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* alsa_mixer: add mixer_index optionDaniel Kahn Gillmor2009-04-281-0/+2
| | | | This allows you to select controls with duplicate names.
* curl: moved proxy settings to "input" blockMax Kellermann2009-04-251-0/+1
| | | | | The old global settings "http_proxy_host", "http_proxy_port", "http_proxy_user" and "http_proxy_password" continue to work.
* player_thread: pause when all audio outputs fail to playMax Kellermann2009-04-251-0/+1
| | | | | | When all audio outputs have been closed due to failures, pause the playback instead of stopping it. This way, the user may resume at the current position after the problem has been dealt with.
* NEWS: flagged some new features "experimental"Max Kellermann2009-04-021-2/+3
| | | | | | | | The "lastfm" input plugin is far from complete, because MPD does not support nesting playlists yet. The "fluidsynth" decoder plugin suffers from shortcomings in the libfluidsynth library: http://www.mail-archive.com/fluid-dev@nongnu.org/msg01099.html
* news: add more miscellaneous missing newsAvuton Olrich2009-04-011-3/+12
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* news: move supports a rangeAvuton Olrich2009-04-011-1/+1
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* news: add pulseaudio mixerAvuton Olrich2009-04-011-1/+3
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* news: Add consume and single commandsAvuton Olrich2009-04-011-2/+5
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* news: trivial modification to make the date line the same as the othersAvuton Olrich2009-03-311-1/+1
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* mpcdec: support the new libmpcdec SV8 APIMax Kellermann2009-03-271-0/+1
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* solaris: new audio output plugin for Solaris /dev/audioMax Kellermann2009-03-161-0/+1
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* pcm_resample: choose the fallback resampler at runtimeMax Kellermann2009-03-141-0/+1
| | | | | | | | | Even if libsamplerate support is enabled, compile the fallback resampler. When the user specifies the option "samplerate_converter=internal", it is chosen in favor of libsamplerate. This may help users with a weak FPU who don't want to compile a custom MPD from source, because the fallback resampler does not use floating point operations.
* player_thread: fill buffer after seekingMax Kellermann2009-03-101-0/+1
| | | | | | After a seek, wait until enough new chunks are decoded before starting playback. If this takes too long, send silence chunks to the audio outputs meanwhile.
* output_thread: wait 10 seconds before reopening after play failureMax Kellermann2009-03-091-0/+1
| | | | | | This is similar to the MPD 0.14 patch "wait 10 seconds before reopening a failed device", which only covered open() failures. This patch adds the same feature for play().
* alsa: better period_time default value for high sample ratesMax Kellermann2009-03-081-0/+1
| | | | | | | | | | | | | | | | | | The default values for buffer_time and period_time were both capped by the hardware limits on practically all chips. The result was a period_time which was half as big as the buffer_time. On some chips, this led to lots of underruns when using a high sample rate (192 kHz), because MPD had very little time to send new samples to ALSA. A period time which is one fourth of the buffer time turned out to be much better. If no period_time is configured, see how much buffer_time the hardware accepts, and try to configure one fourth of it as period_time, instead of hard-coding the default period_time value. This is yet another attempt to provide a solution which is valid for all sound chips. Using the SND_PCM_NONBLOCK flag also seemed to solve the underruns, but put a lot more CPU load to MPD.
* cmdline: Print available protocols when --version is run.Avuton Olrich2009-03-031-0/+1
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* input_lastfm: new input plugin for last.fm radioMax Kellermann2009-03-021-0/+1
| | | | | | | The lastfm input plugin enables MPD to play lastfm:// URLs. This plugin is not complete yet: it plays only the first song in the last.fm playlist, and the playlist parser isn't even implemented properly.
* tag_id3: parse ID3 tags in RIFF/WAV filesMax Kellermann2009-03-021-0/+1
| | | | | Added a small RIFF parser library. Look for an "id3" chunk, and let libid3tag parse it.
* audio_format: allow 32 bit samplesMax Kellermann2009-03-021-0/+1
| | | | | | This is the first patch in a series to enable 32 bit audio samples in MPD. 32 bit samples are more tricky than 24 bit samples, because the integer may overflow when you operate on a sample.
* flac: parse stream tagsMax Kellermann2009-03-011-0/+1
| | | | | Parse the vorbis comments in libflac's metadata_callback and pass them as tag struct to the decoder API.
* audiofile: added 24 bit supportMax Kellermann2009-03-011-0/+1
| | | | | Don't hard code the "bits" parameter to 16. Try to use the input's sample format, if possible.
* output: added option to disable audio outputs by defaultMax Kellermann2009-02-281-0/+1
| | | | | The option "enabled" is on by default. If you specify "enabled no" in an audio_output section, then this device is disabled by default.
* pipe: new audio output plugin which runs a commandMichal Nazarewicz2009-02-281-0/+1
| | | | [mk: adapted to new output plugin API]
* listen: allocate sockaddr_storage struct for accept()Max Kellermann2009-02-281-0/+1
| | | | | | The generic sockaddr struct is too small for some addresses. For accept(), we have to allocate a sockaddr_storage struct on the stack, which is large enough for all addresses.
* song_print: hide HTTP password in playlistMax Kellermann2009-02-271-0/+1
| | | | | | | Added the uri_remove_auth() library function which strips username and password from a HTTP URI, and use it in song_print_url(). This allows you to add HTTP URIs to the playlist including secret username and password, without disclosing it to all MPD clients.
* mvp: fixed default device detectionMax Kellermann2009-02-261-0/+1
| | | | | The check "open()!=0" is wrong, you have to write "open()>=0", because -1 means error, and 0 is a valid file handle.
* mvp: check for reopen errorsMax Kellermann2009-02-251-0/+1
| | | | | When the MVP device has been closed in the cancel() method, and the play() method attempts to reopen it, check for errors.
