The "seeking" flag is not set for the initial seek, and so
decoder_read() could be canceled when another SEEK was emitted during
initial seek.
This fixes several seek problems, for example the one reported for the
FLAC decoder plugin:
https://bugs.musicpd.org/view.php?id=4552
.. instead of doing it after seeking. After seeking, the command had
no effect, because CheckDecoderStartup() waits for all outputs to
finish. This caused a very long delay while seeking and switching
songs (https://bugs.musicpd.org/view.php?id=4534).
Source: mpd
Version: 0.19.14-2
Severity: important
Justification: fails to build form source (but built in the past)
Tags: patch
User: debian-alpha@lists.debian.org
Usertags: alpha
mpd FTBFS on Alpha with a failure in the test suite [1]:
FAIL: test/test_byte_reverse
============================
.F...
!!!FAILURES!!!
Test Results:
Run: 4 Failures: 1 Errors: 0
1) test: ByteReverseTest::TestByteReverse2 (F) line: 58 test/test_byte_reverse.cxx
assertion failed
- Expression: strcmp(result, (const char *)dest) == 0
This occurs because the test suite (in test/test_byte_reversal.cxx)
allocates static char arrays and passes the char arrays to functions
whose respective arguments were declared to be uint16_t *, etc., in
the main code.
This is in the realm of undefined behaviour on architectures with
strict memory alignment requirements. Although the test only fails
on Alpha (because Alpha has a particular CPU load instruction that
gcc likes to use to add bugs ..., ahem, optimise the code on the
assumption of alignment) it is potentially a latent bug for other
architectures with strict alignment requirements.
Since the code is compiled with the c++11 standard I attach a patch
that modifies the test suite to align the non-compliant strings with
the alignas() attribute. The test suite now passes on Alpha with
that patch.
Cheers
Michael
[1] https://buildd.debian.org/status/fetch.php?pkg=mpd&arch=alpha&ver=0.19.14-2&stamp=1461542099
> In file included from src/decoder/DecoderBuffer.cxx:21:0:
> src/decoder/DecoderBuffer.hxx:41:20: error: 'uint8_t' was not declared in this scope
> DynamicFifoBuffer<uint8_t> buffer;
> ^
> src/decoder/DecoderBuffer.hxx:41:27: error: template argument 1 is invalid
> DynamicFifoBuffer<uint8_t> buffer;
> ^
> src/decoder/DecoderBuffer.hxx: In member function 'void DecoderBuffer::Clear()':
> src/decoder/DecoderBuffer.hxx:61:10: error: request for member 'Clear' in '((DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'int'
> buffer.Clear();
> ^
> src/decoder/DecoderBuffer.hxx: In member function 'size_t DecoderBuffer::GetAvailable() const':
> src/decoder/DecoderBuffer.hxx:78:17: error: request for member 'GetAvailable' in '((const DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'const int'
> return buffer.GetAvailable();
> ^
> src/decoder/DecoderBuffer.hxx: In member function 'ConstBuffer<void> DecoderBuffer::Read() const':
> src/decoder/DecoderBuffer.hxx:87:19: error: request for member 'Read' in '((const DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'const int'
> auto r = buffer.Read();
> ^
> src/decoder/DecoderBuffer.hxx:88:27: error: could not convert '{<expression error>, <expression error>}' from '<brace-enclosed initializer list>' to 'ConstBuffer<void>'
> return { r.data, r.size };
> ^
> src/decoder/DecoderBuffer.hxx: In member function 'void DecoderBuffer::Consume(size_t)':
> src/decoder/DecoderBuffer.hxx:105:10: error: request for member 'Consume' in '((DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'int'
> buffer.Consume(nbytes);
> ^
This seems to be caused by a lacking include, fixed by the below patch.
I'm unsure what made this appear now, though, compiler and toolchain
libraries seem to be the same upstream versions that built 0.19.14-1
just fine in late March.
When a reference counter is at its limit, don't allocate a new
TagPoolSlot - that would result in many TagPoolSlot instances with
ref==1. This in turn would make the linked list very very large,
which means quadratic runtime for many operations.
Apparently all other C libraries are not compatible with "constexpr".
Those which are not will get a performance penalty, but at least they
work at all.
MPD does not really take advantage of memory-mapped I/O by generating
data right into the ALSA buffer; using plain snd_pcm_mmap_writei() has
no advantage compared to snd_pcm_writei(). Let's kill this
non-feature.
The initgroups() manpage says we need to check for _BSD_SOURCE. The
thing is that glibc deprecated this macro, and doesn't define it
anymore, effectively breaking all MPD supplementary groups.
The real fix is to check for initgroups() availability at configure
time, instead of relying on the deprecated _BSD_SOURCE macro.
Apply padding only to the fseek(), not to the chunk size. This fixes
bogus "failed to read riff chunk" messages when the last chunk has an
odd size.
See http://bugs.musicpd.org/view.php?id=4486
systemd does not understand LimitRTTIME=-1. For no limit we have to use
the string 'infinity' (see systemd.exec(5)).
Signed-off-by: Christian Hesse <mail@eworm.de>
This reverts commit d7d9dbd2c2 by
reimplementing it with the current MPD API.
3 years ago, I was wrong about the "embcue" plugin being able to
replace this one, because "embcue" reads a tag named "CUESHEET", while
this plugin reads the "CUESHEET" FLAC metablock. There's an important
difference between those two!
Allocate the buffer dynamically using av_malloc(), and free
AVIOContext.buffer in the destructor, as mandated by the libavformat
documentation.
Fixes http://bugs.musicpd.org/view.php?id=4446
Wildcard matches are directly applied to all filenames in
subdirectories without any attempt at matching relative paths.
This change is based on the following feature request:
http://bugs.musicpd.org/view.php?id=3729
Use the first INDEX in each TRACK section, instead of the last, for the
start time. This preserves the original CD layout (including gaps
between tracks), and avoids skipping sections of songs in more exotic
cuesheets (eg musical suite tracks).
