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Adds mixrampdb and mixrampdelay commands. Reads MIXRAP_START and
MIXRAMP_END tags from FLAC files and overlaps instead of crossfading.
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Add an option for each audio output which enables the use of the
hardware mixer, instead of the software volume code.
This is hardware specific, and assumes linear volume control. This is
not the case for hardware mixers which were tested, making this patch
somewhat useless, but we will use it to experiment with the settings,
to find a good solution.
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Apply the replay gain in the output thread. This means a new setting
will be active instantly, without going through the whole music pipe.
And we might have different replay gain settings for each audio output
device.
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Use input_stream.uri.
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Log to stderr instead of the default stdout. We need a pristine
stdout for the conversion result.
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Ensure that the pcm_convert() length argument is aligned to the sample
size.
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This function replaces the replay_gain_info parameter for
decoder_data(). This allows the decoder to announce replay gain
changes, instead of having to pass the same object over and over.
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Major API redesign: don't let the caller allocate the input_stream
object. Let each input plugin allocate its own (derived/extended)
input_stream pointer. The "data" attribute can now be removed, and
all input plugins simply cast the input_stream pointer to their own
structure (with an "struct input_stream base" as the first attribute).
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This is like tag_dup(), but works with an input_stream object instead
of a file path.
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Added attributes start_ms, end_ms. This allows us to address a
portion of a song file (important for CUE support). There is no
support yet for storing these attributes in the state file.
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Remove the data_time parameter from decoder_data(). This patch
eliminates the timestamp counting in most decoder plugins, because the
MPD core will do it automatically by default.
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Use the plugin instead of the glue code in normalize.c. This is used
wrapped inside a "autoconv" filter, to enable normalization for all
input file formats.
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This enables valgrind debugging after an error occurred.
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Enable archive plugin debugging.
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Not used by any plugin currently, but this eliminates the g_error()
call in input_plugin_config(), so it's worth it.
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Log to stderr, not to stdout (which broke PCM output).
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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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Assign default value only if none was given on the command line.
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Unified function for converting an audio_format object to a string,
for log messages and for the "status" command.
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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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Always keep the audio_output object locked within the output thread,
unless a plugin method is called. This fixes several race conditions.
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Don't let the mixer plugin "override" the libpulse callbacks.
Instead, add a "mixer" attribute to the pulse_output struct, and call
the mixer on all interesting events.
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An asterisk means that this attribute should not be enforced, and
stays whatever it used to be. This way, some configuration values
work like masks.
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This is a complete rewrite of the PulseAudio output plugin. It uses
the asynchronous API, which gives us more control over everything.
Additionally, it connects to the PulseAudio server on startup, and
keeps this connection up while MPD runs. During pause, instead of
closing the stream, it enables "cork".
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This allows the mixer object to access its associated audio output
object.
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Based on this API, we will add parsers for EXTM3U, PLS, ASX, last.fm
radio and others.
There is no integration into the MPD core yet. Right now, we have a
command line test program. This is work in progress.
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Don't call g_error(), which will abort the process and dump core.
This patch does not affect all the config_get_X() functions. These
need some more refactoring.
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Don't abort the whole MPD process when the conversion fails. This has
been a denial-of-service attack vector for years.
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It makes no difference right now, but we're about to add an endianness
flag and will want to make sure it's correctly initialised every time.
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This mixer plugin may be used instead of the traditional global
software mixer. It integrates with the "volume" filter plugin, and
can control the software volume of an audio output which has no
hardware mixer.
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This patch adds initial filter support for audio outputs. Each audio
output gets a "filter" attribute, which is used by ao_play_chunk().
The PCM conversion is now performed by convert_filter_plugin.
audio_output.convert_state has been removed.
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Added public methods to get and set the current volume.
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This program runs filter plugins in an isolated environment.
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Renamed all remaining CamelCase functions.
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