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.
Max Kellermann