Fixes 0004355 and 0003359
If the song tag comes from a stream, and MPD playback restarts, MPD
would believe the tag should override the newly received tag. This
makes the previous tag appear stuck. This change passes the song tag
only if it's authoritative - i.e. if it's a song file.
Right after booting, the monotonic clock starts with a very small
value, and AudioOutput::LockUpdate() may believe that the fail_timer
has not recovered yet.
The Connect method can be called between Schedule and lock. In that case, when
locked, the state is already set to CONNECTING of READY and the condition won't
be signaled anymore.
Not initialising granulepos leads to it having arbitrary values in the
encoded stream including possibly negative values which are not valid
and confuse opusdec. Explicitly initialise opus_encoder::granulepos
to avoid that problem.
Requiring this prefix makes the client's intention very clear, but it
was too hard to understand why this prefix was needed. Initially, my
intention was to differentiate from broken clients which prefix relate
URIs with a slash; once MPD allowed that. In the past few years
however, MPD has disallowed that, and there was no significant
breakage (except for the "add /" special case which some clients
apparently still do). So I figure it's about time to define that an
URI that begins with a slash points to an arbitrary file on the file
system.
The file handle is never reset to INVALID_HANDLE_VALUE, and thus the
destructor will assume the operation shall be cancelled and will
delete the temporary file.
This was a major breakage for saving the database file and the state
file.
Build a table of pre-existing tag types before adding new items. The
old way would check HasType() each time, which would return true after
the first instance of that tag type had been added, preventing
duplicate tag types to be merged.
This broke duplicate tag types loaded from the state file, because
this code path uses TagBuilder::Complement().
This is Darwin specific: the previous implementation was causing an integer
overflow when base.numer is very large. On PPC Darwin, the timebase info is 1000000000/33330116 and this is too large for integer arithmetic.
This is Darwin specific: the previous implementation was causing an integer
overflow when base.numer is very large. On PPC Darwin, the timebase info is 1000000000/33330116 and this is too large for integer arithmetic.
The Linux feature allows writing new files to an invisible file, and
then replace the old file. This preserves the old file if we get
interrupted by some event.
Fixes a problem with the "curl" input plugin: IsEOF() always returns
true because the "open" flag was cleared by
CurlInputStream::RequestDone() when end-of-stream was reached. This
flag stays false even when seeking to another position has succeeded.
This patch resets the "open" flag to true after seeking successfully.
NetBSD's pthread_setname_np() prototype is incompatible with the rest
of the world, and it requires to pass the string argument as a
non-const pointer. Instead of working around this misdesign, I hereby
disable the feature on NetBSD.
Here's a change to dynamically allocate the DSD ID3 tag buffer.
Pretty much anything with cover art is going to exceed the existing,
static 4k limit... Here's a change to dynamically allocate the buffer
and sanity check it at some upper limit. I rather arbitrarily pulled
256k out of thin air just to keep a corrupt file from causing it to
trying to allocate a buffer larger than available memory.
When mounting had not yet finished, SocketMonitor::IsDefined() was
always false, due to the workaround at the beginning of the function
that calls SocketMonitor::Steal(). This commit drops the IsDefined()
check because it was never necessary and breaks reconnect.
nfs_destroy_context() will invoke all pending callbacks with
err==-EINTR. In CancellableCallback::Callback(), this will invoke
NfsConnection::DeferClose(), which however is only designed to be
called from nfs_service(). In non-debug mode, this will leak memory
because nfs_close_async() is never called.
Workaround: before nfs_destroy_context(), invoke nfs_close_async() on
all pending file handles.
The method NfsConnection::CancellableCallback::Callback() will always
invoke NfsConnection::Close() on the file handle, even if the void
pointer is not a nfsfh. This can happen if the Open() was not
successful, e.g. when the file does not exist.
MPD used both "album artist" and "albumartist" tags and mapped them to one tag.
This could lead to issues, if a file had both tags, causing MPD to send
a list of albumartists instead of a single one.
Since "album artist" is not a standard tag anyway and even its originators
started to use the proper alternative, its time to say goodbye!
Skipping those songs silently will confuse the client, because
commands specifying the song index within a playlist
(e.g. playlistdelete) will be out of sync.
This copies spl_print()'s behavior to playlist_file_print().
Version 2.5 fixed an API oddity, however it broke API compatibility,
at least with C++. Disable the workaround when a libavformat version
is detected that is recent enough.
The "::" to explicitly refer to the global namespace appeared like a
good idea in C++, but it breaks with C libraries that implement
standard functions using macros (e.g. musl).
This was used by proprietary software. MPD adopted it a few years
ago, which turns out to be a mistake, because it now creates problems
for some MPD users (http://bugs.musicpd.org/view.php?id=4168).
libmp4v2 is licensed under MPL 1.1, which is incompatible with GPLv2.
Unfortunately, this means that we must remove the plugin.
More information can be found in the Debian bug report:
http://bugs.debian.org/767504
Commit d42c0f1dc5 added an OS X-specific
method of calling mpd_main_after_fork(), which uses Grand Central
Dispatch. Since this uses a block literal, it breaks compilation on
compilers which don't support the block extension, e.g. non-Apple
compilers. This affects users on older OS X releases with GCD (which
depend on older Clang releases, or Apple GCCs, which don't support the
C++11 features MPD needs); or which don't support GCD at all (10.5 and
lower).
This patch changes the #ifdef so that the non-GCD code is used
as it was on OS X before this patch if blocks aren't available, via
checking __BLOCKS__ macro.
The old formula calculates the output buffer size with "regular"
rounding (to the nearest integer), however sometimes, that is
insufficient and the last sample cannot be resampled. This causes
audible distortions. By changing the formula to consider the worst
case (always round up), this problem is eliminated.
While seeking, metadata must not be updated. ResponseBoundary() was
added in MPD 0.19.1, but I forgot to add the IsSeeking() check there.
This caused the "seekable" flag to reset.
