| Commit message (Collapse) | Author | Age | Files | Lines |
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This patch seems a bit ugly, maybe it would be a bit cleaner with gotos.
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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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For ffmpeg < 0.5. Copied from libavutil 0.5.
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Conflicts:
NEWS
configure.ac
src/listen.c
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Thanks to clang for complaining.
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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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Fix clang warning.
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Use audio_format_mask_valid() to verify a mask.
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Call print_playlist_result() instead of casting the enum implicitly.
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Fixes build failure on WIN32.
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Fix compiler warning.
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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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<stdbool.h> needs to be included unconditionally from definition of
NDEBUG, since »bool« doesn't get defined otherwise.
Signed-off-by: Andreas Wiese <aw-devel@meterriblecrew.net>
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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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Remove trailing whitespace found by this command:
find -name '*.[ch]' | xargs grep "[[:space:]]$"
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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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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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Cast the constant to dev_t, not to unsigned.
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Fixes the gcc warning "implicit declaration of function 'htons'".
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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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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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Loop over all frames with a specific id, and import all of them - not
just the first one (index 0).
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