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2009-03-26output_plugin: replaced output_plugin.get_mixer() with mixer_pluginMax Kellermann1-19/+1
The mixer core library is now responsible for creating and managing the mixer object. This removes duplicated code from the output plugins.
2009-03-14mixer_api: moved mixer_plugin imports to mixer_list.hMax Kellermann1-1/+1
This patch allows the output plugins to import only mixer_list.h, instead of the full mixer_api.h (which would expose internal structures).
2009-03-14mixer_api: moved functions to mixer_control.cMax Kellermann1-0/+1
mixer_control.h should provide the functions needed to manipulate a mixer, without exposing the internal mixer API (which is provided by mixer_api.h).
2009-03-13all: Update copyright header.Avuton Olrich1-6/+7
This updates the copyright header to all be the same, which is pretty much an update of where to mail request for a copy of the GPL and the years of the MPD project. This also puts all committers under 'The Music Player Project' umbrella. These entries should go individually in the AUTHORS file, for consistancy.
2009-03-10alsa: use snd_pcm_sframes_t instead of intMax Kellermann1-2/+1
snd_pcm_writei() returns the type snd_pcm_sframes_t, not int. Use the correct variable type.
2009-03-10alsa: don't close PCM handle in alsa_recover()Max Kellermann1-13/+3
If the PCM handle gets disconnected, don't close and clear it in alsa_recover(). The MPD core will call alsa_close() anyway. This way, we can always assume that alsa_data.pcm is always valid.
2009-03-08alsa: determine buffer_time if not already knownMax Kellermann1-0/+5
This patch fixes a theoretical (but practically impossible) flaw: the variable "buffer_time" may be uninitialized when it is used. Initialize the variable with snd_pcm_hw_params_get_buffer_time().
2009-03-08alsa: better period_time default value for high sample ratesMax Kellermann1-3/+8
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.
2009-03-03alsa: log period and buffer sizeMax Kellermann1-0/+3
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
2009-03-03alsa: fall back to 32 bit samples if 16 is not supportedMax Kellermann1-1/+2
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.
2009-03-02alsa: fall back to 32 bit samples if 24 is not supportedMax Kellermann1-0/+8
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.
2009-03-01output: removed duplicate debug messages from pluginsMax Kellermann1-4/+0
The MPD core logs the audio format of all audio outputs. Remove the duplicate message from the plugins.
2009-02-26output_plugin: report errors with GErrorMax Kellermann1-20/+35
Use GLib's GError library for reporting output device failures. Note that some init() methods don't clean up properly after a failure, but that's ok for now, because the MPD core will abort anyway.
2009-02-25alsa: fall back to 16 bit audioMax Kellermann1-3/+8
When the sample format is unknown, fall back to 16 bit samples.
2009-02-25alsa: moved code from alsa_open() to alsa_setup()Max Kellermann1-32/+49
Simplify error handling a bit by moving some code into a separate function. This eliminates a good bunch of gotos, but that's not finished yet.
2009-02-25output_plugin: don't pass audio_output object to method init()Max Kellermann1-2/+1
audio_output_get_name() has been removed, which was the only function left in output_api.h. The output plugin doesn't need the audio_output object at all, remove the parameter from the init() method.
2009-02-23output: pass the music chunk pointer as void*, not char*Max Kellermann1-1/+1
The meaning of the chunk depends on the audio format; don't suggest a specific format by declaring the pointer as "char*", pass "void*" instead.
2009-02-23output_api: play() returns a lengthMax Kellermann1-20/+11
The old API required an output plugin to not return until all data passed to the play() method is consumed. Some output plugins have to loop to fulfill that requirement, and may block during that. Simplify these, by letting them consume only part of the buffer: make play() return the length of the consumed data.
2009-02-16output_plugin: replaced method "control()" with "mixer()"Max Kellermann1-8/+9
The output plugin shouldn't know any specifics of the mixer API. Make it return the mixer object, and let the caller deal with it.
2009-01-25use g_free() instead of free()Max Kellermann1-1/+1
On some platforms, g_free() must be used for memory allocated by GLib. This patch intends to correct a lot of occurrences, but is probably not complete.
2009-01-25mixer: removed mixer_configure(), configure mixer in mixer_new()Max Kellermann1-7/+1
Allocate the mixer object when it is configured. Merged mixer_configure() into mixer_new(). mixer_new() was quite useless anyway.
2009-01-25mixer: return a mixer struct pointerMax Kellermann1-7/+7
Don't use statically allocated mixer objects.
2009-01-25conf: allow param==NULLMax Kellermann1-5/+2
Return the default value in the conf_get_block_*() functions when param==NULL was passed. This simplifies a lot of code, because all initialization can be done in one code path, regardless whether configuration is present.