On "list albumartist", songs that have no AlbumArtist tag will use the
Artist tag. However, if AlbumArtist is disabled via
"metadata_to_use", the TagBuilder::AddItem() call is ignored, and
PrintUniqueTag() attempts to print a nullptr string.
This commit fixes the problem by attempting the fallback only if
AlbumArtist is not disabled.
When uri_apply_base() was moved from db/upnp/Util.cpp to
util/UriUtil.cpp, the parameter order was changed, however without
swapping the parameters in the ContentDirectoryService constructor.
Many years ago, FAAD had a serious ABI bug: the NeAACDecInit()
prototype in its header declared the "samplerate" parameter to be
"unsigned long *", but internally, the function assumed it was
"uint32_t *" instead. On 32 bit machines, that was no difference, but
on 64 bit, this left one portion of the return value uninitialized;
and worse, on big-endian, the wrong word was filled. This bug had to
be worked around in MPD (commit 9c4e97a6).
A few months later, the bug was fixed in the FAAD CVS in commit 1.117
on file libfaad/decoder.c; the commit message was:
"Use public headers internally to prevent duplicate declarations"
The commit message was too brief at best; the problem was not
duplicate declarations, but a prototype mismatch. No mention of the
bug fix in the ChangeLog.
The MPD project never learned about this bug fix, and so MPD would
always pass a "uin32_t *" dressed up as a "unsigned long *". Nearly 6
years later, it's about time to fix this second ABI problem. Let's
kill the workaround!
Many years ago, FAAD had a serious ABI bug: the NeAACDecInit()
prototype in its header declared the "samplerate" parameter to be
"unsigned long *", but internally, the function assumed it was
"uint32_t *" instead. On 32 bit machines, that was no difference, but
on 64 bit, this left one portion of the return value uninitialized;
and worse, on big-endian, the wrong word was filled. This bug had to
be worked around in MPD (commit 9c4e97a6).
A few months later, the bug was fixed in the FAAD CVS in commit 1.117
on file libfaad/decoder.c; the commit message was:
"Use public headers internally to prevent duplicate declarations"
The commit message was too brief at best; the problem was not
duplicate declarations, but a prototype mismatch. No mention of the
bug fix in the ChangeLog.
The MPD project never learned about this bug fix, and so MPD would
always pass a "uin32_t *" dressed up as a "unsigned long *". Nearly 6
years later, it's about time to fix this second ABI problem. Let's
kill the workaround!
Pulseaudio expects clients to specify their channel-map if the
default (ALSA) map does not route the audio to the expected speakers.
Many Google results suggest dealing with this by re-routing the audio
channels with the appropriate ALSA plugin, but this will then simply
break any clients which expect the default ALSA mapping.
Virtually all media files and codecs, certainly flac, dca, a52, and of
course anything based on Microsoft's WAVEFORMAT_EXTENSIBLE specification,
assume the layout in the table here:
http://en.wikipedia.org/wiki/Surround_sound#Standard_speaker_channels
Fortunately, pulseaudio directly addresses this with a built-in channel
map for WAVE-EX which can be set automatically in the stream sample-spec.
MPD handles all strings in UTF-8 internally. Those decoders which
read Latin-1 tags are supposed to implement the conversion, instead of
passing Latin-1 to TagBuilder::AddItem(). FixTagString() is simply
the wrong place to do that, and hard-coding Latin-1 is kind of
arbitrary.
The Release Track Id uniquely identifies a recording on a release - that
is, even if a recording appears twice on a release (meaning that the
combination of recording and release id are not enough to figure out
which one it is), the release track id will allow differentiating the two.
The tag names are taken from
https://musicbrainz.org/doc/MusicBrainz_Picard/Tags/Mapping
On NetBSD, PTHREAD_MUTEX_INITIALIZER and PTHREAD_COND_INITIALIZER are
not compatible with C++11 "constexpr" (see Mantis ticket 0004110). As
a workaround, don't ues "constexpr", and use the functions
pthread_mutex_init(), pthread_mutex_destroy(), pthread_cond_init() and
pthread_cond_destroy() instead. This adds some runtime overhead, but
is portable to POSIX implementations that have awkward initializer
macros.
Casting std::numeric_limits<unsigned>::max() to "long" leads to an
overflow if sizeof(unsigned)==sizeof(long), and the result will be -1.
This happens on some 32 bit architectures, for example ARM and WIN32.
Workaround: use std::numeric_limits<int>::max(), which is the largest
signed integer. Since sizeof(long)>=sizeof(int), this will never
overflow.
Fixes Mantis ticket 0004080.
The IsActive() method returned true even if the timer was not active,
after it completed once. This broke the state file timer, and the
state file was not saved periodically.
Read one block at a time. This discards the last partial block, which
cannot be interleaved anyway. Previously, uninitialised memory was
used to interleave the last block, which generated some noise.
When the data chunk size is not a multiple of the frame size, the last
partial frame lead to an endless loop. We fix this by checking
chunk_sze>=frame instead of chunk_sze>0. This way, the partial frame
is simply skipped.
Previously, MPD tried to slurp the whole song file, count the number
of frames and calculate the song duration from that. That however is
extremely expensive for remote files, and will delay playback for a
long time. Workaround: check only the first 128 frames and try to
extrapolate from here. Fixes Mantis ticket 0004035.
Implement a "bulk" edit mode that postpones both UpdateQueuedSong()
and OnModified(). This way, the playlist version gets incremented
only once. More importantly: when adding multiple songs to a queue
that consists of only one song, the first song that got added will
always be played next. By postponing this choice, all newly added
songs get a chance to become the next song. Fixes the second (and
last) part of Mantis ticket 0004005.
Don't restore the current song after shufflung when MPD is stopped
(but still remembers the current song internally). Fixes the first
part of Mantis ticket 0004005.
Turns out the lock-free code using atomics was not thread-safe. The
given callback could be invoked by GLib before the source_id attribute
was assigned. This commit changes the DeferredMonitor class to use a
Mutex to block the event loop until source_id is assigned. This bug
does not exist in the 0.19 branch because it does not use the GLib
main loop anymore.
