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.
If mpd.conf specifies a user, and MPD is invoked by exactly this user,
ignore the "user" setting. Don't bother to look up its groups and
don't attempt to change uid, it won't work anyway.
Use delete_directory() for removing sub directories instead of
dirvec_clear(). This ensures that all memory occupied by
subdirectories of deleted directories is freed.
When a directory is deleted, MPD deleted only the directory from the
database; it did not bother to walk the full tree to free all memory
and to remove deleted songs from the playlist. Replace a
dirvec_delete() with delete_directory().
There are a few problems left in this plugin:
- fluidsynth decodes in real time, while MPD prefers to buffer as
quickly as possible; as a workaround, this plugin uses a timer
object to synchronize with real-time playback
- I don't know yet how fluidsynth tells me when the song has ended
- the "soundfont" configuration setting is not yet documented, and it
will likely change soon (in favor of a per-decoder configuration
block)
Don't define HAVE_FFMPEG if the ffmpeg libraries were found via
pkg-config, but ffmpeg support was disabled because
avcodec_decode_audio2() is not available.
Always assume the buffer is empty before calling the encoder. Always
flush the buffer immediately after there has been added something.
This reduces the risk of buffer overruns, because there will never be
a "rest" in the current buffer.
The non-blocking mode of libshout is sparsely documented, and MPD's
implementation had several bugs. Also removed connect throttling
code, that is done by the MPD core since 0.14.
The "current" variable is used for calculating the seek destination,
and was declared as "int". With very long song files, the 32 bit
integer can overflow. ffmpeg expects an int64_t, which is very
unlikely to overflow. Switch to int64_t.
If avcodec_decode_audio2() returns no output for an AVPacket,
libavcodec may buffer some data, and return a larger chunk of output
later. This patch disables a lot of bogus warnings.
The shout_mp3 encoder had two bugs: when no song was ever played, MPD
segfaulted during cleanup. Second bug: memory leak, each time the
shout device was opened, lame_init() was called again, and
lame_close() is only called once during shutdown.
Fix this by shutting down LAME each time the clear_encoder() method is
called.
Hi -
independently of libmikmod's other problems - there seems
to be a problem in mpd's wrapper: MikMod_Exit() is called
after the first file is decoded, which frees some ressources
within the mikmod library. An attempt to play a second file
leads to a crash. The appended patch fixes this for me.
(I don't know what the "dup" entry is good for - someone
who knows should review that too.)
best regards
Matthias
[mk: removed 3 more MikMod_Exit() invocations]
When the user configures a music_directory with a trailing slash, it
may break playlist loading, because MPD expects a double slash. Chop
off the trailing slash.
ffmpeg_tag_internal() does not look for a few tags that mpd
supports. Most noteably:
comment -> TAG_ITEM_COMMENT -> Description
genre -> TAG_ITEM_GENRE -> WM/Genre (not WM/GenreID)
year -> TAG_ITEM_DATE -> WM/Year
I *think* that this is the last of the tags that AVFormatContext() in
ffmpeg supports that mpd also uses.
This patch implements the MMS protocol, by using libmms. It is quite
experimental: it does not support seeking yet, and it is currently
using synchronous I/O, which causes MPD to hang while waiting for the
server.
When waiting for free space in the ring buffer, the JACK plugin
sleeped 10ms until there is enough space. This delay was too large
for low-latency setups (<10ms), and created a lot of xruns. Work
around that by reducing the sleep time to 1ms.
A proper solution for this would be to use an event based approach,
and we will do it, just not now.
When the connection failed once, you had to restart MPD, because it
never cleared the jack_data.shutdown flag. Instead, it refused to
play anything "because there is no client thread" (which is wrong at
that point).
Added all important id tags from the MusicBrainz wiki:
http://musicbrainz.org/doc/MusicBrainzTag
This should automatically enable its suport in the vorbis and flac
decoder plugins.
When the random mode is toggled, MPD did not clear the queue. Because
of this, MPD continued with the next (random or non-random) song
according to the previous mode. Clear the queued song to fix that.
The null plugin synchronizes the playback so it will happen in real
time. This patch adds a configuration option which disables this: the
playback will then be as fast as possible. This can be useful to
profile MPD.
Max Kellermann