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().
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.
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.
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.
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.
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
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).
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.
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.
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.
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.
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.
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.
The generic sockaddr struct is too small for some addresses. For
accept(), we have to allocate a sockaddr_storage struct on the stack,
which is large enough for all addresses.
Added the uri_remove_auth() library function which strips username
and password from a HTTP URI, and use it in song_print_url(). This
allows you to add HTTP URIs to the playlist including secret username
and password, without disclosing it to all MPD clients.
Avuton Olrich