using ov_test_callback with function CALLBACKS_STREAMONLY will cause
scanning to stop after the comment field. ov_open (and ov_test)
default to CALLBACKS_DEFAULT which scans the file structure causing a
huge slowdown. The speed improvement is huge: It scanned my files
around 10x faster This procedure has been recommended by monthy (main
vorbis developer) and was said to be safe for scanning files.
The recorder plugin writes audio played by MPD to a file. This may be
useful for recording radio streams.
This implementation is incomplete, because support for tags is
missing, and MPD should be able to record each track to a different
file.
MPD checks if every flac (possibly other types as well) file contains
cuesheet on every update, which produces unneeded I/O. My music
collection is on NFS share, so it's quite noticeable. IMHO, it
shouldn't re-read unchanged files, so I wrote simple patch to fix it.
Explicitly make the output thread leave the ao_pause() loop. This
patch is a workaround, and the "pause" flag is not managed in a
thread-safe way, but that's good enough for now.
The function flac_cue_track() first calls FLAC__metadata_object_new(),
then overwrites this pointer with FLAC__metadata_get_cuesheet(). This
allocate two FLAC__StreamMetadata objects, but the first pointer is
lost, and never freed.
Replace decoder_control.notify with decoder_control.mutex and
decoder_control.cond. Lock the mutex on all accesses to
decoder_control.command and decoder_control.state.
For systems that cannot support fork() (like no-mmu Linux), use daemon() if
it is available for the daemonizing code.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Splitted tag_id3_import_frame() into two specialized functions:
tag_id3_import_text() and tag_id3_import_comment(). Use
id3_frame_field() instead of directly accessing id3_frame.fields.
Added a patch to flush out the last.fm input plugin slightly. It
basically turns it into a wrapper for the appropriate plugin. Most
notably metadata is now extracted.
Instead of hard-coding the path "/etc/mpd.conf", use the configured
$(sysconfdir) path. This can be set with:
./configure --sysconfdir=/etc
Note that this changes the default path to "/usr/local/etc/mpd.conf",
given the default prefix "/usr/local". This is actually more correct
than the old default.
Calculate the total play time with the audio_format object each time,
using audio_format_time_to_size(). The function
audioFormatSizeToTime() is not needed anymore, and will be removed
with this patch.
Currently, byteswapping is performed on the format_buffer. This can
go wrong when this buffer is used twice during one run. Add a
separate buffer for swapping the byte order.
Changed function to first close standard input (this may
fail but we don't care) and then try to open /dev/null (this
may fail but it shouldn't on Unix platforms plus we don't
know what to do in such case anyways). Since standard input
has the "zeroth" descriptor number next "open" will use it.
Since there is no "/dev/null" on Windows (It's not even
a valid path!) the second step is skipped if WIN32 is
defined.
As a final touch, since the function consists of merely two
function calls it has been moved to header file and declared
static inline.
[mk: un-inline daemonize_close_stdin()]
When libid3tag is disabled, the libmad decoder plugin is unable to
identify ID3 frames. If the file starts with an (unidentified) ID3
frame, it assumes that the file is not a valid MP3 song. This patch
solves this by adding minimal stubs for the ID3 functions.
The function tag_ape_load() retrieves a 32 bit unsigned integer from
the input file, and passes it to g_malloc(). This is dangerous, and
may be used for a denial of service attack on MPD.
On Windows only daemonize_close_stdin() function does
something. Other functions are either empty or generate an
error. Those have been moved to header file and declared
static inline so compiler can remove the call all together.
The functions were not "const" (they examined values pointed
by arguments passed to them, quoting gcc's doc: "Note that
a function that has pointer arguments and examines the data
pointed to must _not_ be declared 'const'.") but rather
"pure" and still not all of them.
Note also, that even some of the functions declared "pure"
are not pure, however, due to reasons stated in source code
the attribute has been kept.
The "group" configuration option is similar to "user" as it
sets user set what group MPD shall run as. With "user"
option, MPD changed GID to the GID of the user, however,
more control could be desired.