This corrects a major mistake from commit 724a59aa - there was one
small thing that commit was supposed to do, and it failed.
AV_TIME_BASE is not a seek flag.
Don't wait for an optimistic write to fail. This is an improved
workaround for the infamous Raspberry Pi bug (see commit af991765).
It works much better and comes without the negative side effects. The
old workaround is now obsolete.
Move code from CreateConfiguredDatabase() and add XDG support. This
implements an automatic Linux fallback for the setting "db_file" if
none was specified.
This commit adds the NeighborPlugin API which can be used to detect
nearby file servers that can be used by input plugins. This list of
servers is exported using the new "listneighbors" command. The idle
even "neighbor" notifies interested clients when a new neighbor is
found or an existing one is lost.
There's a lot missing currently: protocol&user documentation, and a
way to "mount" remote servers into the music database. Obviously,
some code from the UPnP database plugin can be moved to a neighbor
plugin.
The update thread should not affect the rest of the system, therefore
set "idle" priority, and let it only run when nobody else is using the
resources.
This encoding plugin features a fixed-point mp3 encoder,
with faster encoding on architectures without a FPU.
Right now the encoder is limited to stereo and 16 bit depth.
The bitrate and sample rate can be modified in audio_output.
audio_output {
type "httpd"
name "My shine stream"
encoder "shine"
port "8000"
format "44100:16:2"
bitrate "320" # default: 128
}
If we have libyajl 2.0.1 (without a pkg-config file), our configure.ac
would assume this is the libyajl 1.0 API, because the function
yajl_alloc() exists in both. This commit changes the library check to
the function yajl_parse_complete() which was removed in the 2.0 API.
This fixes build failure with libyajl 2.0.1.
I've created an elementary input plugin that plays sound from the
soundcard, so you can use MPD to listen to anything connected to the
line-in jack, or to Video4Linux FM radio cards that send audio through
the soundcard. There has been a small number of posts here in the
past requesting line-in input, so here is a first, simplistic stab at
it.
The patch adds a new sheme, alsa://, which causes mpd to play data
read directly from a souncdard. It defaults to hw:0,0, but you can
pass any ALSA device name in the URI. So, using mpc for example:
mpc add alsa://
mpc play
will play from device hw:0,0.
To use a diffferent device:
mpc add alsa://hw:1,0
Commit 77c63511 caused MPD to become stuck right after a song change.
The problem was that at some point, the MusicBuffer became full, and
the DecoderThread working on the next song waits for the PlayerThread.
However, the PlayerThread was stuck in a loop of g_usleep() calls, and
never bothered to tell the DecoderThread that the MusicBuffer is not
full anymore. This bug is very old, but its chance to occur went from
nearly 0% to nearly 100%.
The fix is to wake up the DecoderThread before waiting for it. As a
side effect, I replaced the g_usleep() call with a Cond::Wait() call.
Allows big-endian users to configure the fallback byte order to
little-endian. Without this setting, MPD assumes native byte order if
the CD drive can't decide.
Add a new CommandResult code called "FINISH" which, unlike "CLOSE",
will attempt to flush the output buffer. This is a one-shot attempt;
it will do one write, and not try again.
Prior to version 0.3, the "length" callback returned a "long" instead
of AFfileoffset. Now that this API bug fix is a few years old, let's
drop 0.2 support for good.
Migrate from the old curl_multi_perform() API to the newer
curl_multi_socket_action() API (since CURL 7.16).
This allows working around a bug with HTTP redirections with epoll:
when CURL closes a socket and the new one happens to have the same
file number, MPD did not have a chance to remove the old one from
epoll and subsequently attempted to use EPOLL_CTL_MOD, which was not
allowed by epoll, because it's a new socket now.
This prevented using the "volume_normalization" feature with some
codecs (e.g. mp3), because the normalization code requires 16 bit
samples. If the codec happens to deliver formats other than S16, the
AutoConvert filter succeeds to initialize the conversion filter, but
the returned input audio format was wrong.
Share the Mutex between the DecoderThread and the PlayerThread. This
simplifies synchronization between the two threads and fixes a freeze
problem: while the PlayerThread waits for the DeocderThread, it cannot
answer requests from the main thread, and the main thread will block
until the DecoderThread finishes.
This command was removed by commit 206392ad (MPD 0.16), even though it
was been proven useful for some very simple clients. On request, I
add it to the protocol again.
The "loop_count" configuration parameter allows the user to set how
many times a module with backward loops shall loop. "0" (the default)
means a module is not allowed to use backward loops at all. "-1"
enables inifinite looping.
This patch allows the user to configure the mikmod decoder plugin to loop
modules. It adds a configuration parameter to the mikmod decoder called
"loop" which can be "no" (the old behaviour, default) or "yes" to allow
modules to use backward loops.
Player_LoadTitle() returns an aligned pointer in libmikmod-3.2 that
cannot be freed with free(). The correct way to do this now is
MikMod_free() which extracts the original pointer from the buffer and
frees that.
Signed-off-by: Christoph Mende <mende.christoph@gmail.com>
This plugin is cumbersome to support, now that MPD is migrating away
from GLib and the GLib event loop. It has no practical advantages
over the CURL plugin. Soup requires the bloated GType library.
Save the state file 2 minutes after the last change. This reduces the
disruptions by an idle MPD, and MPD can be paged out permanently until
it is used.
Implement the "scan_stream" method that can read tags from any
input_stream object. This requires a FLAC__IOCallbacks implementation
based on the input_stream API.
Improves quality by not squeezing 32 bit samples down to 16 bit, and
then back to 32 bit floating point. Reduces CPU usage by skipping a
conversion step.
Internally the vorbis (non-Tremor) decoder is working in floating
point, and it's not really necessary to cut the output back to 16-bit
if the soundcard or OS supports higher resolution.
