oysteikt/mpd
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1441 Commits

Author SHA1 Message Date
Max Kellermann
d271dd2cce output/alsa: use pcm_export for the DSD-over-USB conversion 2012-03-27 01:22:50 +02:00
Max Kellermann
ebfdd37451 pcm_export: support DSD to DSD-over-USB conversion 
Prepare for removing SAMPLE_FORMAT_DSD_OVER_USB.
 2012-03-27 01:17:11 +02:00
Max Kellermann
f6d6110aaa output/alsa: move pcm_export_open() to caller 
Give the caller more control, prepare for DSD-over-USB improvements.
 2012-03-27 01:17:11 +02:00
Max Kellermann
2f0674807c output/alsa: more debug output 2012-03-27 00:14:37 +02:00
Max Kellermann
167242fec0 output/alsa: add option to enable DSD over USB 2012-03-22 01:30:16 +01:00
Max Kellermann
725fbe946b output/alsa: split the frame_size attribute 
Make it in_frame_size and out_frame_size, to account for packing.
 2012-03-22 01:01:11 +01:00
Max Kellermann
b99ecb4dc9 audio_format: remove the packed S24 format 
For simplicity, the MPD core should not have to deal with packing.  It
is rarely used, and those plugins that need it should use the
pcm_export library instead.
 2012-03-22 00:42:12 +01:00
Max Kellermann
208a96b211 output/alsa: use pcm_export to pack 24 bit samples 2012-03-22 00:33:59 +01:00
Max Kellermann
43774455cc output/oss: use pcm_export to pack 24 bit samples 2012-03-22 00:33:40 +01:00
Max Kellermann
5b61e077e0 pcm_export: add option "pack" 
Converts padded 24 bit samples to packed 24 bit samples.  Will replace
the packed S24 sample format, which is not used internally.
 2012-03-22 00:33:40 +01:00
Max Kellermann
921cc3e5db output/oss: remember the real OSS format 
Improving oss_reopen() by using the very same value that was used
initially.
 2012-03-22 00:33:40 +01:00
Max Kellermann
51dce3d161 output/alsa: simplify setup_format() 2012-03-22 00:29:56 +01:00
Max Kellermann
c10f013fc2 output/alsa: don't pass audio_format to _try_format() 
Let the caller configure the audio_format object.
 2012-03-22 00:24:56 +01:00
Max Kellermann
2874d68bdb output/alsa: simplify alsa_output_try_format_both() 
Merge three functions into one and call get_bitformat() only once.
 2012-03-22 00:23:07 +01:00
Max Kellermann
11e83eb7e7 output/oss: move code to oss_probe_sample_format() 2012-03-21 22:28:18 +01:00
Max Kellermann
170635e3a6 output/{alsa,oss}: move endian code to new library pcm_export 2012-03-21 21:16:50 +01:00
Max Kellermann
8c5ebdff36 audio_format: remove the reverse_endian attribute 
Eliminate support for reverse endian samples from the MPD core.  This
moves a lot of complexity to the plugins that really need it (only
ALSA and CDIO currently).
 2012-03-21 19:31:04 +01:00
Max Kellermann
1c84f324a1 output/oss: always receive host byte order samples 
Don't use audio_format.reverse_endian.
 2012-03-21 19:25:59 +01:00
Max Kellermann
3dba09f339 output/alsa: always receive host byte order samples 
Don't use audio_format.reverse_endian.
 2012-03-21 19:14:05 +01:00
Max Kellermann
766905ba9f output/alsa: merge alsa_data_free() into destructor 2012-03-21 18:58:11 +01:00
Rich Healey
396e97fc94 Fix the build on OSX 2012-03-21 10:32:54 +01:00
Max Kellermann
9c6a4505c8 audio_format: remove the format SAMPLE_FORMAT_DSD_LSBFIRST 
This format is unused since the DSDIFF decoder plugin now reverses the
bit order.
 2012-03-21 09:06:48 +01:00
Max Kellermann
08ce24ec3f audio_format: basic support for DSD-over-USB 2012-03-19 23:49:29 +01:00
Max Kellermann
5016839b90 use g_strerror() instead of strerror() 
Make sure we get a UTF-8 encoded string.
 2012-03-06 22:06:08 +01:00
Kurt Van Dijck
c551c8b31b raop_output: fix raop_session inbalance 
raop_session_free must be called from raop_output_finish,
not from raop_output_remove.
In raop_output_remove, do close the ntp_server & control port.

Signed-off-by: Kurt Van Dijck <kurt.van.dijck@skynet.be>
 2012-03-01 09:51:37 +01:00
Max Kellermann
2516496993 audio_format: add DSD sample format 
Basic support for Direct Stream Digital.  No conversion yet, and no
decoder/output plugin support.