Moreover, the patch changes the way of checking whether no
setuid(2)/setgid(2) is required -- previously user names
were compered, now UID and GIDs are compered (ie. the one we
already have (getuid(2)/getgid(2)) with the one we want to
change to).
The expression "tagLen - size > 0" may result in an integer underflow
and a buffer overflow, when "size" is larger than "tagLen". "size" is
read from the input file, and must not be trusted. This patch changes
the expression to "tagLen > size", which is a lot safer.
The glue_*() functions act as a glue between MPD's main() function and
its libraries. They handle disabled features, and pass validated
configuration options.
Since version 0.14, MPD has been logging to standard error instead of
standard output. The option name should reflect that. The old option
continues to work, we will remove it in a future MPD release.
This encoder plugin is a replacement for the LAME encoder plugin for
those who prefer a "free" (non-patent encumbered) encoder library.
Most of the plugin source code is copied from the LAME encoder plugin,
since the LAME and TwoLAME APIs are nearly the same.
According to the ID3 2.4 documentation, "TOPE" is "Original
artist/performer", not "performer". Removed "TOPE" support. Instead,
map TPE3 ("Conductor/performer refinement") and TPE4 ("Interpreted,
remixed, or otherwise modified by") to "performer".
The tag_id3.c library supports both the documented "TSO2" tag, and the
inofficial TXXX/ALBUMARTISTSORT.
The Vorbis/FLAC decoder automatically supports the new tag, without
further change.
Do all the software volume stuff inside each output thread, not in the
player thread. This allows one software mixer per output device, and
also allows the user to configure the mixer type (hardware or
software) for each audio output.
This moves the global "mixer_type" setting into the "audio_output"
section, deprecating the "mixer_enabled" flag.
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.
Converted the range checks in volume_level_change() to assertions.
Changed all volume types to "unsigned", expect for those which must be
able to indicate error (-1).
When song->mtime was not initialized properly, it was revealed that
strftime() might crash when gmtime_r() returns NULL due to an invalid
time_t input value.
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.
A recent change to the boolean parser introduced a bug: instead of
using the block_param's value with get_bool(), we passed param->value
(which is always NULL in this case).
Some clients have visual feedback for "database update is running".
Using the "database" idle event is unreliable, because it is only
emitted when the database was actually modified. This patch adds the
"update" event, which is emitted when the update is started, and again
when the update is finished, disregarding whether it has been
modified.
The "volume" filter plugin will replace the current software volume
code. One "volume" filter may be attached to each output device.
This will allow the user to use hardware mixers for some devices, and
software mixers for other devices at the same time.
Currently, neither the filter API nor the "volume" plugin is
integrated into MPD.
When the filesystem_charset is changed in mpd.conf, MPD should discard
the old database. In this error branch, MPD did not fill the GError
object properly, and logged a warning message instead, which caused a
segmentation fault.
This patch fixes an assertion failure:
Assertion `order < queue->length' failed.
This happens when the state file is saved, when there is no "current"
song: current==-1, and queue_order_to_position(-1) is called.
When MPD was paused, and the client sent the "stop" command (or
"clear"), a glitch caused MPD to continue playback for a split second.
This was because audio_output_all_cancel() calls
audio_output_all_update(), which reopens all output devices, and
re-ignites the playback loop.
At the moment mpd doesn't store or restore the current track to/from
its state file when the daemon is stopped/started while in 'stopped'
state. I believe the preferred behaviour would be to store and
restore the current track even when the daemon is in stopped state
when shutting down.
I made a small patch to adapt this behaviour. If you believe this is
not the preferred behaviour, maybe this should be realized as a
configuration option. I'm not sure how to do this, but made a small
comment, where one would have to put the option.
Instead of returning an artificial three-state integer, return a
"success" value and put the boolean value into a "bool" pointer.
That's a little bit more overhead, but an API which looks more
natural.
When decoding a local file, the decoder thread tries to run all
matching decoders, until one succeeds. Both file_decode() and
stream_decode() can decode a stream, but MPD closes the stream before
calling file_decode(). Problem is: when this decoder fails, and the
next's stream_decode() method is invoked, the input_stream is still
closed. This patch reopens it.
