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configure.ac: prepare for 0.21
2017-01-25 08:58:07 +01:00
ver 0.21 (not yet released)
client: add tag_mask attribute The "tagtypes" command now has several sub commands which can be used to edit that mask.
2017-02-07 16:52:59 +01:00
* protocol
- "tagtypes" can be used to hide tags
command/Database: add "sort" parameter to "find" and "search" Implement the second part of https://bugs.musicpd.org/view.php?id=3990
2017-02-08 09:22:15 +01:00
- "find" and "search" can sort
ClientProcess: close connection when client sends HTTP request
2014-08-12 14:39:30 +02:00
- close connection when client sends HTTP request
Merge branch 'add-original-year-tag' of git://github.com/tremby/MPD
2017-04-21 18:54:18 +02:00
* tags
- new tag "OriginalDate"
decoder/pcm: support audio/L24 Closes #31
2017-04-22 08:47:49 +02:00
* decoder
NEWS: mention the GME decoder change
2017-09-22 20:00:12 +02:00
- gme: try loading m3u sidecar files
decoder/pcm: support audio/L24 Closes #31
2017-04-22 08:47:49 +02:00
- pcm: support audio/L24 (RFC 3190)
output/alsa: non-blocking mode Use SND_PCM_NONBLOCK, and perform all snd_pcm_writei() calls in the IOThread. Use a lockless queue to copy data from the OutputThread to the IOThread. This rather major change aims to improve MPD's internal latency. All waits are now under MPD's control, instead of blocking inside libasound2. As a side effect, an output's filter is now decoupled from the actual device I/O, which solves a major latency problem with the conversion filter on slow CPUs and small period buffers. See: https://bugs.musicpd.org/view.php?id=3900
2017-01-24 23:10:35 +01:00
* output
- alsa: non-blocking mode
output/shout: use MIME type instead of the encoder plugin name This is more robust, for example it allows using the Shine encoder plugin instead of LAME.
2017-11-10 21:52:54 +01:00
- shout: support the Shine encoder plugin
output/sndio: refuse to build with libroarsndio RoarAudio's sndio emulation has been a source for annoyances. First, their headers turned out to be broken with C++, due to their use of the "new" keyword. Then they used a preprocessor macro to rename "sio_hdl" to something else, effectively disallowing the use of forward declarations. Enough is enough, and I'm removing support for it. RoarAudio users should better use the RoarAudio output plugin.
2017-09-19 19:38:15 +02:00
- sndio: remove support for the broken RoarAudio sndio emulation
NEWS: mention the sndio mixer plugin
2017-09-19 19:38:28 +02:00
* mixer
- sndio: new mixer plugin
configure.ac: prepare for 0.21
2017-01-25 08:58:07 +01:00
release v0.20.12
2017-11-25 19:32:37 +01:00
ver 0.20.12 (2017/11/25)
lib/upnp: adapt to libupnp 1.8 API changes Closes #155
2017-11-16 11:39:11 +01:00
* database
- upnp: adapt to libupnp 1.8 API changes
input/curl: call StartRequest() after setting CURLOPT_RANGE It's not possible to set CURL options after curl_easy_perform(), and thus the CURLOPT_RANGE had no effect.
2017-10-24 21:43:39 +02:00
* input
input/{cdio,ffmpeg,file,smbclient}: unlock the mutex during blocking I/O InputStream::Read() and InputStream::Seek() are called with the mutex locked. That means the implementation must not block, or unlock the mutex before calling into blocking code. Previously, a slow CD drive could stall the whole MPD process, including the main thread, due to this problem. Closes #149
2017-11-13 17:08:00 +01:00
- cdio_paranoia, ffmpeg, file, smbclient: reduce lock contention,
fixing lots of xrun problems
input/curl: call StartRequest() after setting CURLOPT_RANGE It's not possible to set CURL options after curl_easy_perform(), and thus the CURLOPT_RANGE had no effect.
2017-10-24 21:43:39 +02:00
- curl: fix seeking
decoder/vorbis: scale and clip tremor-decoded samples to 15 bits Tremor decoder is unusable since commit 2ee43c4. Sound is distorted to the point where it's nothing but noise. The data from vorbis_synthesis_pcmout() needs to be scaled and clipped for 16 bit sample size. For reference see http://lists.xiph.org/pipermail/tremor/2010-April/001642.html and http://lists.xiph.org/pipermail/vorbis/2006-October/026513.html. Signed-off-by: Marcin Jurkowski <marcin1j@gmail.com>
2017-10-29 00:59:27 +02:00
* decoder
decoder/ffmpeg, test/test_protocol: catch exceptions by reference Work around -Werror=catch-value.
2017-11-12 18:52:11 +01:00
- ffmpeg: fix GCC 8 warning
decoder/vorbis: scale and clip tremor-decoded samples to 15 bits Tremor decoder is unusable since commit 2ee43c4. Sound is distorted to the point where it's nothing but noise. The data from vorbis_synthesis_pcmout() needs to be scaled and clipped for 16 bit sample size. For reference see http://lists.xiph.org/pipermail/tremor/2010-April/001642.html and http://lists.xiph.org/pipermail/vorbis/2006-October/026513.html. Signed-off-by: Marcin Jurkowski <marcin1j@gmail.com>
2017-10-29 00:59:27 +02:00
- vorbis: fix Tremor support
player/Thread: log message when decoder is too slow
2017-10-25 20:26:09 +02:00
* player
- log message when decoder is too slow
encoder/vorbis: default to quality 3 Don't require a quality or bitrate setting. If nothing is set, don't fail startup - just go with a good default. A quality setting of 3 is what "oggenc" defaults to as well.
2017-11-14 11:30:28 +01:00
* encoder
- vorbis: default to quality 3
output/Thread: ensure pending tags are flushed in all cases Fixes hanging playback with soxr resampler. Closes #139, #141
2017-11-04 04:38:49 +01:00
* output
- fix hanging playback with soxr resampler
output/httpd: flush encoder after tag Without the flush, ReadPage() may not return any data, or not all data. This may result in incomplete ddata the new "header" page, corrupting streams with some encoders such as Vorbis. Fixes #145
2017-11-14 12:00:14 +01:00
- httpd: flush encoder after tag; fixes corrupt Vorbis stream
increment version number to 0.20.12
2017-10-24 17:31:55 +02:00
release v0.20.11
2017-10-18 10:14:46 +02:00
ver 0.20.11 (2017/10/18)
storage/curl: support Content-Type application/xml
2017-09-01 11:32:40 +02:00
* storage
- curl: support Content-Type application/xml
ffmpeg plugin: when decoded stream duration is unavailable, attempt fallback to container duration (fix MusicPlayerDaemon/MPD#110)
2017-09-18 07:44:47 +02:00
* decoder
- ffmpeg: more reliable song duration
GME Plugin: fix track numbering GME starts all track indexes at zero, but subtune prefixes start at one. This fixes an off-by-one error during track enumeration.
2017-09-26 15:42:53 +02:00
- gme: fix track numbering
queue/PlaylistControl: keep order list consistency in MoveOrderToCurrent() Our previous use of Queue::SwapOrders() could cause surprising results: - sometimes, the old "current" song would be played again (if the newly selected song had not been played already) - sometimes, the old "current" song would not be played again (if the newly selected song had already been played) This is inconsistent, because it should not depend on whether the newly selected song had already been played. So instead of Queue::SwapOrders() we now use Queue::MoveOrderAfter() and Queue::MoveOrderBefore(), which is more expensive, but also more consistent. It attempts to retain as much from the previous order list as possible, and only moves the newly selected song around.
2017-10-17 19:44:16 +02:00
* improve random song order when switching songs manually
lib/icu/Compare: add fallback using strcasecmp() and strcasestr() Our IcuCaseFold() fallback using strxfrm() is not actually case insensitive. This commit fixes the problem by switching to strcasecmp(). That function is not guaranteed to support UTF-8, but it's the best we can do in this sparse situation. Closes #111
2017-09-20 23:43:06 +02:00
* fix case insensitive search without libicu
Playlist{File,Save}: always use UTF-8 in playlists on Windows Turns out that using CP_ACP is a lousy idea, because only very few Unicode characters can be represented by it. Instead, switch to UTF-8 (which every sane person on other operating system already uses). Closes #102
2017-10-18 10:05:26 +02:00
* fix Unicode file names in playlists on Windows
decoder/Thread: clear the command after catching an exception If an early exception gets caught (e.g. from AllocatedPath::FromUTF8Throw()) before DecoderControl::CommandFinishedLocked() is called, the decoder thread would go in an endless loop, because DecoderCommand::START is still set. Closes #118
2017-09-27 17:08:16 +02:00
* fix endless loop when accessing malformed file names in ZIP files
increment version number to 0.20.11
2017-08-31 19:48:35 +02:00
release v0.20.10
2017-08-24 09:15:43 +02:00
ver 0.20.10 (2017/08/24)
Check for MusicBrainz id3v2 tags in ffmpeg. Addresses #82. Previously, the ffmpeg decoder only checked for the "generic" MusicBrainz metadata keys used in other metadata container formats.
2017-07-20 01:43:17 +02:00
* decoder
- ffmpeg: support MusicBrainz ID3v2 tags
tag/Aiff: the FORM chunk size is big-endian Was broken by commit 8a86460b8f054a42130a4b59e082b66d921d2a1b Closes #87
2017-07-31 13:46:09 +02:00
* tags
- aiff: fix FORM chunk size endianess (is big-endian)
OSX mixer
2017-08-21 18:10:12 +02:00
* mixer
- osx: add a mixer for OSX.
player/Thread: initialize play_audio_format, fixes assertion This fixes an assertion failure caused by resuming playback before the decoder has finished startup.
2017-08-23 17:43:49 +02:00
* fix crash when resuming playback before decoder is ready
util/FormatString: pass the allocated buffer to AllocatedString::Donate() .. and not the stack buffer. This made the AllocatedString destructor crash. Closes #52
2017-08-03 00:25:30 +02:00
* fix crash on Windows
increment version number to 0.20.10
2017-06-15 21:35:23 +02:00
release v0.20.9
2017-06-04 12:51:36 +02:00
ver 0.20.9 (2017/06/04)
decoder/ffmpeg: add support for adx Add ffmpeg decoder support for *.adx files (Sega game console) Closes #60
2017-06-01 19:16:24 +02:00
* decoder
- ffmpeg: support *.adx
system/ByteOrder: fix byte order detection on FreeBSD/aarch64 Patch from https://svnweb.freebsd.org/ports/head/audio/musicpd/files/patch-src_system_ByteOrder.hxx?revision=441921&view=co (with a tiny modification) Closes #59
2017-05-29 20:37:14 +02:00
* fix byte order detection on FreeBSD/aarch64
storage, db, mixer, command: remove more bogus "pure" attributes This commit is similar to 788e3b31e1ab8243386339eaf136ede18f015d55, and removes more "pure" attributes which were placed on functions that could throw exceptions, which is illegal according to clang's understanding of the attribute (but not according to GCC's). GitHub issue #58 was most likely about StorageDirectoryReader::GetInfo() and Storage::GetInfo(), which still had "pure" attributes. Closes #58
2017-06-03 21:54:21 +02:00
* fix more random crashes when compiled with clang
increment version number to 0.20.9
2017-05-29 20:36:08 +02:00
release v0.20.8
2017-05-19 20:10:22 +02:00
ver 0.20.8 (2017/05/19)
output/{osx,haiku,pulse,sles}: add missing "noexcept" Fixes build failure on OS X, closes #44. With the other plugins, that's not critical, because those use the AudioOutputWrapper, which hides this problem.
2017-05-16 07:12:30 +02:00
* output
- osx: fix build failure due to missing "noexcept"
playlist/m3u: support for mime-type `audio-mpegurl` for M3U playlists
2017-05-19 11:28:22 +02:00
* playlist
- m3u: support MIME type `audio/mpegurl`
Main: omit "constexpr" on MIN_BUFFER_SIZE with GCC 4.x
2017-05-16 07:20:47 +02:00
* fix build failure with GCC 4.x
increment version number to 0.20.8
2017-05-16 07:06:57 +02:00
release v0.20.7
2017-05-15 22:51:08 +02:00
ver 0.20.7 (2017/05/15)
storage/FileInfo, db/simple/Directory: use 64 bit for device/inode An ino_t is usually a 64 bit integer, and some file systems (such as Linux's kernel NFS client) really uses the upper 32 bit. This can lead to false positives in the directory loop detection in FindAncestorLoop(). Increasing these two attributes (in StorageFileInfo and Directory) to 64 bit adds little overhead, but makes the check a lot safer.
2017-04-06 09:58:25 +02:00
* database
- simple: fix false positive directory loop detection with NFS
Main: enforce a reasonable minimum audio_buffer_size setting
2017-05-15 22:39:57 +02:00
* enforce a reasonable minimum audio_buffer_size setting
Main: cap buffer_before_play at 80% to prevent deadlock Closes #34
2017-05-15 22:25:39 +02:00
* cap buffer_before_play at 80% to prevent deadlock
*: remove "pure" and "const" attributes from throwing functions The "pure" and "const" attributes are not so well-defined, and a recent clang version implements an optimization which pushes the definition's boundary beyond what I believed it was. clang now assumes that functions declared "pure" cannot throw exceptions, even if they lack the "noexcept" specification. When compiled with this new clang version, MPD will crash randomly if an exception happens to get thrown by such as "pure" function (https://github.com/MusicPlayerDaemon/MPD/issues/41). This commit removes all such misplaced "pure" and "const" attributes, closing #41.
2017-05-08 14:48:58 +02:00
* fix random crashes when compiled with clang
configure.ac: prepare for 0.20.7
2017-03-16 10:50:12 +01:00
release v0.20.6
2017-03-10 16:57:59 +01:00
ver 0.20.6 (2017/03/10)
lib/curl/Request: "ICY 200 OK" is a response boundary header
2017-03-10 16:28:02 +01:00
* input
- curl: fix headers after HTTP redirect to Shoutcast server
decoder/mpcdec: ignore empty frames https://bugs.musicpd.org/view.php?id=4656 describes a crash due to division by zero because frame.samples==0. This should never happen, but apparently can happen after seeking. The best we can do is to just ignore this frame.
2017-03-01 16:12:33 +01:00
* decoder
decoder/Thread: check ENABLE_FFMPEG, not HAVE_FFMPEG This repairs the damage to commit 74dbaade6fa done by commit b3f5b4932c3
2017-03-01 17:06:23 +01:00
- ffmpeg: re-enable as fallback
decoder/mpcdec: ignore empty frames https://bugs.musicpd.org/view.php?id=4656 describes a crash due to division by zero because frame.samples==0. This should never happen, but apparently can happen after seeking. The best we can do is to just ignore this frame.
2017-03-01 16:12:33 +01:00
- mpcdec: fix crash (division by zero) after seeking
decoder/sidplay: make compatible with libsidplayfp < 1.8 https://bugs.musicpd.org/view.php?id=4665
2017-03-10 13:48:52 +01:00
- sidplay: make compatible with libsidplayfp < 1.8
Partition: handle SYNC_WITH_PLAYER before TAG_MODIFIED The TAG_MODIFIED handler (i.e. playlist::TagModified()) works only if the modified song is the current song - something that is not updated until SYNC_WITH_PLAYER is finished. This fixes tag updates right after a new song is started.
2017-03-10 16:11:34 +01:00
* fix stream tags after automatic song change
command/Error: improve libstdc++ 4.9.x detection for std::rethrow_if_nested() workaround
2017-03-01 16:38:22 +01:00
* workaround for GCC 4.9.4 / libstdc++ bug (build failure)
configure.ac: prepare for 0.20.6
2017-03-01 16:12:45 +01:00
release v0.20.5
2017-02-20 21:51:31 +01:00
ver 0.20.5 (2017/02/20)
NEWS: mention ID3 memory leak fix
2017-02-08 08:44:47 +01:00
* tags
- id3: fix memory leak on corrupt ID3 tags
configure.ac: don't require libsidutils when building with libsidplayfp The libsidplayfp fork has merged libsidutils into the main library. The libsidutils we used to link with was part of the original libsidplay project.
2017-02-09 13:09:03 +01:00
* decoder
- sidplay: don't require libsidutils when building with libsidplayfp
output/httpd/IcyMetaDataServer: cast length to unsigned Fixes another buffer overflow: if the stream has a very long title or URL, resulting in a metadata string of more than 2 kB, icy_string[0] is a negative value, which gets casted to size_t - ouch! https://bugs.musicpd.org/view.php?id=4652
2017-02-19 19:11:04 +01:00
* output
- httpd: fix two buffer overflows in IcyMetaData length calculation
mixer/alsa: reset the MultiSocketMonitor in the destructor Fixes potential crash bug.
2017-02-09 20:57:14 +01:00
* mixer
- alsa: fix crash bug
configure.ac: prepare for 0.20.5
2017-02-06 23:28:36 +01:00
release v0.20.4
2017-02-01 21:59:36 +01:00
ver 0.20.4 (2017/02/01)
lib/nfs/Connection: detect libnfs reconnect When rpc_reconnect_requeue() gets called from inside nfs_service(), the NfsInputStream can stall completely because the old socket has been unregistered from epoll automatically, but the new one has never been registered. Therefore, nfs_service() will never be called again. This kludge attempts to detect this condition by checking nfs_which_events()==POLLOUT. https://bugs.musicpd.org/view.php?id=4081
2017-02-01 21:16:50 +01:00
* input
- nfs: fix freeze after reconnect
output/sndio: work around a libroar C++ incompatibility Same as in commit e02d8ad8d29ac, but this time for the sndio plugin which can be emulated by libroar.
2017-02-01 19:53:23 +01:00
* output
- sndio: work around a libroar C++ incompatibility
pcm/SampleFormat: workaround for GCC 4.9 "constexpr" bug GCC 4.9 has incomplete C++14 support. Specifically, it doesn't allow switch/case in "constexpr" functions.
2017-01-27 11:02:58 +01:00
* workaround for GCC 4.9 "constexpr" bug
free() require cstdlib to be included
2017-01-28 16:07:37 +01:00
* fix FreeBSD build failure
configure.ac: prepare for 0.20.4
2017-01-27 08:47:36 +01:00
release v0.20.3
2017-01-25 08:53:16 +01:00
ver 0.20.3 (2017/01/25)
PlaylistFile: "playlistadd" creates new playlist if it does not exist, as documented
2017-01-16 20:55:19 +01:00
* protocol
- "playlistadd" creates new playlist if it does not exist, as documented
db/proxy: make the base class of LibmpdclientError public If the base class is not accessible, the "catching" the base class won't work. This caused the fatal error: terminate called after throwing an instance of 'LibmpdclientError'
2017-01-23 18:28:40 +01:00
* database
- proxy: fix error "terminate called after throwing ..."
db/proxy: make connect errors during startup non-fatal
2017-01-23 18:48:04 +01:00
- proxy: make connect errors during startup non-fatal
lib/upnp: fix bad std::chrono cast libupnp provides seconds, not whatever time unit is used by std::chrono::steady_clock.
2017-01-23 19:16:14 +01:00
* neighbor
- upnp: fix premature expiry
output/Source: reset the ReplayGain serials ion OpenFilter() Each close/open cycle resets the Filter's state, because a new Filter instance is being created. That results in the serials (replay_gain_serial and other_replay_gain_serial) being out of sync with the internal ReplayGainFilter state. So instead of initializing those serials once, we need to initialize them each time we create new ReplayGainFilter instances, i.e. in OpenFilter(). https://bugs.musicpd.org/view.php?id=4632
2017-01-23 17:55:04 +01:00
* replay gain: don't reset ReplayGain levels when unpausing playback
pcm/PcmChannels: silence surround channels when converting from stereo Previously, there was no special code to convert stereo to multi-channel. The generic solution for this was to convert to mono, and then copy the result to all channels. That's a pretty bad solution, but at least something which always renders audio. MPD does something, instead of failing. Now that MPD has proper support for multi-channel (by defining the channel order), we can do better than that. It is a (somewhat) common case to play back stereo music on a DAC which can only do multi-channel. The best approach here is to copy the stereo channels to front-left and front-right, and apply the "silence" pattern to all other channels.
2017-01-19 10:53:41 +01:00
* silence surround channels when converting from stereo
Audio{Format,Parser}: use shortcuts such as "dsd64" in log messages
2017-01-17 09:25:18 +01:00
* use shortcuts such as "dsd64" in log messages
configure.ac: prepare for 0.20.3
2017-01-16 12:03:22 +01:00
release v0.20.2
2017-01-15 01:28:00 +01:00
ver 0.20.2 (2017/01/15)
input/alsa: clear sockets from within IOThread Fixes assertion failure in implicit destructor.
2017-01-13 20:17:16 +01:00
* input
- alsa: fix crash bug
input/alsa: set period_size=buffer_size/4 This way, we have four periods instead of the default of two. With only two periods, we don't get woken up often enough, and we frequently encounter buffer overruns. With four periods, we have more time to breathe, and the buffer overruns magically disappear.
2017-01-14 21:46:17 +01:00
- alsa: fix buffer overruns
decoder/flac: add options "probesize" and "analyzeduration" https://bugs.musicpd.org/view.php?id=3876
2017-01-10 23:05:04 +01:00
* decoder
- flac: add options "probesize" and "analyzeduration"
decoder/Bridge: call PcmConvert::Reset() after seeking
2017-01-11 15:32:57 +01:00
* resampler
- libsamplerate: reset state after seeking
pcm/Export: DSD_U32 quarters the sample rate DSD_U32 packs four bytes instead of one large "sample", thus the sample rate is one quarter of the input sample rate. This fixes a rather critical DSD_U32 playback problem.
2017-01-11 10:06:27 +01:00
* output
output/Thread: reconfigure ConvertFilter for its new input AudioFormat If the input AudioFormat changes but the out_audio_format doesn't change (e.g. because there is a fixed "format" setting in this "audio_output" section), the ConvertFilter needs to be reconfigured. This didn't happen, resulting in awful static noise after changing songs.
2017-01-15 01:18:34 +01:00
- fix static noise after changing to a different audio format
pcm/Export: DSD_U32 quarters the sample rate DSD_U32 packs four bytes instead of one large "sample", thus the sample rate is one quarter of the input sample rate. This fixes a rather critical DSD_U32 playback problem.
2017-01-11 10:06:27 +01:00
- alsa: fix the DSD_U32 sample rate
pcm/Dsd32: fix the byte order The byte order of DSD_U32 was wrong from the start. The oldest bits must be in the MSB, not in the LSB, according to snd_pcm_format_descriptions in alsa-lib.
2017-01-11 22:25:54 +01:00
- alsa: fix the DSD_U32 byte order
output/alsa: support DSD_U16
2017-01-11 22:33:52 +01:00
- alsa: support DSD_U16
output/recorder: fix typo in variable name Fixes the dreaded error "Failed to create : No such file or directory". https://bugs.musicpd.org/view.php?id=4625
2017-01-12 21:36:32 +01:00
- recorder: fix error "Failed to create : No such file or directory"
DetachedSong: compare start_time and end_time in IsSame() This method is used by DecoderControl::IsCurrentSong(), which is used by the player thread to check whether the current decoder instance can be reused to seek. When switching to another song in the same CUE sheet, previously DetachedSong::IsSame() returned true, and thus the old decoder instance was used for the new song, not considering the new end_time. This led to the old decoder quickly quitting.
2017-01-15 00:54:09 +01:00
* playlist
- cue: fix skipping songs
configure.ac: prepare for 0.20.2
2017-01-10 23:01:42 +01:00
release v0.20.1
2017-01-09 18:10:18 +01:00
ver 0.20.1 (2017/01/09)
lib/curl/Global: defer the ReadInfo() call Fixes a crash that can occur due to recursion from InvalidateSockets() to ReadInfo() to CurlRequest callbacks.
2017-01-08 12:40:56 +01:00
* input
- curl: fix crash bug
input/curl: wake up client thread after seek to end of file Call SeekDone() to avoid the freeze bug.
2017-01-09 18:08:33 +01:00
- curl: fix freeze bug
decoder/wavpack: fix crash bug
2017-01-08 14:36:04 +01:00
* decoder
- wavpack: fix crash bug
storage/http: new storage plugin
2017-01-03 07:25:14 +01:00
* storage
- curl: new storage plugin for WebDAV (work in progress)
Rewrite of AlsaMixerPlugin to use volume_mapping Changed AlsaMixerPlugin to use the get and set normalized functions from volume_mapping of alsa-utils/alsamixer Changed volume_mapping set volume to be for all channels and not per channel added volume_mapping files to Makefile.am
2017-01-05 12:09:13 +01:00
* mixer
- alsa: normalize displayed volume according to human perception
filter/AutoConvert: initialize Filter::out_audio_format
2017-01-06 12:44:55 +01:00
* fix crash with volume_normalization enabled
configure.ac: prepare for 0.20.1
2017-01-05 19:36:32 +01:00
release v0.20
2017-01-04 10:49:28 +01:00
ver 0.20 (2017/01/04)
AllCommands: "commands" returns playlist commands only if playlist_directory configured
2014-10-25 23:49:35 +02:00
* protocol
- "commands" returns playlist commands only if playlist_directory configured
DatabaseCommands: add "window" parameter to "search"/"find"
2014-11-18 14:31:27 +01:00
- "search"/"find" have a "window" parameter
{Tag,Song}Print, PlayerCommands: report song duration with milliseconds precision
2014-11-25 17:46:43 +01:00
- report song duration with milliseconds precision
sticker/Match: add operator "EQUALS" Mapped to "=" in the MPD protocol. This is the first operator, initially supporting value matches in the MPD protocol.