The decoder can be trivially modified to bypass its internal
quantisation and produce floating-point output, and a separate
quantisation can be used as appropriate to the platform.
libvorbisidec and libvorbis export the same symbols, which is a
dangerous thing. Since libvorbisenc depends on libvorbis, this can
get nasty, so let's disable the Vorbis encoder unless the user
explicitly wants it.
This commit reimplements the core of the "single" mode. Instead of
doing the detection in the playlist code from the outside, it is moved
to the player thread, which gets a new option called "border_pause".
It will now pause playback exactly at the beginning of the new song,
making the feature more reliable.
Now that the player thread knows what will happen, it can suppress
cross-fading.
Fixes mantis tickets 0003055 and 0003166.
That function is not pure, it writes to error.
When marked as pure, the compiler is allowed to assume it does not do
anything to error, so it can remain NULL, which would result in an
invalid read in print_error().
Ignore APE tags that have no usable tags, and use the ID3 tag instead.
This is useful when the APE tag only contains replay gain, and the
real tags are stored as ID3. This implements feature request Mantis
#0003521.
g_file_test is redefined to be g_file_test_utf8 and thus can't handle
non-ASCII characters. This fix adds simple wrapper (taken from glib)
that fixes encoding and calls g_file_test_utf8. All required inclusions
of glib_compat.h are added as well.
This plugin is horrible code, I mean it. Last year, I tried hard to
fix it, but I figured would take less time to do a full rewrite.
Given that I don't even have any device that supports RAOP, I can't do
that properly. After 16 months, nobody volunteered for fixing it.
Hereby, I delete it, because having no RAOP plugin is better than
having this mess. Sorry.
The existing buffer implementation has a major flaw: it is unable to
re-fill the buffer until it has been consumed completely, leading to
many occasions where the render callback needs to generate silence,
just because the play() implementation was unable to append more
data. The fifo_buffer library handles that well.
Requires YAJL to build, and this doesn't include the necessary
automake changes. Can be built using
./configure CFLAGS="-I/usr/include/yajl" LIBS="-lyajl" --enable-soundcloud
Add the following to your config:
playlist_plugin {
name "soundcloud"
enabled "true"
apikey "c4c979fd6f241b5b30431d722af212e8"
}
Then you can stream from soundcloud using calls like:
mpc load soundcloud://track/<track-id>
mpc load soundcloud://playlist/<playlist-id>
mpc load soundcloud://url/http://soundcloud.com/some/track/or/playlist
For the last case, you can leave off the http:// or
http://soundcloud.com/ .
This was disabled when compiled with a new ffmpeg version. Older
ffmpeg versions used it explicitly, while newer ones may pass it
through from the codec.
This fixes a bug when libsamplerate returns an empty buffer for a very
small input buffer. The caller thinks this is an error, bug there is
no GError object.
This finally enables the new embedded CUE sheet code: when a song file
contains a playlist, it is printed in the "lsinfo" output, so clients
get to know about this.
Use libasound's polling functions, implement a bridge to GSource /
GPollFD and send idle events to clients when an external program
changes the ALSA mixer volume.
Moving songs using either 'move' or 'moveid' to position -1 (after the
current song) would fail for a song which is just before the current
song.
This patch corrects the check to see if the current song is in the range
to be moved. Since the range is from `start` up to `end` (exclusive) the
check was incorrect, but is now fixed.
The implementation of cancel() did not work well: you cannot use
alSourceUnqueueBuffers() to unqueue queued buffers, and our function
openal_unqueue_buffers() left the OpenAL library in a rather undefined
state; nothing was supposed to be queued, but the "filled" variable
was not reset.
The local variable was already divided by 1000, and the return value
was being divided by 1000 again - doh! This caused delays in the
httpd output plugin that were too small by three orders of magnitude,
and the buffer was filled too quickly.
WinAPI explicitly declares filesystem encoding.
It can be determined by GetACP().
Use that instead of Glib routine that always "detects" UTF-8 on Win32,
which is incorrect for MPD case.
Ensure that WINVER is defined early enough, so other system headers
won't fall back to their default value. Specifically, this solves a
build failure (-Werror) with mingw-w64 ("WINVER redefined").
When we have an absolute path that's not inside the music directory,
allow loading it anyway, if we're in "secure" mode (i.e. the client is
connected via UNIX socket).
Right now, a playlist with absolute pathnames can only add songs that
are in the same the directory of the playlist or under it.
If uri is an absolute pathname and base_uri is set,
playlist_check_translate_song() will check that base_uri is a prefix
of uri, excluding every other song in the music directory outside
base_uri.
I think in this case base_uri should be completely ignored (and made
NULL) and uri should just be checked against music root directory.
Previously, the condition "defined(play_audio_format)" was used to see
if an output device has been opened, but if the device had failed on
startup, an assertion failure could occur. This patch adds a separate
flag.
The Naim Uniti does not appear to support icecast-style streaming of FLAC
music but does support the codec from a DLNA server. This change looks for
"transferMode.dlna.org: Streaming" in the HTTP request header and responds
with something the Uniti (and hopefully other DLNA clients) accepts.
The only difference in the DLNA streaming mode is the reponse header and
that icecast metadata is disabled. If a client request indicates both modes
are supported, the DLNA mode is preferred (as the Uniti says it supports
both but then rejects a FLAC ICY stream).
Note: This change may be specific to Naim equipment (the only device it was
tested on). E.g. the hardcoding of Content-Length which works but is not a
logically correct value. The change should be backwards-compatible, so
only those clients requesting a DLNA stream will see any difference.
When playing a CUE track, the player thread waited for the decoder to
become ready, and then sent a SEEK command to the beginning of the CUE
track. If that is near the start of the song file, and the track is
short enough, the decoder could have finished decoding already at that
point, and seeking fails.
This commit makes this initial seek more robust: instead of letting
the player thread deal with the difficult timings, let the decoder API
emulate a SEEK command, and return it to the decoder plugin, as soon
as the plugin finishes its initialization.
Add GMutex, GCond attributes which will be used by callers to
conditionally wait on the stream.
