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More code simplification. Probe all formats, no matter which input
format.
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Remove the debug log messages, because they are duplicate (see
ao_open() in output_thread.c).
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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.
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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.
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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().
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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).
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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.
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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.
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Call snd_config_update_free_global() manually in our finish() method,
don't use atexit().
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The mixer core library is now responsible for creating and managing
the mixer object. This removes duplicated code from the output
plugins.
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This patch allows the output plugins to import only mixer_list.h,
instead of the full mixer_api.h (which would expose internal
structures).
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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).
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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.
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snd_pcm_writei() returns the type snd_pcm_sframes_t, not int. Use the
correct variable type.
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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.
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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().
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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.
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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
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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.
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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.
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The MPD core logs the audio format of all audio outputs. Remove the
duplicate message from the plugins.
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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.
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When the sample format is unknown, fall back to 16 bit samples.
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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.
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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.
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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.
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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.
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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.
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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.
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Allocate the mixer object when it is configured.
Merged mixer_configure() into mixer_new(). mixer_new() was quite
useless anyway.
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Don't use statically allocated mixer objects.
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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.
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All config_get_block_*() functions should accept constant config_param
pointers.
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Document alsa_data members.
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frame_size is a memory size and should be a size_t, not a signed integer.
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Renamed types, functions, variables.
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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>
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Eliminate some more getBlockParam() invocations.
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This replaces lots of getBlockParam() invocations.
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No "force" parameter, pass a default value instead.
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Renamed functions, types, variables.
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