2014-12-12 22:12:19 +01:00
- "sticker find" can match sticker values
command/{Queue,File}Commands: drop "file:///" prefix for absolute paths Requiring this prefix makes the client's intention very clear, but it was too hard to understand why this prefix was needed. Initially, my intention was to differentiate from broken clients which prefix relate URIs with a slash; once MPD allowed that. In the past few years however, MPD has disallowed that, and there was no significant breakage (except for the "add /" special case which some clients apparently still do). So I figure it's about time to define that an URI that begins with a slash points to an arbitrary file on the file system.
2015-02-09 07:55:06 +01:00
- drop the "file:///" prefix for absolute file paths
command/queue: add range parameter to plchanges and plchangesposid
2015-10-27 20:35:40 +01:00
- add range parameter to command "plchanges" and "plchangesposid"
command/CommandError: send verbose error message to client
2015-11-11 19:49:17 +01:00
- send verbose error message to client
input/curl: remove duplicate InitEasy() call
2016-12-19 14:38:50 +01:00
* input
- curl: fix memory leak
ApeTag: drop support for non-standard tag "album artist" This was used by proprietary software. MPD adopted it a few years ago, which turns out to be a mistake, because it now creates problems for some MPD users (http://bugs.musicpd.org/view.php?id=4168).
2014-11-25 17:07:08 +01:00
* tags
vorbiscomments: Remove support for non-standard "album artist" tags. MPD used both "album artist" and "albumartist" tags and mapped them to one tag. This could lead to issues, if a file had both tags, causing MPD to send a list of albumartists instead of a single one. Since "album artist" is not a standard tag anyway and even its originators started to use the proper alternative, its time to say goodbye!
2014-12-12 18:28:11 +01:00
- ape, ogg: drop support for non-standard tag "album artist"
affected filetypes: vorbis, flac, opus & all files with ape2 tags
(most importantly some mp3s)
TagId3: remove the "id3v1_encoding" setting Drop support for misencoded tags. People should not be using ID3v1 anyway.
2015-06-26 00:08:29 +02:00
- id3: remove the "id3v1_encoding" setting; by definition, all ID3v1 tags
are ISO-Latin-1
decoder/thread: load APE replay gain from remote files
2016-02-23 10:57:03 +01:00
- ape: support APE replay gain on remote files
TagStream: add TagBuilder overload with ScanGenericTags() fallback This commit adds support for APE/ID3 tags from NFS/SMB files. See http://bugs.musicpd.org/view.php?id=4270
2016-02-26 13:23:42 +01:00
- read ID3 tags from NFS/SMB
decoder/ffmpeg: support ReplayGain and MixRamp
2014-12-10 13:05:16 +01:00
* decoder
decoder/Thread: log all errors
2015-12-31 13:38:27 +01:00
- improved error logging
decoder/Thread: pass InputStream errors to DecoderControl::error Give MPD clients access to the error condition.
2015-12-31 13:30:35 +01:00
- report I/O errors to clients
NEWS: mention ffmpeg/MixRamp
2014-12-10 14:01:12 +01:00
- ffmpeg: support ReplayGain and MixRamp
decoder/ffmpeg: support stream tags
2014-12-11 10:39:49 +01:00
- ffmpeg: support stream tags
decoder/gme: add option "accuracy"
2015-10-26 16:48:38 +01:00
- gme: add option "accuracy"
decoder/gme: provide the TRACK tag
2016-11-22 12:19:37 +01:00
- gme: provide the TRACK tag
decoder/DecoderPlugin: container_scan() returns forward_list<string>
2016-11-22 09:33:52 +01:00
- gme: faster scanning
decoder/mad: reduce memory usage while scanning tags Allocate the frame pointers only when actually decoding the file.
2015-10-19 13:12:04 +02:00
- mad: reduce memory usage while scanning tags
decoder/mpcdec: read the bit rate This was lost in commit 8ead8f7ea, because this was never implemented for the new API.
2015-08-05 23:33:11 +02:00
- mpcdec: read the bit rate
decoder/pcm: add MIME type "audio/x-mpd-float"
2016-06-10 22:59:45 +02:00
- pcm: support audio/L16 (RFC 2586) and audio/x-mpd-float
decoder/DecoderPlugin: container_scan() returns forward_list<string>
2016-11-22 09:33:52 +01:00
- sidplay: faster scanning
decoder/wavpack: implement WavpackStreamReader64 if available
2016-12-10 00:24:54 +01:00
- wavpack: large file support
decoder/wavpack: basic DSD support (WavPack 5) Enable OPEN_DSD_AS_PCM if available. No OPEN_DSD_NATIVE yet. https://bugs.musicpd.org/view.php?id=4606
2016-12-09 21:03:47 +01:00
- wavpack: support DSD (WavPack 5)
decoder/wavpack: implement scan_stream()
2016-12-09 23:14:58 +01:00
- wavpack: archive support
playlist/embcue: fix last track
2015-08-05 23:37:57 +02:00
* playlist
CueParser.cxx: ignore INDEX after first per track Use the first INDEX in each TRACK section, instead of the last, for the start time. This preserves the original CD layout (including gaps between tracks), and avoids skipping sections of songs in more exotic cuesheets (eg musical suite tracks). Fixes 0004355 and 0003359
2015-08-23 04:19:55 +02:00
- cue: don't skip pregap
playlist/embcue: fix last track
2015-08-05 23:37:57 +02:00
- embcue: fix last track
playlist/flac: re-add the plugin This reverts commit d7d9dbd2c2798b6ff39acb07a3bed38b6f99e283 by reimplementing it with the current MPD API. 3 years ago, I was wrong about the "embcue" plugin being able to replace this one, because "embcue" reads a tag named "CUESHEET", while this plugin reads the "CUESHEET" FLAC metablock. There's an important difference between those two!
2015-10-26 13:16:01 +01:00
- flac: new plugin which reads the "CUESHEET" metadata block
Set pulseaudio channel map to WAVE-EX 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.
2014-10-17 00:06:27 +02:00
* output
pcm/Order: new library to convert from FLAC to ALSA channel order This new library is integrated in the PcmExport class and (if enabled) converts MPD's channel order (= FLAC channel order) to ALSA channel order. This fixes: http://bugs.musicpd.org/view.php?id=3147 and http://bugs.musicpd.org/view.php?id=3255
2015-10-27 00:22:22 +01:00
- alsa: fix multi-channel order
output/alsa: remove option "use_mmap" MPD does not really take advantage of memory-mapped I/O by generating data right into the ALSA buffer; using plain snd_pcm_mmap_writei() has no advantage compared to snd_pcm_writei(). Let's kill this non-feature.
2016-02-28 09:30:59 +01:00
- alsa: remove option "use_mmap"
output/alsa: probe DSD_U32 if DSD_U8 fails See http://bugs.musicpd.org/view.php?id=4485
2016-02-28 09:54:20 +01:00
- alsa: support DSD_U32
output/alsa: disable DoP if it fails See http://bugs.musicpd.org/view.php?id=4496
2016-02-28 10:15:02 +01:00
- alsa: disable DoP if it fails
output/jack: use jack_ringbuffer_get_write_vector() Reduce number of libjack calls.
2014-12-24 22:58:25 +01:00
- jack: reduce CPU usage
Set pulseaudio channel map to WAVE-EX 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.
2014-10-17 00:06:27 +02:00
- pulse: set channel map to WAVE-EX
output/recorder: implement send_tag()
2014-12-26 14:57:21 +01:00
- recorder: record tags
output/recorder: dynamic file name
2015-01-10 08:58:31 +01:00
- recorder: allow dynamic file names
Add sndio output plugin
2016-06-21 19:28:41 +02:00
- sndio: new output plugin
mixer/null: new mixer plugin
2014-12-02 18:16:33 +01:00
* mixer
- null: new plugin
Config: add section "resampler"
2015-01-21 20:42:41 +01:00
* resampler
- new block "resampler" in configuration file
replacing the old "samplerate_converter" setting
pcm/SoxrResampler: allow multi-threaded resampling
2015-01-21 20:33:13 +01:00
- soxr: allow multi-threaded resampling
NEWS: add "player" section
2016-12-13 22:39:29 +01:00
* player
- reset song priority on playback
output/Thread: wake up the player thread periodically while playing Without this, the pipe would run empty very often, which may result in an xrun if the roundtrip to the PlayerThread and back takes too long. By waking up the PlayerThread before the pipe runs empty, we make MPD much more latency tolerant, which is a major optimization.
2016-12-13 22:39:20 +01:00
- reduce xruns
fs/io/FileOutputStream: use O_TMPFILE if available The Linux feature allows writing new files to an invisible file, and then replace the old file. This preserves the old file if we get interrupted by some event.
2015-01-05 20:24:59 +01:00
* write database and state file atomically
LogBackend: remove character set support Always write UTF-8 to the log file.
2015-06-22 22:33:37 +02:00
* always write UTF-8 to the log file.
configure.ac: make GLib optional by default Only very few (minor) features still depend on GLib.
2014-12-04 23:55:11 +01:00
* remove dependency on GLib
configure.ac: support libsystemd (instead of the older libsystemd-daemon)
2015-03-20 22:12:43 +01:00
* support libsystemd (instead of the older libsystemd-daemon)
db/plugins/ProxyDatabasePlugin: add "keepalive" parameter
2015-01-15 23:33:10 +01:00
* database
- proxy: add TCP keepalive option
update: apply .mpdignore matches to subdirectories Wildcard matches are directly applied to all filenames in subdirectories without any attempt at matching relative paths. This change is based on the following feature request: http://bugs.musicpd.org/view.php?id=3729
2015-09-29 19:39:07 +02:00
* update
- apply .mpdignore matches to subdirectories
configure.ac: enable C++14
2016-11-18 08:41:47 +01:00
* switch the code base to C++14
- GCC 4.9 or clang 3.4 (or newer) recommended
configure.ac: prepare for 0.20
2014-10-25 00:29:51 +02:00
release v0.19.21
2016-12-13 10:54:04 +01:00
ver 0.19.21 (2016/12/13)
decoder/ffmpeg: fix double free bug From the avformat_open_input() API documentation: "Note that a user-supplied AVFormatContext will be freed on failure." https://bugs.musicpd.org/view.php?id=4607
2016-12-13 08:34:05 +01:00
* decoder
- ffmpeg: fix crash bug
test/test_queue_priority: fix unit test failure after recent "setprio" change
2016-12-13 08:36:42 +01:00
* fix unit test failure after recent "setprio" change
systemd: add user unit The user unit omits the "ProtectKernelModules" setting which fails with modular kernels: Failed at step CAPABILITIES spawning /usr/bin/mpd: Operation not permitted It is unfortunate that systemd (version 232) is unable to reduce its own capabilities, because this requires us to split system and user units. https://bugs.musicpd.org/view.php?id=4608
2016-12-13 08:41:55 +01:00
* systemd: add user unit
configure.ac: prepare for 0.19.21
2016-12-13 08:31:21 +01:00
release v0.19.20
2016-12-09 20:02:07 +01:00
ver 0.19.20 (2016/12/09)
Queue: "setprio" re-enqueues old song if priority has been raised This commit changes a minor queue priority design to something which makes a little bit more sense. Previously, a song that had already been played would only be re-enqueued if its priority had just been raised above the current song's. This means that if it was already above, it was not re-enqueued. That is a surprising behavior, because users expect a song to be played when its priority is raised. Now the song is always re-enqueued if its priority is raised (and above the current song's - no matter if it has already been above before). https://bugs.musicpd.org/view.php?id=4592
2016-12-09 13:02:26 +01:00
* protocol
- "setprio" re-enqueues old song if priority has been raised
decoder/ffmpeg: ignore empty packets An empty packet would be a command for avcodec_send_packet() to finalize the codec. Fixes https://bugs.musicpd.org/view.php?id=4588
2016-10-26 18:06:56 +02:00
* decoder
- ffmpeg: ignore empty packets
decoder/pcm: round buffer size down to nearest frame size https://bugs.musicpd.org/view.php?id=4599
2016-11-17 21:58:27 +01:00
- pcm: fix corruption bug with partial frames (after short read)
decoder/sidplay: fix playback speed with libsidplayfp https://bugs.musicpd.org/view.php?id=4577
2016-10-27 20:24:16 +02:00
- sidplay: fix playback speed with libsidplayfp
output/winmm: 8 bit playback is not supported Everything must be S16.
2016-11-07 08:53:57 +01:00
* output
- winmm: fix 8 bit playback
{input,output}/alsa: fix gcc 7.0 -Wimplicit-fallthrough
2016-11-16 19:50:38 +01:00
* fix gcc 7.0 -Wimplicit-fallthrough
systemd: more paranoid security settings
2016-12-09 10:36:02 +01:00
* systemd: paranoid security settings
configure.ac: prepare for 0.19.20
2016-09-27 22:04:46 +02:00
release v0.19.19
2016-08-23 10:19:10 +02:00
ver 0.19.19 (2016/08/23)
decoder/wildmidi: support libWildMidi 0.4
2016-08-15 10:08:35 +02:00
* decoder
decoder/ffmpeg: call avcodec_parameters_to_context() These bug reports describe problems with some FFmpeg codecs: https://bugs.musicpd.org/view.php?id=4564 https://bugs.musicpd.org/view.php?id=4568 https://bugs.musicpd.org/view.php?id=4572 According to the FFmpeg bug tracker, a call to avcodec_parameters_to_context() is required after avcodec_alloc_context3(): https://trac.ffmpeg.org/ticket/5781 This requirement was previously undocumented.
2016-08-23 09:37:19 +02:00
- ffmpeg: bug fix for FFmpeg 3.1 support
decoder/wildmidi: support libWildMidi 0.4
2016-08-15 10:08:35 +02:00
- wildmidi: support libWildMidi 0.4
Support S24_P32/S32/FLOAT sample formats on Pulse This is based on a patch from Ian Scott in 2014. It was never committed, so I figured I'd fix the outstanding issue and resubmit it. https://www.mail-archive.com/mpd-devel%40musicpd.org/msg00139.html
2016-08-14 18:45:32 +02:00
* output
- pulse: support 32 bit, 24 bit and floating point playback
system/ByteOrder: gssupport non-x86 NetBSD
2015-04-06 01:29:48 +02:00
* support non-x86 NetBSD
Compiler.h: work around clang 3.9 warning -Wexpansion-to-defined Check {GCC,CLANG}_VERSION==0 or >0 instead of using defined(), which may render undefined behavior.
2016-08-23 09:55:59 +02:00
* fix clang 3.9 warnings
configure.ac: prepare for 0.19.19
2016-08-15 10:00:03 +02:00
release v0.19.18
2016-08-05 18:45:23 +02:00
ver 0.19.18 (2016/08/05)
decoder/ffmpeg: log detailed error message
2014-12-18 22:07:38 +01:00
* decoder
decoder/ffmpeg: copy the AVPacket in ffmpeg_send_packet() Revert commit 70495aad by rewriting it. Turns out, in old FFmpeg versions, copying the AVPacket is necessary.
2016-07-28 19:41:05 +02:00
- ffmpeg: fix crash with older FFmpeg versions (< 3.0)
decoder/ffmpeg: log detailed error message
2014-12-18 22:07:38 +01:00
- ffmpeg: log detailed error message
decoder/ffmpeg: use AVCodecParameters on FFmpeg 3.1 The AVCodecContext attribute is deprecated.
2016-07-27 15:06:42 +02:00
- ffmpeg: support FFmpeg 3.1
configure.ac: detect libsidplay2 and related with pkg-config The comment about requiring libtool from 7 years ago is obsolete these days.
2016-07-29 15:21:38 +02:00
- sidplay: detect libsidplay2 with pkg-config
decoder/sidplay: log detailed error message
2016-07-29 16:59:17 +02:00
- sidplay: log detailed error message
decoder/sidplay: read the "date" tag
2016-07-29 17:47:08 +02:00
- sidplay: read the "date" tag
decoder/sidplay: allow building with libsidplayfp instead of libsidplay2 https://bugs.musicpd.org/view.php?id=4558
2016-07-29 15:31:02 +02:00
- sidplay: allow building with libsidplayfp instead of libsidplay2
output/shout: rename "encoding" to "encoder" The user manual specifies "encoder", which is consistent with other output plugins. "encoding" should be deprecated.
2016-07-29 10:52:03 +02:00
* output
- shout: recognize setting "encoder" instead of "encoding"
decoder/Thread: delete the InputStream on error Fixes memory leak after stream failure. See https://bugs.musicpd.org/view.php?id=4562
2016-08-05 18:15:30 +02:00
* fix memory leak after stream failure
lib/nfs/Manager: add Compare(ManagedConnection, ManagedConnection) Required for Boost 1.61, which uses that overload in a BOOST_ASSERT().
2016-08-05 18:06:07 +02:00
* fix build failure with Boost 1.61
Compiler.h: require gcc 4.7 or newer The ScopeExit library uses C++11 initializers, which gcc 4.6 does not support. Let's kill support for this ancient incomplete C++11 compiler, nobody should be using it anymore.
2016-07-29 09:52:23 +02:00
* require gcc 4.7 or newer
configure.ac: prepare for 0.19.18
2016-07-27 15:07:15 +02:00
release v0.19.17
2016-07-09 00:40:57 +02:00
ver 0.19.17 (2016/07/09)
decoder/flac: evaluate all possible FLAC__stream_decoder_get_state() values Stop after all fatal errors. This fixes assertion failures in libFLAC.
2016-07-05 19:27:40 +02:00
* decoder
- flac: fix assertion failure while seeking
decoder/flac: handle unknown duration correctly If the duration is unknown, pass SignedSongTime::Negative(), as documented for decoder_initialized().
2016-07-08 22:33:49 +02:00
- flac: fix stream duration indicator
decoder/API: check initial_seek_running in _check_cancel_read() The "seeking" flag is not set for the initial seek, and so decoder_read() could be canceled when another SEEK was emitted during initial seek. This fixes several seek problems, for example the one reported for the FLAC decoder plugin: https://bugs.musicpd.org/view.php?id=4552
2016-07-06 15:37:11 +02:00
- fix seek problems in several plugins
PlayerThread: make chunk allocation error non-fatal in SendSilence() Fixes abort after seeking on fast machines.
2016-07-05 17:44:45 +02:00
* fix spurious seek error "Failed to allocate silence buffer"
filter/ReplayGain: skip PcmVolume if a mixer is set Previously, volume was applied twice: once by PcmVolume, and again by the hardware mixer.
2016-07-01 21:17:52 +02:00
* replay gain: fix "replay_gain_handler mixer" setting
PlayerThread: use PcmSilence() in SendSilence() No change for regular PCM, but DSD uses 0x69 now.
2016-07-05 17:55:59 +02:00
* DSD: use 0x69 as silence pattern
client/ClientRead: call Break() before Close() Referencing the attribute "partition" is illegal after Close(), because Close() deletes "this".
2016-07-07 13:52:33 +02:00
* fix use-after-free bug on "close" and "kill"
configure.ac: prepare for 0.19.17
2016-07-01 21:16:14 +02:00
release v0.19.16
2016-06-13 18:59:07 +02:00
ver 0.19.16 (2016/06/13)
player/Thread: cancel outputs before seeking .. instead of doing it after seeking. After seeking, the command had no effect, because CheckDecoderStartup() waits for all outputs to finish. This caused a very long delay while seeking and switching songs (https://bugs.musicpd.org/view.php?id=4534).
2016-06-13 09:13:56 +02:00
* faster seeking
NEWS: add missing lines
2016-05-11 17:09:46 +02:00
* fix system include path order
* add missing DocBook file to tarball
configure.ac: prepare for 0.19.16
2016-05-02 08:57:53 +02:00
release v0.19.15
2016-04-30 14:21:52 +02:00
ver 0.19.15 (2016/04/30)
decoder/Thread: use "ffmpeg" as fallback instead of "mad" Adds support for stream codecs which havn't been explicitly listed in ffmpeg_mime_types.
2016-03-30 00:58:48 +02:00
* decoder
decoder/ffmpeg: use av_packet_unref() instead of av_free_packet() av_free_packet() was deprecated in FFmpeg 3.0.
2016-04-12 21:15:05 +02:00
- ffmpeg: support FFmpeg 3.0
decoder/Thread: use "ffmpeg" as fallback instead of "mad" Adds support for stream codecs which havn't been explicitly listed in ffmpeg_mime_types.
2016-03-30 00:58:48 +02:00
- ffmpeg: use as fallback instead of "mad" if no plugin matches
decoder/opus: support bigger OpusTags packets Required for OpusTags packets which contain artwork. See https://bugs.musicpd.org/view.php?id=4520
2016-04-19 13:05:42 +02:00
- opus: support bigger OpusTags packets
notify: use "constexpr" only with glibc The Mutex and Cond constructors are only "constexpr" with glibc, and this is what this #ifdef is about. Backport of commit 459a812a See http://bugs.musicpd.org/view.php?id=4511
2015-09-17 22:56:35 +02:00
* fix more build failures on non-glibc builds due to constexpr Mutex
DecoderBuffer: add missing include > In file included from src/decoder/DecoderBuffer.cxx:21:0: > src/decoder/DecoderBuffer.hxx:41:20: error: 'uint8_t' was not declared in this scope > DynamicFifoBuffer<uint8_t> buffer; > ^ > src/decoder/DecoderBuffer.hxx:41:27: error: template argument 1 is invalid > DynamicFifoBuffer<uint8_t> buffer; > ^ > src/decoder/DecoderBuffer.hxx: In member function 'void DecoderBuffer::Clear()': > src/decoder/DecoderBuffer.hxx:61:10: error: request for member 'Clear' in '((DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'int' > buffer.Clear(); > ^ > src/decoder/DecoderBuffer.hxx: In member function 'size_t DecoderBuffer::GetAvailable() const': > src/decoder/DecoderBuffer.hxx:78:17: error: request for member 'GetAvailable' in '((const DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'const int' > return buffer.GetAvailable(); > ^ > src/decoder/DecoderBuffer.hxx: In member function 'ConstBuffer<void> DecoderBuffer::Read() const': > src/decoder/DecoderBuffer.hxx:87:19: error: request for member 'Read' in '((const DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'const int' > auto r = buffer.Read(); > ^ > src/decoder/DecoderBuffer.hxx:88:27: error: could not convert '{<expression error>, <expression error>}' from '<brace-enclosed initializer list>' to 'ConstBuffer<void>' > return { r.data, r.size }; > ^ > src/decoder/DecoderBuffer.hxx: In member function 'void DecoderBuffer::Consume(size_t)': > src/decoder/DecoderBuffer.hxx:105:10: error: request for member 'Consume' in '((DecoderBuffer*)this)->DecoderBuffer::buffer', which is of non-class type 'int' > buffer.Consume(nbytes); > ^ This seems to be caused by a lacking include, fixed by the below patch. I'm unsure what made this appear now, though, compiler and toolchain libraries seem to be the same upstream versions that built 0.19.14-1 just fine in late March.
2016-04-24 22:00:56 +02:00
* fix build failure due to missing include
Bug#822848: mpd FTBFS on Alpha; misaligned arrays in the test suite Source: mpd Version: 0.19.14-2 Severity: important Justification: fails to build form source (but built in the past) Tags: patch User: debian-alpha@lists.debian.org Usertags: alpha mpd FTBFS on Alpha with a failure in the test suite [1]: FAIL: test/test_byte_reverse ============================ .F... !!!FAILURES!!! Test Results: Run: 4 Failures: 1 Errors: 0 1) test: ByteReverseTest::TestByteReverse2 (F) line: 58 test/test_byte_reverse.cxx assertion failed - Expression: strcmp(result, (const char *)dest) == 0 This occurs because the test suite (in test/test_byte_reversal.cxx) allocates static char arrays and passes the char arrays to functions whose respective arguments were declared to be uint16_t *, etc., in the main code. This is in the realm of undefined behaviour on architectures with strict memory alignment requirements. Although the test only fails on Alpha (because Alpha has a particular CPU load instruction that gcc likes to use to add bugs ..., ahem, optimise the code on the assumption of alignment) it is potentially a latent bug for other architectures with strict alignment requirements. Since the code is compiled with the c++11 standard I attach a patch that modifies the test suite to align the non-compliant strings with the alignas() attribute. The test suite now passes on Alpha with that patch. Cheers Michael [1] https://buildd.debian.org/status/fetch.php?pkg=mpd&arch=alpha&ver=0.19.14-2&stamp=1461542099
2016-04-28 12:22:50 +02:00
* fix unit test on Alpha
configure.ac: prepare for 0.19.15
2016-03-30 00:30:39 +02:00
release v0.19.14
2016-03-18 18:26:58 +01:00
ver 0.19.14 (2016/03/18)
decoder/opus: limit tag size to 64 kB
2016-03-06 23:26:48 +01:00
* decoder
decoder/dsdiff: fix off-by-one buffer overflow
2016-03-06 23:28:29 +01:00
- dsdiff: fix off-by-one buffer overflow
decoder/opus: limit tag size to 64 kB
2016-03-06 23:26:48 +01:00
- opus: limit tag size to 64 kB
archive/iso9660: check path buffer bounds
2016-03-07 14:18:39 +01:00
* archive
- iso9660: fix buffer overflow
tag/TagPool: optimize _dup_item() When a reference counter is at its limit, don't allocate a new TagPoolSlot - that would result in many TagPoolSlot instances with ref==1. This in turn would make the linked list very very large, which means quadratic runtime for many operations.
