| Commit message (Collapse) | Author | Age | Files | Lines |
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This patch fixes several imports to use pkg-config instead of certain
esoteric tests.
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string_toupper() and strDupToUpper() were not able to deal with
character sets other than US-ASCII. Use GLib's g_utf8_casefold()
for strings.
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Moved the musicDir variable and its initialization code from path.c to
mapper.c.
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When the music directory is not mounted yet, let MPD start anyway.
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GLib provides an easier API for character set conversion than iconv().
Use g_convert() / g_convert_with_fallback() for all character
conversions. We should optimize the path.h API later to return a
newly allocated buffer, so we can just pass GLib's return value.
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GLib is a nice and portable utility library. We are going to use it
from now on, and eliminate a lot of duplicated code from MPD. Why
invent the wheel again and again?
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Indentation was broken in tag_id3.c: it used 4 spaces instead of tabs.
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Use memchr() instead of manually traversing the input buffer. Update
the client's properties after all commands have been processed. Check
for buffer overflow once.
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The caller already knows the protocol family, and we can eliminate the
complicated switch statement in establishListen() if we just pass this
information. This seems more robust.
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getaddrinfo() is more robust and has proper IPv6 support. The new
code tries to bind to all IP addresses returned by getaddrinfo().
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"idle" waits until something noteworthy happens on the server,
e.g. song change, playlist modified, database updated. This allows
clients to keep up to date without polling.
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These macros are not available in older libasound versions (1.0.13
fails, 1.0.16 is ok). Ignore the configuration if the constants are
not defined.
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Instead of the uninitialized local variable "s", I should have used
"uri".
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The OSS constants overflow a signed integer, use unsigned instead.
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C enums are nicer than CPP macros.
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Added mpd.conf options for disabling automatic resamling, sample
format and channel conversion. This way, users may choose to override
ALSA's automatic resampling, and use libsamplerate instead.
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This git branch has become a real MPD fork now. Time to change the
package name to the code name "mpd-mk". Set the version number to
"0.14~git" to mark this as a non-released version.
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Don't follow relative symlinks which point into the music directory.
This allows you to organize music with symbolic links, without MPD
managing separate copies of each song.
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The mapper code has replaced these functions.
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The mapper library maps directory and song objects to file system
paths. With this central library, the code mixture in path.c should
be cleaned up, and we will be able to add neat features like aliasing.
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playlist_print_song() and playlist_print_uri() handle charset
conversion and (optional) music directory prefixing.
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song_get_url() doesn't modify the song object.
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Found by Valgrind while looking for another bug...
Hmm.. I should really just make this code generic since
they're duplicated...
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isMusic() used to be a very inefficient function: with every
invocation, it did another stat() on the specified file. There is
only one caller, do the stat() there manually and use hasMusicSuffix()
instead of isMusic().
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By always creating the parent directory, we can use delete_name_in()
without further lookups. The parents which may non exist will be
pruned later. An update request for a non-existing or empty directory
should be quite unusual, so this doesn't add any measurable overhead.
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In order to optimize buffer usage, pass only the base file name to
updateInDirectory(). This way, updateInDirectory() may choose when to
allocate a larger buffer for the full path.
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delete_name_in() is similar to delete_path(), but it does not need to
look up the parent directory.
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It is invalid to pass a path with the wrong dirname to dirvec_find().
To be able to find a subdirectory only by its basename, compare only
the basename of both paths.
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directory_get_name() returns the base name of the directory.
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The only caller of deletePlaylist() appends PLAYLIST_FILE_SUFFIX, so
we can be sure it's already there. We don't need to stat the file,
since unlink() does all the checking.
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Commit 80a2c937 broke resume after pause: it cleared the
input_audio_format when it attempted to simplify a complicated
expression. Don't clear it, just assign input_audio_format if a new
format was specified.
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The inline functions directory_is_empty() and directory_get_path()
don't modify the object - pass constant object pointers to them.
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We only need to lock sv->nr changes to prevent traversals ( why
it's called "nr_lock"). free(3) is a "slow" function on my
system; so we can avoid unnecessarily holding a lock long for
longer than needed.
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Previously only updates with subdirectories being
specified could be queued. No harm in queueing
full updates.
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This way we avoid unnecessary heap allocations.
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Not all compilers support struct packing, and those that don't
shouldn't be punished for it.
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Added configuration parameter "protocol" which lets the user choose
from 3 shout protocols. This adds support for real shoutcast servers.
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Case insensitivity isn't helpful, and comparing only the first 3 bytes
of a configured value may encourage users to supply wrong or
misleading values.
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If the sample format isn't supported by the device (i.e. 24 bit on
low-end sound chips), fall back to 16 bit output. There is code in
pcm_utils.c which converts PCM data to 16 bit.
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When calculating the conversion buffer size, don't hard-code the
formulas for only mono<->stereo.
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Convert any number of channels to stereo. In fact, this isn't really
stereo, it's rater mono blown up to stereo. This patch should only
make it possible to play 5.1 files at all; "real" conversion to stereo
should be implemented, but for now, this is better than nothing.
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In order to be able to deal with non-trivial conversions,
pcm_convertChannels() needs to know both the input and the output
channel count. Simplify buffer allocation in that function.
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Moved code from pcm_convertChannels() to pcm_convert_channels_1_to_2()
and pcm_convert_channels_2_to_1(). Improved the quality of
pcm_convert_channels_2_to_1() by calculating the arithmetic mean value
of both samples.
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Move code which loads configuration to alsa_configure(). This removes
one indent level.
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Pass int16_t pointers instead of char pointers to functions which can
deal with 16 bit audio only.
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buffered_before_play was copied to struct player because it was used
to disable buffering when seeking. Instead of mainaining a copy of
this number, move just the flag to the player struct.
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Renamed audio_configFormat to configured_audio_format. Renamed
audio_buffer.format to input_audio_format. Simplified its
initialization in openAudioDevice().
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