3915 Commits

Author SHA1 Message Date
Max Kellermann
554a34fb95 alsa: better period_time default value for high sample rates
The default values for buffer_time and period_time were both capped by
the hardware limits on practically all chips.  The result was a
period_time which was half as big as the buffer_time.  On some chips,
this led to lots of underruns when using a high sample rate (192 kHz),
because MPD had very little time to send new samples to ALSA.

A period time which is one fourth of the buffer time turned out to be
much better.  If no period_time is configured, see how much
buffer_time the hardware accepts, and try to configure one fourth of
it as period_time, instead of hard-coding the default period_time
value.

This is yet another attempt to provide a solution which is valid for
all sound chips.  Using the SND_PCM_NONBLOCK flag also seemed to solve
the underruns, but put a lot more CPU load to MPD.
2009-03-08 03:55:01 +01:00
Max Kellermann
27193d8402 output_all: fix boolean short circuit in update()
Sometimes, audio_output_update() isn't called for the second device
when the first one has succeeded.  The patch
"audio_output_all_update() returns bool" broke it, because the boolean
evaluation ended after the first "true".
2009-03-07 23:48:28 +01:00
Max Kellermann
fc6d836a2d player_thread: moved code to player_check_decoder_startup() 2009-03-07 23:11:43 +01:00
Max Kellermann
bd6bcfb676 music_pipe: refuse to push empty chunks
Added two assertions.
2009-03-07 21:41:25 +01:00
Max Kellermann
85cc46ad6f decoder_internal: don't push empty chunk into pipe
When the decoder chunk is empty in decoder_flush_chunk(), don't push
it into the music pipe - return it to the music buffer instead.  An
empty chunk in the pipe wastes resources for no advantage.
2009-03-07 21:41:23 +01:00
Max Kellermann
eb2e3a554d chunk: added music_chunk_is_empty() 2009-03-07 21:40:27 +01:00
Max Kellermann
f8aebc52b5 music_pipe: poison music_chunk.next
The value of music_chunk.next is undefined for a chunk returned by
music_pipe_shift().  For more pedantic debugging, poison the reference
before returning the chunk.
2009-03-07 21:40:13 +01:00
Max Kellermann
39d3521956 music_pipe: added music_pipe_peek()
music_pipe_peek() is similar to music_pipe_shift(), but doesn't remove
the chunk.  This allows it to be used with a "const" music_pipe.
2009-03-07 19:56:31 +01:00
Max Kellermann
b13cd03f75 output_all: audio_output_all_update() returns bool
audio_output_all_update() returns true when there is at least open
output device which is open.
2009-03-07 19:55:57 +01:00
David Guibert
498ec26f25 pulse_mixer: allow mpd to reconnect to the pulse mixer
This patch follows the commit 21bb10f4b.

>From Max Kellermann:
> I removed the daemonization changes in main.c.  Please explain why you
> changed that.  If you need it for some reason, make that a separate
> patch with a good description of your rationale.

> That's the biggest flaw of your code: it opens the mixer device in the
> init() method, while the open() method is empty.  When the pulse
> daemon is not available (either during MPD startup or when it dies
> while MPD runs), the plugin will not even attempt to reconnect to
> pulse.  Please move the code to the open() method, to make that work.

I changed the daemonize call as the fork losts the connection to the
pulse server. According to your remark, the init() method should be
moved to the open() ones.

With the modification, mpd is able to reconnect the pulse mixer after
restarting the pulseaudio daemon.

Signed-off-by: David Guibert <david.guibert@gmail.com>
Signed-off-by: Max Kellermann <max@duempel.org>
2009-03-07 19:55:09 +01:00
Max Kellermann
5ffb2dd88c pulse_mixer: added missing copyright header 2009-03-07 15:59:29 +01:00
Max Kellermann
b1137fe81a pulse_mixer: added GLib log domain
Shorten some log messages, let GLib add the "pulse_mixer" prefix.
2009-03-07 15:59:26 +01:00
Max Kellermann
6069cafda0 pulse: clean up includes
Don't include output_api.h - this is not an output plugin.  Added
missing explicit conf.h and string.h includes.
2009-03-07 15:59:22 +01:00
David Guibert
21bb10f4bf pulse mixer
This patch introduces the mixer for the pulse output.

Technically speaking, the pulse index is needed to get or set
the volume. You must define callback fonctions to get this index since
the pulse output in mpd is done using the simpe api. The pulse simple api
does not provide the index of the newly defined output.

