This has been replaced by the last.fm playlist plugin.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.
Qball Cow