renamed functions, types, variables.
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It returns bool, not int.
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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.
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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
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All what's left in pcm_utils.h is the pcm_range() utility function,
which is only used internally by pcm_volume and pcm_mix.
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"#ifdef G_BYTE_ORDER == G_BIG_ENDIAN" cannot work, of course.
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g_strsplit() is more portable than strtok_r().
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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.
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Use GLib's G_GNUC_UNUSED instead of gcc.h's mpd_unused.
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Use GLib's G_GNUC_UNUSED instead of macros from gcc.h.
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least MIN(od->bufferSize, size) free bytes in the buffer. Thus MIN(od->bufferSize - od->len, size) is always equal to MIN(od->bufferSize, size).
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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.
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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.
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One my_usleep() invocation remains, until we find out if we can delete
it.
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Cast AudioBuffer.mData to a "unsigned char*" before adding "curpos".
This fixes a gcc warning.
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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.
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"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.
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It's easier to work with an int16_t* pointer here.
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If the shout plugin is disabled, shout_plugin.c isn't compiled at all,
no need to check the macro definition.
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When ao_open_live() failed, MPD would ignore the error code in
"errno". Make it print a meaningful error message.
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The function audioOutputAo_error() did not implement all possible
libao error codes. Support the rest of them, and fall back to
strerror().
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The output thread automatically waits some time before retrying to
open the device. Don't duplicate this check in the pulse plugin.
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The pulse plugin crashed with a segmentation fault when the pulse
server was killed.
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Fix prototypes and unused variables.
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The OS X output plugin wasn't adapted to the new output plugin yet,
because I had no Mac to test...
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When using autodetection, AlsaData.device wasn't properly initialized
with NULL. This broke autodetection randomly.
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