Several users had problems with binding MPD to "localhost". The cause
was duplicate /etc/hosts entries: the resolver library returns
127.0.0.1 twice, and of course, MPD attempts to bind to "both" of
them. This patch makes failures non-fatal, given that at least one
address was bound successfully. This is a workaround; users should
rather fix their /etc/hosts file.
When client_defer_output() aborts the connection to the client,
client_write_output() called client_write_deferred() anyway. This
caused an assertion failure. Fix it by checking for the "expired"
flag again after client_defer_output() returns.
When the decoder is finished, break out of the player loop only after
another player.pipe check. We did check the pipe size a few lines
above, but that check was kind of racy.
When a music_chunk only contains a tag but no PCM data, play_chunk()
returns true without freeing the chunk. The caller now assumes that
the chunk is moved into some music_pipe and does not bother to free it
either.
To check for leaked music_chunk objects, free the music buffer on
CLOSE_AUDIO. This invokes an assertion check which ensures that all
chunks have been returned to the buffer.
Instead of returning the local variable "ret" which is always true at
this point, hard-code the "true" return value, because that might be
more readable.
If a file is removed the library, next time mpd will try to play it it
will result in an error 'ERROR: problems decoding some/file.ogg'.
Nothing is written in log files (verbose mode or not)
[mk: append strerror(errno)]
Commit f78cddb4 introduced a regression: when the playlist reached its
end, MPD did not reset the "current song" pointer anymore after stop.
Add a "current = -1" code line.
The only pc_seek() caller clears the error, rendering the check
useless. Even if the previous PLAY command resulted in a player
error, this check is not very useful.
Flush the encoder before calling encoder_tag(). The first page
generated by the encoder after sending the tag will be the new
"header" page, which is sent to all HTTP clients when they connect.
This is a little bit specific to the vorbis encoder, but there are no
other encoders which support tags (yet).
When a new tag is set, end the current stream and begin a new one.
Use vorbis_analysis_headerout() to write a full ogg header. This
fixes a problem with icecast: after a song change in MPD, icecast
stops forwarding ogg packets to its clients.
When a song was in the database twice (which shouldn't happen), and
the first song had no tag items, MPD calledd tag_free(NULL). Add a
check to that source location, and an assertion to tag_free().
libvorbis goes into a very long loop if we try to add data after a
flush was invoked by vorbis_analysis_wrote(0). This seems to be a
problem with the internal end-of-stream marker. Thus, we cannot reuse
the vorbis_dsp_state object.
When the decoder thread has a pending command, send the STOP command
to cancel this command. Send STOP again if the decoder thread is
still running after that, just in case the decoder thread has executed
the previous command (which was overwritten).
Using two different kinds of locks may result in a race condition with
a deadlock. The libpulse callbacks need no locks at all, because the
mainloop object can be assumed to be already locked.
When all audio outputs have been closed due to failures, pause the
playback instead of stopping it. This way, the user may resume
at the current position after the problem has been dealt with.
When no audio outputs could be opened while seeking, leave MPD seeked
at that position and pause playback. The user may continue from this
point at any time, as soon as the audio outputs are fixed. The old
behaviour triggered an assertion failure: the failure wasn't passed
properly to the do_play() function, which attempted to play audio
chunks.
snd_config_update_free_global() frees cached ALSA configuration. This
keeps valgrind a little bit more quiet. This patch moves the call
from the open() method into the finish() method, which seems more
natural: it allows the use of the config cache, and improves the
cleanup phase.
[mk: folded with patch "Put icy related functions in extra source
files"; moved icy_server.c from HAVE_CURL to ENABLE_HTTPD_OUTPUT;
removed an unused variable]
When a new song starts playing, send its tag (song->tag) to the music
pipe. This allows output plugins to render tags for all songs, not
only those with embedded tags understood by the decoder plugin.
db_get_song was being called once for all sub-handlers, but with the
addition of the find command, we don't have a URI coming in, so doing
db_get_song once won't work anymore.
