| Commit message (Collapse) | Author | Age | Files | Lines |
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Without the ogg_stream_reset() call, the "e_o_s" flag never gets
reset, and libogg writes EOS packets over and over.
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Conflicts:
NEWS
configure.ac
src/output/jack_plugin.c
src/update.c
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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.
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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.
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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".
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See code comment.
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According to the Solaris dsp manpage, AFMT_S24_PACKED is
little-endian:
http://download.oracle.com/docs/cd/E19963-01/821-1475/6nmf5baot/index.html
The Minix soundcard.h header says the same.
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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)
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this needs to be done for the end of songs to be detected.
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When the configure options were moved around for 0.16, the order was
changed, and the Tremor check broke.
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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."
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When set, MPD will not auto-start playback on startup; it will be in
"paused" state.
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Added a new optional parameter for the shout plugin called "url".
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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.
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Remote Audio Output Protocol (RAOP), for Apple devices.
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Conflicts:
NEWS
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A bit of automake magic (see info automake "Per-Object Flags").
Compile-tested.
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Conflicts:
NEWS
configure.ac
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Conflicts:
NEWS
configure.ac
src/directory.h
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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.
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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.
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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.
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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.
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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.
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It is known to crash instantly.
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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
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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
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Loop over all frames with a specific id, and import all of them - not
just the first one (index 0).
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One APE tag may contain more than one value, separated by null bytes.
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Based on the APE reader.
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