Remove the (now-useless) plugin method buffer(), wait on GCond
instead. Lock the input_stream before each method call. Do the same
with the playlist plugins.
D'oh, we were reading 16 bit integers instead of 32 bit integers!
That caused silence when trying to play a 32 bit input file on a 24
bit sound card (e.g. USB sound chips with 24 bit packed samples).
Don't abort the configure script when avahi could not be
auto-detected. It previously did, because there was no custom "fail"
action for PKG_CHECK_MODULES.
The output thread could hang indefinitely after finishing CANCEL,
because it could have missed the signal while the output was not
unlocked in ao_command_finished().
This patch removes the wait() call after CANCEL, and adds the flag
"allow_play" instead. While this flag is set, playback is skipped.
With this flag, there will not be any excess wait() call after the
pipe has been cleared.
This patch fixes a bug that causes mpd to discontinue playback after
seeking, due to the race condition described above.
To demonstrate the new I/O thread. libsoup is well-integrated into
the GLib main loop, which made this plugin pretty easy to write.
As a side effect, we have to initialize the I/O thread in all debug
programs that use the input API.
This warning should only be logged when we really received something.
When the client disconnects, G_IO_IN is triggered, and the read
returns G_IO_STATUS_EOF.
In the "vorbis" plugin, this is a copy of the old flush() method,
while flush() gets a lot of code remove, it just sets the "flush" flag
and nothing else. It doesn't start a new stream now, which should fix
a few problems in some players.
This makes FreeBSD detect libogg correctly. The '==' operator is an
undocumented GNU extension to test(1) and cannot be relied upon to
exist and do the right thing. POSIX mandates string comparisons to be
done using "test foo = bar".
From http://bugs.debian.org/513291
"In mpd.conf, the "admin" permission covers updating the db and
killing mpd.
"Since there are quite some usecases in which the user can upload
music to the mpd's directory by means of anonymous FTP or so, it is
desirable that any user may issue a db update, while killing the mpd
should not be possible.
"I'd suggest to remove the db update from the admin group and either
add it to "control" or an own group."
With mono sound, jack_sample_size is smaller than frame_size (4 vs 2
bytes), and "space/jack_sample_size==0". That means mpd_jack_play()
will return 0, although no error has occurred.
Version 1.0.0 of the libao library added a new field to the
ao_sample_format struct. It is a char * named matrix. When
an ao_sample_format is allocated on the stack, this field contains
garbage. The proper course is to insure that is initialized to NULL.
NULL indicates that we do not want any mapping.
The struct is now initialized using a static initializer, and this
technique is compatible with all known versions of libao.
This fixes the following valgrind warning occuring on the first call of
httpd_output_read_page:
==20124== Conditional jump or move depends on uninitialised value(s)
==20124== at 0x425E65: httpd_output_read_page (httpd_output_plugin.c:240)
==20124== by 0x426087: httpd_output_open (httpd_output_plugin.c:279)
==20124== by 0x41D862: ao_open (output_plugin.h:206)
==20124== by 0x41E133: audio_output_task (output_thread.c:590)
I wanted mpd to play a mp3 stream from a music website. The stream is
only available to subscribers, which restriction is enforced through
normal http authentication. However, the URL I get from the website
is not the final URL of the stream, but a generic URL which points to
the real one through a redirect (code 301). Thus, I cannot predict
the final URL, and so I cannot use the username:password hack to force
the authentication, and mpd (libcurl on mpds behalf) fails to grab the
stream.
libcurl allows the option CURLOPT_NETRC to be set and then the
credentials can be stored in the good old .netrc file (in this case it
would be ~mpd/.netrc, of course). But mpd doesn't set this option. I
think it should.
When a music_chunk to be crossfaded consists only of a tag,
cross-fading is not possible, and led to an assertion failure. This
patch just discards those, as if cross-fading was not enabled.
During the whole output thread, the audio_output object is locked, and
it is only unlocked while waiting for the GCond and while running a
plugin method. The error handler in ao_play_chunk() attempted to lock
the object again, which was code from MPD 0.15.x which should have
been removed a long time ago.
Until the decoder plugin has called decoder_initialized(), the player
may not submit seek commands. This however could occur with a slow
decoder and a CUE file with a virtual song offset. This patch adds
another check.
When you don't explicitly set an output sample rate, liblame tries to
guess an output sample rate from the input sample rate. You would
think that this "guessing" consists of just setting both equal, but
that is not the case. For 44.1kHz at 96kbit/s, liblame chooses
32kHz. This patch explicitly configures the output sample rate, to
stop the bad guessing.
This is a MPD 0.16 regression: when playing a 24 bit file, the switch
to 16 bit was made only partially, after mBytesPerPacket and
mBytesPerFrame had already been applied.
That means mBytesPerFrame referred to 24 bit, and mBitsPerChannel
referred to 16 bits. Of course, that cannot work.
Rename the "version" struct, because it seems to be a reserved name on
Solaris:
"src/decoder/mad_decoder_plugin.c", line 550: (enum) tag redeclared: version
cc: acomp failed for src/decoder/mad_decoder_plugin.c
Add new config parameter 'device' to audio_output type "osx":
- if not supplied or set to "default", open default device
- if set to "system", open system device
- otherwise 'device' should be an audio device name: mpd will find and
open the specified audio device, falling back to the default
device if it's not found
After popular demand, I've switched the order of "artist" and "title"
in the stream title. There is no standard, and there is no reliable
way to parse those from the stream title.
When one song is played twice, and the decoder is working on the
second "instance", but the first should be seeked, the check in
player_seek_decoder() may assume that it can reuse the decoder without
exchanging pipes. The last thing was the mistake: the pipe pointer
was different, which led to an assertion failure. This patch adds
another check which exchanges the player pipe.
Change the assertion on "fail_timer==NULL" in OPEN to a runtime check.
This assertion crashed when the output thread failed while the player
thread was calling audio_output_open().