2016-03-14 13:08:04 +01:00
* fix quadratic runtime bug in the tag pool
thread/Posix{Mutex,Cond}: use "constexpr" only with glibc Apparently all other C libraries are not compatible with "constexpr". Those which are not will get a performance penalty, but at least they work at all.
2015-08-25 12:46:12 +02:00
* fix build failures on non-glibc builds due to constexpr Mutex
configure.ac: prepare for 0.19.14
2016-03-01 21:22:42 +01:00
release v0.19.13
2016-02-23 22:06:13 +01:00
ver 0.19.13 (2016/02/23)
tag/{aiff,riff}: fix ID3 chunk padding Apply padding only to the fseek(), not to the chunk size. This fixes bogus "failed to read riff chunk" messages when the last chunk has an odd size. See http://bugs.musicpd.org/view.php?id=4486
2016-02-19 17:31:20 +01:00
* tags
- aiff, riff: fix ID3 chunk padding
Add TAK as a supported FFmpeg format.
2016-02-13 03:49:10 +01:00
* decoder
- ffmpeg: support the TAK codec
queue/Playlist: move only the tag items in TagModified() Fixes disappearing duration of remote songs during playback. See http://bugs.musicpd.org/view.php?id=4492
2016-02-23 21:01:55 +01:00
* fix disappearing duration of remote songs during playback
configure.ac, unix/Daemon: check for initgroups() at configure time The initgroups() manpage says we need to check for _BSD_SOURCE. The thing is that glibc deprecated this macro, and doesn't define it anymore, effectively breaking all MPD supplementary groups. The real fix is to check for initgroups() availability at configure time, instead of relying on the deprecated _BSD_SOURCE macro.
2016-02-23 20:13:34 +01:00
* initialize supplementary groups with glibc 2.19+
configure.ac: prepare for 0.19.13
2016-02-19 17:30:01 +01:00
release v0.19.12
2015-12-15 21:54:42 +01:00
ver 0.19.12 (2015/12/15)
fix mpd crash on invalid utf8 stream title
2015-11-14 21:18:41 +01:00
* fix assertion failure on malformed UTF-8 tag
Main: fix build failure on non-Linux systems
2015-11-10 08:38:53 +01:00
* fix build failure on non-Linux systems
fix LimitRTTIME in systemd unit file systemd does not understand LimitRTTIME=-1. For no limit we have to use the string 'infinity' (see systemd.exec(5)). Signed-off-by: Christian Hesse <mail@eworm.de>
2015-12-01 08:53:35 +01:00
* fix LimitRTTIME in systemd unit file
configure.ac: prepare for 0.19.12
2015-11-10 08:33:50 +01:00
release v0.19.11
2015-10-27 10:42:20 +01:00
ver 0.19.11 (2015/10/27)
tag/ApeLoader: fix buffer overflow after unterminated key
2015-10-16 14:40:46 +02:00
* tags
- ape: fix buffer overflow
decoder/ffmpeg: fix crash due to wrong avio_alloc_context() call Allocate the buffer dynamically using av_malloc(), and free AVIOContext.buffer in the destructor, as mandated by the libavformat documentation. Fixes http://bugs.musicpd.org/view.php?id=4446
2015-10-26 13:06:29 +01:00
* decoder
- ffmpeg: fix crash due to wrong avio_alloc_context() call
gme: don't loop forever, fall back to GME's default play length Fixes http://bugs.musicpd.org/view.php?id=4432
2015-10-26 17:16:20 +01:00
- gme: don't loop forever, fall back to GME's default play length
encoder/flac: fix crash with 32 bit playback Copy to encoder->audio_format *after* adjusting the sample format to S24_P32. Fixes http://bugs.musicpd.org/view.php?id=4433
2015-10-16 18:05:34 +02:00
* encoder
- flac: fix crash with 32 bit playback
output/Command: flush the mixer cache when enabling/disabling output Fixes mixer lag (http://bugs.musicpd.org/view.php?id=4425).
2015-10-26 16:29:07 +01:00
* mixer
- fix mixer lag after enabling/disabling output
configure.ac: prepare for 0.19.11
2015-10-16 14:55:00 +02:00
release v0.19.10
2015-06-21 15:51:50 +02:00
ver 0.19.10 (2015/06/21)
input/smbclient: fix DFF playback
2015-05-29 22:37:49 +02:00
* input
input/curl: trigger the condition variable in RequestDone() Fixes deadlock on small responses.
2015-06-19 16:45:38 +02:00
- curl: fix deadlock on small responses
input/smbclient: fix DFF playback
2015-05-29 22:37:49 +02:00
- smbclient: fix DFF playback
decoder/ffmpeg: skip unwanted samples after seeking When seeking to the beginning of a packet, skip the samples that come before the desired time stamp.
2015-06-19 18:02:10 +02:00
* decoder
- ffmpeg: improve seeking accuracy
DecoderThread: set Decoder::song_tag only for local files If the song tag comes from a stream, and MPD playback restarts, MPD would believe the tag should override the newly received tag. This makes the previous tag appear stuck. This change passes the song tag only if it's authoritative - i.e. if it's a song file.
2015-06-21 15:02:14 +02:00
- fix stuck stream tags
OpusEncoderPlugin: initialise granulepos so we end up with sane values Not initialising granulepos leads to it having arbitrary values in the encoded stream including possibly negative values which are not valid and confuse opusdec. Explicitly initialise opus_encoder::granulepos to avoid that problem.
2015-03-23 19:14:54 +01:00
* encoder
- opus: fix bogus granulepos
OutputControl: fix fail_timer check right after booting Right after booting, the monotonic clock starts with a very small value, and AudioOutput::LockUpdate() may believe that the fail_timer has not recovered yet.
2015-06-20 15:37:19 +02:00
* output
- fix failure to open device right after booting
storage/nfs: fix deadlock when connecting The Connect method can be called between Schedule and lock. In that case, when locked, the state is already set to CONNECTING of READY and the condition won't be signaled anymore.
2015-05-09 15:59:31 +02:00
* neighbor
- nfs: fix deadlock when connecting
PlayerThread: start the decoder on PlayerCommand::QUEUE Fixes missing SongBorder() call, which causes "single" mode breakage.
2015-06-21 15:38:48 +02:00
* fix "single" mode breakage due to queue edits
configure.ac: prepare for 0.19.10
2015-03-25 18:02:35 +01:00
release v0.19.9
2015-02-06 17:08:25 +01:00
ver 0.19.9 (2015/02/06)
decoder/DsdLib: raise ID3 tag limit to 1 MB A bug report was submitted with a 600 kB ID3 tag that could not be read by MPD.
2015-01-29 08:20:14 +01:00
* decoder
- dsdiff, dsf: raise ID3 tag limit to 1 MB
TagBuilder: allow adding duplicate tag types in Complement() Build a table of pre-existing tag types before adding new items. The old way would check HasType() each time, which would return true after the first instance of that tag type had been added, preventing duplicate tag types to be merged. This broke duplicate tag types loaded from the state file, because this code path uses TagBuilder::Complement().
2015-02-06 12:25:34 +01:00
* playlist: fix loading duplicate tag types from state file
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2015-02-06 17:07:47 +01:00
* despotify: remove defunct plugin
Avoid integer overflow in MonotonicClock{S,MS,US} This is Darwin specific: the previous implementation was causing an integer overflow when base.numer is very large. On PPC Darwin, the timebase info is 1000000000/33330116 and this is too large for integer arithmetic.
2015-01-26 06:54:16 +01:00
* fix clock integer overflow on OS X
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2015-02-06 17:07:47 +01:00
* fix gcc 5.0 warnings
thread/Name: include stdio.h if HAVE_PRCTL Caused a build failure with uClibc because snprintf() was not available.
2015-01-21 20:40:56 +01:00
* fix build failure with uClibc
Test the existence of strndup(3) before using it. This can eliminate the ad-hoc "#ifdef WIN32" and can also support other platforms lacking it as well (including Darwin 9).
2015-01-26 05:02:15 +01:00
* fix build failure on non-POSIX operating systems
Makefile.am: move org_musicpd_Bridge.h to BUILT_SOURCES
2015-01-30 23:39:06 +01:00
* fix dependency issue on parallel Android build
fs/io/FileOutputStream: don't auto-delete file on WIN32 The file handle is never reset to INVALID_HANDLE_VALUE, and thus the destructor will assume the operation shall be cancelled and will delete the temporary file. This was a major breakage for saving the database file and the state file.
2015-02-06 14:36:55 +01:00
* fix database/state file saving on Windows
configure.ac: prepare for 0.19.9
2015-01-16 17:26:50 +01:00
release v0.19.8
2015-01-14 23:11:29 +01:00
ver 0.19.8 (2015/01/14)
input/mms: limit the mmsx_read() size
2014-12-23 20:34:45 +01:00
* input
input/async: reset the "open" flag after seeking successfully Fixes a problem with the "curl" input plugin: IsEOF() always returns true because the "open" flag was cleared by CurlInputStream::RequestDone() when end-of-stream was reached. This flag stays false even when seeking to another position has succeeded. This patch resets the "open" flag to true after seeking successfully.
2015-01-06 12:46:28 +01:00
- curl: fix bug after rewinding from end-of-file
input/mms: limit the mmsx_read() size
2014-12-23 20:34:45 +01:00
- mms: reduce delay at the beginning of playback
DSF ID3 tags hitting 4k size limit Here's a change to dynamically allocate the DSD ID3 tag buffer. Pretty much anything with cover art is going to exceed the existing, static 4k limit... Here's a change to dynamically allocate the buffer and sanity check it at some upper limit. I rather arbitrarily pulled 256k out of thin air just to keep a corrupt file from causing it to trying to allocate a buffer larger than available memory.
2014-12-23 07:26:55 +01:00
* decoder
- dsdiff, dsf: allow ID3 tags larger than 4 kB
decoder/ffmpeg: support interleaved floating point
2014-12-23 20:51:08 +01:00
- ffmpeg: support interleaved floating point
Merge branch 'v0.18.x' into v0.19.x
2014-12-26 14:26:31 +01:00
* fix clang 3.6 warnings
thread/Name: disable pthread_setname_np() on NetBSD NetBSD's pthread_setname_np() prototype is incompatible with the rest of the world, and it requires to pass the string argument as a non-const pointer. Instead of working around this misdesign, I hereby disable the feature on NetBSD.
2015-01-06 12:08:36 +01:00
* fix build failure on NetBSD
configure.ac: prepare for 0.19.8
2014-12-23 09:42:17 +01:00
release v0.19.7
2014-12-17 19:20:54 +01:00
ver 0.19.7 (2014/12/17)
lib/nfs/Connection: fix crash while canceling a failing Open() The method NfsConnection::CancellableCallback::Callback() will always invoke NfsConnection::Close() on the file handle, even if the void pointer is not a nfsfh. This can happen if the Open() was not successful, e.g. when the file does not exist.
2014-12-14 15:09:55 +01:00
* input
- nfs: fix crash while canceling a failing file open operation
lib/nfs/Connection: fix memory leak (and assertion failure) nfs_destroy_context() will invoke all pending callbacks with err==-EINTR. In CancellableCallback::Callback(), this will invoke NfsConnection::DeferClose(), which however is only designed to be called from nfs_service(). In non-debug mode, this will leak memory because nfs_close_async() is never called. Workaround: before nfs_destroy_context(), invoke nfs_close_async() on all pending file handles.
2014-12-14 15:56:53 +01:00
- nfs: fix memory leak on connection failure
lib/nfs/Connection: fix reconnect after mount failure When mounting had not yet finished, SocketMonitor::IsDefined() was always false, due to the workaround at the beginning of the function that calls SocketMonitor::Steal(). This commit drops the IsDefined() check because it was never necessary and breaks reconnect.
2014-12-14 22:48:46 +01:00
- nfs: fix reconnect after mount failure
lib/nfs/Connection: implement mount timeout
2014-12-14 23:27:57 +01:00
- nfs: implement mount timeout (60 seconds)
storage/nfs: add timeout
2014-12-14 21:16:34 +01:00
* storage
- nfs: implement I/O timeout (60 seconds)
playlist/Print: don't skip non-existent songs in "listplaylist" Skipping those songs silently will confuse the client, because commands specifying the song index within a playlist (e.g. playlistdelete) will be out of sync. This copies spl_print()'s behavior to playlist_file_print().
2014-12-09 13:36:48 +01:00
* playlist
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2014-12-17 19:19:13 +01:00
- embcue: fix filename suffix detection
playlist/Print: don't skip non-existent songs in "listplaylist" Skipping those songs silently will confuse the client, because commands specifying the song index within a playlist (e.g. playlistdelete) will be out of sync. This copies spl_print()'s behavior to playlist_file_print().
2014-12-09 13:36:48 +01:00
- don't skip non-existent songs in "listplaylist"
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2014-12-17 19:19:13 +01:00
* decoder
- ffmpeg: fix time stamp underflow
util/HugeAllocator: disable MEM_LARGE_PAGES on Windows MEM_LARGE_PAGES does not appear to work. Instead, MEM_RESERVE appears to be necessary. Until I figure this out, this large pages are disabled.
2014-12-12 09:53:44 +01:00
* fix memory allocator bug on Windows
configure.ac: prepare for 0.19.7
2014-12-09 13:09:03 +01:00
release v0.19.6
2014-12-08 15:12:41 +01:00
ver 0.19.6 (2014/12/08)
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2014-12-08 15:03:09 +01:00
* decoder
- ffmpeg: support FFmpeg 2.5
* fix build failure with musl
android: update libFLAC to 1.3.1 Due to security vulnerabilities.
2014-11-28 20:16:33 +01:00
* android
- update libFLAC to 1.3.1
android/build.py: update FFmpeg to 2.5
2014-12-08 15:05:49 +01:00
- update FFmpeg to 2.5
configure.ac: prepare for 0.19.6
2014-11-28 20:13:57 +01:00
release v0.19.5
2014-11-26 20:05:45 +01:00
ver 0.19.5 (2014/11/26)
lib/nfs/Manager: defer NfsConnection destruction Avoids a crash that occurs when NfsConnection::OnSocketReady() dereferences itself before returning.
2014-11-25 10:42:52 +01:00
* input
- nfs: fix crash on connection failure
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2014-11-26 20:01:25 +01:00
* archive
- zzip: fix crash after seeking
decoder/{dsdiff,dsf,opus}: fix deadlock while seeking
2014-11-24 08:54:30 +01:00
* decoder
- dsdiff, dsf, opus: fix deadlock while seeking
decoder/mp4v2: remove because of incompatible license libmp4v2 is licensed under MPL 1.1, which is incompatible with GPLv2. Unfortunately, this means that we must remove the plugin. More information can be found in the Debian bug report: http://bugs.debian.org/767504
2014-11-24 22:18:31 +01:00
- mp4v2: remove because of incompatible license
configure.ac: prepare for 0.19.5
2014-11-24 07:34:34 +01:00
release v0.19.4
2014-11-18 21:40:52 +01:00
ver 0.19.4 (2014/11/18)
QueueCommands: workaround for buggy clients that send "add /"
2014-11-18 21:26:31 +01:00
* protocol
- workaround for buggy clients that send "add /"
decoder/opus: add MIME types audio/ogg and application/ogg
2014-11-12 15:14:34 +01:00
* decoder
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2014-11-18 21:38:44 +01:00
- ffmpeg: support opus
decoder/opus: add MIME types audio/ogg and application/ogg
2014-11-12 15:14:34 +01:00
- opus: add MIME types audio/ogg and application/ogg
release v0.18.18 -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAABAgAGBQJUa62OAAoJECNuiljG20US1CkP/3YaAg76hJDaeFAyi7/10lOs yI8tPmpQk8S+nuclsBJeIr+9xNRsjb1ebW4tI9uBRxqMT10SF7rMCH/qAmgnS9tQ 6M48hF0o9xegjz4x5BosEJjXtiekYN068Xi/hk+H27guVy14f6Xx7oRpdr5oI/O0 F83D1V2+vs6FuOyn42UQQNHVlYMac5SScHoSFomnwRYWI2wb2ay4H5Dlykb5BUl0 tTjJ2lYwN18vUoMxjmCNS+bLd7jag6FZ/PXntB1zXe22R17eAIuFZAMKaQvnQtUi QZjFPhpd+mBsZLfokB3belyNNb4F6qSG2W6lZrFRB+SU/LqnY2AL/dxPJD9eiufs 5jEIOw9hNBBKo217+hkPwp5pAF50u11tKXUXbKGeR42zAO0zll/+g2afVwoqdAfD llPDyQyXiB3tmA1bgJD7u9gaud3UaaXoxajPb5LnmklzGxiN5u1rnoo+E5ppBvQ4 31wPHGpzsjxEC7Xn9hErPP17jYXY1HT/xRb8XZNrFqJVZxSyjqQlDWomhCWDcvPc Du1PyKaQ08zgElEM5trwC9RjJxQdr0sckQcTnYC/0ksfJ7Nhh8M2yepEI/gmWbnO JBQ7YzpQhE9oMMH8jzVLLdC44yF6QvXcGQ3D9eUfwvPk9Z70mHlNI3Q6kbZ76U5j RHLp+oSaxFUXfPdciZaw =GO+k -----END PGP SIGNATURE----- Merge tag 'v0.18.18' into v0.19.x
2014-11-18 21:38:44 +01:00
* fix crash on failed filename charset conversion
* fix local socket detection from uid=0 (root)
configure.ac: prepare for 0.19.4
2014-11-12 15:16:07 +01:00
release v0.19.3
2014-11-11 11:21:42 +01:00
ver 0.19.3 (2014/11/11)
tag/Set: do AlbumArtist/Artist fallback only if AlbumArtist is not disabled On "list albumartist", songs that have no AlbumArtist tag will use the Artist tag. However, if AlbumArtist is disabled via "metadata_to_use", the TagBuilder::AddItem() call is ignored, and PrintUniqueTag() attempts to print a nullptr string. This commit fixes the problem by attempting the fallback only if AlbumArtist is not disabled.
2014-11-08 19:25:01 +01:00
* protocol
- fix "(null)" result string to "list" when AlbumArtist is disabled
lib/upnp/ContentDirectoryService: swap uri_apply_base() parameters When uri_apply_base() was moved from db/upnp/Util.cpp to util/UriUtil.cpp, the parameter order was changed, however without swapping the parameters in the ContentDirectoryService constructor.
2014-11-07 18:42:51 +01:00
* database
- upnp: fix breakage due to malformed URIs
input/curl: ignore ResponseBoundary() while seeking While seeking, metadata must not be updated. ResponseBoundary() was added in MPD 0.19.1, but I forgot to add the IsSeeking() check there. This caused the "seekable" flag to reset.
2014-11-10 08:45:19 +01:00
* input
- curl: another fix for redirected streams
input/AsyncInputStream: set Error when seeking unseekable Fixes crash in the "audiofile" decoder while logging the seek error.
2014-11-07 13:55:23 +01:00
* decoder
- audiofile: fix crash while playing streams
decoder/audiofile: fix bit rate calculation
2014-11-10 09:00:50 +01:00
- audiofile: fix bit rate calculation
decoder/ffmpeg: support opus
2014-11-07 19:22:26 +01:00
- ffmpeg: support opus
decoder/opus: fix mistyped LoadEOSPacket() return value
2014-11-11 11:18:51 +01:00
- opus: fix bogus duration on streams
decoder/opus: support chained streams
2014-11-11 07:41:01 +01:00
- opus: support chained streams
decoder/opus: improved error logging
2014-11-11 10:56:36 +01:00
- opus: improved error logging
pcm/SoxrResampler: round output buffer size up The old formula calculates the output buffer size with "regular" rounding (to the nearest integer), however sometimes, that is insufficient and the last sample cannot be resampled. This causes audible distortions. By changing the formula to consider the worst case (always round up), this problem is eliminated.
2014-11-10 22:52:17 +01:00
* fix distorted audio with soxr resampler
Main: fix compilation on OS X using non-Apple compilers Commit d42c0f1dc5063d50a62817b63a1c2a4507c46071 added an OS X-specific method of calling mpd_main_after_fork(), which uses Grand Central Dispatch. Since this uses a block literal, it breaks compilation on compilers which don't support the block extension, e.g. non-Apple compilers. This affects users on older OS X releases with GCD (which depend on older Clang releases, or Apple GCCs, which don't support the C++11 features MPD needs); or which don't support GCD at all (10.5 and lower). This patch changes the #ifdef so that the non-GCD code is used as it was on OS X before this patch if blocks aren't available, via checking __BLOCKS__ macro.
2014-11-11 04:38:22 +01:00
* fix build failure on Mac OS X with non-Apple compilers
configure.ac: prepare for 0.19.3
2014-11-04 09:24:09 +01:00
release v0.19.2
2014-11-02 13:46:32 +01:00
ver 0.19.2 (2014/11/02)
input/curl: forget Content-Length (and more) after redirect Fixes playback of redirected streams.
2014-11-02 12:59:45 +01:00
* input
- curl: fix redirected streams
playlist/m3u: recognize the file suffix ".m3u8"
2014-10-26 08:14:16 +01:00
* playlist
PlaylistFile: don't allow empty playlist name
2014-10-31 14:59:27 +01:00
- don't allow empty playlist name
TextInputStream: don't ignore unterminated last line
2014-10-28 22:10:47 +01:00
- m3u: don't ignore unterminated last line
playlist/m3u: recognize the file suffix ".m3u8"
2014-10-26 08:14:16 +01:00
- m3u: recognize the file suffix ".m3u8"
decoder/ffmpeg: recognize MIME type audio/aacp
2014-02-09 16:47:45 +01:00
* decoder
Decoder, Playlist: ignore URI query string for plugin detection Use the new uri_get_suffix() overload that removes the query string.
2014-11-01 13:20:39 +01:00
- ignore URI query string for plugin detection
decoder/faad: remove workaround for ancient libfaad2 ABI bug 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!
2014-10-25 20:42:50 +02:00
- faad: remove workaround for ancient libfaad2 ABI bug
decoder/ffmpeg: recognize MIME type audio/aacp
2014-02-09 16:47:45 +01:00
- ffmpeg: recognize MIME type audio/aacp
decoder/mad: fix negative replay gain values Negating an unsigned integer does not work.
2014-10-28 22:22:30 +01:00
- mad: fix negative replay gain values
OutputThread: close the output plugin after filter failure Fixes memory leak because ao_plugin_close() never gets called.
2014-10-23 23:19:40 +02:00
* output
- fix memory leak after filter initialization error
OutputThread: fall back to PCM if given DSD sample rate is not supported Works around the "PCM conversion from f to dsd is not implemented" error message that prevents DSD playback.
2014-10-25 21:25:49 +02:00
- fall back to PCM if given DSD sample rate is not supported
pcm/PcmConvert: assign {src,dest}_format at the end Fixes assertion failure in destructor by not assigning {src,dest}_format when an error occurs.
2014-10-23 22:31:54 +02:00
* fix assertion failure on unsupported PCM conversion
configure.ac: auto-disable plugins that require GLib when --disable-glib is used
2014-10-24 23:33:10 +02:00
* auto-disable plugins that require GLib when --disable-glib is used
configure.ac: prepare for 0.19.2
2014-10-23 21:57:04 +02:00
release v0.19.1
2014-10-19 01:03:17 +02:00
ver 0.19.1 (2014/10/19)
ThreadInputStream: call ThreadRead() inside the thread instead of Read() Fixes deadlock bug in the "mms" plugin.
2014-10-11 21:52:38 +02:00
* input
- mms: fix deadlock bug
playlist/extm3u: strip first line for #EXTM3U detection
2014-10-17 20:45:18 +02:00
* playlist
- extm3u: fix Extended M3U detection
TextInputStream: manually shift the buffer before reading Fixes truncated lines in m3u and cue files (regression by commit 67958f7).
2014-10-19 00:50:52 +02:00
- m3u, extm3u, cue: fix truncated lines
NEWS: add missing line
2014-10-11 21:48:52 +02:00
* fix build failure on Mac OS X
Makefile.am: distribute systemd/mpd.socket The file systemd/mpd.service.in is being distributed implicitly, but systemd/mpd.socket is not and needs to be added to EXTRA_DIST.
2014-10-12 08:38:52 +02:00
* add missing file systemd/mpd.socket to tarball
configure.ac: prepare for 0.19.1
2014-10-11 20:25:19 +02:00
release v0.19
2014-10-10 23:49:31 +02:00
ver 0.19 (2014/10/10)
command: add commands "addtagid", "cleartagid"
2013-12-02 12:00:32 +01:00
* protocol
doc: document the mount/neighbor commands
2014-10-10 23:20:50 +02:00
- new commands "addtagid", "cleartagid", "listfiles", "listmounts",
"listneighbors", "mount", "rangeid", "unmount"
command: "lsinfo" and "readcomments" allowed for remote files
2013-12-29 14:40:24 +01:00
- "lsinfo" and "readcomments" allowed for remote files
neighbor: new subsystem to detect file servers on the local network This commit adds the NeighborPlugin API which can be used to detect nearby file servers that can be used by input plugins. This list of servers is exported using the new "listneighbors" command. The idle even "neighbor" notifies interested clients when a new neighbor is found or an existing one is lost. There's a lot missing currently: protocol&user documentation, and a way to "mount" remote servers into the music database. Obviously, some code from the UPnP database plugin can be moved to a neighbor plugin.
