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* playlist/cue: fill song.start_ms, .end_msMax Kellermann2009-12-271-0/+1
| | | | This patch adds practical usefulness to the CUE playlist plugin.
* output_init: use the normalize filter pluginMax Kellermann2009-12-251-0/+1
| | | | | | Use the plugin instead of the glue code in normalize.c. This is used wrapped inside a "autoconv" filter, to enable normalization for all input file formats.
* archive/iso: renamed plugin to "iso9660"Max Kellermann2009-12-161-0/+1
| | | | Based on libiso9660.
* archive/zip: renamed plugin to "zzip"Max Kellermann2009-12-151-0/+2
| | | | This plugin is based on libzzip.
* Merge branch 'v0.15.x'Max Kellermann2009-12-151-0/+6
|\ | | | | | | | | | | | | | | Conflicts: src/archive/bz2_plugin.c src/archive_api.h src/input/file_input_plugin.c test/run_input.c
| * iso, zip: fixed memory leak in destructorMax Kellermann2009-12-151-0/+1
| | | | | | | | Free the "context" pointer in the method archive_plugin.close().
| * archive: close archive when stream is closedMax Kellermann2009-12-151-0/+2
| | | | | | | | | | | | | | | | Fixes a memory leak: the "archive" input plugin opens the archive, but never closes it. This patch moves the responsibility for doing that to archive_plugin.open_stream(). This is an slight internal API change, but it is the simplest and least intrusive fix for the memory leak.
| * input/archive: close the archive file on errorMax Kellermann2009-12-151-0/+1
| | | | | | | | Fixed memory leak in error handler.
| * input/file: don't fall back to parent directoryMax Kellermann2009-12-151-0/+2
| | | | | | | | | | This code has never made any sense, and has broken some of the archive plugin.
* | Merge branch 'v0.15.x'Max Kellermann2009-12-141-0/+8
|\| | | | | | | | | Conflicts: src/decoder/ffmpeg_plugin.c
| * mixer: explicitly close all mixers on shutdownMax Kellermann2009-12-081-0/+1
| | | | | | | | | | | | Mixers with the "global" flag set aren't closed automatically when the output device is closed. Thus, they might still be open when MPD shuts down.
| * mapper: apply filesystem_charset to playlistsMax Kellermann2009-12-081-0/+1
| | | | | | | | | | | | | | | | This fixes an inconsistency in the stored playlist subsystem: when obtaining the list of playlists (listplaylist, listplaylistinfo), the file names in the playlist directory are converted to UTF-8 (according to filesystem_charset), but when saving or loading playlists, the filesystem_charset setting was ignored.
| * command: verify playlist name in the "rm" commandMax Kellermann2009-12-081-0/+1
| | | | | | | | Call spl_valid_name() in spl_delete().
| * mapper: fix memory leak when playlist_directory is not setMax Kellermann2009-12-081-0/+1
| | | | | | | | Don't allocate the file name before the playlist_dir==NULL check.
| * tag_id3: fix ID3v1 charset conversionsvitoos2009-11-301-0/+2
| | | | | | | | | | If we define id3v1_encoding, then the tags are not added to the database.
| * ffmpeg: don't try to force stereoMax Kellermann2009-11-301-0/+2
| | | | | | | | | | | | The plugin code tried to force libavcodec to supply stereo samples. That however has never actually worked. By removing this code, we are able to play surround files for the first time.
* | compress: upgraded to AudioCompress 2.0J. Shagam2009-12-021-0/+1
| | | | | | | | | | | | | | | | Copied sources from http://beesbuzz.biz/code/audiocompress/AudioCompress-2.0.tar.gz [mk: created this patch under fluffy's name and fixed some gcc signed/unsigned comparison warnings]
* | pcm_mix: implemented 32 bit supportMax Kellermann2009-11-191-1/+1
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* | pcm_volume: implemented 32 bit supportMax Kellermann2009-11-191-0/+1
| | | | | | | | Support 32 bit samples with software mixer.
* | Merged release 0.15.6 from branch 'v0.15.x'Max Kellermann2009-11-191-1/+8
|\| | | | | | | | | | | | | Conflicts: NEWS configure.ac
| * Modify version string to post-release version 0.15.7~gitAvuton Olrich2009-11-181-0/+3
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| * mpd version 0.15.6release-0.15.6Avuton Olrich2009-11-181-1/+1
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| * decoder/flac: fixed NULL pointer dereference in CUE codeMax Kellermann2009-11-181-0/+1
| | | | | | | | The function flac_vtrack_tnum() was missing a strrchr()==NULL check.
| * id3: allow 4 MB RIFF/AIFF tagsMax Kellermann2009-11-151-0/+2
| | | | | | | | | | | | | | Allow RIFF/AIFF ID3 tags up to 4 MB (old limit was 256 kB). This might still be too small for some users, and when somebody complains, we might do something more clever (like streaming the data into libid3tag?).
| * decoder/ffmpeg: align the output bufferMax Kellermann2009-11-151-0/+1
| | | | | | | | | | | | On some platforms, libavcodec wants the output buffer aligned to 16 bytes (because it uses SSE/Altivec internally). It will segfault when you don't obey this rule.
* | encoder: introducing flac encoder pluginViliam Mateicka2009-11-171-0/+2
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* | decoder/mikmod: sample rate is configurableMax Kellermann2009-11-141-0/+1
| | | | | | | | The new option "sample_rate" sets the sample rate for libmikmod.