[mk: fixed initialization order]
On both locations, the result of write() can be ignored safely. In
event_pipe_emit_fast(), that can only be "EAGAIN", which means that
the pipe buffer is full - no further notification required. In
client_init(), that would be a fatal connection error, which would be
caught by the next event.
This patch fixes gcc warnings.
Cuesheets are often saved as vorbis comment
flac files (CUESHEET=.. case doesn't matter).
We can parse this now and use the information to
tag the subtracks (from the embedded cuesheets).
With these methods a tag struct can be created
from the cdtext information in a cue sheet.
The methods depend on a cue parsing library.
Reading from strings (char*) as well as from
a file (FILE*) is supported.
This is a little ugly, but as nextSongInPlaylist is both called when
queued is update (in case playlist ended) and for user "next" command,
there isn't any other (simple) solution
Previous cast to float didn't have any effect because one value is uint
and the other is a floating type but the number itself is even..
This caused some tracks to end before they were really at an end.
The strings passed to tag_pool_get_item() are not null-terminated, and
the caller passes the string length. Don't assume it is
null-terminated anyway by using strcmp().
There are numerous race conditions between the libpulse thread
(pulse_mixer.c callbacks) and the rest of MPD. Protect the volatile
attributes of the pulse_mixer struct with a mutex to fix that.
Don't mess with pulse_mixer.volume for setting the volume. This
variable should only be used to transfer the current volume from
sink_input_vol() to pulse_mixer_get_volume().
The pa_context_get_sink_input_info() function is asynchronous, and
after it returns, libpulse does not guarantee that the operation has
completed yet; in fact, it is not likely. Explicitly wait for the
operation to complete.
The code for the new pulse_wait_for_operation() function was inspired
by mplayer and xine code.
The function mixer_failed() expects the mixer mutex to be already
locked, yet it calls mixer_close(), which attempts to lock the mutex
again, deadlocking itself.
The smartstop feature is a way to tell mpd to stop playing after
current song.
This patche provides:
- 'state' command returns 'smartstop' state (1 or 0)
- 'smartstop' can activate or not the smartstop state
- when song is terminated, mpd stops playing and smartstop is set to 0
This patch fixes a longer delay when moving around songs in the
playlist. The main thread wants to enqueue a new "next" song into the
player thread, but the player thread is waiting inside
audio_output_all_wait() for the output threads. Use
player_control.notify there, so audio_output_all_wait() gets woken up
by the main thread, too.
The functions playPlaylist() and seekSongInPlaylist() expect a song
position, not a song order number. Don't convert the "current"
variable with queue_position_to_order().
The move command now accepts a range for the first argument, in the same
form as other range commands, e.g. move 15:17 3. The first song in the
range is placed at the destination position. Note that as with other
range commands, the range is inclusive on the left only; this example
would move only songs 15 and 16, not 17.
[mk: fixed signed/unsigned warnings; use G_MAXUINT instead of
UINT_MAX]
The pm->volume attribute was allocated in pulse_mixer_init(), but is
never freed. This leaks memory. Instead of adding the g_free() call
to pulse_mixer_finish(), let's just make "volume" a static attribute
of the pulse_mixer struct. That is easier to deal with.
Nobody needs to modify these strings. We can make them const, and
convert config_dup_block_string() to config_get_block_string(). This
also fixes memory leaks in the pulse mixer.
Remember if a mixer object is open or closed. Don't call open() again
if it is already open. This guarantees that the mixer plugin is
always called in a consistent state, and we will be able to remove
lots of checks from the implementations.
To support mixers which are automatically opened even if the audio
output is still closed (to set the volume before playback starts),
this patch also adds the "global" flag to the mixer_plugin struct.
Both ALSA and OSS set this flag, while PULSE does not.
As a side effect, the previous patch added the mixer==NULL checks. It
is now illegal to call mixer functions with a NULL argument. Convert
the runtime checks to assertions.