2014-01-18 16:36:42 +01:00
- "listneighbors" lists file servers on the local network
PlaylistFile: use class SongLoader
2014-02-03 23:30:34 +01:00
- "playlistadd" supports file:///
Idle: error out when unrecognized idle event was specified Implements the error checks missing in commit 0bad8406
2014-02-09 08:05:02 +01:00
- "idle" with unrecognized event name fails
db/Helpers: "list" on album artist falls back to the artist tag
2014-04-24 10:48:52 +02:00
- "list" on album artist falls back to the artist tag
db/Count: implement grouping
2014-04-26 23:15:31 +02:00
- "list" and "count" allow grouping
SongFilter: new filter "modified-since"
2014-08-11 22:08:26 +02:00
- new "search"/"find" filter "modified-since"
ArgParser: allow fractional seconds in ParseCommandArg(SongTime)
2014-08-29 23:46:38 +02:00
- "seek*" allows fractional position
AllCommands: close connection after syntax error Stop HTTP clients from exploiting MPD via forged POST requests.
2014-08-12 15:33:45 +02:00
- close connection after syntax error
UPnP database plugin [mk: renamed source files, applied coding style, reduced bloat, using MPD's threading library, using MPD's error reporting and logging library and refactoring, fixed lots of bugs]
2013-11-01 19:26:01 +01:00
* database
db/proxy: forward "idle" events Send "idle" to the other MPD whenever there's nothing else to do and forward incoming "idle database" events to all our MPD clients.
2014-01-10 23:44:03 +01:00
- proxy: forward "idle" events
db/proxy: implement Update()
2014-07-29 23:16:46 +02:00
- proxy: forward the "update" command
db/proxy: copy "Last-Modified" from remote directories
2014-01-22 23:07:21 +01:00
- proxy: copy "Last-Modified" from remote directories
db/simple: compress the database file using gzip
2014-08-11 21:30:49 +02:00
- simple: compress the database file using gzip
UPnP database plugin [mk: renamed source files, applied coding style, reduced bloat, using MPD's threading library, using MPD's error reporting and logging library and refactoring, fixed lots of bugs]
2013-11-01 19:26:01 +01:00
- upnp: new plugin
db/update: cancel the update on shutdown
2014-02-27 16:36:11 +01:00
- cancel the update on shutdown
Main: support arbitrary storage plugins
2014-02-08 00:04:29 +01:00
* storage
- music_directory can point to a remote file server
storage/nfs: new storage plugin
2014-02-08 14:03:25 +01:00
- nfs: new plugin
Main: support arbitrary storage plugins
2014-02-08 00:04:29 +01:00
- smbclient: new plugin
http -> https for SoundCloud plugin
2014-01-02 12:09:40 +01:00
* playlist
SongPrint: use DetachedSong::GetDuration() Fixes the bogus duration of the last track in a CUE sheet.
2014-07-12 18:41:15 +02:00
- cue: fix bogus duration of the last track
QueueSave: use the long format to save partial songs Previously, only streams were saved with range and tags, but this is necessary for all "partial" songs (e.g. CUE tracks).
2014-07-12 19:36:39 +02:00
- cue: restore CUE tracks from state file
http -> https for SoundCloud plugin
2014-01-02 12:09:40 +01:00
- soundcloud: use https instead of http
NEWS: mention default soundcloud API key Missing from commit 48c96bba
2014-02-07 22:38:04 +01:00
- soundcloud: add default API key
SongUpdate: read tags from songs in an archive Add the TagStream.cxx library, similar to TagFile.cxx, and use it to load tags from song files inside archives.
2013-12-29 16:34:16 +01:00
* archive
- read tags from songs in an archive
add draft ALSA input plugin I've created an elementary input plugin that plays sound from the soundcard, so you can use MPD to listen to anything connected to the line-in jack, or to Video4Linux FM radio cards that send audio through the soundcard. There has been a small number of posts here in the past requesting line-in input, so here is a first, simplistic stab at it. The patch adds a new sheme, alsa://, which causes mpd to play data read directly from a souncdard. It defaults to hw:0,0, but you can pass any ALSA device name in the URI. So, using mpc for example: mpc add alsa:// mpc play will play from device hw:0,0. To use a diffferent device: mpc add alsa://hw:1,0
2013-12-15 17:52:21 +01:00
* input
- alsa: new input plugin
input/curl: options "verify_peer" and "verify_host"
2014-07-11 16:39:42 +02:00
- curl: options "verify_peer" and "verify_host"
input/ffmpeg: update offset after seeking
2014-08-18 09:52:53 +02:00
- ffmpeg: update offset after seeking
input/ffmpeg: use av_strerror() Generate more detailed error messages.
2014-08-18 20:20:51 +02:00
- ffmpeg: improved error messages
input/mms: move blocking I/O to thread
2014-03-16 11:14:16 +01:00
- mms: non-blocking I/O
input/nfs: new plugin using libnfs
2014-02-06 07:29:26 +01:00
- nfs: new input plugin
input/smbclient: new input plugin
2013-12-29 00:36:57 +01:00
- smbclient: new input plugin
pcm/Volume: improved dithering Instead of just adding a rectangular random value before shifting back to the normal scale, use the existing PcmDither library.
2013-12-22 17:39:26 +01:00
* filter
- volume: improved software volume dithering
decoder: vorbis, flac, opus: honor DESCRIPTION= tag in Xiph-based files as a comment to the song (see https://www.xiph.org/vorbis/doc/v-comment.html for Details)
2014-06-27 21:14:17 +02:00
* decoder:
- vorbis, flac, opus: honor DESCRIPTION= tag in Xiph-based files as a comment to the song
decoder/audiofile: implement scan_stream() instead of scan_file()
2014-07-10 09:23:56 +02:00
- audiofile: support scanning remote files
decoder/audiofile: log libaudiofile errors
2014-07-10 09:32:33 +02:00
- audiofile: log libaudiofile errors
Report bitrate for DSF and DSDIFF DSD decoders
2014-04-04 12:43:54 +02:00
- dsdiff, dsf: report bit rate
decoder/dsdiff: implement seeking
2014-08-23 15:27:21 +02:00
- dsdiff, dsf: implement seeking
decoder/dsf: Allow up to DSD512. Enable DSD rates based on Fs=48kHz
2014-07-04 14:15:09 +02:00
- dsf: support DSD512
decoder/dsf: fix multi-channel files The plugin was horribly bugged for files that were not stereo.
2014-08-22 06:56:58 +02:00
- dsf: support multi-channel files
decoder/dsf: fix big-endian bugs
2014-08-23 13:35:29 +02:00
- dsf: fix big-endian bugs
decoder/dsf: fix noise at end of malformed file 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.
2014-08-21 17:46:25 +02:00
- dsf: fix noise at end of malformed file
decoder/mpg123: support ID3v2, ReplayGain and MixRamp
2014-09-24 20:08:44 +02:00
- mpg123: support ID3v2, ReplayGain and MixRamp
decoder/sndfile: implement scan_stream() instead of scan_file()
2014-07-11 22:09:35 +02:00
- sndfile: support scanning remote files
decoder/sndfile: support more tag types
2014-07-11 22:01:11 +02:00
- sndfile: support tags "comment", "album", "track", "genre"
decoder/sndfile: support float and 16 bit samples Support these PCM formats natively, instead of letting libsndfile convert everything to 32 bit.
2014-09-19 21:40:22 +02:00
- sndfile: native floating point playback
- sndfile: optimized 16 bit playback
decoder/Mp4v2: add MP4v2 decoder plugin This plugin uses the MP4v2 library to play mp4/m4a files. It is limited to file_decode.
2014-08-07 19:52:56 +02:00
- mp4v2: support playback of MP4 files.
Shine encoding plugin This encoding plugin features a fixed-point mp3 encoder, with faster encoding on architectures without a FPU. Right now the encoder is limited to stereo and 16 bit depth. The bitrate and sample rate can be modified in audio_output. audio_output { type "httpd" name "My shine stream" encoder "shine" port "8000" format "44100:16:2" bitrate "320" # default: 128 }
2014-01-12 21:40:29 +01:00
* encoder:
- shine: new encoder plugin
output/alsa: support native DSD playback Translate SampleFormat::DSD to SND_PCM_FORMAT_DSD_U8, which was added to alsa-lib 1.0.27.1.
2014-08-23 16:12:18 +02:00
* output
- alsa: support native DSD playback
output/alsa, pcm: rename "DSD over USB" to "DoP" The standard has been renamed since the early draft that was implemented in MPD.
2014-08-31 16:12:26 +02:00
- alsa: rename "DSD over USB" to "DoP"
Main: run the OS X native event loop
2014-10-10 19:20:53 +02:00
- osx: fix hang after (un)plugging headphones
NEWS: add "threads" section
2014-01-30 18:28:08 +01:00
* threads:
- the update thread runs at "idle" priority
- the output thread runs at "real-time" priority
Main, OutputThread: increase kernel timer slack (Linux) Allows the kernel to combine timer wakeups with other processes, reducing power usage.
2014-01-30 18:37:30 +01:00
- increase kernel timer slack on Linux
NEWS: mention thread names
2014-01-30 18:31:34 +01:00
- name each thread (for debugging)
NEWS: add group "configuration"
2014-03-01 22:51:38 +01:00
* configuration
- allow playlist directory without music directory
fs/StandardDirectory: add GetUserCacheDir() Move code from CreateConfiguredDatabase() and add XDG support. This implements an automatic Linux fallback for the setting "db_file" if none was specified.
2014-03-01 22:50:11 +01:00
- use XDG to auto-detect "music_directory" and "db_file"
Tag: support "AlbumSort" The new tag is supported by all decoders that use the tag name table, and the ID3v2 tag "TSOA" maps to it.
2014-09-29 18:55:59 +02:00
* add tags "AlbumSort", "MUSICBRAINZ_RELEASETRACKID"
TagString: remove ISO-Latin-1 fallback 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.
2014-10-10 20:53:08 +02:00
* disable global Latin-1 fallback for tag values
pcm/SoxrResampler: new resampler option using libsoxr
2013-11-10 19:17:16 +01:00
* new resampler option using libsoxr
pcm/PcmFormat: ARM NEON optimizations for float->s16 This is nearly 4 times faster than the "portable" algorithm.
2014-03-14 23:45:27 +01:00
* ARM NEON optimizations
systemd: add socket activation files
2014-02-12 21:22:36 +01:00
* install systemd unit for socket activation
first Android release Finally, MPD runs on Android. For some small value of "runs". Very much work left, too much to describe.
2014-02-18 08:33:06 +01:00
* Android port
NEWS: prepare for 0.19
2013-12-01 19:12:58 +01:00
release v0.18.23
2015-02-06 17:04:14 +01:00
ver 0.18.23 (2015/02/06)
{playlist,input}/despotify: remove defunct plugin
2015-01-26 09:55:31 +01:00
* despotify: remove defunct plugin
Avoid integer overflow in MonotonicClock{S,MS,US} This is Darwin specific: the previous implementation was causing an integer overflow when base.numer is very large. On PPC Darwin, the timebase info is 1000000000/33330116 and this is too large for integer arithmetic.
2015-01-26 06:54:16 +01:00
* fix clock integer overflow on OS X
ClientProcess: cast enum to int before passing to printf() Fixes gcc5 warning.
2015-01-23 16:50:11 +01:00
* fix gcc 5.0 warnings
configure.ac: prepare for 0.18.23
2015-01-23 16:46:48 +01:00
NEWS: fix v0.18.22 release date
2015-02-01 12:22:17 +01:00
ver 0.18.22 (2015/01/14)
util/{ASCII,UriUtil}, ...: work around -Wtautological-pointer-compare New in clang 3.6.
2014-12-26 13:40:17 +01:00
* fix clang 3.6 warnings
configure.ac: prepare for 0.18.22
2014-12-26 13:23:04 +01:00
release v0.18.21
2014-12-17 19:13:47 +01:00
ver 0.18.21 (2014/12/17)
playlist/embcue: fix filename suffix detection The definition of the playlist_plugin struct member of the embcue plugin was incorrect.
2014-12-16 18:43:05 +01:00
* playlist
- embcue: fix filename suffix detection
decoder/ffmpeg: detect and fix negative time stamps Works around assertion failure due to something that appears to be a (minor) FFmpeg bug.
2014-12-11 10:50:20 +01:00
* decoder
- ffmpeg: fix time stamp underflow
configure.ac: prepare for 0.18.21
2014-12-15 00:39:52 +01:00
release v0.18.20
2014-12-08 14:57:17 +01:00
ver 0.18.20 (2014/12/08)
decoder/ffmpeg: support FFmpeg 2.5 Version 2.5 fixed an API oddity, however it broke API compatibility, at least with C++. Disable the workaround when a libavformat version is detected that is recent enough.
2014-12-08 14:25:34 +01:00
* decoder
- ffmpeg: support FFmpeg 2.5
thread/Id: drop "::" prefix before pthread function names The "::" to explicitly refer to the global namespace appeared like a good idea in C++, but it breaks with C libraries that implement standard functions using macros (e.g. musl).
2014-12-08 14:17:17 +01:00
* fix build failure with musl
configure.ac: prepare for 0.18.20
2014-12-08 14:13:20 +01:00
release v0.18.19
2014-11-26 19:58:48 +01:00
ver 0.18.19 (2014/11/26)
archive/zzip: fix inverted error handler Set the Error when zzip_seek()==-1 and not on success. Fixes a crash after seeking.
2014-11-24 22:08:11 +01:00
* archive
- zzip: fix crash after seeking
configure.ac: prepare for 0.18.19
2014-11-24 22:08:28 +01:00
release v0.18.18
2014-11-18 21:34:03 +01:00
ver 0.18.18 (2014/11/18)
decoder/ffmpeg: support opus
2014-11-07 19:22:26 +01:00
* decoder
- ffmpeg: support opus
Construct a Null AllocatedPath if the filename conversion into UTF8 failed
2014-11-11 17:03:29 +01:00
* fix crash on failed filename charset conversion
Client: assume uid==0 is local socket A negative uid value means it's not a "local socket" (PF_LOCAL). uid==0 means user "root" connected.
2014-11-18 20:56:27 +01:00
* fix local socket detection from uid=0 (root)
configure.ac: prepare for 0.18.1
2014-11-10 17:59:06 +01:00
release v0.18.17
2014-11-02 13:06:20 +01:00
ver 0.18.17 (2014/11/02)
playlist/m3u: recognize the file suffix ".m3u8"
2014-10-26 08:14:16 +01:00
* playlist
PlaylistFile: don't allow empty playlist name
2014-10-31 14:59:27 +01:00
- don't allow empty playlist name
playlist/m3u: recognize the file suffix ".m3u8"
2014-10-26 08:14:16 +01:00
- m3u: recognize the file suffix ".m3u8"
decoder/ffmpeg: recognize MIME type audio/aacp
2014-02-09 16:47:45 +01:00
* decoder
Decoder, Playlist: ignore URI query string for plugin detection Use the new uri_get_suffix() overload that removes the query string.
2014-11-01 13:20:39 +01:00
- ignore URI query string for plugin detection
decoder/faad: remove workaround for ancient libfaad2 ABI bug 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!
2014-10-25 20:42:50 +02:00
- faad: remove workaround for ancient libfaad2 ABI bug
decoder/ffmpeg: recognize MIME type audio/aacp
2014-02-09 16:47:45 +01:00
- ffmpeg: recognize MIME type audio/aacp
configure.ac: prepare for 0.18.17
2014-11-02 11:41:40 +01:00
release v0.18.16
2014-09-26 10:57:04 +02:00
ver 0.18.16 (2014/09/26)
configure.ac: fix DSD breakage due to typo
2014-09-26 10:55:53 +02:00
* fix DSD breakage due to typo in configure.ac
configure.ac: prepare for 0.18.16
2014-09-26 10:55:43 +02:00
release v0.18.15
2014-09-26 09:41:40 +02:00
ver 0.18.15 (2014/09/26)
command/list: reset used size after the list has been processed
2014-09-16 12:54:16 +02:00
* command
- list: reset used size after the list has been processed
OutputThread: retain negative mix ratio Fixes MixRamp breakage.
2014-09-18 13:50:23 +02:00
* fix MixRamp
thread/Posix{Cond,Mutex}: don't ues PTHREAD_*_INITIALIZER on NetBSD 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.
2014-09-13 11:26:17 +02:00
* work around build failure on NetBSD
configure.ac: prepare for 0.18.15
2014-09-13 11:14:41 +02:00
release v0.18.14
2014-09-11 19:09:49 +02:00
ver 0.18.14 (2014/09/11)
protocol/ArgParser: fix integer overflow in parse_range() 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.
2014-09-04 14:38:55 +02:00
* protocol
- fix range parser bug on certain 32 bit architectures
decoder/audiofile: fix crash after seeking Log call was added to the wrong branch. Fixes regression by commit ca1a1149
2014-09-06 19:32:10 +02:00
* decoder
- audiofile: fix crash after seeking
decoder/ffmpeg: pass MIME type to ffmpeg/libav version 11 That attribute was uninitialized before, which could crash libavformat. See Debian bug 760669
2014-09-07 22:05:33 +02:00
- ffmpeg: fix crash with ffmpeg/libav version 11
DecoderThread: clear the pipe when handling late SEEK See code comment. Fixes assertion failure in decoder_command_finished().
2014-09-07 21:31:10 +02:00
- fix assertion failure after seeking
configure.ac: prepare for 0.18.14
2014-09-03 19:59:26 +02:00
release v0.18.13
2014-08-31 14:50:23 +02:00
ver 0.18.13 (2014/08/31)
PlaylistControl: use SeekSongOrder(current) to keep current song The "current" attribute is a "song order", not a "song position". This is usually the same - except in random mode. Fixes Mantis ticket 0004073.
2014-08-31 14:44:20 +02:00
* protocol
- don't change song on "seekcur" in random mode
release v0.18.13 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1 iQIcBAABAgAGBQJUAxsyAAoJECNuiljG20USmRAP/2J/+kV5TRfqr7acFKh/aqip RAefHTvHpYuAUDOi3WDVSZcmMnaYGKslZhbXhqchbRfR2oMwYhR+5gHIoxOp+EcL 09UF1fJfUnjg6B2zz6tggIWR53Dwxckg4xEaKtI2Wi3Gz+9Q/CiNlpWLJc0ONA5e 9gcz8pNK3oN0Em9c1KXahZKArzGdOGeytfSI3PLE+QnRiEyEDz1+StxIuyl72vr4 ar2EMa9Vdvb7ouNK0g8WqaEOzYyNDXhklIqc+zALQJGThY1xj6OCfQ8JVagJhzwB UDuYfX1FylkN8LhF5pNzH1F0ANMozWcqeZlLLph6vuKkVBD7WUX4X1t82bwUlFDB ppdA5pPkiXsqYnuhHML4GqL7LlI2wsMxUHK7v+A9+9hfGN9HIoFyOSLBjB1pyI93 yi8IPOq2a1BJQfQ2c69693ZS7WPxU6vOFKucKrk9PBlr2900SlvpMzIpZr1PMnGV hoU8eLHOIW86wi5Ea2ycFbFyI1ZoyrKJmzcVJjPjZUT05jDV6bA2Gi1IG6gIhX1q DiBfXUre+GISkLS4r8GcbG1FIXcv7fGjZ6Vols5di8lpYQ5oBu5aw2TEf8zukrHC LzPRa2l6Mhe2OI7a/mmTaEYDINgCL//46hOU5mIbmB0XZS8Jf+D1/HBzqKpeqO7f idENtbelqQdSsR5UU+R0 =c2vC -----END PGP SIGNATURE----- Merge tag 'v0.18.13'
2014-08-31 15:01:12 +02:00
decoder/ffmpeg: use avcodec_descriptor_get() to determine codec name In version 11, both ffmpeg and libav deprecate AVCodecContext::codec_name. The function avcodec_descriptor_get() has been introduced long ago.
2014-08-13 18:40:39 +02:00
* decoder
decoer/dsdiff: fix endless loop on malformed file Same bug as in the previous commit.
2014-08-21 12:48:03 +02:00
- dsdiff, dsf: fix endless loop on malformed file
decoder/ffmpeg: use avcodec_descriptor_get() to determine codec name In version 11, both ffmpeg and libav deprecate AVCodecContext::codec_name. The function avcodec_descriptor_get() has been introduced long ago.
2014-08-13 18:40:39 +02:00
- ffmpeg: support ffmpeg/libav version 11
decoder/gme: fix song duration The unit of gme_info_t::length is milliseconds, not centiseconds.
2014-08-29 23:03:29 +02:00
- gme: fix song duration
output/alsa: fix endless loop at end of file in dsd_usb mode
2014-08-31 13:58:04 +02:00
* output
- alsa: fix endless loop at end of file in dsd_usb mode
event/TimeoutMonitor: reset "active" flag before invoking OnTimeout() 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.
2014-08-24 13:13:12 +02:00
* fix state file saver
system/ByteOrder: <endian.h> is a non-standard header that only Linux provides.
2014-08-23 14:27:44 +02:00
* fix build failure on Darwin
configure.ac: prepare for 0.18.13
2014-08-02 08:45:44 +02:00
release v0.18.12
2014-07-30 10:30:17 +02:00
ver 0.18.12 (2014/07/30)
db/proxy: use mpd_song_get_{start,end}() only with libmpdclient >= 2.3
2014-06-23 08:57:51 +02:00
* database
- proxy: fix build failure with libmpdclient 2.2
db/proxy: fall back to recursive walk on old libmpdclient/MPD Error message was 'too few arguments for "find"' because the "base" constraint was not supported, and no other constraints remained.
2014-06-23 09:12:51 +02:00
- proxy: fix add/search and other commands with libmpdclient < 2.9
decoder/audiofile: use decoder_read() .. instead of InputStream::LockRead(). The former is cancellable.
2014-07-09 18:46:58 +02:00
* decoder
- audiofile: improve responsiveness
decoder/audiofile: use decoder_read_full() Works around WAV stream playback bug, because libaudiofile does not like partial reads (Mantis 0004028).
2014-07-09 19:05:20 +02:00
- audiofile: fix WAV stream playback
decoder/dsd: use decoder_read_full() where appropriate Addresses Mantis ticket 0004015. [mk: use decoder_read_full() only when needed, and a few formal changes]
2014-07-09 19:18:36 +02:00
- dsdiff, dsf: fix stream playback
decoder/dsdiff: fix metadata parser bug (uninitialized variables)
2014-07-12 20:41:21 +02:00
- dsdiff: fix metadata parser bug (uninitialized variables)
decoder/faad: estimate song duration for remote files 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.
2014-07-11 22:52:26 +02:00
- faad: estimate song duration for remote files
decoder/sndfile: use decoder_read() .. instead of InputStream::LockRead(). The former is cancellable.
2014-07-11 20:50:44 +02:00
- sndfile: improve responsiveness
Playlist: randomize next song when enabling "random" mode while not playing 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.
2014-07-11 19:41:39 +02:00
* randomize next song when enabling "random" mode while not playing
PlaylistEdit: postpone UpdateQueuedSong() when adding multiple songs 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.
2014-07-11 20:01:53 +02:00
* randomize next song when adding to single-song queue
configure.ac: prepare for 0.18.12
2014-06-23 09:14:35 +02:00
release v0.18.11
2014-05-12 18:20:26 +02:00
ver 0.18.11 (2014/05/12)
decoder/OggUtil: allow skipping up to 32 kB after seek Fixes missing song length on high-latency Opus files. According to tests with 320 kbit/s opus files with 60ms packets, we need to skip up to 29 kB.
2014-04-29 11:56:05 +02:00
* decoder
- opus: fix missing song length on high-latency files
DeferredMonitor: fix race condition when using GLib event loop Turns out the lock-free code using atomics was not thread-safe. The given callback could be invoked by GLib before the source_id attribute was assigned. This commit changes the DeferredMonitor class to use a Mutex to block the event loop until source_id is assigned. This bug does not exist in the 0.19 branch because it does not use the GLib main loop anymore.
2014-04-26 22:11:23 +02:00
* fix race condition when using GLib event loop (non-Linux)
configure.ac: prepare for 0.18.11
2014-04-26 22:08:08 +02:00
release v0.18.10
2014-04-10 13:36:38 +02:00
ver 0.18.10 (2014/04/10)
decoder/gme: fix memory leak in container_scan()
2014-03-06 13:12:39 +01:00
* decoder
decoder/ffmpeg: pass AVSEEK_FLAG_ANY to av_seek_frame() This corrects a major mistake from commit 724a59aa - there was one small thing that commit was supposed to do, and it failed. AV_TIME_BASE is not a seek flag.
2014-03-18 09:10:36 +01:00
- ffmpeg: fix seeking bug
decoder/ffmpeg: handle unknown stream start time
2014-03-18 08:19:05 +01:00
- ffmpeg: handle unknown stream start time
decoder/gme: fix memory leak in container_scan()
2014-03-06 13:12:39 +01:00
- gme: fix memory leak
decoder/sndfile: work around libsndfile bug on partial read
2014-04-09 23:58:56 +02:00
- sndfile: work around libsndfile bug on partial read
PlaylistEdit: don't interrupt playback when current song gets deleted
2014-04-09 23:10:14 +02:00
* don't interrupt playback when current song gets deleted
configure.ac: prepare for 0.18.10
2014-03-06 13:08:30 +01:00
release v0.18.9
2014-03-02 11:25:01 +01:00
ver 0.18.9 (2014/03/02)
AllCommands: "findadd" requires the "add" permission
2014-02-27 23:08:22 +01:00
* protocol
- "findadd" requires the "add" permission
output/alsa: call snd_pcm_prepare() after snd_pcm_drop() Don't wait for an optimistic write to fail. This is an improved workaround for the infamous Raspberry Pi bug (see commit af991765). It works much better and comes without the negative side effects. The old workaround is now obsolete.