So callback fonctions are associated to the pulse context.
The list of all the sink input is then retreived.
Then we select the name of the mpd pulse output and control
its volume by its associated index number.

Signed-off-by: Patrice Linel <patnathanael@gmail.com>
Signed-off-by: David Guibert <david.guibert@gmail.com>

[mk: fixed whitespace errors and broke long lines; removed
daemonization changes from main.c]
2009-03-07 15:59:20 +01:00
Max Kellermann
a547d24eb2 mixer: check for init() failures
When the init() method of a mixer plugin fails, mixer_new()
dereferences the NULL pointer.
2009-03-07 15:50:26 +01:00
Max Kellermann
5e0acec118 curl: reverse GLIB_CHECK_VERSION()
The GLIB_CHECK_VERSION() macro was used improperly, which broke build
on GLib < 2.14.  Add a "!" for negation.
2009-03-06 15:42:33 +01:00
Max Kellermann
4c3ce9ef1c socket_util: check if IN6_IS_ADDR_V4MAPPED is defined
On some systems, the macro IN6_IS_ADDR_V4MAPPED() is not available.
Don't try to convert IPv6 to their IPV4 equivalents in this case.
2009-03-06 10:09:10 +01:00
Max Kellermann
01cf7feac7 pipe: added music_buffer, rewrite music_pipe
Turn the music_pipe into a simple music_chunk queue.  The music_chunk
allocation code is moved to music_buffer, and is now managed with a
linked list instead of a ring buffer.  Two separate music_pipe objects
are used by the decoder for the "current" and the "next" song, which
greatly simplifies the cross-fading code.
2009-03-06 00:42:03 +01:00
Max Kellermann
000b2d4f3a music_pipe: added music_pipe_push()
Added music_pipe_allocate(), music_pipe_push() and
music_pipe_cancel().  Those functions allow the caller (decoder thread
in this case) to do its own chunk management.  The functions
music_pipe_flush() and music_pipe_tag() can now be removed.
2009-03-06 00:42:01 +01:00
Max Kellermann
10be8a8714 playlist_control: fix requeue after seek
The queue update after a seek was wrong: the queued song is cleared by
a successful seek.  This caused queue/cross-fading problems after a
seek.
2009-03-06 00:41:59 +01:00
Max Kellermann
b0fcce65d8 flac: explicitly check for STOP command
After the decoder command was obtained, don't wait until libflac
detects EOF (as a side effect), quit the decoder immediately.  This
check was missing completely.
2009-03-05 18:20:43 +01:00
Max Kellermann
efd606337e flac: check command after flac_process_single() failure
When the MPD core sends the decoder a command while
flac_process_single() is executed, this function fails.  Abort the
decoder only if not seeking.  This fixes a seeking bug.
2009-03-05 18:20:41 +01:00
Max Kellermann
74a2813d78 music_chunk: added music_chunk_write(), music_chunk_expand()
Moved some code from music_pipe_write() and music_pipe_expand().  Only
music_chunk.c should access the music_chunk internals.
2009-03-05 17:37:11 +01:00
Max Kellermann
c655f804a9 music_pipe: moved struct music_chunk to chunk.h 2009-03-03 22:23:25 +01:00
Max Kellermann
1063c1f2e3 alsa: log period and buffer size
Log the real period and buffer size.  This might be useful when
debugging xruns.  Note that the same information is available in
/proc/asound/card*/pcm*p/sub*/hw_params
2009-03-03 22:19:37 +01:00
Avuton Olrich
3e5a445467 ls: Print output of supported uri to fp rather than stdout.
Since there are no other callers than stdout, this wouldn't be a
problem, but since there maybe in the future go ahead and fix it.
2009-03-03 13:12:39 -08:00
Viliam Mateicka
3b76ca7186 ffmpeg: fix version comparision for av_get_bits_per_sample_format() implemetation
function was implemented in the version we are comparing to so there must be higher or equal
2009-03-03 21:30:55 +01:00
Viliam Mateicka
c89482de65 ffmpeg: support for new metadata api 2009-03-03 21:30:46 +01:00
Avuton Olrich
e7f034dcef cmdline: Print available protocols when --version is run. 2009-03-03 21:25:19 +01:00
Max Kellermann
0f64e658fd alsa: fall back to 32 bit samples if 16 is not supported
There are a few high-end devices (e.g. ICE1724) which cannot even play