Added support for a new optional configuration setting for the httpd output
named "bind_to_address". Setting it to a specific IP address (v4 or v6) will
cause the httpd output to bind to that address exclusively. Supporting
multiple addresses in parallel is future work.
This implements the feature requests #2998 and #2646.
Clear the notification before finishing the CANCEL command, so the
notify_wait() after that will always wait for the right notification,
sent by audio_output_all_cancel().
Some users reported that MPD crashes when using a new CURL version
with the threaded DNS resolver enabled. It seems that
curl_multi_fdset() returns no file descriptor when the DNS resolver
runs in another thread, so MPD does not have any event to wait for.
On the CURL mailing list, somebody suggested to sleep for a fixed
amount of time. This is not an elegant solution, because daemons
should never have to sleep without waiting for an event. I hope the
CURL developers will review the API and remove the threaded DNS
resolver.
Meanwhile, I'm removing the assertion in question, to allow those
unfortunate users running the latest CURL version to continue using
MPD.
In libwildmidi 0.2.3, the function WildMidi_SampledSeek() was removed,
without changing the SO name. This patch adds an autoconf check for
that function. Fall back to WildMidi_FastSeek() if
WildMidi_SampledSeek() is not available anymore.
libavformat 0.6 does not pass the original URI pointer to the "open"
method, which leads to a crash because MPD was using a dirty hack to
pass a pointer to that method.
This patch switches to av_open_input_stream() with a custom
ByteIOContext class, instead of doing the URI string hack with
av_open_input_file().
Loosely based on a patch from Jasper St. Pierre.
I've attached a patch that will make file URIs work on operating systems
that provide the getpeereid() function call to check the user ID of the
peer connected to a UNIX domain socket.
I took this tag name from a MusePack sample file I got from a user.
It is not documented in the APE specification:
http://wiki.hydrogenaudio.org/index.php?title=APE_key
People seem to be using undocumented extensions to the specification
anyway, and the best we can do is attempt to support them.
I took these tag names from a MusePack sample file I got from a user.
These are not documented in the APE specification:
http://wiki.hydrogenaudio.org/index.php?title=APE_key
People seem to be using undocumented extensions to the specification
anyway, and the best we can do is attempt to support them.
Without libid3tag, we were trying to skip the ID3 frame (since
0.15.2). Its length however was not calculated at all, we were just
dropping everything from the current input buffer. This lead to the
first few seconds of the file being skipped. This patch attempts to
calculate the ID3v2 frame size with the formula from:
http://www.id3.org/id3v2.4.0-structure 3.1 and 6.2
"When playing musepack files with mpd v0.15.8, rg seems to have no effect.
Using sample file below, mpd says 'computing ReplayGain album scale with gain 122.879997, peak 0.549150'.
One thing though, if I build mpd against old libmpcdec-1.2.6, rg works
as expected: 'computing ReplayGain album scale with gain 16.820000,
peak 0.099765'"
Previously, tags of the new song being cross-faded in were sent
immediately. That can cause wrong information being displayed,
because the "previous" song might send its tag at the end again,
overriding the "next" song's tag. This patch saves & merges the tag
of the next song, and sends it when cross-fading is finished, and the
next song really starts.
"There is a bug in fixed-point musepack (musepack_src_r435) playback.
In floating-point audio is OK but in fixed audio is distorted. I have
made a patch for this"
With single+repeat enabled, it is expected that MPD repeats the
current song over andd over. With random mode also enabled, this
didn't work, because the song order was shuffled internally. This
patch adds a special check for this case.
This is a very basic check, which only ensures that the path does not
begin with a slash, doesn't have double slashes and the special names
"." and ".." are forbidden.
Removed the decoder_command_finished() call at the end of
mp3_decode(). This is invalid, because decoder_command_finished() has
already been called in mp3_read().
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.
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.
When all plugins have failed, MPD used to fall back to the "mad"
decoder plugin, to handle those radio streams without a Content-Type
response header. This however leads to unexpected results (garbage
being played) when the stream isn't really mp3. Since we care little
about "bad" streams, we shouldn't have hacks which have bad side
effects.
Let's get rid of this hack now! Only try to "mad" plugin if there was
no match at all (Content-Type, path suffix) and no other plugin has
been tried.
To allow libavformat to detect the format of the input file, append
the suffix of the input file to the URL of the virtual stream. This
specifically enables the "shorten" codec, which is supported by
libavformat/raw.c, detected only by the suffix.
The patch "input/file: don't fall back to parent directory" introduced
a regression: when trying to play a CUE track, decoder_run_song()
tries to open the file as a stream first, but this fails, because the
path is virtual.
This patch fixes decoder_run_song() (instead of reverting the previous
patch) to accept input_stream_open() failures if the song is a local
file. It passes the responsibility to handle non-existing files to
the decoder's file_decode() method.
This replaces the rewinding buffer code from the CURL input plugin.
It is more generic, and allows rewinding even when the server sends
Icy-Metadata (which would have been too difficult to implement within
the CURL plugin).
This is a rather complex patch for the stable branch (v0.15.x), but it
fixes a serious problem: the "vorbis" decoder plugin was unable to
play streams with Icy-Metadata, because it couldn't rewind the stream
after detecting the codec (Vorbis vs. FLAC).
When collecting tag values for the result set, add all of a song's tag
values of the searched type. This affects the "list" command.
Previously, "list" only considered the first tag value of a song.
Don't clear the music pipe when seeking has failed - check the
"seeking" flag instead of "command==SEEK". Clear the "seeking" flag
in decoder_seek_error().
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.
Fixes a memory leak: the "archive" input plugin opens the archive, but
never closes it. This patch moves the responsibility for doing that
to archive_plugin.open_stream(). This is an slight internal API
change, but it is the simplest and least intrusive fix for the memory
leak.
This fixes an inconsistency in the stored playlist subsystem: when
obtaining the list of playlists (listplaylist, listplaylistinfo), the
file names in the playlist directory are converted to UTF-8 (according
to filesystem_charset), but when saving or loading playlists, the
filesystem_charset setting was ignored.