2009-01-25conf: const pointers in block get functionsMax Kellermann1-2/+2
All config_get_block_*() functions should accept constant config_param pointers.
2009-01-25alsa: added commentsMax Kellermann1-3/+21
Document alsa_data members.
2009-01-25alsa: frame_size is size_t, not intMax Kellermann1-1/+1
frame_size is a memory size and should be a size_t, not a signed integer.
2009-01-25alsa: no CamelCaseMax Kellermann1-93/+103
Renamed types, functions, variables.
2009-01-25alsa: fix option parsing and restore default period_timeDan McGee1-3/+6
Two bugs here led to a large number of interrupts being generated on the sound card when ALSA output is being used. Because we specify no default period_time, the sound card gives us 3000 interrupts/sec rather than a more sane 20 or 30. This completes the revert of dd7711 already started by 4ca24f. The larger bug was in the change to config_get_block_unsigned() and using 0 as the default value for both 'buffer_time' and 'period_time'. This means any pre-setting of these options in newAlsaData() gets wiped out. Add a new default for period_time, and ensure default values for buffer_time and period_time are used if none are provided by the user. Signed-off-by: Dan McGee <dan@archlinux.org> [mk: set defaults in newAlsaData() to fix auto-configuration; renamed "_MS" back to "_US" because ALSA expects microseconds, not milliseconds] Signed-off-by: Max Kellermann <max@duempel.org>
2009-01-18conf: added config_get_block_unsigned()Max Kellermann1-6/+2
Eliminate some more getBlockParam() invocations.
2009-01-18conf: added config_get_block_string()Max Kellermann1-2/+1
This replaces lots of getBlockParam() invocations.
2009-01-17conf: replaced getBoolBlockParam() with config_get_block_bool()Max Kellermann1-6/+5
No "force" parameter, pass a default value instead.
2009-01-17conf: no CamelCase, part IMax Kellermann1-5/+6
Renamed functions, types, variables.
2009-01-10Introducing mixer apiViliam Mateicka1-7/+8
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
2009-01-07pcm_utils: moved conversion code to pcm_convert.cMax Kellermann1-1/+1
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.
2009-01-04renaming mixer.h to mixer_api.hViliam Mateicka1-1/+1
2009-01-02alsa: use GLib instead of utils.hMax Kellermann1-3/+2
2009-01-01output plugins: don't include gcc.hMax Kellermann1-2/+2
Use GLib's G_GNUC_UNUSED instead of gcc.h's mpd_unused.
2008-12-31Moving mixers to audio outputsViliam Mateicka1-1/+21
2008-12-29alsa: use GLib loggingMax Kellermann1-29/+30
2008-12-01alsa: reverted the default buffer_time to 500 msMax Kellermann1-1/+5
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.
2008-11-03alsa: initialize "device" with NULLMax Kellermann1-2/+1
When using autodetection, AlsaData.device wasn't properly initialized with NULL. This broke autodetection randomly.
2008-11-01alsa, jack: no const pointers for allocated stringsMax Kellermann1-17/+25
Make the pointers "device" and "name" non-const, so we don't need the xfree() hack. The default value is expressed as NULL.
2008-10-29output: close device on play errorMax Kellermann1-1/+0
When an output plugin fails to play a chunk, close it. This replaces various manual close() calls in nearly all plugins.
2008-10-29output: use bool for return values and flagsMax Kellermann1-9/+10
Don't return 0/-1 on success/error, but true/false. Instead of int, use bool for storing flags.
2008-10-26output: don't compile plugins which are disabledMax Kellermann1-11/+3
Don't compile the sources of disabled output plugins at all.
2008-10-26renamed src/audioOutputs/ to src/output/Max Kellermann1-0/+0
Again, no CamelCase in the directory name.
2008-10-14alsa: added #ifdefs around SND_PCM_NO_AUTO_xxxMax Kellermann1-0/+6
These macros are not available in older libasound versions (1.0.13 fails, 1.0.16 is ok). Ignore the configuration if the constants are not defined.
2008-10-14alsa: optionally disable resampling and othersMax Kellermann1-1/+15
Added mpd.conf options for disabling automatic resamling, sample format and channel conversion. This way, users may choose to override ALSA's automatic resampling, and use libsamplerate instead.
2008-10-12alsa: fall back to 16 bit outputMax Kellermann1-0/+11
If the sample format isn't supported by the device (i.e. 24 bit on low-end sound chips), fall back to 16 bit output. There is code in pcm_utils.c which converts PCM data to 16 bit.
2008-10-12alsa: moved code to alsa_configure()Max Kellermann1-13/+18
Move code which loads configuration to alsa_configure(). This removes one indent level.
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