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* output/httpd: implement "pause"Max Kellermann2010-08-311-0/+19
| | | | | Send silence to all connected clients while paused, to avoid connection interruption.
* output/httpd: forced flush after 32 kB of input dataMax Kellermann2010-08-312-0/+23
| | | | | Avoid buffer underruns on the streaming client, if the encoder is "too efficient" (e.g. when encoding silence while paused).
* database, ...: remove EINTR checks after stdio callsMax Kellermann2010-07-251-1/+1
| | | | | | MPD doesn't have child processes anymore, and thus we're not expecting to receive SIGCHLD very often. Since hard disk access isn't interrupted by signals anyway, we don't need those excessive checks.
* output/httpd: add libwrap supportTim Phipps2010-06-061-0/+28
| | | | | | libwrap support is in MPD but only for the control port. This patch adds support for the http port. The code is copied from src/client_new.c
* output/win32: new output plugin for Windows WaveMax Kellermann2010-05-201-0/+296
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* input/file, output/{fifo,recorder}: add O_BINARY to open() flagsMax Kellermann2010-05-202-5/+6
| | | | Windows compatibility.
* output/oss: 24 bit support via OSS4Max Kellermann2010-05-141-0/+34
| | | | Added #ifdefs on the macros AFMT_S24_NE and other OSS4 extensions.
* output/oss: rewrite of the audio format detectionMax Kellermann2010-05-131-301/+324
| | | | | Removed the "supported" / "unsupported" arrays, and auto-detect on every open() call, just like the ALSA output plugin.
* output/oss: use the *_NE macrosMax Kellermann2010-05-131-8/+2
| | | | Removed the macro AFMT_S16_MPD.
* output/httpd: added name/genre/website configurationJames Pike2010-04-053-5/+25
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* Add support for building httpd_output plugin for win32Avuton Olrich2010-03-222-0/+10
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* output/jack: drain the ring buffers during pauseMax Kellermann2010-03-101-0/+7
| | | | | If we're not doing this, and a new song is played after pause ends, then you will hear the rest of the previous song.
* output/jack: synchronize all channelsMax Kellermann2010-03-101-9/+26
| | | | | Always use the same number of samples from each channel's ring buffer. This ensures that all channels are kept in sync.
* output/pulse: clear the "mainloop" attribute on errorMax Kellermann2010-01-171-2/+3
| | | | | | | | When enabling the pulse device fails, clear po->mainloop after pa_threaded_mainloop_free() has finished. This is important for the assertions. Two wrong g_free() calls were also removed.
* output/alsa: support packed 24 bit samplesMax Kellermann2010-01-171-0/+13
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* output/alsa: probe all sample formats in a loopMax Kellermann2010-01-161-36/+34
| | | | | More code simplification. Probe all formats, no matter which input format.
* output/alsa: merged code into alsa_output_try_format()Max Kellermann2010-01-161-51/+71
| | | | | Remove the debug log messages, because they are duplicate (see ao_open() in output_thread.c).
* output/alsa: pass sample_format to get_bitformat()Max Kellermann2010-01-161-3/+3
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* output/alsa: moved code to alsa_output_setup_format()Max Kellermann2010-01-161-72/+80
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* Update copyright notices.Avuton Olrich2009-12-3119-19/+19
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* httpd: use get_mime_type to determine encoder contentViliam Mateicka2009-12-031-7/+6
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* audio_format: changed "bits" to "enum sample_format"Max Kellermann2009-12-029-65/+108
| | | | | | This patch prepares support for floating point samples (and probably other formats). It changes the meaning of the "bits" attribute from a bit count to a symbolic value.
* output/openal: use audio_format_to_string()Max Kellermann2009-11-151-3/+3
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* include config.h in all sourcesMax Kellermann2009-11-1216-8/+22
| | | | | | After we've been hit by Large File Support problems several times in the past week (which only occur on 32 bit platforms, which I don't have), this is yet another attempt to fix the issue.
* configure.ac: require GLib 2.12Max Kellermann2009-11-101-0/+1
| | | | | | Drop the required GLib version from 2.16 to 2.12, because many current systems still don't have GLib 2.16. This requires several new compatibility functions in glib_compat.h.
* fd_util: removed creat_cloexec()Max Kellermann2009-11-104-7/+8
| | | | Add a "mode" argument to open_cloexec() instead.
* output/alsa: fill period buffer with silence before drainingMax Kellermann2009-11-091-3/+47
| | | | | | | | | ALSA passes full period buffers to the hardware. If an application doesn't finish writing a period, libasound will nonetheless send the partial buffer (with undefined trailing data). This causes noise at the end of playback. This patch attempts to track the current position within the period buffer, and generates silence at the end, before calling snd_pcm_drain().
