This patch introduces the mixer for the pulse output.
Technically speaking, the pulse index is needed to get or set
the volume. You must define callback fonctions to get this index since
the pulse output in mpd is done using the simpe api. The pulse simple api
does not provide the index of the newly defined output.
So callback fonctions are associated to the pulse context.
The list of all the sink input is then retreived.
Then we select the name of the mpd pulse output and control
its volume by its associated index number.
Signed-off-by: Patrice Linel <patnathanael@gmail.com>
Signed-off-by: David Guibert <david.guibert@gmail.com>
[mk: fixed whitespace errors and broke long lines; removed
daemonization changes from main.c]
Turn the music_pipe into a simple music_chunk queue. The music_chunk
allocation code is moved to music_buffer, and is now managed with a
linked list instead of a ring buffer. Two separate music_pipe objects
are used by the decoder for the "current" and the "next" song, which
greatly simplifies the cross-fading code.
Added music_pipe_allocate(), music_pipe_push() and
music_pipe_cancel(). Those functions allow the caller (decoder thread
in this case) to do its own chunk management. The functions
music_pipe_flush() and music_pipe_tag() can now be removed.
After the decoder command was obtained, don't wait until libflac
detects EOF (as a side effect), quit the decoder immediately. This
check was missing completely.
When the MPD core sends the decoder a command while
flac_process_single() is executed, this function fails. Abort the
decoder only if not seeking. This fixes a seeking bug.
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
The lastfm input plugin enables MPD to play lastfm:// URLs. This
plugin is not complete yet: it plays only the first song in the
last.fm playlist, and the playlist parser isn't even implemented
properly.
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.
There is nothing 24 bit specific in the pcm_dither_24 struct. Since
we want to reuse the struct for 32 bit dithering, let's drop the "_24"
suffix from the struct name.
Some 24 bit code can be reused. The 32 bit variant has to use 64 bit
integers, because 32 bit integers could overflow. This may be a
performance hit on 32 bit CPUs.
This is the first patch in a series to enable 32 bit audio samples in
MPD. 32 bit samples are more tricky than 24 bit samples, because the
integer may overflow when you operate on a sample.
audio_valid_sample_format() verifies the number of channels. Let's
just say up to 8 channels is allowed (which is possible with some
consumer sound chips). I don't know if there are bigger cards, and
since I cannot test it, I'll limit it to 8 for now.
On some hardware, reading the mixer value from hardware is an
expensive operation, and MPD has to do it for every client. Throttle
access to the hardware, cache the result for one second.
time() is not a monotonic timer, and MPD might get confused by clock
skews. clock_gettime() provides a monotonic clock, but is not
portable to non-POSIX systems (i.e. Windows). This patch uses GLib's
GTimer API, which aims to be portable.
If an input_stream is not seekable, libaudiofile fails to play at all:
Audio File Library: unrecognized audio file format [error 0]
Since we know in advance whether the input_stream is seekable, just
refuse to play on a non-seekable stream.
The generic sockaddr struct is too small for some addresses. For
accept(), we have to allocate a sockaddr_storage struct on the stack,
which is large enough for all addresses.
Create the socket_util.c library, the first function is
sockaddr_to_string(): it converts a sockaddr struct to a string
containing the IP address in a human-readable form.
When checking whether database entries have been deleted, don't check
if an archive file is a directory (G_FILE_TEST_IS_DIR), use
G_FILE_TEST_IS_REGULAR for this case instead. To determine if a
"struct directory" is an archive, check for device==DEVICE_INARCHIVE.
This is always false after loading the database, so this patch is not
complete yet.
Remember the modification time of each directory. This is important
for archives (which are virtual directories right now), but may also
be useful for an automatic update mechanism.
Added the uri_remove_auth() library function which strips username
and password from a HTTP URI, and use it in song_print_url(). This
allows you to add HTTP URIs to the playlist including secret username
and password, without disclosing it to all MPD clients.
Since we introduced a GLib logging domain, the "client" string appears
twice in the log lines:
client: client 0: command returned 0
Removed the second one, now it looks like this:
client: [0] command returned 0
Still not quite good, but better than before.
MPD used to be silent when it could stat() a directory, but could not
opendir() it to read its contents. This caused a lot of support
headache with users who have wrong file permissions. Add another
warning message.
There's no point in declaring num_items as a uint8_t, it doesn't save
any space, due to padding. This allows us to lift the articial "255
items" limitation.
The warning message "problems opening audio device while playing ..."
does not help at all, and should be removed. At this point, the real
error message has already been logged by the output thread.
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.
Don't call AudioOutputUnitStart() in the play() method, do it after
the device has been opened. We can eliminate the "started" property
now, because the device is always started when it's open.
ao_play() gets PCM data in the in_audio_format, and converts it to
out_audio_format. Comparing the input data with out_audio_format is
wrong.
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