| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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It used to ignore the decoder_data() return value.
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paused
Use a shortcut in playlist_seek_song(), don't call
playlist_play_order() because that would reset the "paused" state.
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Don't calculate the song duration when the sample rate is 0 (division
by zero crash).
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Keep those when scanning for empty directories. The check in
playlist_vector_is_empty() was missing.
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g_path_get_dirname() returns "." when there is no directory name in
the given path. This patch adds a workaround for that.
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avcodec_decode_audio3() has been added in libavformat 52.25.0, and the
predecessor avcodec_decode_audio2() has been deprecated.
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fixes build with lavc 53.
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Conflicts:
NEWS
configure.ac
src/listen.c
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Otherwise OGGs can't be played.
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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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Use audio_format_mask_valid() to verify a mask.
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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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A bit of automake magic (see info automake "Per-Object Flags").
Compile-tested.
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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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