16 bit audio.  Try the 32 bit fallback, which we already implemented
for 24 bit.
2009-03-03 09:38:20 +01:00
Eric Wollesen
b8ebb748c9 Add sticker list command.
[mk: merged memory leak patch; fixed indentation (tabs); fixed
documentation typo]
2009-03-03 07:49:23 +01:00
Max Kellermann
4220e6b0ad input_lastfm: new input plugin for last.fm radio
The lastfm input plugin enables MPD to play lastfm:// URLs.  This
plugin is not complete yet: it plays only the first song in the
last.fm playlist, and the playlist parser isn't even implemented
properly.
2009-03-02 23:11:31 +01:00
Max Kellermann
cfb350f4f0 input: pass config_param to input_plugin.init()
Allow input plugins to configure with an "input" block in mpd.conf.
Also allow the user to disable a plugin completely.
2009-03-02 23:08:17 +01:00
Max Kellermann
9a350acf04 input_plugin: added methods init(), finish()
Instead of hard-coding the plugin global initialization in
input_stream_global_init(), make it walk the plugin list and
initialize all plugins.
2009-03-02 20:45:50 +01:00
Max Kellermann
36d24fb7ea input: moved plugins to ./src/input/
Create a sub directory for input plugins.
2009-03-02 20:40:31 +01:00
Max Kellermann
2e51365ea4 input_stream: moved struct input_plugin to input_plugin.h
Start to separate private from public input_stream API.
2009-03-02 20:13:08 +01:00
Viliam Mateicka
8694574f63 ffmpeg: use ffmpeg's sampleformat for output format 2009-03-02 20:12:36 +01:00
Viliam Mateicka
60a5b5562b fixing unused parameter warning 2009-03-02 19:00:21 +01:00
Viliam Mateicka
57d836da49 fixing unsigned to signed comparision
[mk: cast off_t to uint32_t; same fix for aiff.c]
2009-03-02 18:59:59 +01:00
Viliam Mateicka
406b0403a5 mixer: adding code to optionally disable all hw mixers 2009-03-02 18:57:49 +01:00
Max Kellermann
2f438e5d23 tag_id3: parse ID3 tags in AIFF files
Added a small AIFF parser library, code copied from the RIFF parser
(big-endian integers).  Look for an "ID3" chunk, and let libid3tag
parse it.
2009-03-02 18:12:44 +01:00
Max Kellermann
336f624277 tag_id3: parse ID3 tags in RIFF/WAV files
Added a small RIFF parser library.  Look for an "id3" chunk, and let
libid3tag parse it.
2009-03-02 18:00:46 +01:00
Max Kellermann
72176db429 alsa: fall back to 32 bit samples if 24 is not supported
Some sound chips/drivers (e.g. Intel HDA) don't support 24 bit
samples, they want to get 32 bit instead.  Now that MPD's PCM library
supports 32 bit, add a 32 bit fallback when 24 bit is not supported.
2009-03-02 16:41:38 +01:00
Max Kellermann
a5a15beac2 pcm_convert: added 32 bit support
All PCM sub libraries have 32 bit support now.  Add support to the
glue function pcm_convert().
2009-03-02 16:41:10 +01:00
Max Kellermann
3165e26f9a pcm_format: added conversion from 32 bit
Support converting 32 bit samples to any other supported sample
format.
2009-03-02 16:41:08 +01:00
Max Kellermann
d4e4c57b8d pcm_format: added pcm_convert_to_32()
Added code to convert all other sample formats to 32 bit.
2009-03-02 16:39:54 +01:00
Max Kellermann
d24f2ba5ee pcm_dither: added pcm_dither_32_to_16()
For 32 bit dithering, reuse the 24 bit dithering code, but apply a 8
bit right shift first.
2009-03-02 16:37:11 +01:00
Max Kellermann
78e08f655a pcm_dither: renamed struct pcm_dither_24 to struct pcm_dither
There is nothing 24 bit specific in the pcm_dither_24 struct.  Since
we want to reuse the struct for 32 bit dithering, let's drop the "_24"
suffix from the struct name.
2009-03-02 16:37:05 +01:00
Max Kellermann
d9c1434298 pcm_resample: use 24 bit resampling code for 32 bit samples
Resampling 32 bit samples is the same as resampling 24 bit samples -
both are stored in the int32_t type.
2009-03-02 16:37:00 +01:00
Max Kellermann
1b31f52285 pcm_channels: added implementation for 32 bit samples
Some 24 bit code can be reused.  The 32 bit variant has to use 64 bit
integers, because 32 bit integers could overflow.  This may be a
performance hit on 32 bit CPUs.
2009-03-02 16:36:49 +01:00