* | decoder/wavpack: allow more than 2 channelsMax Kellermann2009-11-111-0/+1
| | | | | | | | | | Remove the OPEN_2CH_MAX option. MPD's support for surround sound is still clunky, but we're working on it.
* | decoder/wavpack: activate 32 bit supportMax Kellermann2009-11-111-0/+1
| | | | | | | | | | | | | | | | | | | | MPD has been supporting 32 bit samples since version 0.15. This patch changes one check, and removes the 32->24 conversion code. Note that WavPack floating point samples have 32 bits, and MPD doesn't have a special check for floating point - therefore, this WavPack plugin still returns 24 bit integer samples as before (until we have float support in the MPD core).
* | Merge remote branch 'origin/v0.15.x'Max Kellermann2009-11-111-0/+2
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| * decoder/flac: fixed CUE seeking range checkMax Kellermann2009-11-111-0/+1
| | | | | | | | | | | | If flac_container_decode() gets a seek destination which is out of range, it ignores the SEEK command (never finishes it). This leads to MPD lockup, because the player thread waits for completion.
| * oggflac: rewind stream after FLAC detectionMax Kellermann2009-11-111-0/+1
| | | | | | | | | | The oggflac plugin has been completely broken for quite a while and nobody has noticed - maybe we should remove it?
* | configure.ac: require GLib 2.12Max Kellermann2009-11-101-1/+1
| | | | | | | | | | | | Drop the required GLib version from 2.16 to 2.12, because many current systems still don't have GLib 2.16. This requires several new compatibility functions in glib_compat.h.
* | Merge branch 'v0.15.x'Max Kellermann2009-11-101-0/+7
|\| | | | | | | | | | | Conflicts: src/input/lastfm_input_plugin.c src/song_save.c
| * zzip: require libzzip 0.13Max Kellermann2009-11-101-0/+2
| | | | | | | | We need the function zzip_file_stat().
| * input/mms: require libmms 0.4Max Kellermann2009-11-101-0/+1
| | | | | | | | We're using API functions which are not available in 0.3.
| * sticker: added fallback for sqlite3_prepare_v2()Max Kellermann2009-11-101-0/+1
| | | | | | | | This function was not present in SQLite < 3.4.
| * input/lastfm: fixed variable name in GLib<2.16 code pathMax Kellermann2009-11-101-0/+2
| | | | | | | | Should be "lastfm_user", not "lastfm_username".
| * song_save: increased maximum line length to 32 kBMax Kellermann2009-11-011-0/+1
| | | | | | | | | | | | | | The line buffer had a fixed size of 5 kB, and was allocated on the stack. This was too small for some users. As a hotfix, we're increasing the buffer size to 32 kB now, allocated on the heap. In MPD 0.16, we'll switch to dynamic allocation.
* | output/alsa: fill period buffer with silence before drainingMax Kellermann2009-11-091-0/+1
| | | | | | | | | | | | | | | | | | ALSA passes full period buffers to the hardware. If an application doesn't finish writing a period, libasound will nonetheless send the partial buffer (with undefined trailing data). This causes noise at the end of playback. This patch attempts to track the current position within the period buffer, and generates silence at the end, before calling snd_pcm_drain().
* | player_thread: drain audio outputs at the end of the playlistMax Kellermann2009-11-091-0/+2
| | | | | | | | | | | | | | | | When there's no queued song, and the current one has finished playing, first make sure that the hardware outputs have really finished playing the last chunk: call the drain() method in all audio outputs. Without this patch, MPD stopped playback shortly before the ALSA sound card had finished playing.
* | set the close-on-exec flag on all file descriptorsMax Kellermann2009-11-071-0/+1
| | | | | | | | | | | | | | | | | | | | | | Added the "fd_util" library, which attempts to use the new thread-safe Linux system calls pipe2(), accept4() and the options O_CLOEXEC, SOCK_CLOEXEC. Without these, it falls back to FD_CLOEXEC, which is not thread safe. This is particularly important for the "pipe" output plugin (and others, such as JACK/PulseAudio), because we were heavily leaking file descriptors to child processes.
* | log: redirect stdout/stderr to /dev/null if syslog is usedMax Kellermann2009-11-071-0/+1
| | | | | | | | | | Don't hold a file descriptor on root's tty when syslog is used for logging.
* | output/jack: added option "server_name"Max Kellermann2009-11-071-1/+1
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* | output/jack: dynamic source port listMax Kellermann2009-11-061-1/+1
| | | | | | | | | | | | Same as the previous patch: create up to 16 configured source ports. The plugin tries to do its best at guessing the right combination for the given input file, the number of source and destination ports.
* | output/jack: renamed option "ports" to "destination_ports"Max Kellermann2009-11-061-0/+1
| | | | | | | | Be more clear which kind of port should be configured here.
* | playlist: added extm3u pluginMax Kellermann2009-11-061-1/+1
| | | | | | | | | | This new plugin parses extm3u files. Files without the "#EXTM3U" header are still parsed by the plain old "m3u" plugin.
* | output/httpd: bind port when output is enabledMax Kellermann2009-11-051-1/+2
| | | | | | | | | | | | Implement the methods enable() and disable(). Bind the HTTP port in the enable() method, but reject all incoming connections until the output is opened.
* | output/jack: support mono inputMax Kellermann2009-11-051-0/+1
| | | | | | | | | | When MPD plays a mono song (audio_format.channel==1), connect only one source port to both destination ports.
* | output/jack: clear ring buffers before activatingMax Kellermann2009-11-051-0/+1
| | | | | | | | | | | | After playback has stopped, the ring buffers may still contain samples. These will be played when playback is started the next time. We should clear the buffers each time.
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