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* 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.
* | output/jack: use jack_client_open() instead of jack_client_new()Max Kellermann2009-11-051-0/+1
| | | | | | | | | | | | | | | | jack_client_new() is deprecated. This requires libjack 0.100 (released nearly 5 years ago). We havn't been testing older libjack versions anyway. As a side effect, there is the new option "autostart".
* | output/jack: added option "client_name"Max Kellermann2009-11-051-0/+1
| | | | | | | | | | | | Instead of using MPD's audio output name (setting "name"), use a separate configuration option. Change the default to "Music Player Daemon".
* | database: rescan after metadata_to_use changeMax Kellermann2009-11-041-0/+1
| | | | | | | | | | | | Store a list of supported tag items in the database. When loading a database which does not have a matching list, we must rescan in order to get the missing information.
* | text_file: allocate line buffers dynamicallyMax Kellermann2009-11-011-0/+1
| | | | | | | | | | | | | | Use a single GString buffer object in all functions loading the database. Enlarge it automatically for long lines. This eliminates the maximum line length for tag values. There is still an upper limit of 512 kB to prevent denial of service, but that's reasonable I guess.
* | Merge branch 'v0.15.x'Max Kellermann2009-10-311-0/+7
|\| | | | | | | | | | | | | | | Conflicts: NEWS configure.ac src/decoder/ffmpeg_plugin.c src/update.c
| * decoder/ffmpeg: convert metadataMax Kellermann2009-10-281-0/+2
| | | | | | | | | | | | Convert the metadata with the libavformat function av_metadata_conv(). This ensures that canonical tag names are provided by libavformat, and we can remove the "artist" vs "author" workaround.
| * update: delete ignored symlinks from databaseMax Kellermann2009-10-271-0/+1
| | | | | | | | | | | | When you disable the "follow_outside_symlinks" or the "follow_inside_symlinks" setting, the next update should remove the now-ignored files from the database.
| * output_thread: check again if output is open on PAUSEMax Kellermann2009-10-211-0/+1
| | | | | | | | | | Basically the same as the 0.15.5 patch "check again if output is open on CANCEL". Same race condition, same fix.
| * Modify version string to post-release version 0.15.6~gitAvuton Olrich2009-10-181-0/+3
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* | songvec: sort songs by album name first, then disc/track numberMax Kellermann2009-10-311-0/+1
| | | | | | | | | | When the songs of two albums are in the same directory, all songs of an album should be right next to each others.
* | output: consistently lock audio output objectsMax Kellermann2009-10-291-0/+1
| | | | | | | | | | Always keep the audio_output object locked within the output thread, unless a plugin method is called. This fixes several race conditions.
* | output/alsa: don't recover on CANCELMax Kellermann2009-10-291-0/+1
| | | | | | | | | | | | The recovery is for nothing if we get CLOSE afterwards. Let's not recover in the cancel() method, and let the next play() call sort it out.
* | playlist: new ASX playlist pluginMax Kellermann2009-10-211-1/+1
| | | | | | | | Based on the XSPF playlist plugin.
* | audio_format: wildcards allowed in audio_format configurationMax Kellermann2009-10-211-0/+1
| | | | | | | | | | | | An asterisk means that this attribute should not be enforced, and stays whatever it used to be. This way, some configuration values work like masks.
* | output/jack: connect to server on MPD startupMax Kellermann2009-10-211-0/+1
| | | | | | | | | | .. and keep up the JACK connection while MPD runs. Allocate the ring buffers on the first open, and free them at MPD exit.
* | output/jack: implement the "pause" methodMax Kellermann2009-10-211-0/+1
| | | | | | | | Don't disconnect from JACK during pause.
* | pulse: code rewrite using the asynchronous libpulse APIMax Kellermann2009-10-211-0/+1
| | | | | | | | | | | | | | | | This is a complete rewrite of the PulseAudio output plugin. It uses the asynchronous API, which gives us more control over everything. Additionally, it connects to the PulseAudio server on startup, and keeps this connection up while MPD runs. During pause, instead of closing the stream, it enables "cork".
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