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8e331cfc65
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456 Commits

Author SHA1 Message Date
Max Kellermann
8e331cfc65 gcc.h: re-add gcc_const and gcc_pure 
Remove GLib dependency from some headers.
 2012-08-02 19:12:18 +02:00
Max Kellermann
edf811fa02 DatabaseVisitor, ...: use GError forward declaration 2012-08-02 19:12:06 +02:00
Max Kellermann
5dfc0918c3 require GLib 2.16 
GLib 2.16 was released more than 4 years ago.  Let's remove some cruft
from the glib_compat.h header, and avoid new cruft to it.
 2012-07-10 01:28:52 +02:00
Max Kellermann
457d98c860 output/raop: delete the RAOP plugin 
This plugin is horrible code, I mean it.  Last year, I tried hard to
fix it, but I figured would take less time to do a full rewrite.
Given that I don't even have any device that supports RAOP, I can't do
that properly.  After 16 months, nobody volunteered for fixing it.
Hereby, I delete it, because having no RAOP plugin is better than
having this mess.  Sorry.
 2012-05-29 23:15:41 +02:00
Max Kellermann
89377556cd output/alsa: multiply writei() result with out_frame_size 
.. and not in_frame_size, because this relates to the frame size being
sent to ALSA.  pcm_export_source_size() will then turn it back into
the in_frame_size scale.
 2012-04-23 21:54:09 +02:00
Max Kellermann
df1c5ce314 pcm_export: add _frame_size() 
Move code from the ALSA output plugin.
 2012-04-23 21:54:09 +02:00
Max Kellermann
a0e4b6e266 output/alsa: fix out_frame_size formula, multiply with channels 
The hard-coded "3 bytes" was wrong because it ignored the number of
channels.
 2012-04-23 21:54:08 +02:00
Max Kellermann
c22cbbf828 Merge branch 'v0.16.x' 
Conflicts:
	src/output/osx_plugin.c
	src/text_input_stream.c
 2012-04-05 00:45:39 +02:00
Max Kellermann
5acee73fc8 encoder/vorbis: generate end-of-stream packet when playback ends 
Add the encoder_plugin method end().  This is important for the
recorder plugin.
 2012-04-05 00:21:53 +02:00
Max Kellermann
a9edf85a69 output/jack: check for connection failure before starting playback 2012-04-04 21:40:56 +02:00
Max Kellermann
e7a1862517 output/jack: workaround for libjack1 crash bug 2012-04-04 21:38:29 +02:00
Max Kellermann
8ff0197a43 output/osx: use the fifo_buffer library instead of rolling own 
The existing buffer implementation has a major flaw: it is unable to
re-fill the buffer until it has been consumed completely, leading to
many occasions where the render callback needs to generate silence,
just because the play() implementation was unable to append more
data.  The fifo_buffer library handles that well.
 2012-03-28 21:51:17 +02:00
Max Kellermann
a2b5db0003 audio_format: remove SAMPLE_FORMAT_DSD_OVER_USB 
DSD-over-USB should not be a MPD core format, because it is not a
"natural" format; it is just a temnporary over-the-wire format.  This
format has been implemented in pcm_export, and does not need to be
supported by pcm_convert.
 2012-03-27 01:22:50 +02:00
Max Kellermann
2803ec2e96 output/alsa: support 32 bit DSD-over-USB 2012-03-27 01:22:50 +02:00
Max Kellermann
ddd4f675a2 pcm_export: implement 24 to 32 bit conversion 
For 32 bit DSD-over-USB support.
 2012-03-27 01:22:50 +02:00
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