 2012-03-01 01:15:22 +01:00
Max Kellermann
c616165f81 Merge branch 'v0.16.x' 
Conflicts:
	NEWS
	configure.ac
	src/decoder/ffmpeg_decoder_plugin.c
	test/read_tags.c
 2012-02-13 19:15:18 +01:00
Max Kellermann
9c92afa5fe output/winmm: remove pointless NULL check 
pcm_buffer_get() cannot ever return NULL.
 2012-02-13 18:10:36 +01:00
Max Kellermann
95b2df8261 output/osx: fix memory leak after AudioUnitSetProperty() failure 2012-01-04 20:47:41 +01:00
Max Kellermann
1ebadf8620 output/osx: implement 32 bit playback 2011-12-24 18:18:42 +01:00
Max Kellermann
9179d9592d output/osx: allocate the device in enable() 
Keep the device open as long as the output is enabled, but initialize
it only when playback starts.
 2011-12-24 18:15:24 +01:00
Max Kellermann
43c5058682 Merge branch 'v0.16.x' 2011-12-24 18:15:17 +01:00
Max Kellermann
97b4a6b51f output/osx: clear render buffer when there's not enough data 
When we don't have enough data, generate some silence, hoping the
input buffer will fill soon.  Reducing the render buffer size is not
legal.
 2011-12-24 17:59:36 +01:00
Max Kellermann
f405d27c56 output/osx: remove sleep call from render callback 
Blocking inside the render callback is forbidden, and this sleep call
didn't make any sense.
 2011-12-24 17:56:10 +01:00
Max Kellermann
78c4351e04 output/openal: improve synchronization 
This plugin's use of the "Timer" library was wrong; it added the same
amount of virtual data in every iteration in _play(), but did not
actually play something.  This created an artificial, but useless,
delay.  This patch implements the method _cancel(), and implements
hard-coded sleep values.  This is only slightly better, but does not
attempt to look sane.
 2011-12-13 22:02:05 +01:00
Max Kellermann
0a427890fe output/openal: move code to inline functions 2011-12-13 22:00:18 +01:00
Max Kellermann
e735abe334 output/openal: use alGetSourcei(AL_BUFFER) to force-unqueue buffers 
The implementation of cancel() did not work well: you cannot use
alSourceUnqueueBuffers() to unqueue queued buffers, and our function
openal_unqueue_buffers() left the OpenAL library in a rather undefined
state; nothing was supposed to be queued, but the "filled" variable
was not reset.
 2011-12-13 21:59:10 +01:00
Max Kellermann
c0070b2f13 output/openal: make attribute "filled" unsigned 2011-12-13 21:58:13 +01:00
Max Kellermann
cfedc6e9b4 output/openal: remove bogus format check from _open() 
The expression "!format" does not make sense, and cannot occur.
 2011-12-13 21:58:09 +01:00
Max Kellermann
b66d7f7e0b output/fifo: implement output_plugin method delay() 2011-12-13 21:58:07 +01:00
Max Kellermann
e01df06cd7 output/null: implement output_plugin method delay() 2011-12-13 21:58:02 +01:00
Max Kellermann
6584897b69 output/null: don't initialize the "timer" attribute in _init() 
Unnecessary overhead.
 2011-12-13 21:57:59 +01:00
Max Kellermann
aa4f45b9a5 Merge branch 'v0.16.x' 
Conflicts:
	NEWS
	configure.ac
 2011-12-13 21:57:44 +01:00
Max Kellermann
96ad5b8444 output/openal: force 16 bit playback, as 8 bit doesn't work 
The OpenAL specification says that AL_FORMAT_MONO8 and
AL_FORMAT_STEREO8 expect unsigned 8 bit samples, but MPD uses unsigned
samples.
 2011-12-13 21:32:19 +01:00
Denis Krjuchkov
33232face9 winmm_output_plugin: fail if wrong device specified instead of using fallback. 
Silently choosing default is misleading and can cause hours of investigation.
It's better to fail immediately telling user what is wrong with config.
 2011-12-13 19:12:33 +01:00
Max Kellermann
545685bc32 audio_format: basic support for floating point samples 
Support for conversion from float to 16, 24 and 32 bit integer
samples.
 2011-10-20 02:32:39 +02:00
Max Kellermann
8dd83a2cf3 output/roar: move code to _use_audio_format() 2011-10-08 14:51:19 +02:00
Max Kellermann
b43bf4dd74 Merge branch 'v0.16.x' 2011-10-08 14:50:44 +02:00
Max Kellermann
5ed0eb51d1 output/openal: auto-fallback to mono if channel count is unsupported 
.. instead of failing playback completely.
 2011-10-08 14:41:22 +02:00
Max Kellermann
72a1ca3b99 output/alsa: remove "default" case from switch 
Allow gcc to warn when a new format isn't supported.
 2011-10-08 14:41:11 +02:00