2014-03-02 11:12:09 +01:00
* output
- alsa: improved workaround for noise after manual song change
configure.ac: fix linker failure when libvorbis/libogg are static Link libvorbisfile first, followed to libvorbis and finally libogg. This order is necessary because libvorbisfile depends on libvorbis.
2014-02-18 18:39:19 +01:00
* decoder
- vorbis: fix linker failure when libvorbis/libogg are static
configure.ac: link the Vorbis encoder with libogg Fixes another linker failure. Similar to commit ea406875
2014-02-17 19:42:38 +01:00
* encoder
- vorbis: fix another linker failure
event/SignalMonitor: unblock signals after fork Fixes hanging child process in the "pipe" output plugin.
2014-02-18 19:13:50 +01:00
* output
- pipe: fix hanging child process due to blocked signals
event/SignalMonitor: fix build failure due to missing signal.h include
2014-03-02 10:21:31 +01:00
* fix build failure due to missing signal.h include
configure.ac: prepare for 0.18.9
2014-02-09 22:58:14 +01:00
release v0.18.8
2014-02-07 00:06:17 +01:00
ver 0.18.8 (2014/02/07)
decoder/ffmpeg: support libav v10_alpha1
2014-01-15 11:13:50 +01:00
* decoder
- ffmpeg: support libav v10_alpha1
configure.ac: link the Vorbis encoder with libvorbis Since the encoder plugin uses a libvorbis function (and not only libvorbisenc functions), we need to link with libvorbis explicitly.
2014-02-06 21:32:50 +01:00
* encoder
- vorbis: fix linker failure
doc/user: document the RoarAudio output plugin
2014-02-06 21:46:29 +01:00
* output
- roar: documentation
IcyMetadataParser: more robust tag parser Allow semicolons and single quotes in the stream title. This is not part of any specification, but found in real life.
2014-01-21 14:51:35 +01:00
* more robust Icy-Metadata parser
Queue: rename struct queue to Queue Works around a build failure on Solaris because annoyingly, Solaris reserves the name "queue". This rename was pending anyway.
2014-01-20 08:57:22 +01:00
* fix Solaris build failure
configure.ac: prepare for 0.18.8
2014-01-15 11:22:06 +01:00
NEWS: fix 0.18.7 release year
2014-02-07 00:06:28 +01:00
ver 0.18.7 (2014/01/13)
configure.ac: improved check for libyajl 1.0 If we have libyajl 2.0.1 (without a pkg-config file), our configure.ac would assume this is the libyajl 1.0 API, because the function yajl_alloc() exists in both. This commit changes the library check to the function yajl_parse_complete() which was removed in the 2.0 API. This fixes build failure with libyajl 2.0.1.
2013-12-29 14:12:33 +01:00
* playlist
playlist/pls: don't free stack buffer
2014-01-08 19:50:44 +01:00
- pls: fix crash after parser error
configure.ac: improved check for libyajl 1.0 If we have libyajl 2.0.1 (without a pkg-config file), our configure.ac would assume this is the libyajl 1.0 API, because the function yajl_alloc() exists in both. This commit changes the library check to the function yajl_parse_complete() which was removed in the 2.0 API. This fixes build failure with libyajl 2.0.1.
2013-12-29 14:12:33 +01:00
- soundcloud: fix build failure with libyajl 2.0.1
decoder/faad: fix memory leak
2014-01-08 22:11:00 +01:00
* decoder
- faad: fix memory leak
configure.ac: reject libmpcdec SV7 in configure script Look for symbol "mpc_demux_init" which does not exist in SV7. This avoids build failures when SV7 was found by configure.ac.
2014-01-11 21:02:12 +01:00
- mpcdec: reject libmpcdec SV7 in configure script
daemon: no initgroups() when already running as the configured user We can assume that initgroups() would be a no-op in that case, however initgroups() is not allowed for unprivileged users anyway.
2013-12-29 13:58:39 +01:00
* daemon: don't initialize supplementary groups when already running
as the configured user
configure.ac: prepare for 0.18.7
2013-12-29 10:40:59 +01:00
release v0.18.6
2013-12-24 12:01:01 +01:00
ver 0.18.6 (2013/12/24)
input/cdio_paranoia: support libcdio-paranoia 0.90
2013-11-27 08:24:34 +01:00
* input
- cdio_paranoia: support libcdio-paranoia 0.90
riff: recognize upper-case "ID3" chunk name Some tagging libraries (eg. TagLib) produce that variant.
2013-12-19 09:32:01 +01:00
* tags
- riff: recognize upper-case "ID3" chunk name
fix FfmpegDecoderPlugin to use relative timestamps
2013-12-20 21:45:07 +01:00
* decoder
- ffmpeg: use relative timestamps
NEWS: add openal line
2013-12-10 19:19:27 +01:00
* output
- openal: fix build failure on Mac OS X
output/osx: fix build failure
2013-12-10 19:16:37 +01:00
- osx: fix build failure
mixer/alsa: use BlockingCall() instead of EventLoop::AddCall() This is safer, and works without epoll(). Fixes a build failure with uClibc, which does not support epoll().
2013-12-13 14:35:36 +01:00
* mixer
- alsa: fix build failure with uClibc
OutputControl: update both ReplayGainFilters The "mode" of the second ReplayGainFilter was never set, and thus replay gain was never applied to the new song during cross-fade.
2013-12-24 11:51:37 +01:00
* fix replay gain during cross-fade
SongUpdate: accept files without metadata If the file was recognized by a decoder plugin, accept it - don't require metadata.
2013-12-10 19:34:35 +01:00
* accept files without metadata
configure.ac: prepare for 0.18.6
2013-11-27 08:24:54 +01:00
release v0.18.5
2013-11-23 18:08:39 +01:00
ver 0.18.5 (2013/11/23)
Mapper: _get_music_directory_utf8() may return nullptr If no music_directory is configured, return nullptr instead of an empty string. This fixes a crash when db_file is configured without music_directory.
2013-11-22 00:02:17 +01:00
* configuration
- fix crash when db_file is configured without music_directory
Stats: print db statistics only if db is available Fixes crash on "stats" in certain configurations.
2013-11-22 00:23:17 +01:00
- fix crash on "stats" without db_file/music_directory
Stats: auto-reload statistics with proxy plugin
2013-11-21 23:30:49 +01:00
* database
- proxy: auto-reload statistics
db/proxy: implement method GetUpdateStamp()
2013-11-22 00:45:27 +01:00
- proxy: provide "db_update" in "stats" response
input/curl: work around stream resume bug (fixed in libcurl 7.32.0)
2013-11-23 11:59:32 +01:00
* input
- curl: work around stream resume bug (fixed in libcurl 7.32.0)
configure.ac: auto-detect fluidsynth by default The option "--enable-fluidsynth" was documented to be "auto" by default, when it was really "no".
2013-11-21 22:50:58 +01:00
* decoder
- fluidsynth: auto-detect by default
PcmResampleLibsamplerate: clip 24 bit data Using pcm_resample_lsr_32() for 24 bit samples works, but may cause 24 bit overflows. This commit makes 24 bit a special case with explicit clipping.
2013-11-22 22:50:29 +01:00
* clip 24 bit data from libsamplerate
system/ByteOrder: testing for endianness in a more general way Fixes build tests on ia64 and mipsel by testing for endianness in a more general / portable way.
2013-11-21 22:56:45 +01:00
* fix ia64, mipsel and other little-endian architectures
NEWS: add version 0.18.5 change log
2013-11-18 11:40:00 +01:00
* fix build failures due to missing includes
* fix build failure with static libmpdclient
release v0.18.4
2013-11-13 18:26:01 +01:00
ver 0.18.4 (2013/11/13)
decoder/dsdiff: fix byte order bug
2013-11-09 17:02:49 +01:00
* decoder
- dsdiff: fix byte order bug
filter/route, ...: add missing stdlib.h includes
2013-11-11 08:17:29 +01:00
* fix build failures due to missing includes
util/RefCount: no "constexpr" with libc++ Not supported by libc++.
2013-11-11 08:26:16 +01:00
* libc++ compatibility
configure.ac: prepare for 0.18.4
2013-11-09 09:52:51 +01:00
release v0.18.3
2013-11-08 12:55:12 +01:00
ver 0.18.3 (2013/11/08)
PlayerThread: fix stuck MPD after song change (0.18.2 regression) Commit 77c63511 caused MPD to become stuck right after a song change. The problem was that at some point, the MusicBuffer became full, and the DecoderThread working on the next song waits for the PlayerThread. However, the PlayerThread was stuck in a loop of g_usleep() calls, and never bothered to tell the DecoderThread that the MusicBuffer is not full anymore. This bug is very old, but its chance to occur went from nearly 0% to nearly 100%. The fix is to wake up the DecoderThread before waiting for it. As a side effect, I replaced the g_usleep() call with a Cond::Wait() call.
2013-11-08 11:54:30 +01:00
* fix stuck MPD after song change (0.18.2 regression)
configure.ac: prepare for 0.18.3
2013-11-08 11:57:37 +01:00
release v0.18.2
2013-11-07 18:41:10 +01:00
ver 0.18.2 (2013/11/07)
ClientRead: "close" flushes the output buffer Add a new CommandResult code called "FINISH" which, unlike "CLOSE", will attempt to flush the output buffer. This is a one-shot attempt; it will do one write, and not try again.
2013-11-06 22:01:06 +01:00
* protocol:
- "close" flushes the output buffer
input/curl: fix bug with redirected streams Migrate from the old curl_multi_perform() API to the newer curl_multi_socket_action() API (since CURL 7.16). This allows working around a bug with HTTP redirections with epoll: when CURL closes a socket and the new one happens to have the same file number, MPD did not have a chance to remove the old one from epoll and subsequently attempted to use EPOLL_CTL_MOD, which was not allowed by epoll, because it's a new socket now.
2013-11-06 19:01:50 +01:00
* input:
input/cdio_paranoia: add setting "default_byte_order" Allows big-endian users to configure the fallback byte order to little-endian. Without this setting, MPD assumes native byte order if the CD drive can't decide.
2013-11-07 18:17:54 +01:00
- cdio_paranoia: add setting "default_byte_order"
input/curl: fix bug with redirected streams Migrate from the old curl_multi_perform() API to the newer curl_multi_socket_action() API (since CURL 7.16). This allows working around a bug with HTTP redirections with epoll: when CURL closes a socket and the new one happens to have the same file number, MPD did not have a chance to remove the old one from epoll and subsequently attempted to use EPOLL_CTL_MOD, which was not allowed by epoll, because it's a new socket now.
2013-11-06 19:01:50 +01:00
- curl: fix bug with redirected streams
playlist/pls: fix reversed song order Remove the forward_list::reverse() call. It was not necessary, because pls_parser() already reads the playlist in reverse order.
2013-11-05 18:22:34 +01:00
* playlist:
- pls: fix reversed song order
decoder/dsf: enable DSD128
2013-11-04 13:20:09 +01:00
* decoder:
configure.ac: require libaudiofile 0.3 due to API breakage Prior to version 0.3, the "length" callback returned a "long" instead of AFfileoffset. Now that this API bug fix is a few years old, let's drop 0.2 support for good.
2013-11-06 20:04:21 +01:00
- audiofile: require libaudiofile 0.3 due to API breakage
decoder/dsf: enable DSD128
2013-11-04 13:20:09 +01:00
- dsf: enable DSD128
PlayerThread: enable buffering when starting playback For some reason, this got lost in commit 975deca8. Re-enabling this fixes stuttering at the beginning of radio streams.
2013-11-06 20:14:38 +01:00
* enable buffering when starting playback (regression fix)
NEWS: add missing line
2013-11-05 17:38:41 +01:00
* fix build failures due to missing includes
system/ByteOrder: fix big-endian support D'oh!
2013-11-05 22:08:53 +01:00
* fix big-endian support
configure.ac: prepare for 0.18.2
2013-11-05 11:08:17 +01:00
release v0.18.1
2013-11-04 23:46:14 +01:00
ver 0.18.1 (2013/11/04)
ClientRead: always ignore whitespace at the end of the line
2013-11-04 18:11:15 +01:00
* protocol:
- always ignore whitespace at the end of the line
system/resolver: return path of UNIX domain sockets getnameinfo() doesn't work well - it always returns "localhost".
2013-11-04 18:39:42 +01:00
* networking:
- log UNIX domain path names instead of "localhost"
event/ServerSocket: open sockets in the order they were configured Use a std::list which can be appended at the end.
2013-11-04 20:10:46 +01:00
- open listener sockets in the order they were configured
event/ServerSocket: don't abort if IPv6 is not available First check if an IPv6 socket can be created.
2013-11-04 23:36:00 +01:00
- don't abort if IPv6 is not available
output/alsa: avoid endless loop in Raspberry Pi workaround See code comment.
2013-11-04 23:26:24 +01:00
* output:
- alsa: avoid endless loop in Raspberry Pi workaround
filter/AutoConvert: modify child_audio_format, not in_audio_format This prevented using the "volume_normalization" feature with some codecs (e.g. mp3), because the normalization code requires 16 bit samples. If the codec happens to deliver formats other than S16, the AutoConvert filter succeeds to initialize the conversion filter, but the returned input audio format was wrong.
2013-11-04 07:59:15 +01:00
* filter:
- autoconvert: fix "volume_normalization" with mp3 files
Makefile.am: distribute missing files
2013-11-04 08:08:54 +01:00
* add missing files to source tarball
configure.ac: prepare for 0.18.1
2013-11-04 07:40:52 +01:00
release v0.18
2013-10-31 00:09:19 +01:00
ver 0.18 (2013/10/31)
Listen: Allow tilde paths for socket.
2013-08-25 15:51:41 +02:00
* configuration:
- allow tilde paths for socket
fs/Charset: default filesystem charset is UTF-8 Implement a fast path for UTF-8 which leaves fs_charset empty, and don't assign a value to fs_charset if there's no configuration.
2013-10-17 22:42:22 +02:00
- default filesystem charset is UTF-8 instead of ISO-8859-1
increase default buffer size to 4 MB 2 MB was too small for cross-fading a 24 bit file. Increasing to 4 MB is still not too large for weak machines, but is enough for cross-fading.
2013-10-30 21:46:34 +01:00
- increase default buffer size to 4 MB
Commands: new command "toggleoutput"
2013-08-04 23:20:56 +02:00
* protocol:
command: new command "readcomments" lists arbitrary file tags
2013-10-25 19:35:40 +02:00
- new command "readcomments" lists arbitrary file tags
Commands: new command "toggleoutput"
2013-08-04 23:20:56 +02:00
- new command "toggleoutput"
SongFilter: "any" does not match file name According to the protocol documentation, matching the file name was wrong. This removes some awkward special-case code.
2013-10-29 19:54:40 +01:00
- "find"/"search" with "any" does not match file name
SongFilter: add special keyword "base" Restores the features from the previous draft commands "findin" / "searchin".
2013-10-29 18:54:34 +01:00
- "search" and "find" with base URI (keyword "base")
SongFilter: search for album artist falls back to the artist tag Implement Mantis ticket 0003646.
2013-09-26 19:25:13 +02:00
- search for album artist falls back to the artist tag
OtherCommands: re-add the "volume" command This command was removed by commit 206392ad (MPD 0.16), even though it was been proven useful for some very simple clients. On request, I add it to the protocol again.
2013-10-25 00:11:10 +02:00
- re-add the "volume" command
playlist/lastfm: remove defunct Last.fm support This plugin has been defunct ever since Last.fm changed their protocol. Since there is no volunteer willing to fix the plugin, I'm removing it now.
2013-09-28 14:00:07 +02:00
* input:
input/curl: enable https
2013-09-03 13:07:33 +02:00
- curl: enable https
input/soup: plugin removed This plugin is cumbersome to support, now that MPD is migrating away from GLib and the GLib event loop. It has no practical advantages over the CURL plugin. Soup requires the bloated GType library.
2013-04-17 21:46:16 +02:00
- soup: plugin removed
playlist/lastfm: remove defunct Last.fm support This plugin has been defunct ever since Last.fm changed their protocol. Since there is no volunteer willing to fix the plugin, I'm removing it now.
2013-09-28 14:00:07 +02:00
* playlist:
- lastfm: remove defunct Last.fm support
src/decoder/opus: new decoder plugin for the Opus codec Using libopus and libogg.
2012-09-04 09:26:18 +02:00
* decoder:
NEWS: mention the AdPlug decoder
2012-09-25 23:30:55 +02:00
- adplug: new decoder plugin using libadplug
decoder/dsf: don't play junk at the end of the "data" chunk
2013-10-28 23:01:13 +01:00
- dsf: don't play junk at the end of the "data" chunk
decoder/ffmpeg: require ffmpeg/libav 0.8 Now that even Debian stable has picked up version 0.8, we can safely make this a requirement, and remove a lot of old cruft.
2013-07-31 01:01:21 +02:00
- ffmpeg: drop support for pre-0.8 ffmpeg
{decoder,encoder}/flac: drop support for libFLAC 1.1
2012-10-02 10:04:44 +02:00
- flac: require libFLAC 1.2 or newer
decoder/flac: support FLAC files inside archives Implement the "scan_stream" method that can read tags from any input_stream object. This requires a FLAC__IOCallbacks implementation based on the input_stream API.
2012-10-04 07:09:31 +02:00
- flac: support FLAC files inside archives
src/decoder/opus: new decoder plugin for the Opus codec Using libopus and libogg.
2012-09-04 09:26:18 +02:00
- opus: new decoder plugin for the Opus codec
decoder/vorbis: skip 16 bit quantisation, provide float samples Internally the vorbis (non-Tremor) decoder is working in floating point, and it's not really necessary to cut the output back to 16-bit if the soundcard or OS supports higher resolution. The decoder can be trivially modified to bypass its internal quantisation and produce floating-point output, and a separate quantisation can be used as appropriate to the platform.
2012-09-25 21:08:32 +02:00
- vorbis: skip 16 bit quantisation, provide float samples
decoder/mikmod: Add loop configuration parameter This patch allows the user to configure the mikmod decoder plugin to loop modules. It adds a configuration parameter to the mikmod decoder called "loop" which can be "no" (the old behaviour, default) or "yes" to allow modules to use backward loops.
2013-10-18 04:12:36 +02:00
- mikmod: add "loop" configuration parameter
decoder/modplug: Add "loop_count" parameter The "loop_count" configuration parameter allows the user to set how many times a module with backward loops shall loop. "0" (the default) means a module is not allowed to use backward loops at all. "-1" enables inifinite looping.
2013-10-19 15:39:04 +02:00
- modplug: add "loop_count" configuration parameter
decoder/mp4ff: delete obsolete plugin The underlying library has been obsolete for many years.
2013-01-28 20:43:01 +01:00
- mp4ff: obsolete plugin removed
encoder/opus: new encoder plugin for the Opus codec
2012-10-01 20:15:15 +02:00
* encoder:
- opus: new encoder plugin for the Opus codec
encoder/vorbis: accept floating point input samples Improves quality by not squeezing 32 bit samples down to 16 bit, and then back to 32 bit floating point. Reduces CPU usage by skipping a conversion step.
2012-10-02 08:19:45 +02:00
- vorbis: accept floating point input samples
output: new option "tags" may be used to disable sending tags Implements Mantis ticket 0003340.
2012-09-25 23:28:53 +02:00
* output:
- new option "tags" may be used to disable sending tags to output
output/alsa: workaround for noise after manual song change Workaround for driver bug observed on the Raspberry Pi, see code comment for details.
2013-02-04 15:41:33 +01:00
- alsa: workaround for noise after manual song change
output/ffado: remove broken plugin This plugin has been in MPD for three years, and it has never worked. Enough!
2013-04-17 00:44:02 +02:00
- ffado: remove broken plugin
output/httpd: support HEAD requests
2013-10-25 00:32:48 +02:00
- httpd: support HEAD requests
output/mvp: remove obsolete plugin The hardware is obsolete, and the product does not exist anymore on the Hauppauge web site. Let's see if anybody complains about the removal.
2013-04-17 00:33:37 +02:00
- mvp: remove obsolete plugin
configure.ac: disable the OS X output plugin by default There hasn't been a maintainer for this plugin for years. It may even fail to build.
2013-10-28 22:25:58 +01:00
- osx: disabled by default because it's unmaintained and unsupported
player_control: add GError attribute Rewrite of the pc_get_error_message() function, now using a GError object instead of the complicated "errored_song" attribute.
2012-08-08 22:18:08 +02:00
* improved decoder/output error reporting
StateFile: schedule timer only after a change Save the state file 2 minutes after the last change. This reduces the disruptions by an idle MPD, and MPD can be paged out permanently until it is used.
2013-04-08 22:31:51 +02:00
* eliminate timer wakeup on idle MPD
DecoderControl: convert mutex and client_cond to a reference Share the Mutex between the DecoderThread and the PlayerThread. This simplifies synchronization between the two threads and fixes a freeze problem: while the PlayerThread waits for the DeocderThread, it cannot answer requests from the main thread, and the main thread will block until the DecoderThread finishes.
2013-10-28 10:09:04 +01:00
* fix unresponsive MPD while waiting for stream
release v0.18
2013-10-31 00:09:19 +01:00
* port of the source code to C++11
Merge branch 'v0.17.x'
2013-01-16 20:05:26 +01:00
release 0.17.6
2013-10-14 21:15:55 +02:00
ver 0.17.6 (2013/10/14)
mixer/alsa: handle ENODEV Fixes busy loop when USB sound device gets unplugged (Mantis bug #3824).
2013-10-01 16:33:07 +02:00
* mixer:
- alsa: fix busy loop when USB sound device gets unplugged
decoder/modplug: fix include directory
2013-10-14 21:12:47 +02:00
* decoder:
- modplug: fix build with Debian package 1:0.8.8.4-4
load_file
2013-10-14 20:54:05 +02:00
* stored playlists:
- fix loading playlists with references to local files
stored_playlist: use fs_charset_to_utf8() for URLs
2013-10-14 21:11:20 +02:00
- obey filesystem_charset for URLs
prepare 0.17.6
2013-10-01 16:29:54 +02:00
release 0.17.5
2013-08-04 14:21:45 +02:00
ver 0.17.5 (2013/08/04)
commmand: fix URI argument in playlistadd
2013-08-01 08:42:22 +02:00
* protocol:
- fix "playlistadd" with URI
playlist_edit: fix "move" relative to current when there is no current song Fixes Mantis #3770.
2013-08-01 09:28:03 +02:00
- fix "move" relative to current when there is no current song
decoder/mikmod: use MikMod_free() to free the title on libmikmod-3.2 Player_LoadTitle() returns an aligned pointer in libmikmod-3.2 that cannot be freed with free(). The correct way to do this now is MikMod_free() which extracts the original pointer from the buffer and frees that. Signed-off-by: Christoph Mende <mende.christoph@gmail.com>
2013-04-20 08:29:11 +02:00
* decoder:
FfmpegDecoderPlugin: add application/flv to mime types list
2013-07-02 22:16:46 +02:00
- ffmpeg: support "application/flv"
decoder/mikmod: use MikMod_free() to free the title on libmikmod-3.2 Player_LoadTitle() returns an aligned pointer in libmikmod-3.2 that cannot be freed with free(). The correct way to do this now is MikMod_free() which extracts the original pointer from the buffer and frees that. Signed-off-by: Christoph Mende <mende.christoph@gmail.com>
2013-04-20 08:29:11 +02:00
- mikmod: adapt to libmikmod 3.2
configure.ac: detect system "ar"
2013-08-01 09:13:41 +02:00
* configure.ac:
- detect system "ar"
prepare 0.17.5
2013-04-16 23:17:38 +02:00
release 0.17.4
2013-04-08 21:38:40 +02:00
ver 0.17.4 (2013/04/08)
command: allow to omit END in ranges (START:END) If END is omitted, mpd should use the highest possible value instead of raising an error. This partially reverts 52e9cab1c1743f64a7
2013-02-18 22:57:39 +01:00
* protocol:
- allow to omit END in ranges (START:END)
player_control: don't emit IDLE_PLAYER before audio format is known Eliminates one IDLE_PLAYER call in playlist_control, and add two new ones to player_thread. Fixes Mantis bug 3636.
2013-04-08 21:22:06 +02:00
- don't emit IDLE_PLAYER before audio format is known
decoder/ffmpeg: support float planar audio
2013-01-16 18:02:58 +01:00
* decoder:
- ffmpeg: support float planar audio (ffmpeg 1.1)
ffmpeg decoder plugin: do not allocate an AVFrame on stack. AVFrame must be allocated with avcodec_alloc_frame().
2013-03-22 07:05:00 +01:00
- ffmpeg: fix AVFrame allocation
player_thread: implement missing "idle" events on output errors When all outputs fail, MPD automatically pauses playback, but did not emit the IDLE_PLAYER event. Thus, clients were not woken up. (Mantis 3665)
2013-02-04 10:15:34 +01:00
* player:
- implement missing "idle" events on output errors
clock: add missing include
2013-02-27 19:42:32 +01:00
* clock: fix build failure
Modify version string to post-release version 0.17.4~git
2013-01-07 01:47:09 +01:00
mpd version 0.17.3
2013-01-07 01:47:09 +01:00
ver 0.17.3 (2013/01/06)
output/shout: fix memory leak in error handler
2012-10-01 23:18:33 +02:00
* output:
OSX: Set mDataByteSize correctly on AudioBuffers during render.