The plugin code tried to force libavcodec to supply stereo samples.
That however has never actually worked. By removing this code, we are
able to play surround files for the first time.
Allow RIFF/AIFF ID3 tags up to 4 MB (old limit was 256 kB). This
might still be too small for some users, and when somebody complains,
we might do something more clever (like streaming the data into
libid3tag?).
On some platforms, libavcodec wants the output buffer aligned to 16
bytes (because it uses SSE/Altivec internally). It will segfault when
you don't obey this rule.
MPD has been supporting 32 bit samples since version 0.15. This patch
changes one check, and removes the 32->24 conversion code.
Note that WavPack floating point samples have 32 bits, and MPD doesn't
have a special check for floating point - therefore, this WavPack
plugin still returns 24 bit integer samples as before (until we have
float support in the MPD core).
If flac_container_decode() gets a seek destination which is out of
range, it ignores the SEEK command (never finishes it). This leads to
MPD lockup, because the player thread waits for completion.
Drop the required GLib version from 2.16 to 2.12, because many current
systems still don't have GLib 2.16. This requires several new
compatibility functions in glib_compat.h.
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().
When there's no queued song, and the current one has finished playing,
first make sure that the hardware outputs have really finished playing
the last chunk: call the drain() method in all audio outputs. Without
this patch, MPD stopped playback shortly before the ALSA sound card
had finished playing.
Added the "fd_util" library, which attempts to use the new thread-safe
Linux system calls pipe2(), accept4() and the options O_CLOEXEC,
SOCK_CLOEXEC. Without these, it falls back to FD_CLOEXEC, which is
not thread safe.
This is particularly important for the "pipe" output plugin (and
others, such as JACK/PulseAudio), because we were heavily leaking file
descriptors to child processes.
Same as the previous patch: create up to 16 configured source ports.
The plugin tries to do its best at guessing the right combination for
the given input file, the number of source and destination ports.
Implement the methods enable() and disable(). Bind the HTTP port in
the enable() method, but reject all incoming connections until the
output is opened.
After playback has stopped, the ring buffers may still contain
samples. These will be played when playback is started the next
time. We should clear the buffers each time.
jack_client_new() is deprecated. This requires libjack 0.100
(released nearly 5 years ago). We havn't been testing older libjack
versions anyway.
As a side effect, there is the new option "autostart".
Store a list of supported tag items in the database. When loading a
database which does not have a matching list, we must rescan in order
to get the missing information.
Use a single GString buffer object in all functions loading the
database. Enlarge it automatically for long lines. This eliminates
the maximum line length for tag values. There is still an upper limit
of 512 kB to prevent denial of service, but that's reasonable I guess.
The line buffer had a fixed size of 5 kB, and was allocated on the
stack. This was too small for some users. As a hotfix, we're
increasing the buffer size to 32 kB now, allocated on the heap. In
MPD 0.16, we'll switch to dynamic allocation.
Convert the metadata with the libavformat function av_metadata_conv().
This ensures that canonical tag names are provided by libavformat, and
we can remove the "artist" vs "author" workaround.
When you disable the "follow_outside_symlinks" or the
"follow_inside_symlinks" setting, the next update should remove the
now-ignored files from the database.
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".
Don't initialize "vc" and "cs" with FLAC__metadata_object_new(); that
value is overwritten by FLAC__metadata_get_tags() and
FLAC__metadata_get_cuesheet().
When the player thread unpauses, it sends CANCEL to the output thread,
after having checked that the output is still open. Problem is when
the output thread closes the device before it can process the CANCEL
command - race condition. This patch adds another "open" check inside
the output thread.
This has been replaced by the last.fm playlist plugin. The input
plugin has never worked well, and was just a playground to experiment
with the last.fm radio protocol.
When the connection is lost while buffering, the CURL input plugin may
enter an endless loop, because it does not check the EOF condition.
This patch makes fill_buffer() return success only if there's at least
one buffer, which is enough of a check.x
Accidently, MPD has been using several GLib 2.16 functions for a
while, and nobody noticed yet. To simplify the code base, let's bump
the minimum GLib version for MPD to 2.16. That version is old enough,
and it's reasonable to expect users to have it.
On 32 bit systems with large file support enabled (i.e. "sizeof(off_t)
> sizeof(size_t)") gcc emits a warning because a size_t cast to off_t
can never become negative.
When there is no Content-Type response header, try the "mad" decoder
plugin. It uesd to be named "mp3", and we forgot to change the
fallback name in decoder_thread.c.
When a received chunk of data has only icy-metadata, there was no
usable data left for input_curl_read() to return, and thus it returned
0 bytes. "0" however is a special value for "end of file" or
"error". This patch makes input_curl_read() read more data from the
socket, until the read request can be fulfilled (or until there's
really EOF).
No more CD player emulation. The current behaviour of "previous" is
difficult for a client to predict, because it does not definitely know
the current position within the song. If a client wants to restart
the current song, it can always send "playid".
If nothing has changed since the last save, don't save the state
file. Saving will spin up the hard drive, which is undesirable on
hosts where MPD is idling in background.
Usually, we read our "artist" tag from ffmpeg's "author" tag. In some
cases however (e.g. APE), this tag is named "artist". This patch
implements a fallback: if no "author" is found, MPD tries to use
"artist".
When the ID3 tag in an AAC file is larger than the current buffer, the
function decoder_buffer_consume() aborts. By using the new function
decoder_buffer_skip() instead, we can safely skip the ID3 tag.
This patch implements a light-weight inotify library, and watches all
directories below the music directory. It updates all directories
where files changed after a delay of 5 seconds.
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.
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.
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.
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.
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.
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.
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.
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.
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 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.
The "lastfm" input plugin is far from complete, because MPD does not
support nesting playlists yet. The "fluidsynth" decoder plugin
suffers from shortcomings in the libfluidsynth library:
http://www.mail-archive.com/fluid-dev@nongnu.org/msg01099.html
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.
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.
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().
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.
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.
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.