* fd_util: added O_NONBLOCK functionsMax Kellermann2009-11-081-1/+2
| | | | | Changed the wrappers for pipe(), socket(), accept(). On WIN32, this does not work for pipe().
* set the close-on-exec flag on all file descriptorsMax Kellermann2009-11-076-12/+22
| | | | | | | | | | | Added the "fd_util" library, which attempts to use the new thread-safe Linux system calls pipe2(), accept4() and the options O_CLOEXEC, SOCK_CLOEXEC. Without these, it falls back to FD_CLOEXEC, which is not thread safe. This is particularly important for the "pipe" output plugin (and others, such as JACK/PulseAudio), because we were heavily leaking file descriptors to child processes.
* output/jack: added option "server_name"Max Kellermann2009-11-071-1/+8
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* output/httpd: moved code to httpd_output_bind()Max Kellermann2009-11-071-28/+48
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* output/jack: free source port names on exitMax Kellermann2009-11-071-0/+3
| | | | Make valgrind happy.
* output/jack: dynamic source port listMax Kellermann2009-11-061-37/+74
| | | | | | Same as the previous patch: create up to 16 configured source ports. The plugin tries to do its best at guessing the right combination for the given input file, the number of source and destination ports.
* output/jack: dynamic destination port listMax Kellermann2009-11-061-25/+71
| | | | | Support up to 16 configured destination ports, that should really be enough for everybody.
* output/jack: renamed option "ports" to "destination_ports"Max Kellermann2009-11-061-1/+9
| | | | Be more clear which kind of port should be configured here.
* output/jack: renamed "output ports" to "destination ports"Max Kellermann2009-11-061-18/+20
| | | | Use the same name as in the libjack API documentation.
* output/httpd: bind port when output is enabledMax Kellermann2009-11-052-13/+49
| | | | | | Implement the methods enable() and disable(). Bind the HTTP port in the enable() method, but reject all incoming connections until the output is opened.
* output/jack: support mono inputMax Kellermann2009-11-051-13/+41
| | | | | When MPD plays a mono song (audio_format.channel==1), connect only one source port to both destination ports.
* output/jack: clear ring buffers before activatingMax Kellermann2009-11-051-1/+6
| | | | | | After playback has stopped, the ring buffers may still contain samples. These will be played when playback is started the next time. We should clear the buffers each time.
* output/jack: use jack_client_open() instead of jack_client_new()Max Kellermann2009-11-051-4/+23
| | | | | | | | jack_client_new() is deprecated. This requires libjack 0.100 (released nearly 5 years ago). We havn't been testing older libjack versions anyway. As a side effect, there is the new option "autostart".
* output/jack: added option "client_name"Max Kellermann2009-11-051-3/+2
| | | | | | Instead of using MPD's audio output name (setting "name"), use a separate configuration option. Change the default to "Music Player Daemon".
* alsa_plugin.c: workaround snd_pcm_drain bugJeffrey Middleton2009-11-021-1/+2
| | | | | | | Reintroduce a fix from commit 52a0653 (Warren Dukes): "don't call snd_pcm_drain unless we're already in the RUNNING state". This prevents ALSA with dmix from sometimes hanging when snd_pcm_drain is called, e.g. when moving from one song to the next (as in mantis issue 2634).
* httpd: add config option to limit number of clientsViliam Mateicka2009-10-292-3/+20
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* output_plugin: added method "drain"Max Kellermann2009-10-291-3/+9
| | | | | | | drain() is the opposite of cancel(): it waits until all data in the buffer has finished playing. Instead of implicitly draining in the close() method like the ALSA plugin has been doing it forever, let the output thread decide whether to drain or to cancel.
* output/alsa: don't recover on CANCELMax Kellermann2009-10-291-1/+1
| | | | | | The recovery is for nothing if we get CLOSE afterwards. Let's not recover in the cancel() method, and let the next play() call sort it out.
* output/pulse: initialize pulse_output.mixerMax Kellermann2009-10-291-0/+1
| | | | This variable was uninitialized and led to crashes.
* include pulse/version.h for PA_CHECK_VERSIONAlam Arias2009-10-271-0/+2
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* output/jack: implement methods enable()/disable()Max Kellermann2009-10-231-16/+24
| | | | Don't connect to JACK before MPD has daemonized.
* output/pulse: implement methods enable()/disable()Max Kellermann2009-10-231-9/+35
| | | | Don't connect to PulseAudio before MPD has daemonized.
* output_plugin: added methods enable() and disable()Max Kellermann2009-10-231-0/+2
| | | | | | | With these methods, an output plugin can allocate some global resources only if it is actually enabled. The method enable() is called after daemonization, which allows for more sophisticated resource allocation during that method.
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