When the decoder initialization has not been completed yet, all calls
to dc_seek() will fail, because dc.seekable is not initialized yet.
Wait for the decoder to complete its initialization, i.e. until it has
called decoder_initialized().
Don't start playback as soon as the "current" song is being loaded
from the state file. That is unclean, and leads to an obscure bug: in
repeat mode, when the song is started (which is yet the last song in
the list), the playlist code marked the very first song in the
playlist as "next" song, because the end of the playlist was wrapped.
It's easier to set up the playback after all songs have been loaded,
and after the random/repeat mode has been set.
Use audio_output_client_notify instead of g_usleep(1ms) in
audio_output_all_wait() to synchronize with the output_thread. Signal
the audio_output_client_notify object in ao_play().
There was a deadlock between the output thread and the player thread:
when the output thread failed (and closed itself) while the player
thread worked with the audio_output object, MPD could crash.
The config_audio_format used to contain the configured audio format,
which is copied to out_audio_format. Let's convert the former to a
boolean, which indicates whether out_audio_format was already set.
This simplifies some code and saves a few bytes.
On 2009/03/17 Max Kellermann<max@duempel.org> wrote:
> There doesn't seem to be an "official" standard. I'd say: search for
> TITLE[1] first (the most explicit form), then TITLE1, and finally fall
> back to TITLE. This makes sure MPD supports every possible standard,
> without breaking.
I've also added some additional checks to make sure entry is long
enough.
when the mixer is closed,
- the mainloop is stopped.
- the context is disconnected.
- then the mainloop is freed.
Signed-off-by: David Guibert <david.guibert@gmail.com>
The cue sheet embedded in a flac file doen't contain any information
about track titles and similar. There are three possibilities: Use an
external cue sheet that includes these information, use a tag CUESHEET
with a cue sheet including these information or use tags. I think the
latter is the best option and is already used by other projects.
Let's get rid of the "shout" plugin, and the awfully complicated
icecast daemon setup! MPD can do better if it's doing the HTTP server
stuff on its own. This new plugin has several advantages:
- easier to set up - only one daemon, no password settings, no mount
settings
- MPD controls the encoder and thus already knows the packet
boundaries - icecast has to parse them
- MPD doesn't bother to encode data while nobody is listening
This implementation is very experimental (no header parsing, ignores
request URI, no icy-metadata, ...). It should be able to suport
several encoders in parallel in the future (with different bit rates,
different codec, ...), to make MPD the perfect streaming server. Once
MPD gets multi-player support, we can even mount several different
radio stations on one server.
Converted the ogg_page attribute from the vorbis_encoder struct to a
local function of vorbis_encoder_read(). This simplifies some code,
because we don't need to check the page anymore before using it.
Add the "flush" flag, and defer the ogg_stream_flush() call. Call
ogg_stream_pageout() or ogg_stream_flush() (depending on the "flush"
flag) in vorbis_encoder_read(). This prevents the ogg_page from
getting overwritten by consecutive ogg_stream_pageout() calls.
Even if libsamplerate support is enabled, compile the fallback
resampler. When the user specifies the option
"samplerate_converter=internal", it is chosen in favor of
libsamplerate. This may help users with a weak FPU who don't want to
compile a custom MPD from source, because the fallback resampler does
not use floating point operations.
Added diversion functions to pcm_resample.c. These check which
resampler is enabled at compile time (libsamplerate or fallback).
This prepares the following patch.
In some rare cases, there was a race condition between the output
thread and the main thread: when you disable/enable an output device
in the main thread, this caused a crash in the output thread. Protect
the whole mixer struct with a GMutex to prevent that.
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.
When the destination chunk was empty in cross_fade_apply(), it had no
audio_format attached (an attribute which is only used for assertion
in the debug build). cross_fade_apply() should assign it the
audio_format of the second chunk (if available), otherwise MPD will
crash.
When there are chunks which are not yet finished,
audio_output_all_check() returned the size of its music pipe minus
one. I can't remember exactly why I subtracted 1 from the return
value, it must have had something to do with a former meaning of this
function. Now it induces assertion failures.