2012-10-02 17:18:41 +02:00
- osx: fix pops during playback
output/recorder: fix write() error check We can only check for negative values if the variable is signed.
2012-10-02 00:00:56 +02:00
- recorder: fix I/O error check
output/shout: fix memory leak in error handler
2012-10-01 23:18:33 +02:00
- shout: fix memory leak in error handler
output/{recorder,shout}: call encoder_read() in a loop This is necessary for Ogg packets that span more than one page.
2012-10-01 23:50:50 +02:00
- recorder, shout: support Ogg packets that span more than one page
decoder/ffmpeg: ignore negative time stamps Works around assertion failure in decoder_timestamp().
2012-10-05 16:35:45 +02:00
* decoder:
- ffmpeg: ignore negative time stamps
decoder/ffmpeg: support planar audio Implements Mantis feature request 3582.
2012-10-05 15:27:04 +02:00
- ffmpeg: support planar audio
cue_parser: fix memory leak
2013-01-03 21:02:59 +01:00
* playlist:
- cue: fix memory leak
cue_parser: fix CUE files with only one track Track whether _finish() has been called, and deliver all partial results then. Fixes Mantis ticket 0003621.
2013-01-03 21:21:01 +01:00
- cue: fix CUE files with only one track
Modify version string to post-release version 0.17.3~git
2012-09-30 12:27:38 +02:00
mpd version 0.17.2
2012-09-30 12:27:38 +02:00
ver 0.17.2 (2012/09/30)
client_file: remove pure attribute from client_allow_file(). That function is not pure, it writes to error. When marked as pure, the compiler is allowed to assume it does not do anything to error, so it can remain NULL, which would result in an invalid read in print_error().
2012-08-12 18:58:50 +02:00
* protocol:
- fix crash in local file check
decoder/fluidsynth: remove throttle (requires libfluidsynth 1.1) The libfluidsynth API is now sane, and does not require real-time decoding.
2012-08-15 00:29:38 +02:00
* decoder:
- fluidsynth: remove throttle (requires libfluidsynth 1.1)
decoder/fluidsynth: stop playback at end of file Use libfluidsynth's new function fluid_player_get_status().
2012-08-15 00:47:10 +02:00
- fluidsynth: stop playback at end of file
decoder/fluidsynth: check if file is really a MIDI Use fluid_is_midifile() to verify the file format.
2012-08-15 00:35:56 +02:00
- fluidsynth: check MIDI file format while scanning
decoder/fluidsynth: add "sample_rate" setting
2012-08-15 00:57:32 +02:00
- fluidsynth: add sample rate setting
decoder/wavpack: support all APEv2 tags WavPack tags are always APEv2, by definition. Reuse the tag_table from tag_ape.c, instead of rolling our own.
2012-09-25 09:37:16 +02:00
- wavpack: support all APEv2 tags
timer: use monotonic clock if available
2012-08-14 19:02:26 +02:00
* output:
- httpd: use monotonic clock, avoid hiccups after system clock adjustment
output/httpd: fix throttling bug after resuming playback Reset the timer when paused and no client is connected. This fixes Mantis ticket 0003527.
2012-08-14 21:39:33 +02:00
- httpd: fix throttling bug after resuming playback
playlist/cue: map "PERFORMER" to "artist" or "album artist" Implements Mantis ticket 0003549.
2012-08-25 09:56:14 +02:00
* playlist:
- cue: map "PERFORMER" to "artist" or "album artist"
mapper: fix non-UTF8 music directory name Duplicate the music_dir variable: one encoded in UTF-8, and another one using the configured filesystem character set. This fixes an ancient MPD bug.
2012-08-13 23:37:50 +02:00
* mapper: fix non-UTF8 music directory name
mapper: fix potential crash in file permission check
2012-08-15 22:19:03 +02:00
* mapper: fix potential crash in file permission check
playlist_song: fix user-after-free bug
2012-08-15 23:57:38 +02:00
* playlist: fix use-after-free bug
playlist_print: fix memory leak
2012-08-15 23:59:52 +02:00
* playlist: fix memory leak
queue_save: save song priorities
2012-08-21 19:17:14 +02:00
* state_file: save song priorities
player_thread: disable cross-fading in "single" mode This commit reimplements the core of the "single" mode. Instead of doing the detection in the playlist code from the outside, it is moved to the player thread, which gets a new option called "border_pause". It will now pause playback exactly at the beginning of the new song, making the feature more reliable. Now that the player thread knows what will happen, it can suppress cross-fading. Fixes mantis tickets 0003055 and 0003166.
2012-08-25 08:44:31 +02:00
* player: disable cross-fading in "single" mode
update_walk: fix unsafe readlink() usage
2012-09-03 22:41:04 +02:00
* update: fix unsafe readlink() usage
configure.ac: don't auto-detect the Vorbis encoder when Tremor is enabled libvorbisidec and libvorbis export the same symbols, which is a dangerous thing. Since libvorbisenc depends on libvorbis, this can get nasty, so let's disable the Vorbis encoder unless the user explicitly wants it.
2012-09-25 10:41:39 +02:00
* configure.ac:
- don't auto-detect the vorbis encoder when Tremor is enabled
Modify version string to post-release version 0.17.2~git
2012-08-01 04:05:52 +02:00
mpd version 0.17.1
2012-08-01 04:05:52 +02:00
ver 0.17.1 (2012/07/31)
command: require appropriate permissions for searchadd{,pl}
2012-06-29 00:05:33 +02:00
* protocol:
- require appropriate permissions for searchadd{,pl}
aiff: support the AIFC format
2012-07-10 01:53:46 +02:00
* tags:
- aiff: support the AIFC format
tag_ape: return false if no usable tag was found Ignore APE tags that have no usable tags, and use the ID3 tag instead. This is useful when the APE tag only contains replay gain, and the real tags are stored as ID3. This implements feature request Mantis #0003521.
2012-07-29 18:44:03 +02:00
- ape: check for ID3 if no usable APE tag was found
cue_parser: support file types "MP3", "AIFF" These two strings are common "FILE" types.
2012-07-26 23:01:40 +02:00
* playlist:
- cue: support file types "MP3", "AIFF"
output_init: put the "convert" filter at the end of the list No, really! This fixes a regression of commit 74617389, which changed the order of filter plugins.
2012-07-10 01:14:43 +02:00
* output:
- fix noisy playback with conversion and software volume
configure.ac: increment version number to 0.17.1
2012-07-09 22:50:32 +02:00
mpd version 0.17
2012-06-27 08:12:21 +02:00
ver 0.17 (2012/06/27)
protocol: support client-to-client communication
2011-01-29 09:26:22 +01:00
* protocol:
- support client-to-client communication
command: "update" and "rescan" need only "CONTROL" permission From http://bugs.debian.org/513291 "In mpd.conf, the "admin" permission covers updating the db and killing mpd. "Since there are quite some usecases in which the user can upload music to the mpd's directory by means of anonymous FTP or so, it is desirable that any user may issue a db update, while killing the mpd should not be possible. "I'd suggest to remove the db update from the admin group and either add it to "control" or an own group."
2011-02-27 23:35:00 +01:00
- "update" and "rescan" need only "CONTROL" permission
command: new command "seekcur" For simpler seeking within current song.
2011-12-24 11:20:02 +01:00
- new command "seekcur" for simpler seeking within current song
command: new command "config"
2012-02-13 19:59:41 +01:00
- new command "config" dumps location of music directory
command: add optional range parameter to "load"
2012-02-09 23:52:51 +01:00
- add range parameter to command "load"
db_print: print extra "playlist" object for embedded CUE sheets This finally enables the new embedded CUE sheet code: when a song file contains a playlist, it is printed in the "lsinfo" output, so clients get to know about this.
2012-02-12 18:41:25 +01:00
- print extra "playlist" object for embedded CUE sheets
Documentation for commands searchadd, searchaddpl, and for tcp keepalive config options
2012-06-27 09:26:05 +02:00
- new commands "searchadd", "searchaddpl"
configure.ac: hook the cdio_paranoia input plugin
2010-12-21 22:23:01 +01:00
* input:
- cdio_paranoia: new input plugin to play audio CDs
input/curl: enable CURLOPT_NETRC I wanted mpd to play a mp3 stream from a music website. The stream is only available to subscribers, which restriction is enforced through normal http authentication. However, the URL I get from the website is not the final URL of the stream, but a generic URL which points to the real one through a redirect (code 301). Thus, I cannot predict the final URL, and so I cannot use the username:password hack to force the authentication, and mpd (libcurl on mpds behalf) fails to grab the stream. libcurl allows the option CURLOPT_NETRC to be set and then the credentials can be stored in the good old .netrc file (in this case it would be ~mpd/.netrc, of course). But mpd doesn't set this option. I think it should.
2011-01-29 08:43:30 +01:00
- curl: enable CURLOPT_NETRC
input_stream: non-blocking I/O Add GMutex, GCond attributes which will be used by callers to conditionally wait on the stream. Remove the (now-useless) plugin method buffer(), wait on GCond instead. Lock the input_stream before each method call. Do the same with the playlist plugins.
2011-09-14 21:46:41 +02:00
- curl: non-blocking I/O
input/soup: new input plugin based on libsoup To demonstrate the new I/O thread. libsoup is well-integrated into the GLib main loop, which made this plugin pretty easy to write. As a side effect, we have to initialize the I/O thread in all debug programs that use the input API.
2011-08-24 03:23:12 +02:00
- soup: new input plugin based on libsoup
tag_rva2: support separate album/track replay gain
2012-04-23 22:15:23 +02:00
* tags:
- RVA2: support separate album/track replay gain
mpg123 decoder: implement seeking
2011-01-21 14:25:52 +01:00
* decoder:
- mpg123: implement seeking
decoder/ffmpeg: drop support for pre-0.5 ffmpeg All modern distros ship 0.5, so there's no need to support old and buggy ffmpeg versions.
2011-03-31 12:55:18 +02:00
- ffmpeg: drop support for pre-0.5 ffmpeg
mpd version 0.17
2012-06-27 08:12:21 +02:00
- ffmpeg: support WebM
decoder/oggflac: delete this obsolete plugin libOggFLAC has been deprecated for 5 years now, and we havn't been testing it for a long time. Let's delete it.
2011-09-01 19:29:04 +02:00
- oggflac: delete this obsolete plugin
decoder/dsdiff: new decoder plugin Doesn't seem to work yet, getting just noise from a test file. Seeking isn't implemented yet.
2011-10-03 12:52:15 +02:00
- dsdiff: new decoder plugin
osx_output: allow user to specify other audio devices. Add new config parameter 'device' to audio_output type "osx": - if not supplied or set to "default", open default device - if set to "system", open system device - otherwise 'device' should be an audio device name: mpd will find and open the specified audio device, falling back to the default device if it's not found
2010-12-21 04:21:46 +01:00
* output:
output/alsa: add option to enable DSD over USB
2012-03-21 10:36:19 +01:00
- alsa: support DSD-over-USB (dCS suggested standard)
Add support for streaming to a DLNA client The Naim Uniti does not appear to support icecast-style streaming of FLAC music but does support the codec from a DLNA server. This change looks for "transferMode.dlna.org: Streaming" in the HTTP request header and responds with something the Uniti (and hopefully other DLNA clients) accepts. The only difference in the DLNA streaming mode is the reponse header and that icecast metadata is disabled. If a client request indicates both modes are supported, the DLNA mode is preferred (as the Uniti says it supports both but then rejects a FLAC ICY stream). Note: This change may be specific to Naim equipment (the only device it was tested on). E.g. the hardcoding of Content-Length which works but is not a logically correct value. The change should be backwards-compatible, so only those clients requesting a DLNA stream will see any difference.
2011-09-29 00:18:38 +02:00
- httpd: support for streaming to a DLNA client
output/openal: use alGetSourcei(AL_BUFFER) to force-unqueue buffers The implementation of cancel() did not work well: you cannot use alSourceUnqueueBuffers() to unqueue queued buffers, and our function openal_unqueue_buffers() left the OpenAL library in a rather undefined state; nothing was supposed to be queued, but the "filled" variable was not reset.
2011-12-13 21:45:26 +01:00
- openal: improve buffer cancellation
osx_output: allow user to specify other audio devices. Add new config parameter 'device' to audio_output type "osx": - if not supplied or set to "default", open default device - if set to "system", open system device - otherwise 'device' should be an audio device name: mpd will find and open the specified audio device, falling back to the default device if it's not found
2010-12-21 04:21:46 +01:00
- osx: allow user to specify other audio devices
output/osx: implement 32 bit playback
2011-12-24 18:18:42 +01:00
- osx: implement 32 bit playback
output/shout: add possibility to set url Added a new optional parameter for the shout plugin called "url".
2011-02-04 14:14:29 +01:00
- shout: add possibility to set url
RoarAudio output plugin
2011-02-08 00:17:58 +01:00
- roar: new output plugin for RoarAudio
winmm_output_plugin: fail if wrong device specified instead of using fallback. Silently choosing default is misleading and can cause hours of investigation. It's better to fail immediately telling user what is wrong with config.
2011-10-23 08:59:31 +02:00
- winmm: fail if wrong device specified instead of using default device
mixer/alsa: listen for external volume changes Use libasound's polling functions, implement a bridge to GSource / GPollFD and send idle events to clients when an external program changes the ALSA mixer volume.
2012-01-05 00:48:54 +01:00
* mixer:
- alsa: listen for external volume changes
playlist_song: allow references to songs outside the musicdir When we have an absolute path that's not inside the music directory, allow loading it anyway, if we're in "secure" mode (i.e. the client is connected via UNIX socket).
2011-10-06 22:57:14 +02:00
* playlist:
- allow references to songs outside the music directory
cue_parser: new line based CUE sheet parser To replace libcue, the unmaintained and crashy library.
2012-02-10 00:12:29 +01:00
- new CUE parser, without libcue
A soundcloud playlist plugin. Requires YAJL to build, and this doesn't include the necessary automake changes. Can be built using ./configure CFLAGS="-I/usr/include/yajl" LIBS="-lyajl" --enable-soundcloud Add the following to your config: playlist_plugin { name "soundcloud" enabled "true" apikey "c4c979fd6f241b5b30431d722af212e8" } Then you can stream from soundcloud using calls like: mpc load soundcloud://track/<track-id> mpc load soundcloud://playlist/<playlist-id> mpc load soundcloud://url/http://soundcloud.com/some/track/or/playlist For the last case, you can leave off the http:// or http://soundcloud.com/ .
2012-02-27 13:19:45 +01:00
- soundcloud: new plugin for accessing soundcloud.com
playlist_state: add option "restore_paused" When set, MPD will not auto-start playback on startup; it will be in "paused" state.
2011-02-23 09:09:48 +01:00
* state_file: add option "restore_paused"
cue_tag: show CUE track numbers
2011-05-09 21:26:36 +02:00
* cue: show CUE track numbers
server_socket: use resolve_host_port() instead of getaddrinfo() Allow port specification in "bind_to_address" settings.
2011-09-20 20:52:49 +02:00
* allow port specification in "bind_to_address" settings
audio_format: basic support for floating point samples Support for conversion from float to 16, 24 and 32 bit integer samples.
2011-10-08 10:25:06 +02:00
* support floating point samples
listen: implement systemd socket activation
2012-02-13 20:48:51 +01:00
* systemd socket activation
cmdline: consistent plugin listings
2012-06-12 22:29:04 +02:00
* improve --version output
Work around incorrect g_file_test() behavior on Win32 g_file_test is redefined to be g_file_test_utf8 and thus can't handle non-ASCII characters. This fix adds simple wrapper (taken from glib) that fixes encoding and calls g_file_test_utf8. All required inclusions of glib_compat.h are added as well.
2012-06-03 09:00:11 +02:00
* WIN32: fix renaming of stored playlists with non-ASCII names
Modify version string to post-release version 0.16.9~git
2012-04-05 03:37:47 +02:00
mpd version 0.16.8
2012-04-05 03:37:47 +02:00
ver 0.16.8 (2012/04/04)
pcm_buffer: pcm_buffer_get() never returns NULL This fixes a bug when libsamplerate returns an empty buffer for a very small input buffer. The caller thinks this is an error, bug there is no GError object.
2012-02-13 18:05:42 +01:00
* fix for libsamplerate assertion failure
decoder_api: check state before emitting initial seek command This fixes seeking in the vorbis decoder during MPD startup.
2012-02-13 18:27:43 +01:00
* decoder:
- vorbis (and others): fix seeking at startup
decoder/ffmpeg: read the "year" tag This was disabled when compiled with a new ffmpeg version. Older ffmpeg versions used it explicitly, while newer ones may pass it through from the codec.
2012-02-13 19:05:39 +01:00
- ffmpeg: read the "year" tag
encoder/vorbis: generate end-of-stream packet before tag Don't reset the ogg_stream_state object, because this discards the end-of-stream packet that was just added.
2012-04-04 23:10:38 +02:00
* encoder:
- vorbis: generate end-of-stream packet before tag
encoder/vorbis: generate end-of-stream packet when playback ends Add the encoder_plugin method end(). This is important for the recorder plugin.
2012-04-05 00:03:38 +02:00
- vorbis: generate end-of-stream packet when playback ends
output/osx: use the fifo_buffer library instead of rolling own The existing buffer implementation has a major flaw: it is unable to re-fill the buffer until it has been consumed completely, leading to many occasions where the render callback needs to generate silence, just because the play() implementation was unable to append more data. The fifo_buffer library handles that well.
2012-03-28 21:51:17 +02:00
* output:
output/jack: check for connection failure before starting playback
2012-04-04 21:40:56 +02:00
- jack: check for connection failure before starting playback
output/jack: workaround for libjack1 crash bug
2012-04-04 21:38:29 +02:00
- jack: workaround for libjack1 crash bug
output/osx: use the fifo_buffer library instead of rolling own The existing buffer implementation has a major flaw: it is unable to re-fill the buffer until it has been consumed completely, leading to many occasions where the render callback needs to generate silence, just because the play() implementation was unable to append more data. The fifo_buffer library handles that well.
2012-03-28 21:51:17 +02:00
- osx: fix stuttering due to buffering bug
text_input_stream: detect end-of-file Fixes endless loop when the last line of a text file was not terminated (bug 3470).
2012-03-19 19:51:19 +01:00
* fix endless loop in text file reader
update: properly skip symlinks in path that is to be updated.
2012-04-04 08:56:45 +02:00
* update: skip symlinks in path that is to be updated
Modify version string to post-release version 0.16.8~git
2012-02-04 23:41:59 +01:00
mpd version 0.16.7
2012-02-04 23:41:59 +01:00
ver 0.16.7 (2012/02/04)
input/ffmpeg: use the new AVIOContext API URLContext is deprecated.
2011-05-09 18:22:59 +02:00
* input:
- ffmpeg: support libavformat 0.7
decoder/ffmpeg: use AVIOContext instead of ByteIOContext
2011-05-09 21:16:43 +02:00
* decoder:
decoder/ffmpeg: use AV_SAMPLE_FMT_* if available Implements support for libavcodec 0.9, which removes the compatibility macros SAMPLE_FMT_*
2012-02-03 09:55:25 +01:00
- ffmpeg: support libavformat 0.8, libavcodec 0.9
decoder/ffmpeg: support all MPD tags Use the tag_item_names table to look up the names of all MPD tags, and remove the duplicate entries from ffmpeg_tag_maps.
2012-02-03 09:07:31 +01:00
- ffmpeg: support all MPD tags
timer: fix time unit mixup in timer_delay() The local variable was already divided by 1000, and the return value was being divided by 1000 again - doh! This caused delays in the httpd output plugin that were too small by three orders of magnitude, and the buffer was filled too quickly.
2011-12-13 21:02:33 +01:00
* output:
- httpd: fix excessive buffering
output/openal: force 16 bit playback, as 8 bit doesn't work The OpenAL specification says that AL_FORMAT_MONO8 and AL_FORMAT_STEREO8 expect unsigned 8 bit samples, but MPD uses unsigned samples.
2011-12-13 21:32:19 +01:00
- openal: force 16 bit playback, as 8 bit doesn't work
output/osx: remove sleep call from render callback Blocking inside the render callback is forbidden, and this sleep call didn't make any sense.
2011-12-24 17:56:10 +01:00
- osx: remove sleep call from render callback
output/osx: clear render buffer when there's not enough data When we don't have enough data, generate some silence, hoping the input buffer will fill soon. Reducing the render buffer size is not legal.
2011-12-24 17:54:41 +01:00
- osx: clear render buffer when there's not enough data
Playlist: fix bug in moving after current song Moving songs using either 'move' or 'moveid' to position -1 (after the current song) would fail for a song which is just before the current song. This patch corrects the check to see if the current song is in the range to be moved. Since the range is from `start` up to `end` (exclusive) the check was incorrect, but is now fixed.
2011-12-19 21:15:00 +01:00
* fix moving after current song
Modify version string to post-release version 0.16.7~git
2011-12-01 14:44:53 +01:00
mpd version 0.16.6
2011-12-01 14:44:53 +01:00
ver 0.16.6 (2011/12/01)
decoder_api: cancel initial seek when song is not seekable Fixes assertion failure.
2011-11-27 19:19:43 +01:00
* decoder:
- fix assertion failure when resuming streams
decoder/ffmpeg: work around bogus channel count Initialize the audio_format before calling avcodec_open(), because avcodec_open() will fill bogus values.
2011-11-27 23:39:21 +01:00
- ffmpeg: work around bogus channel count
encoder/flac: use fifo_buffer instead of pcm_buffer This fixes a buffer corruption bug; pcm_buffer is not designed to be a persistent buffers, and will discard anything between two consecutive calls.
2011-11-28 08:00:10 +01:00
* encoder:
encoder/null: use fifo_buffer instead of pcm_buffer This fixes a buffer corruption bug; pcm_buffer is not designed to be a persistent buffers, and will discard anything between two consecutive calls.
2011-11-28 08:27:59 +01:00
- flac, null, wave: fix buffer corruption bug
encoder/wave: support packed 24 bit samples Convert to padded 24 bit samples, instead of falling back to 16 bit.
2011-11-28 09:25:42 +01:00
- wave: support packed 24 bit samples
mapper: fix the bogus "not a directory" error message Use stat() instead of g_file_test() to detect other types of errors, such as "permission denied".
2011-11-28 09:37:05 +01:00
* mapper: fix the bogus "not a directory" error message
mapper: check "r" permission on music directory Yet another common support case.
2011-11-28 09:56:03 +01:00
* mapper: check "x" and "r" permissions on music directory
log: print reason for failure
2011-11-28 09:31:43 +01:00
* log: print reason for failure
event_pipe: fix WIN32 regression The event pipe is not a socket, and the patch that introduced g_io_channel_new_socket() to the event pipe library was wrong.
2011-10-13 09:08:37 +02:00
* event_pipe: fix WIN32 regression
configure.ac: define WINVER Ensure that WINVER is defined early enough, so other system headers won't fall back to their default value. Specifically, this solves a build failure (-Werror) with mingw-w64 ("WINVER redefined").
2011-10-13 09:23:32 +02:00
* define WINVER in ./configure
path: autodetect filesystem encoding on Win32 WinAPI explicitly declares filesystem encoding. It can be determined by GetACP(). Use that instead of Glib routine that always "detects" UTF-8 on Win32, which is incorrect for MPD case.
2011-10-21 20:57:30 +02:00
* WIN32: autodetect filesystem encoding
Modify version string to post-release version 0.16.6~git
2011-10-09 13:44:51 +02:00
mpd version 0.16.6
2011-12-01 14:44:53 +01:00
ver 0.16.5 (2011/10/09)
configure.ac: fix solaris result display Wrong variable name.
2011-09-19 07:42:11 +02:00
* configure.ac
- disable assertions in the non-debugging build
- show solaris plugin result correctly
configure.ac: add option --enable-solaris-output Allow enabling the plugin explicitly without running Solaris, to test the build.
2011-09-19 07:39:27 +02:00
- add option --enable-solaris-output
pcm_format: fix 32-to-24 bit conversion (the "silence" bug) D'oh, we were reading 16 bit integers instead of 32 bit integers! That caused silence when trying to play a 32 bit input file on a 24 bit sound card (e.g. USB sound chips with 24 bit packed samples).
2011-09-08 23:47:32 +02:00
* pcm_format: fix 32-to-24 bit conversion (the "silence" bug)
input/rewind: copy the MIME type only once Reduce heap usage by reducing the number of malloc() / free() calls.
2011-09-15 20:24:15 +02:00
* input:
- rewind: reduce heap usage
decoder/ffmpeg: higher precision timestamps
2011-09-15 21:07:32 +02:00
* decoder:
- ffmpeg: higher precision timestamps
decoder/ffmpeg: don't require key frame for seeking Use flag AV_TIME_BASE.
2011-09-15 21:32:29 +02:00
- ffmpeg: don't require key frame for seeking
decoder_api: emulate SEEK command for initial seek to CUE track When playing a CUE track, the player thread waited for the decoder to become ready, and then sent a SEEK command to the beginning of the CUE track. If that is near the start of the song file, and the track is short enough, the decoder could have finished decoding already at that point, and seeking fails. This commit makes this initial seek more robust: instead of letting the player thread deal with the difficult timings, let the decoder API emulate a SEEK command, and return it to the decoder plugin, as soon as the plugin finishes its initialization.
2011-09-21 23:17:34 +02:00
- fix CUE track seeking
output/openal: auto-fallback to mono if channel count is unsupported .. instead of failing playback completely.
