| Commit message (Collapse) | Author | Files | Lines |
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Use audio_format_frame_size() instead of
channels*audio_format_sample_size().
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Pass the music chunk as a "const void *" to the encoder, instead of a
"const char *". Actually, both encoders currently expect 16 bit
samples, passing a 8-bit character is rather pointless.
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For simplification, moved the PCM conversion code to
pcm16_to_ogg_buffer(). Work with a int16_t pointer instead of a char
pointer.
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writeSize is a memory size and its type should thus be size_t. This
allows us to remove two explicit casts.
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Nobody needs these debug messages anymore.
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Always assume the buffer is empty before calling the encoder. Always
flush the buffer immediately after there has been added something.
This reduces the risk of buffer overruns, because there will never be
a "rest" in the current buffer.
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Don't duplicate the tag received by the send_metadata() method - send
it to the shout server directly.
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Removed the manual timer synchronization from the shout plugin.
libshout's shout_sync() function does it for us.
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The non-blocking mode of libshout is sparsely documented, and MPD's
implementation had several bugs. Also removed connect throttling
code, that is done by the MPD core since 0.14.
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When shout_data.tag!=NULL, there is a "tag to send". The tag_to_send
flag is redundant.
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That variable is set in handle_shout_error(), but is never read.
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The shout_mp3 encoder had two bugs: when no song was ever played, MPD
segfaulted during cleanup. Second bug: memory leak, each time the
shout device was opened, lame_init() was called again, and
lame_close() is only called once during shutdown.
Fix this by shutting down LAME each time the clear_encoder() method is
called.
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Make valgrind a little bit happier: free the global lame_data struct
in the finish() method.
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Move the "while" loop which checks for commands to the caller
ao_pause(). This simplifies the pause() method, and lets us remove
audio_output_is_pending().
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The function is only used by the MVP output plugin, and this one call
is wrong.
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If no ports are configured, don't overwrite the (NULL) configuration
with the port names of the first JACK server. If the server changes
after a JACK reconnect, MPD won't attempt to auto-detect again.
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Currently, the JACK plugin manipulates the audio_format struct which
was passed to the open() method. This is very likely to break,
because the plugin must not permanently store this pointer. After
this patch, MPD ignores sample rate changes. It looks like other
software is doing the same, and I guess this is a non-issue.
This patch converts the audio_format pointer within jack_data into a
static audio_format struct.
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jack_set_info_function() is not provided by older libjack versions.
Attempt to detect if it is available.
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Use jack_set_info_function() to install an info callback. Don't let
libjack print them to stderr.
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Return false from mpd_jack_play(), let the MPD core close the device.
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Don't leave uninitialized bytes in the jack_data struct.
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When MPD stops playback, close the JACK client connection.
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The "bps" attribute is calculated, but never used.
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Return true/false instead of 1/-1.
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Preparation for supporting other channel numbers than stereo: use
loops instead of duplicating code for the second channel. Most
likely, gcc will unroll these loops, so the binary won't be any
different.
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When jack_get_ports() returns NULL, we cannot have any ports to
connect to, and the device cannot play anything.
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libjack's jack_port_name() function returns the effective port name,
we don't need to do it manually.
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Do the global libjack initialization in the global plugin
initialization function.
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When waiting for free space in the ring buffer, the JACK plugin
sleeped 10ms until there is enough space. This delay was too large
for low-latency setups (<10ms), and created a lot of xruns. Work
around that by reducing the sleep time to 1ms.
A proper solution for this would be to use an event based approach,
and we will do it, just not now.
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When the connection failed once, you had to restart MPD, because it
never cleared the jack_data.shutdown flag. Instead, it refused to
play anything "because there is no client thread" (which is wrong at
that point).
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If the ring buffers are allocated after jack_activate(),
mpd_jack_process() might segfault because it attempts to access them.
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Call jack_port_register() before jack_activate().
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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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The null plugin synchronizes the playback so it will happen in real
time. This patch adds a configuration option which disables this: the
playback will then be as fast as possible. This can be useful to
profile MPD.
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Free memory in the finish() method to make valgrind happy.
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Renamed functions and variables.
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Eliminate some more getBlockParam() invocations.
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This replaces lots of getBlockParam() invocations.
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Don't modify the configured string.
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Determine the audio output name with audio_output_get_name() instead
of obtaining the name from the configuration again.
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No "force" parameter, pass a default value instead.
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Renamed functions, types, variables.
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