* mvp: fall back to 16 bit audio samplesMax Kellermann2009-02-251-0/+1
| | | | | | Looks like the MVP audio output only supports 16 and 24 bit audio samples. If MPD generates any other sample formats, force it to use 16 bit.
* mvp: fall back to stereoMax Kellermann2009-02-251-0/+1
| | | | When the channel count is greater than 2, fall back to stereo sound.
* decoder_list: added configuration option to disable decoder pluginsMax Kellermann2009-02-151-0/+1
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* daemon: ignore "user" setting if already running as that userMax Kellermann2009-02-151-0/+1
| | | | | | If mpd.conf specifies a user, and MPD is invoked by exactly this user, ignore the "user" setting. Don't bother to look up its groups and don't attempt to change uid, it won't work anyway.
* mpd version 0.14.2Avuton Olrich2009-02-141-3/+1
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* update: free deleted subdirectoriesMax Kellermann2009-02-121-0/+1
| | | | | | Use delete_directory() for removing sub directories instead of dirvec_clear(). This ensures that all memory occupied by subdirectories of deleted directories is freed.
* update: recursively purge deleted directoriesMax Kellermann2009-02-121-0/+1
| | | | | | | When a directory is deleted, MPD deleted only the directory from the database; it did not bother to walk the full tree to free all memory and to remove deleted songs from the playlist. Replace a dirvec_delete() with delete_directory().
* aac: fix stream metadataMax Kellermann2009-02-121-0/+1
| | | | | Pass the input_stream object to decoder_data(). Without it, the MPD core does not see stream tags.
* wildmidi: new decoder plugin for MIDI filesMax Kellermann2009-02-121-0/+1
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* fluidsynth: new decoder plugin for MIDI filesMax Kellermann2009-02-121-0/+1
| | | | | | | | | | | | | | There are a few problems left in this plugin: - fluidsynth decodes in real time, while MPD prefers to buffer as quickly as possible; as a workaround, this plugin uses a timer object to synchronize with real-time playback - I don't know yet how fluidsynth tells me when the song has ended - the "soundfont" configuration setting is not yet documented, and it will likely change soon (in favor of a per-decoder configuration block)
* sidplay: new decoder plugin for playing C64 SID filesMax Kellermann2009-02-111-0/+1
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* ffmpeg: added TTA supportMax Kellermann2009-02-111-1/+2
| | | | | The ffmpeg library supports the "True Audio Codec". The entry in ffmpeg_suffixes was missing.
* configure.ac: define HAVE_FFMPEG after all checksMax Kellermann2009-02-091-0/+2
| | | | | | Don't define HAVE_FFMPEG if the ffmpeg libraries were found via pkg-config, but ffmpeg support was disabled because avcodec_decode_audio2() is not available.
* shout: clear buffer before calling the encoderMax Kellermann2009-02-091-0/+1
| | | | | | | Always assume the buffer is empty before calling the encoder. Always flush the buffer immediately after there has been added something. This reduces the risk of buffer overruns, because there will never be a "rest" in the current buffer.
* shout: don't postpone metadataMax Kellermann2009-02-091-0/+1
| | | | | Don't duplicate the tag received by the send_metadata() method - send it to the shout server directly.
* shout: use libshout's synchronizationMax Kellermann2009-02-091-0/+1
| | | | | Removed the manual timer synchronization from the shout plugin. libshout's shout_sync() function does it for us.
* shout: switch to blocking modeMax Kellermann2009-02-091-0/+1
| | | | | | The non-blocking mode of libshout is sparsely documented, and MPD's implementation had several bugs. Also removed connect throttling code, that is done by the MPD core since 0.14.
* ffmpeg: fixed seek integer overflowMax Kellermann2009-02-031-0/+1
| | | | | | | The "current" variable is used for calculating the seek destination, and was declared as "int". With very long song files, the 32 bit integer can overflow. ffmpeg expects an int64_t, which is very unlikely to overflow. Switch to int64_t.
* ffmpeg: check if the time stamp is validMax Kellermann2009-02-031-0/+1
| | | | | | When ffmpeg cannot estimate the elapsed time, it sets AVPacket.pts=AV_NOPTS_VALUE. Our ffmpeg decoder plugin did not check for that special value.
* ffmpeg: don't warn of empty packet outputMax Kellermann2009-02-031-0/+1
| | | | | | If avcodec_decode_audio2() returns no output for an AVPacket, libavcodec may buffer some data, and return a larger chunk of output later. This patch disables a lot of bogus warnings.
* shout_mp3: call lame_close() in clear_encoder() methodMax Kellermann2009-02-021-0/+1
| | | | | | | | | | The shout_mp3 encoder had two bugs: when no song was ever played, MPD segfaulted during cleanup. Second bug: memory leak, each time the shout device was opened, lame_init() was called again, and lame_close() is only called once during shutdown. Fix this by shutting down LAME each time the clear_encoder() method is called.
* player_thread: set player error when output device failsMax Kellermann2009-02-021-0/+1
| | | | | | When the output device fails to play a chunk, set pc.error to PLAYER_ERROR_AUDIO. This way, the playlist knows that it should not queue the next song.
* mikmod: call MikMod_Exit() only in the finish() methodMatthias Drochner2009-01-301-0/+1
| | | | | | | | | | | | | | | Hi - independently of libmikmod's other problems - there seems to be a problem in mpd's wrapper: MikMod_Exit() is called after the first file is decoded, which frees some ressources within the mikmod library. An attempt to play a second file leads to a crash. The appended patch fixes this for me. (I don't know what the "dup" entry is good for - someone who knows should review that too.) best regards Matthias [mk: removed 3 more MikMod_Exit() invocations]
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