After adding the container_scan() method the update_regular_file() method was quite hard to read.
Now there's update_container_file() which deals with container files.
That way normal container files (i.e. without embedded tracks) are handled by the old code like a regular file.
This will fix some of the odd behaviour observed.
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.
After a seek, wait until enough new chunks are decoded before starting
playback. If this takes too long, send silence chunks to the audio
outputs meanwhile.
When the audio outputs are closed, also clear the audio format. If we
don't do this, every call to audio_output_all_update() will open the
device, even if it's meant to be paused.
When playback is unpaused, pass the audio_format to
audio_output_all_open(). Don't assume that output_all.c remembers the
previous audio format. Also check if there has been an audio format
yet.
Check audio_output.command after each sub-chunk has been played. It
discards the rest of the chunk, but since all commands make the device
stop anyway, this is not a problem, but part of the improvement. This
improves the latency of audio output commands.
A larger chunk size means less overhead for managing them. 4 kB seems
to be a reasonable choice: it contains 23 ms of 44.1 kHz 16 bit stereo
data, or 3 ms of 192 kHz 24 bit stereo data. The original value of
1020 seemed to be too small, there were quite a lot of system calls
and context switches.
Instead of passing individual buffers to audio_output_all_play(), pass
music_chunk objects. Append all those chunks asynchronously to a
music_pipe instance. All output threads may then read chunks from
this pipe. This reduces MPD's internal latency by an order of
magnitude.
When a PAUSE command is received while the decoder starts, don't open
the audio device when the decoder becomes ready. It's pointless,
because MPD will close if after that.
If the header valgrind/memcheck.h is available, add
VALGRIND_MAKE_MEM_NOACCESS() and VALGRIND_MAKE_MEM_UNDEFINED()
support, which enables nice warnings in the valgrind memory checker.
This is similar to the MPD 0.14 patch "wait 10 seconds before
reopening a failed device", which only covered open() failures. This
patch adds the same feature for play().
Until now every flac file got removed unconditionally (and then re-added)
whenever the update command was issued. Now there is a check if we need
to that, so the file will only be removed if there is a embedded cuesheet
in that file
So far only seekpoints are supported, so no proper tagging yet
except for track number and track length.
Tagging should be done by parsing the cue sheet which
is often embedded as vorbis comment in flac files.
Furthermore the pathname should be configurable like "%A - %t - %T",
where %A means Artist, %t track number and %T Title or so.
In !NDEBUG, remember which audio_format is stored in every chunk and
every pipe. Check the audio_format of every new data block appended
to the music_chunk, and the format of every new chunk appended to the
music_pipe.
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().
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.
Sometimes, audio_output_update() isn't called for the second device
when the first one has succeeded. The patch
"audio_output_all_update() returns bool" broke it, because the boolean
evaluation ended after the first "true".
When the decoder chunk is empty in decoder_flush_chunk(), don't push
it into the music pipe - return it to the music buffer instead. An
empty chunk in the pipe wastes resources for no advantage.
The value of music_chunk.next is undefined for a chunk returned by
music_pipe_shift(). For more pedantic debugging, poison the reference
before returning the chunk.
This patch follows the commit 21bb10f4b.
>From Max Kellermann:
> I removed the daemonization changes in main.c. Please explain why you
> changed that. If you need it for some reason, make that a separate
> patch with a good description of your rationale.
> That's the biggest flaw of your code: it opens the mixer device in the
> init() method, while the open() method is empty. When the pulse
> daemon is not available (either during MPD startup or when it dies
> while MPD runs), the plugin will not even attempt to reconnect to
> pulse. Please move the code to the open() method, to make that work.
I changed the daemonize call as the fork losts the connection to the
pulse server. According to your remark, the init() method should be
moved to the open() ones.
With the modification, mpd is able to reconnect the pulse mixer after
restarting the pulseaudio daemon.
Signed-off-by: David Guibert <david.guibert@gmail.com>
Signed-off-by: Max Kellermann <max@duempel.org>
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.