2011-10-08 14:39:40 +02:00
* output:
- openal: auto-fallback to mono if channel count is unsupported
player_thread: handle SEEK while not playing
2011-10-05 22:13:13 +02:00
* player:
- make seeking to CUE track more reliable
- the "seek" command works when MPD is stopped
player_control: auto-start playback when seeking is requested Now that the player thread can handle SEEK commands while not (yet) playing, we can remove the "pc.state" check from pc_seek().
2011-10-05 22:24:53 +02:00
- restore song position from state file (bug fix)
player_thread: add flag "output_open", fixes assertion failure Previously, the condition "defined(play_audio_format)" was used to see if an output device has been opened, but if the device had failed on startup, an assertion failure could occur. This patch adds a separate flag.
2011-10-06 20:30:46 +02:00
- fix crash that sometimes occurred when audio device fails on startup
playlist_song: fix absolute path support in playlists Right now, a playlist with absolute pathnames can only add songs that are in the same the directory of the playlist or under it. If uri is an absolute pathname and base_uri is set, playlist_check_translate_song() will check that base_uri is a prefix of uri, excluding every other song in the music directory outside base_uri. I think in this case base_uri should be completely ignored (and made NULL) and uri should just be checked against music root directory.
2011-10-06 22:21:24 +02:00
- fix absolute path support in playlists
fd_util: add function close_socket() Wrap close(), use closesocket() on WIN32/WinSock.
2011-09-19 21:04:19 +02:00
* WIN32: close sockets properly
Install systemd service file if systemd is available
2011-09-30 08:37:36 +02:00
* install systemd service file if systemd is available
Modify version string to post-release version 0.16.5~git
2011-09-02 02:58:29 +02:00
mpd version 0.16.4
2011-09-02 02:58:28 +02:00
ver 0.16.4 (2011/09/01)
configure.ac: use MPD_AUTO_PKG to detect avahi Don't abort the configure script when avahi could not be auto-detected. It previously did, because there was no custom "fail" action for PKG_CHECK_MODULES.
2011-09-01 10:07:58 +02:00
* don't abort configure when avahi is not found
configure.ac: auto-detect libmad without pkg-config The pkg-config file was added by the Debian package maintainers, and unfortunately, the rest of the world doesn't have it.
2011-09-01 09:22:18 +02:00
* auto-detect libmad without pkg-config
NEWS: fix memory leaks
2011-07-18 22:47:51 +02:00
* fix memory leaks
playlist_control: don't resume playback when seeking to another song while paused Use a shortcut in playlist_seek_song(), don't call playlist_play_order() because that would reset the "paused" state.
2011-07-20 11:33:51 +02:00
* don't resume playback when seeking to another song while paused
update_walk: apply follow_inside_symlinks to absolute symlinks
2011-07-20 07:05:10 +02:00
* apply follow_inside_symlinks to absolute symlinks
output_thread: reimplement CANCEL synchronization The output thread could hang indefinitely after finishing CANCEL, because it could have missed the signal while the output was not unlocked in ao_command_finished(). This patch removes the wait() call after CANCEL, and adds the flag "allow_play" instead. While this flag is set, playback is skipped. With this flag, there will not be any excess wait() call after the pipe has been cleared. This patch fixes a bug that causes mpd to discontinue playback after seeking, due to the race condition described above.
2011-09-01 07:13:21 +02:00
* fix playback discontinuation after seeking
input/curl: limit the receive buffer size
2011-08-23 20:46:51 +02:00
* input:
- curl: limit the receive buffer size
input/curl: implement a hard-coded timeout of 10 seconds Be sure to stop the operation at some point when the server isn't responding.
2011-08-26 19:28:09 +02:00
- curl: implement a hard-coded timeout of 10 seconds
ffmpeg: workaround for semantic API change in recent ffmpeg versions
2011-07-03 14:54:56 +02:00
* decoder:
- ffmpeg: workaround for semantic API change in recent ffmpeg versions
decoder/flac: validate the sample rate when scanning the tag Don't calculate the song duration when the sample rate is 0 (division by zero crash).
2011-07-03 14:57:56 +02:00
- flac: validate the sample rate when scanning the tag
wavpack: obey all decoder commands, stop at CUE track border It used to ignore the decoder_data() return value.
2011-07-20 12:30:56 +02:00
- wavpack: obey all decoder commands, stop at CUE track border
encoder_plugin: add method pre_tag() In the "vorbis" plugin, this is a copy of the old flush() method, while flush() gets a lot of code remove, it just sets the "flush" flag and nothing else. It doesn't start a new stream now, which should fix a few problems in some players.
2011-07-20 20:54:34 +02:00
* encoder:
- vorbis: don't send end-of-stream on flush
output/alsa: fix SIGFPE when alsa announces a period size of 0
2011-07-20 06:54:51 +02:00
* output:
- alsa: fix SIGFPE when alsa announces a period size of 0
output/httpd: don't warn on client disconnect This warning should only be logged when we really received something. When the client disconnects, G_IO_IN is triggered, and the read returns G_IO_STATUS_EOF.
2011-08-23 17:51:28 +02:00
- httpd: don't warn on client disconnect
output/osx: don't drain the buffer when closing Eliminate an unnecessary source of deadlocks.
2011-09-01 18:13:05 +02:00
- osx: don't drain the buffer when closing
pulse/output: fix deadlock when resuming the stream Unlock the mainloop in all code paths.
2011-08-23 22:44:43 +02:00
- pulse: fix deadlock when resuming the stream
output/pulse: fix deadlock when the stream was suspended Check if the stream is suspended; wake up the main loop when it becomes suspended.
2011-08-23 22:43:08 +02:00
- pulse: fix deadlock when the stream was suspended
Modify version string to post-release version 0.16.4~git
2011-06-04 16:37:34 +02:00
mpd version 0.16.3
2011-06-04 16:37:33 +02:00
ver 0.16.3 (2011/06/04)
audio_parser: fix assertion failure in audio format mask parser Use audio_format_mask_valid() to verify a mask.
2011-03-23 22:22:51 +01:00
* fix assertion failure in audio format mask parser
playlist_song: fix NULL pointer dereference
2011-05-09 18:03:45 +02:00
* fix NULL pointer dereference in playlist parser
playlist_song: fix playlist files in base music directory g_path_get_dirname() returns "." when there is no directory name in the given path. This patch adds a workaround for that.
2011-05-09 18:00:41 +02:00
* fix playlist files in base music directory
directory: allow directories with just playlists Keep those when scanning for empty directories. The check in playlist_vector_is_empty() was missing.
2011-05-09 21:37:43 +02:00
* database: allow directories with just playlists
decoder/ffmpeg: don't use deprecated CODEC_TYPE_AUDIO with new lavc fixes build with lavc 53.
2011-05-02 20:53:15 +02:00
* decoder:
decoder/ffmpeg: use avcodec_decode_audio3() if available avcodec_decode_audio3() has been added in libavformat 52.25.0, and the predecessor avcodec_decode_audio2() has been deprecated.
2011-04-12 08:16:57 +02:00
- ffmpeg: support libavcodec 0.7
Modify version string to post-release version 0.16.3~git
2011-03-19 01:43:11 +01:00
mpd version 0.16.2
2011-03-19 01:43:11 +01:00
ver 0.16.2 (2011/03/18)
configure.ac: fix bashism in tremor test This makes FreeBSD detect libogg correctly. The '==' operator is an undocumented GNU extension to test(1) and cannot be relied upon to exist and do the right thing. POSIX mandates string comparisons to be done using "test foo = bar".
2011-02-08 05:44:00 +01:00
* configure.ac:
- fix bashism in tremor test
configure.ac: fix tremor configure test When the configure options were moved around for 0.16, the order was changed, and the Tremor check broke.
2011-01-28 12:21:03 +01:00
* decoder:
- tremor: fix configure test
Set fadeout in gme_decoder_plugin. Due to the nature of the gme library, this needs to be done for the end of songs to be detected.
2011-02-03 00:25:35 +01:00
- gme: detect end of song
encoder/vorbis: reset the Ogg stream after flush Without the ogg_stream_reset() call, the "e_o_s" flag never gets reset, and libogg writes EOS packets over and over.
2011-03-16 19:13:46 +01:00
* encoder:
- vorbis: reset the Ogg stream after flush
output/httpd: initialize unflushed_input This fixes the following valgrind warning occuring on the first call of httpd_output_read_page: ==20124== Conditional jump or move depends on uninitialised value(s) ==20124== at 0x425E65: httpd_output_read_page (httpd_output_plugin.c:240) ==20124== by 0x426087: httpd_output_open (httpd_output_plugin.c:279) ==20124== by 0x41D862: ao_open (output_plugin.h:206) ==20124== by 0x41E133: audio_output_task (output_thread.c:590)
2011-02-04 10:22:00 +01:00
* output:
- httpd: fix uninitialized variable
output/httpd: include sys/socket.h for AF_UNIX
2011-02-08 05:44:00 +01:00
- httpd: include sys/socket.h
output/oss: AFMT_S24_PACKED is little-endian According to the Solaris dsp manpage, AFMT_S24_PACKED is little-endian: http://download.oracle.com/docs/cd/E19963-01/821-1475/6nmf5baot/index.html The Minix soundcard.h header says the same.
2011-02-28 00:00:41 +01:00
- oss: AFMT_S24_PACKED is little-endian
output/oss: disable 24 bit playback on FreeBSD See code comment.
2011-02-28 00:09:45 +01:00
- oss: disable 24 bit playback on FreeBSD
Modify version string to post-release version 0.16.2~git
2011-01-10 03:00:12 +01:00
copyright year 2011
2011-01-29 10:13:54 +01:00
ver 0.16.1 (2011/01/09)
audio_check: fix parameter in prototype
2010-12-21 07:24:43 +01:00
* audio_check: fix parameter in prototype
add void casts to suppress "result unused" warnings (clang)
2010-12-21 08:05:36 +01:00
* add void casts to suppress "result unused" warnings (clang)
configure.ac: disable the FFADO plugin by default It is known to crash instantly.
2010-12-22 07:25:39 +01:00
* input:
- ffado: disable by default
decoder/mad: work around build failure on Solaris Rename the "version" struct, because it seems to be a reserved name on Solaris: "src/decoder/mad_decoder_plugin.c", line 550: (enum) tag redeclared: version cc: acomp failed for src/decoder/mad_decoder_plugin.c
2010-12-21 04:21:53 +01:00
* decoder:
- mad: work around build failure on Solaris
Makefile.am: resolve modplug vs. libsndfile cflags/headers conflict A bit of automake magic (see info automake "Per-Object Flags"). Compile-tested.
2011-01-09 18:21:27 +01:00
- resolve modplug vs. libsndfile cflags/headers conflict
output/solaris: add missing parameter to open_cloexec() call
2010-12-21 07:27:35 +01:00
* output:
- solaris: add missing parameter to open_cloexec() cal
output/osx: fix up audio format first, then apply it to device This is a MPD 0.16 regression: when playing a 24 bit file, the switch to 16 bit was made only partially, after mBytesPerPacket and mBytesPerFrame had already been applied. That means mBytesPerFrame referred to 24 bit, and mBitsPerChannel referred to 16 bits. Of course, that cannot work.
2011-01-07 17:15:37 +01:00
- osx: fix up audio format first, then apply it to device
player_thread: discard empty chunks while cross-fading When a music_chunk to be crossfaded consists only of a tag, cross-fading is not possible, and led to an assertion failure. This patch just discards those, as if cross-fading was not enabled.
2011-01-07 23:45:51 +01:00
* player_thread: discard empty chunks while cross-fading
player_thread: fix assertion failure due to early seek Until the decoder plugin has called decoder_initialized(), the player may not submit seek commands. This however could occur with a slow decoder and a CUE file with a virtual song offset. This patch adds another check.
2011-01-07 22:51:50 +01:00
* player_thread: fix assertion failure due to early seek
output_thread: fix double lock During the whole output thread, the audio_output object is locked, and it is only unlocked while waiting for the GCond and while running a plugin method. The error handler in ao_play_chunk() attempted to lock the object again, which was code from MPD 0.15.x which should have been removed a long time ago.
2011-01-07 23:08:18 +01:00
* output_thread: fix double lock
Modify version string to post-release version 0.16.1~git
2010-12-21 07:29:31 +01:00
mpd version 0.16
2010-12-11 13:19:49 +01:00
ver 0.16 (2010/12/11)
song_print: send song modification time to client Added the response line "Last-Modified", which sends the modification time in ISO 8601. The same was already implemented for playlists.
2009-07-05 08:40:29 +02:00
* protocol:
- send song modification time to client
idle: added "update" event Some clients have visual feedback for "database update is running". Using the "database" idle event is unreliable, because it is only emitted when the database was actually modified. This patch adds the "update" event, which is emitted when the update is started, and again when the update is finished, disregarding whether it has been modified.
2009-07-05 08:46:53 +02:00
- added "update" idle event
command: removed the "volume" command This command has been deprecated more than 5 years ago (0.10.0). Its implementation is a kludge, let's remove it now.
2009-07-06 21:51:00 +02:00
- removed the deprecated "volume" command
command: add "findadd" command.
2009-08-25 13:43:22 +02:00
- added the "findadd" command
command: range support for "delete"
2009-09-30 23:13:13 +02:00
- range support for "delete"
playlist_control: "previous" really plays the previous song No more CD player emulation. The current behaviour of "previous" is difficult for a client to predict, because it does not definitely know the current position within the song. If a client wants to restart the current song, it can always send "playid".
2009-10-08 20:33:50 +02:00
- "previous" really plays the previous song
doc/protocol.xml: "addid" with negative position is deprecated
2009-10-09 10:18:02 +02:00
- "addid" with negative position is deprecated
playlist: added extm3u plugin This new plugin parses extm3u files. Files without the "#EXTM3U" header are still parsed by the plain old "m3u" plugin.
2009-11-06 00:41:42 +01:00
- "load" supports remote playlists (extm3u, pls, asx, xspf, lastfm://)
command: allow changing replay gain mode on-the-fly The new command "replay_gain_mode" allows the user to switch the replay gain mode on-the-fly. No more mpd.conf editing.
2009-10-17 22:58:19 +02:00
- allow changing replay gain mode on-the-fly
command: omitting the range end is possible When the range end is missing, then the maximum possible value is assumed.
2009-10-18 01:24:49 +02:00
- omitting the range end is possible
command: "update" checks if the path is malformed This is a very basic check, which only ensures that the path does not begin with a slash, doesn't have double slashes and the special names "." and ".." are forbidden.
2010-02-27 19:01:17 +01:00
- "update" checks if the path is malformed
archive/zip: renamed plugin to "zzip" This plugin is based on libzzip.
2009-12-15 20:29:44 +01:00
* archive:
archive/iso: renamed plugin to "iso9660" Based on libiso9660.
2009-12-16 16:06:16 +01:00
- iso: renamed plugin to "iso9660"
archive/zip: renamed plugin to "zzip" This plugin is based on libzzip.
2009-12-15 20:29:44 +01:00
- zip: renamed plugin to "zzip"
input/lastfm: use metadata Added a patch to flush out the last.fm input plugin slightly. It basically turns it into a wrapper for the appropriate plugin. Most notably metadata is now extracted.
2009-07-28 16:41:50 +02:00
* input:
input/lastfm: removed obsolete last.fm input plugin This has been replaced by the last.fm playlist plugin. The input plugin has never worked well, and was just a playground to experiment with the last.fm radio protocol.
2009-10-13 19:43:56 +02:00
- lastfm: obsolete plugin removed
input/ffmpeg: new input plugin using libavformat's "avio" library
2010-05-18 20:48:52 +02:00
- ffmpeg: new input plugin using libavformat's "avio" library
tag: added tag "AlbumArtistSort" The tag_id3.c library supports both the documented "TSO2" tag, and the inofficial TXXX/ALBUMARTISTSORT. The Vorbis/FLAC decoder automatically supports the new tag, without further change.
2009-07-07 07:36:25 +02:00
* tags:
Implement ArtistSort tag
2009-07-09 14:28:08 +02:00
- added tags "ArtistSort", "AlbumArtistSort"
tag_id3: revised "performer" tag support According to the ID3 2.4 documentation, "TOPE" is "Original artist/performer", not "performer". Removed "TOPE" support. Instead, map TPE3 ("Conductor/performer refinement") and TPE4 ("Interpreted, remixed, or otherwise modified by") to "performer".
2009-07-07 08:00:21 +02:00
- id3: revised "performer" tag support
tag_id3: support multiple values Loop over all frames with a specific id, and import all of them - not just the first one (index 0).
2010-12-07 18:05:44 +01:00
- id3: support multiple values
Merge release 0.15.11 from branch 'v0.15.x Conflicts: NEWS configure.ac src/decoder/ffmpeg_decoder_plugin.c src/decoder/mikmod_decoder_plugin.c src/decoder/mp4ff_decoder_plugin.c src/decoder/vorbis_decoder_plugin.c src/directory_print.c src/playlist_control.c src/tag_ape.c
2010-07-19 23:49:15 +02:00
- ape: MusicBrainz tags
tag_ape: support multiple values One APE tag may contain more than one value, separated by null bytes.
2010-11-24 08:59:04 +01:00
- ape: support multiple values
ffmpeg: support multiple tags Call av_metadata_get() in a loop.
2009-06-25 08:43:59 +02:00
* decoders:
decoder_thread: don't try a plugin twice (MIME type & suffix) Manage a linked list of plugins which were already tried.
2010-01-17 16:47:04 +01:00
- don't try a plugin twice (MIME type & suffix)
decoder_thread: don't fall back to "mad" unless no plugin matches When all plugins have failed, MPD used to fall back to the "mad" decoder plugin, to handle those radio streams without a Content-Type response header. This however leads to unexpected results (garbage being played) when the stream isn't really mp3. Since we care little about "bad" streams, we shouldn't have hacks which have bad side effects. Let's get rid of this hack now! Only try to "mad" plugin if there was no match at all (Content-Type, path suffix) and no other plugin has been tried.
2010-01-17 17:08:56 +01:00
- don't fall back to "mad" unless no plugin matches
ffmpeg: support multiple tags Call av_metadata_get() in a loop.
2009-06-25 08:43:59 +02:00
- ffmpeg: support multiple tags
ffmpeg_plugin: convert metadata to generic format
2009-08-03 17:40:22 +02:00
- ffmpeg: convert metadata to generic format
decoder/ffmpeg: implement the libavutil log callback Pass everything to the GLib logging library. No direct stderr access.
2010-02-27 19:32:59 +01:00
- ffmpeg: implement the libavutil log callback
decoder/sndfile: new decoder plugin based on libsndfile
2009-07-07 08:58:51 +02:00
- sndfile: new decoder plugin based on libsndfile
decoder/flac: removed CUE sheet support This feature has been moved to the "flac" playlist plugin.
2010-01-06 08:04:57 +01:00
- flac: moved CUE sheet support to a playlist plugin
decoder/flac: support streams without STREAMINFO block
2010-01-06 09:54:09 +01:00
- flac: support streams without STREAMINFO block
decoder/mikmod: sample rate is configurable The new option "sample_rate" sets the sample rate for libmikmod.
2009-11-14 02:03:03 +01:00
- mikmod: sample rate is configurable
decoder/mpg123: new decoder plugin based on libmpg123 Still missing: - seeking - tags - streaming - encodings other than MPG123_ENC_SIGNED_16
2009-08-26 20:08:13 +02:00
- mpg123: new decoder plugin based on libmpg123
decoder/sidplay: subtunes
2009-08-30 19:48:56 +02:00
- sidplay: support sub-tunes
decoder/sidplay: implemented songlength database [mk: added autoconf test; fixed songlen_data_size type]
2009-08-30 19:49:04 +02:00
- sidplay: implemented songlength database
decoder/sidplay: support seeking
2009-08-30 19:49:16 +02:00
- sidplay: support seeking
decoder/sidplay: play monaural SID tunes in mono
2010-10-27 21:16:24 +02:00
- sidplay: play monaural SID tunes in mono
decoder/sidplay: play mus, str, prg, x00 files
2010-10-27 21:18:43 +02:00
- sidplay: play mus, str, prg, x00 files
decoder/wavpack: activate 32 bit support 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).
2009-11-11 21:49:00 +01:00
- wavpack: activate 32 bit support
decoder/wavpack: allow more than 2 channels Remove the OPEN_2CH_MAX option. MPD's support for surround sound is still clunky, but we're working on it.
2009-11-11 23:03:20 +01:00
- wavpack: allow more than 2 channels
decoder/mp4ff: rename plugin "mp4" to "mp4ff" The underlying library is named "libmp4ff". To reduce confusion, rename the plugin to a more specific name.
2010-05-30 22:33:15 +02:00
- mp4ff: rename plugin "mp4" to "mp4ff"
encoder/twolame: new encoder plugin based on libtwolame This encoder plugin is a replacement for the LAME encoder plugin for those who prefer a "free" (non-patent encumbered) encoder library. Most of the plugin source code is copied from the LAME encoder plugin, since the LAME and TwoLAME APIs are nearly the same.
2009-07-14 23:07:41 +02:00
* encoders:
- twolame: new encoder plugin based on libtwolame
encoder: introducing flac encoder plugin
2009-11-17 19:41:35 +01:00
- flac: new encoder plugin based on libFLAC
- wave: new encoder plugin for PCM WAV format
output/recorder: new output plugin for recording radio streams The recorder plugin writes audio played by MPD to a file. This may be useful for recording radio streams. This implementation is incomplete, because support for tags is missing, and MPD should be able to record each track to a different file.
2009-08-24 18:57:06 +02:00
* output:
- recorder: new output plugin for recording radio streams
output/httpd: bind port when output is enabled Implement the methods enable() and disable(). Bind the HTTP port in the enable() method, but reject all incoming connections until the output is opened.
2009-11-05 23:47:29 +01:00
- alsa: don't recover on CANCEL
output/alsa: fill period buffer with silence before draining 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().
2009-11-09 22:22:31 +01:00
- alsa: fill period buffer with silence before draining
doc: documented the "openal" output plugin settings
2009-09-06 21:34:57 +02:00
- openal: new output plugin
pulse: announce "media.role=music" This allows PulseAudio to do some advanced tweaks.
2009-10-20 21:02:10 +02:00
- pulse: announce "media.role=music"
output/pulse: renamed context to "Music Player Daemon" This looks nicer in the PulseAudio manager than just "mpd".
2009-10-20 21:07:38 +02:00
- pulse: renamed context to "Music Player Daemon"
pulse: code rewrite using the asynchronous libpulse API 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".
2009-10-21 10:30:42 +02:00
- pulse: connect to server on MPD startup, implement pause
output/jack: use jack_client_open() instead of jack_client_new() 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".
2009-11-05 20:01:18 +01:00
- jack: require libjack 0.100
output/jack: implement the "pause" method Don't disconnect from JACK during pause.
2009-10-21 18:33:05 +02:00
- jack: don't disconnect during pause
output/jack: connect to server on MPD startup .. and keep up the JACK connection while MPD runs. Allocate the ring buffers on the first open, and free them at MPD exit.
2009-10-21 21:37:11 +02:00
- jack: connect to server on MPD startup
output/jack: added option "server_name"
2009-11-07 17:26:21 +01:00
- jack: added options "client_name", "server_name"
output/jack: clear ring buffers before activating 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.
2009-11-05 20:01:50 +01:00
- jack: clear ring buffers before activating
output/jack: renamed option "ports" to "destination_ports" Be more clear which kind of port should be configured here.
2009-11-06 01:54:58 +01:00
- jack: renamed option "ports" to "destination_ports"
output/jack: dynamic source port list 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.
2009-11-06 18:58:35 +01:00
- jack: support more than two audio channels
output/httpd: bind port when output is enabled Implement the methods enable() and disable(). Bind the HTTP port in the enable() method, but reject all incoming connections until the output is opened.
2009-11-05 23:47:29 +01:00
- httpd: bind port when output is enabled
output/httpd: added name/genre/website configuration
2010-04-05 13:11:10 +02:00
- httpd: added name/genre/website configuration
output/httpd: implement "pause" Send silence to all connected clients while paused, to avoid connection interruption.
2010-08-06 07:23:17 +02:00
- httpd: implement "pause"
output/httpd: bind_to_address support (including IPv6) Added support for a new optional configuration setting for the httpd output named "bind_to_address". Setting it to a specific IP address (v4 or v6) will cause the httpd output to bind to that address exclusively. Supporting multiple addresses in parallel is future work. This implements the feature requests #2998 and #2646.
2010-09-25 15:00:43 +02:00
- httpd: bind_to_address support (including IPv6)
output/oss: 24 bit support via OSS4 Added #ifdefs on the macros AFMT_S24_NE and other OSS4 extensions.
2010-05-13 15:42:20 +02:00
- oss: 24 bit support via OSS4
output/win32: new output plugin for Windows Wave
2010-05-18 22:56:42 +02:00
- win32: new output plugin for Windows Wave
output/httpd: implement delay()
2010-11-05 09:42:14 +01:00
- shout, httpd: more responsive to control commands
audio_format: wildcards allowed in audio_format configuration 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.
2009-10-21 23:01:04 +02:00
- wildcards allowed in audio_format configuration
output: consistently lock audio output objects Always keep the audio_output object locked within the output thread, unless a plugin method is called. This fixes several race conditions.
2009-10-29 17:06:40 +01:00
- consistently lock audio output objects
player_thread: drain audio outputs at the end of the playlist 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.
