Many years ago, FAAD had a serious ABI bug: the NeAACDecInit()
prototype in its header declared the "samplerate" parameter to be
"unsigned long *", but internally, the function assumed it was
"uint32_t *" instead. On 32 bit machines, that was no difference, but
on 64 bit, this left one portion of the return value uninitialized;
and worse, on big-endian, the wrong word was filled. This bug had to
be worked around in MPD (commit 9c4e97a6).
A few months later, the bug was fixed in the FAAD CVS in commit 1.117
on file libfaad/decoder.c; the commit message was:
"Use public headers internally to prevent duplicate declarations"
The commit message was too brief at best; the problem was not
duplicate declarations, but a prototype mismatch. No mention of the
bug fix in the ChangeLog.
The MPD project never learned about this bug fix, and so MPD would
always pass a "uin32_t *" dressed up as a "unsigned long *". Nearly 6
years later, it's about time to fix this second ABI problem. Let's
kill the workaround!
Many years ago, FAAD had a serious ABI bug: the NeAACDecInit()
prototype in its header declared the "samplerate" parameter to be
"unsigned long *", but internally, the function assumed it was
"uint32_t *" instead. On 32 bit machines, that was no difference, but
on 64 bit, this left one portion of the return value uninitialized;
and worse, on big-endian, the wrong word was filled. This bug had to
be worked around in MPD (commit 9c4e97a6).
A few months later, the bug was fixed in the FAAD CVS in commit 1.117
on file libfaad/decoder.c; the commit message was:
"Use public headers internally to prevent duplicate declarations"
The commit message was too brief at best; the problem was not
duplicate declarations, but a prototype mismatch. No mention of the
bug fix in the ChangeLog.
The MPD project never learned about this bug fix, and so MPD would
always pass a "uin32_t *" dressed up as a "unsigned long *". Nearly 6
years later, it's about time to fix this second ABI problem. Let's
kill the workaround!
Pulseaudio expects clients to specify their channel-map if the
default (ALSA) map does not route the audio to the expected speakers.
Many Google results suggest dealing with this by re-routing the audio
channels with the appropriate ALSA plugin, but this will then simply
break any clients which expect the default ALSA mapping.
Virtually all media files and codecs, certainly flac, dca, a52, and of
course anything based on Microsoft's WAVEFORMAT_EXTENSIBLE specification,
assume the layout in the table here:
http://en.wikipedia.org/wiki/Surround_sound#Standard_speaker_channels
Fortunately, pulseaudio directly addresses this with a built-in channel
map for WAVE-EX which can be set automatically in the stream sample-spec.
MPD handles all strings in UTF-8 internally. Those decoders which
read Latin-1 tags are supposed to implement the conversion, instead of
passing Latin-1 to TagBuilder::AddItem(). FixTagString() is simply
the wrong place to do that, and hard-coding Latin-1 is kind of
arbitrary.
The Release Track Id uniquely identifies a recording on a release - that
is, even if a recording appears twice on a release (meaning that the
combination of recording and release id are not enough to figure out
which one it is), the release track id will allow differentiating the two.
The tag names are taken from
https://musicbrainz.org/doc/MusicBrainz_Picard/Tags/Mapping
On NetBSD, PTHREAD_MUTEX_INITIALIZER and PTHREAD_COND_INITIALIZER are
not compatible with C++11 "constexpr" (see Mantis ticket 0004110). As
a workaround, don't ues "constexpr", and use the functions
pthread_mutex_init(), pthread_mutex_destroy(), pthread_cond_init() and
pthread_cond_destroy() instead. This adds some runtime overhead, but
is portable to POSIX implementations that have awkward initializer
macros.
Casting std::numeric_limits<unsigned>::max() to "long" leads to an
overflow if sizeof(unsigned)==sizeof(long), and the result will be -1.
This happens on some 32 bit architectures, for example ARM and WIN32.
Workaround: use std::numeric_limits<int>::max(), which is the largest
signed integer. Since sizeof(long)>=sizeof(int), this will never
overflow.
Fixes Mantis ticket 0004080.
The IsActive() method returned true even if the timer was not active,
after it completed once. This broke the state file timer, and the
state file was not saved periodically.
Read one block at a time. This discards the last partial block, which
cannot be interleaved anyway. Previously, uninitialised memory was
used to interleave the last block, which generated some noise.
When the data chunk size is not a multiple of the frame size, the last
partial frame lead to an endless loop. We fix this by checking
chunk_sze>=frame instead of chunk_sze>0. This way, the partial frame
is simply skipped.
Previously, MPD tried to slurp the whole song file, count the number
of frames and calculate the song duration from that. That however is
extremely expensive for remote files, and will delay playback for a
long time. Workaround: check only the first 128 frames and try to
extrapolate from here. Fixes Mantis ticket 0004035.
Implement a "bulk" edit mode that postpones both UpdateQueuedSong()
and OnModified(). This way, the playlist version gets incremented
only once. More importantly: when adding multiple songs to a queue
that consists of only one song, the first song that got added will
always be played next. By postponing this choice, all newly added
songs get a chance to become the next song. Fixes the second (and
last) part of Mantis ticket 0004005.
Don't restore the current song after shufflung when MPD is stopped
(but still remembers the current song internally). Fixes the first
part of Mantis ticket 0004005.
Max Kellermann