2009-11-09 22:22:27 +01:00
* player:
- drain audio outputs at the end of the playlist
volume: removed support for legacy mixer configuration The top-level "mixer_device" and "mixer_control" options have been deprecated by MPD 0.15, and it's safe to remove them in MPD 0.16.
2009-06-25 08:37:51 +02:00
* mixers:
- removed support for legacy mixer configuration
output: use the software mixer plugin Do all the software volume stuff inside each output thread, not in the player thread. This allows one software mixer per output device, and also allows the user to configure the mixer type (hardware or software) for each audio output. This moves the global "mixer_type" setting into the "audio_output" section, deprecating the "mixer_enabled" flag.
2009-07-06 22:00:50 +02:00
- reimplemented software volume as mixer+filter plugin
- per-device software/hardware mixer setting
command: added new "status" line with more precise "elapsed time"
2009-06-25 08:37:34 +02:00
* commands:
- added new "status" line with more precise "elapsed time"
update: obey .mpdignore files
2009-10-16 18:11:43 +02:00
* update:
- automatically update the database with Linux inotify
- support .mpdignore files in the music directory
songvec: sort songs by album name first, then disc/track number When the songs of two albums are in the same directory, all songs of an album should be right next to each others.
2009-10-31 17:36:56 +01:00
- sort songs by album name first, then disc/track number
database: rescan after metadata_to_use change 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.
2009-11-04 18:48:22 +01:00
- rescan after metadata_to_use change
compress: upgraded to AudioCompress 2.0 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]
2009-12-02 18:11:53 +01:00
* normalize: upgraded to AudioCompress 2.0
output_init: use the normalize filter plugin 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.
2009-12-14 21:36:25 +01:00
- automatically convert to 16 bit samples
replay_gain: reimplement as a filter plugin Apply the replay gain in the output thread. This means a new setting will be active instantly, without going through the whole music pipe. And we might have different replay gain settings for each audio output device.
2010-02-14 17:04:39 +01:00
* replay gain:
- reimplemented as a filter plugin
replay_gain: fall back to track gain if album gain is unavailable
2010-02-17 08:13:34 +01:00
- fall back to track gain if album gain is unavailable
replay_gain: optionally use hardware mixer to apply replay gain Add an option for each audio output which enables the use of the hardware mixer, instead of the software volume code. This is hardware specific, and assumes linear volume control. This is not the case for hardware mixers which were tested, making this patch somewhat useless, but we will use it to experiment with the settings, to find a good solution.
2010-02-16 08:55:37 +01:00
- optionally use hardware mixer to apply replay gain
replay_gain: added mode "auto"
2010-04-25 13:42:23 +02:00
- added mode "auto"
replay_gain_ape: parse replay gain from APE tags Based on the APE reader.
2010-11-18 22:25:11 +01:00
- parse replay gain from APE tags
conf: log unused/unknown block parameters
2009-06-19 09:02:14 +02:00
* log unused/unknown block parameters
conf: removed the deprecated "error_file" option This option was deprecated by the 0.15 release. This patch makes this option invalid.
2009-08-24 17:12:15 +02:00
* removed the deprecated "error_file" option
playlist_state: save state when stopped At the moment mpd doesn't store or restore the current track to/from its state file when the daemon is stopped/started while in 'stopped' state. I believe the preferred behaviour would be to store and restore the current track even when the daemon is in stopped state when shutting down. I made a small patch to adapt this behaviour. If you believe this is not the preferred behaviour, maybe this should be realized as a configuration option. I'm not sure how to do this, but made a small comment, where one would have to put the option.
2009-06-26 09:26:12 +02:00
* save state when stopped
cmdline: renamed option "--stdout" to "--stderr" Since version 0.14, MPD has been logging to standard error instead of standard output. The option name should reflect that. The old option continues to work, we will remove it in a future MPD release.
2009-07-15 18:27:32 +02:00
* renamed option "--stdout" to "--stderr"
cmdline: removed options --create-db and --no-create-db Both options are deprecated, and should not be used anymore. Many users get confused by their presence.
2009-09-20 23:31:35 +02:00
* removed options --create-db and --no-create-db
state_file: save only if something has changed If nothing has changed since the last save, don't save the state file. Saving will spin up the hard drive, which is undesirable on hosts where MPD is idling in background.
2009-10-08 15:22:39 +02:00
* state_file: save only if something has changed
text_file: allocate line buffers dynamically 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.
2009-11-01 15:37:16 +01:00
* database: eliminated maximum line length
log: redirect stdout/stderr to /dev/null if syslog is used Don't hold a file descriptor on root's tty when syslog is used for logging.
2009-11-07 17:48:57 +01:00
* log: redirect stdout/stderr to /dev/null if syslog is used
set the close-on-exec flag on all file descriptors 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.
2009-11-07 18:55:16 +01:00
* set the close-on-exec flag on all file descriptors
pcm_mix: implemented 32 bit support
2009-11-19 21:00:54 +01:00
* pcm_volume, pcm_mix: implemented 32 bit support
audio_format: support packed 24 bit samples
2010-01-15 09:23:36 +01:00
* support packed 24 bit samples
playlist/cue: fill song.start_ms, .end_ms This patch adds practical usefulness to the CUE playlist plugin.
2009-12-26 12:03:27 +01:00
* CUE sheet support
Add support for MixRamp tags Adds mixrampdb and mixrampdelay commands. Reads MIXRAP_START and MIXRAMP_END tags from FLAC files and overlaps instead of crossfading.
2010-03-21 18:21:47 +01:00
* support for MixRamp tags
cmdline: obey $(sysconfdir) for default mpd.conf location Instead of hard-coding the path "/etc/mpd.conf", use the configured $(sysconfdir) path. This can be set with: ./configure --sysconfdir=/etc Note that this changes the default path to "/usr/local/etc/mpd.conf", given the default prefix "/usr/local". This is actually more correct than the old default.
2009-07-28 16:17:18 +02:00
* obey $(sysconfdir) for default mpd.conf location
configure.ac: build with large file support by default This fixes mpg123 support.
2009-10-08 15:24:59 +02:00
* build with large file support by default
Makefile.am: added unit tests for the archive plugins
2009-12-31 11:11:14 +01:00
* added test suite ("make check")
configure.ac: require GLib 2.12 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.
2009-11-10 21:14:22 +01:00
* require GLib 2.12
client: optionally use libwrap
2010-01-20 21:04:07 +01:00
* added libwrap support
playlist: make single mode 'sticky'
2010-09-28 12:52:52 +02:00
* make single mode 'sticky'
Modify version string to post-release version 0.16~git
2009-06-24 03:43:57 +02:00
Modify version string to post-release version 0.15.17~git
2011-03-14 04:27:33 +01:00
ver 0.15.17 (2011/??/??)
encoder/vorbis: reset the Ogg stream after flush Without the ogg_stream_reset() call, the "e_o_s" flag never gets reset, and libogg writes EOS packets over and over.
2011-03-16 19:13:46 +01:00
* encoder:
- vorbis: reset the Ogg stream after flush
configure.ac: Enable HAVE_OGG_COMMON when using libtremor Otherwise OGGs can't be played.
2011-03-20 17:25:52 +01:00
* decoders:
- vorbis: fix tremor support
Modify version string to post-release version 0.15.17~git
2011-03-14 04:27:33 +01:00
mpd version 0.15.16
2011-03-14 04:27:33 +01:00
ver 0.15.16 (2011/03/13)
Insure proper initialization of stack-allocated struct. Version 1.0.0 of the libao library added a new field to the ao_sample_format struct. It is a char * named matrix. When an ao_sample_format is allocated on the stack, this field contains garbage. The proper course is to insure that is initialized to NULL. NULL indicates that we do not want any mapping. The struct is now initialized using a static initializer, and this technique is compatible with all known versions of libao.
2011-02-13 02:37:28 +01:00
* output:
- ao: initialize the ao_sample_format struct
output/jack: fix crash with mono playback With mono sound, jack_sample_size is smaller than frame_size (4 vs 2 bytes), and "space/jack_sample_size==0". That means mpd_jack_play() will return 0, although no error has occurred.
2011-02-27 23:26:50 +01:00
- jack: fix crash with mono playback
encoder/lame: explicitly configure the output sample rate When you don't explicitly set an output sample rate, liblame tries to guess an output sample rate from the input sample rate. You would think that this "guessing" consists of just setting both equal, but that is not the case. For 44.1kHz at 96kbit/s, liblame chooses 32kHz. This patch explicitly configures the output sample rate, to stop the bad guessing.
2011-01-07 19:37:39 +01:00
* encoders:
- lame: explicitly configure the output sample rate
update: log all file permission problems
2011-01-31 07:19:34 +01:00
* update: log all file permission problems
Modify version string to post-release version 0.15.16~git
2010-12-21 20:19:49 +01:00
mpd version 0.15.15
2010-11-08 18:48:28 +01:00
ver 0.15.15 (2010/11/08)
input/rewind: fix assertion failure The assertion added in MPD 0.15.14 was too much, it failed when the MIME type of a stream was NULL.
2010-11-08 10:37:09 +01:00
* input:
- rewind: fix assertion failure
output/shout: artist comes first in stream title After popular demand, I've switched the order of "artist" and "title" in the stream title. There is no standard, and there is no reliable way to parse those from the stream title.
2010-11-08 18:46:14 +01:00
* output:
- shout: artist comes first in stream title
Modify version string to post-release version 0.15.15~git
2010-11-06 22:42:03 +01:00
mpd version 0.15.14
2010-11-06 22:42:02 +01:00
ver 0.15.14 (2010/11/06)
player_thread: fix assertion failure due to wrong music pipe on seek When one song is played twice, and the decoder is working on the second "instance", but the first should be seeked, the check in player_seek_decoder() may assume that it can reuse the decoder without exchanging pipes. The last thing was the mistake: the pipe pointer was different, which led to an assertion failure. This patch adds another check which exchanges the player pipe.
2010-11-05 19:24:42 +01:00
* player_thread: fix assertion failure due to wrong music pipe on seek
output_thread: fix assertion failure due to race condition in OPEN Change the assertion on "fail_timer==NULL" in OPEN to a runtime check. This assertion crashed when the output thread failed while the player thread was calling audio_output_open().
2010-11-04 21:30:16 +01:00
* output_thread: fix assertion failure due to race condition in OPEN
input/rewind: fix double free bug Duplicate the "mime" attribute of the inner input_stream object, instead of copying the pointer.
2010-11-05 18:23:48 +01:00
* input:
- rewind: fix double free bug
decoder/{mp4ff,ffmpeg}: add extension ".m4b" (audio book) Same as ".m4a".
2010-11-05 02:01:35 +01:00
* decoders:
- mp4ff, ffmpeg: add extension ".m4b" (audio book)
Modify version string to post-release version 0.15.14~git
2010-10-10 18:57:57 +02:00
mpd version 0.15.13
2010-10-10 18:57:52 +02:00
ver 0.15.13 (2010/10/10)
output_thread: fix race condition after CANCEL command Clear the notification before finishing the CANCEL command, so the notify_wait() after that will always wait for the right notification, sent by audio_output_all_cancel().
2010-08-19 11:05:24 +02:00
* output_thread: fix race condition after CANCEL command
Correctly terminate stream_title. This caused random data to be send via icy-server if the played song had no tags.
2010-09-08 13:19:59 +02:00
* output:
- httpd: fix random data in stream title
output/httpd: MIME type audio/ogg for Ogg Vorbis RFC 5334 10.3 defines the MIME type "audio/ogg". We could use "application/ogg" as well, but we know for sure that we only emit audio data.
2010-10-03 16:22:03 +02:00
- httpd: MIME type audio/ogg for Ogg Vorbis
rewind_input_plugin: Update MIME not only once The assumption that MIME type is set only once is not valid with CURL, as URL redirections may update the MIME type. This fixes bug #3044.
2010-09-22 22:20:50 +02:00
* input:
- rewind: update MIME not only once
input/rewind: enable for MMS
2010-09-28 12:56:47 +02:00
- rewind: enable for MMS
Modify version string to post-release version 0.15.13~git
2010-07-21 05:19:22 +02:00
mpd version 0.15.12
2010-07-20 22:57:57 +02:00
ver 0.15.12 (2010/07/20)
input/curl: remove assertion after curl_multi_fdset() Some users reported that MPD crashes when using a new CURL version with the threaded DNS resolver enabled. It seems that curl_multi_fdset() returns no file descriptor when the DNS resolver runs in another thread, so MPD does not have any event to wait for. On the CURL mailing list, somebody suggested to sleep for a fixed amount of time. This is not an elegant solution, because daemons should never have to sleep without waiting for an event. I hope the CURL developers will review the API and remove the threaded DNS resolver. Meanwhile, I'm removing the assertion in question, to allow those unfortunate users running the latest CURL version to continue using MPD.
2010-07-20 22:49:29 +02:00
* input:
- curl: remove assertion after curl_multi_fdset()
tag_rva2: set "gain", not "peak" RVA2 tags only store the "gain" value, there is no "peak" attribute.
2010-07-20 22:32:55 +02:00
* tags:
- rva2: set "gain", not "peak"
decoder/wildmidi: support version 0.2.3 In libwildmidi 0.2.3, the function WildMidi_SampledSeek() was removed, without changing the SO name. This patch adds an autoconf check for that function. Fall back to WildMidi_FastSeek() if WildMidi_SampledSeek() is not available anymore.
2010-07-20 18:26:13 +02:00
* decoders:
- wildmidi: support version 0.2.3
Modify version string to post-release version 0.15.12~git
2010-07-15 02:33:29 +02:00
mpd version 0.15.11
2010-07-15 02:33:28 +02:00
ver 0.15.11 (2010/06/14)
tag_ape: support album artist I took this tag name from a MusePack sample file I got from a user. It is not documented in the APE specification: http://wiki.hydrogenaudio.org/index.php?title=APE_key People seem to be using undocumented extensions to the specification anyway, and the best we can do is attempt to support them.
2010-05-30 22:29:48 +02:00
* tags:
- ape: support album artist
decoder/mp4ff: support tags "albumartist", "band" I'm not sure if mapping "band" to TAG_PERFORMER is correct, but it might be better than nothing.
2010-06-30 21:22:13 +02:00
* decoders:
decoder/mp4ff: support tag "album artist" We already supported "albumartist", but it seems some folks also use "album artist" (with a space).
2010-06-30 21:55:03 +02:00
- mp4ff: support tags "album artist", "albumartist", "band"
decoder/mikmod: fix memory leak The return value of Player_LoadTitle() is allocated with malloc(), and must be freed by the caller.
2010-06-30 21:37:36 +02:00
- mikmod: fix memory leak
decoder/vorbis: handle uri==NULL This fixes a theoretical crash, which has never occurred in practice.
2010-06-30 23:38:49 +02:00
- vorbis: handle uri==NULL
decoder/ffmpeg: free URI, fix memory leak Free the string allocated by decoder_get_uri().
2010-06-30 23:40:04 +02:00
- ffmpeg: fix memory leak
decoder/ffmpeg: free AVFormatContext on error Fix a memory leak in some code paths.
2010-01-18 11:05:15 +01:00
- ffmpeg: free AVFormatContext on error
ffmpeg: read more metadata.
2010-02-02 12:35:08 +01:00
- ffmpeg: read more metadata
decoder/ffmpeg: fix libavformat 0.6 by using av_open_input_stream() libavformat 0.6 does not pass the original URI pointer to the "open" method, which leads to a crash because MPD was using a dirty hack to pass a pointer to that method. This patch switches to av_open_input_stream() with a custom ByteIOContext class, instead of doing the URI string hack with av_open_input_file(). Loosely based on a patch from Jasper St. Pierre.
2010-06-30 23:27:45 +02:00
- ffmpeg: fix libavformat 0.6 by using av_open_input_stream()
playlist: emit IDLE_OPTIONS when resetting single mode
2010-06-30 21:40:33 +02:00
* playlist: emit IDLE_OPTIONS when resetting single mode
Make get_remote_uid() work on BSD I've attached a patch that will make file URIs work on operating systems that provide the getpeereid() function call to check the user ID of the peer connected to a UNIX domain socket.
2010-06-30 21:42:01 +02:00
* listen: make get_remote_uid() work on BSD
Modify version string to post-release version 0.15.11~git
2010-05-30 17:59:00 +02:00
mpd version 0.15.10
2010-05-30 17:59:00 +02:00
ver 0.15.10 (2010/05/30)
input/mms: fix memory leak in error handler
2010-05-18 20:57:57 +02:00
* input:
- mms: fix memory leak in error handler
input/mms: initialize the "eof" attribute
2010-05-18 21:11:00 +02:00
- mms: initialize the "eof" attribute
decoder/mad: properly calculate ID3 size without libid3tag Without libid3tag, we were trying to skip the ID3 frame (since 0.15.2). Its length however was not calculated at all, we were just dropping everything from the current input buffer. This lead to the first few seconds of the file being skipped. This patch attempts to calculate the ID3v2 frame size with the formula from: http://www.id3.org/id3v2.4.0-structure 3.1 and 6.2
2010-04-13 07:51:29 +02:00
* decoders:
- mad: properly calculate ID3 size without libid3tag
Modify version string to post-release version 0.15.10~git
2010-03-22 01:25:18 +01:00
mpd version 0.15.9
2010-03-22 01:25:18 +01:00
ver 0.15.9 (2010/03/21)
decoder/mad: fix crash when seeking at end of song Removed the decoder_command_finished() call at the end of mp3_decode(). This is invalid, because decoder_command_finished() has already been called in mp3_read().
2010-02-27 18:35:31 +01:00
* decoders:
- mad: fix crash when seeking at end of song
decoder/mpcdec: fix negative shift on fixed-point samples "There is a bug in fixed-point musepack (musepack_src_r435) playback. In floating-point audio is OK but in fixed audio is distorted. I have made a patch for this"
2010-03-17 17:54:21 +01:00
- mpcdec: fix negative shift on fixed-point samples
decoder/mpcdec: fix replay gain formula with v8 "When playing musepack files with mpd v0.15.8, rg seems to have no effect. Using sample file below, mpd says 'computing ReplayGain album scale with gain 122.879997, peak 0.549150'. One thing though, if I build mpd against old libmpcdec-1.2.6, rg works as expected: 'computing ReplayGain album scale with gain 16.820000, peak 0.099765'"
2010-03-12 18:04:46 +01:00
- mpcdec: fix replay gain formula with v8
playlist: fix single+repeat in random mode With single+repeat enabled, it is expected that MPD repeats the current song over andd over. With random mode also enabled, this didn't work, because the song order was shuffled internally. This patch adds a special check for this case.
2010-03-07 18:58:44 +01:00
* playlist: fix single+repeat in random mode
player_thread: postpone song tags during cross-fade Previously, tags of the new song being cross-faded in were sent immediately. That can cause wrong information being displayed, because the "previous" song might send its tag at the end again, overriding the "next" song's tag. This patch saves & merges the tag of the next song, and sends it when cross-fading is finished, and the next song really starts.
2010-03-17 21:57:35 +01:00
* player: postpone song tags during cross-fade
Modify version string to post-release version 0.15.9~git
2010-01-18 02:28:39 +01:00
mpd version 0.15.8
2010-01-18 02:28:39 +01:00
ver 0.15.8 (2010/01/17)
input/rewind: new input_stream wrapper to allow stream rewinding This replaces the rewinding buffer code from the CURL input plugin. It is more generic, and allows rewinding even when the server sends Icy-Metadata (which would have been too difficult to implement within the CURL plugin). This is a rather complex patch for the stable branch (v0.15.x), but it fixes a serious problem: the "vorbis" decoder plugin was unable to play streams with Icy-Metadata, because it couldn't rewind the stream after detecting the codec (Vorbis vs. FLAC).
2009-12-29 23:55:40 +01:00
* input:
- curl: allow rewinding with Icy-Metadata
decoder/{ffmpeg,flac,vorbis}: added more flac/vorbis MIME types Support deprecated MIME types such as "audio/x-ogg". Support new types such as "audio/flac".
2009-12-29 22:33:46 +01:00
* decoders:
- ffmpeg, flac, vorbis: added more flac/vorbis MIME types
decoder/ffmpeg: append file name suffix to virtual stream URL To allow libavformat to detect the format of the input file, append the suffix of the input file to the URL of the virtual stream. This specifically enables the "shorten" codec, which is supported by libavformat/raw.c, detected only by the suffix.
2010-01-17 02:35:15 +01:00
- ffmpeg: enabled libavformat's file name extension detection
NEWS: added missing entry
2010-01-16 18:41:34 +01:00
* dbUtils: return empty tag value only if no value was found
decoder_thread: fix CUE track playback The patch "input/file: don't fall back to parent directory" introduced a regression: when trying to play a CUE track, decoder_run_song() tries to open the file as a stream first, but this fails, because the path is virtual. This patch fixes decoder_run_song() (instead of reverting the previous patch) to accept input_stream_open() failures if the song is a local file. It passes the responsibility to handle non-existing files to the decoder's file_decode() method.
2010-01-16 19:20:11 +01:00
* decoder_thread: fix CUE track playback
queue: don't repeat current song in consume mode Check consume mode in queue_next_order(), because the current song would be deleted as soon as it's finished; it cannot be played again.
2010-01-16 20:58:24 +01:00
* queue: don't repeat current song in consume mode
Modify version string to post-release version 0.15.8~git
2009-12-27 17:31:47 +01:00
mpd version 0.15.7
2009-12-27 17:31:47 +01:00
ver 0.15.7 (2009/12/27)
archive: close archive when stream is closed 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.
2009-12-15 19:45:50 +01:00
* archive:
- close archive when stream is closed
iso, zip: fixed memory leak in destructor Free the "context" pointer in the method archive_plugin.close().
2009-12-15 19:56:38 +01:00
- iso, zip: fixed memory leak in destructor
input/file: don't fall back to parent directory This code has never made any sense, and has broken some of the archive plugin.
2009-12-15 18:59:03 +01:00
* input:
- file: don't fall back to parent directory
input/archive: close the archive file on error Fixed memory leak in error handler.
2009-12-15 19:31:50 +01:00
- archive: fixed memory leak in error handler
tag_id3: fix ID3v1 charset conversion If we define id3v1_encoding, then the tags are not added to the database.
2009-11-30 17:42:46 +01:00
* tags:
- id3: fix ID3v1 charset conversion
ffmpeg: don't try to force stereo 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.
2009-11-30 09:59:05 +01:00
* decoders:
decoder_thread: eliminate jitter after seek failure Don't clear the music pipe when seeking has failed - check the "seeking" flag instead of "command==SEEK". Clear the "seeking" flag in decoder_seek_error().
2009-12-26 01:10:23 +01:00
- eliminate jitter after seek failure
ffmpeg: don't try to force stereo 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.
2009-11-30 09:59:05 +01:00
- ffmpeg: don't try to force stereo
decoder/wavpack: allow fine-grained seeking First multiply the floating point return value of decoder_seek_where(), then cast to integer.
2009-12-26 00:35:05 +01:00
- wavpack: allow fine-grained seeking
mixer: explicitly close all mixers on shutdown 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.
2009-12-08 08:47:47 +01:00
* mixer: explicitly close all mixers on shutdown
mapper: fix memory leak when playlist_directory is not set Don't allocate the file name before the playlist_dir==NULL check.
2009-12-08 08:06:10 +01:00
* mapper: fix memory leak when playlist_directory is not set
mapper: apply filesystem_charset to playlists 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.
2009-12-08 08:17:35 +01:00
* mapper: apply filesystem_charset to playlists
command: verify playlist name in the "rm" command Call spl_valid_name() in spl_delete().
2009-12-08 08:30:33 +01:00
* command: verify playlist name in the "rm" command
dbUtils: return multiple tag values per song When collecting tag values for the result set, add all of a song's tag values of the searched type. This affects the "list" command. Previously, "list" only considered the first tag value of a song.
2009-12-27 16:49:04 +01:00
* database: return multiple tag values per song
Modify version string to post-release version 0.15.7~git
2009-11-19 03:49:25 +01:00
mpd version 0.15.6
2009-11-19 03:49:25 +01:00
ver 0.15.6 (2009/11/18)
input/lastfm: fixed variable name in GLib<2.16 code path Should be "lastfm_user", not "lastfm_username".
2009-11-10 20:54:17 +01:00
* input:
- lastfm: fixed variable name in GLib<2.16 code path
input/mms: require libmms 0.4 We're using API functions which are not available in 0.3.
2009-11-10 20:57:10 +01:00
- input/mms: require libmms 0.4
zzip: require libzzip 0.13 We need the function zzip_file_stat().
2009-11-10 21:00:10 +01:00
* archive:
- zzip: require libzzip 0.13
id3: allow 4 MB RIFF/AIFF tags 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?).
2009-11-15 18:44:53 +01:00
* tags:
- id3: allow 4 MB RIFF/AIFF tags
decoder/ffmpeg: convert metadata 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.
2009-10-28 22:12:22 +01:00
* decoders:
- ffmpeg: convert metadata
decoder/ffmpeg: align the output buffer 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.
2009-11-15 17:39:09 +01:00
- ffmpeg: align the output buffer
oggflac: rewind stream after FLAC detection The oggflac plugin has been completely broken for quite a while and nobody has noticed - maybe we should remove it?
2009-11-11 08:14:37 +01:00
- oggflac: rewind stream after FLAC detection
decoder/flac: fixed CUE seeking range check 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.
2009-11-11 08:55:55 +01:00
- flac: fixed CUE seeking range check
decoder/flac: fixed NULL pointer dereference in CUE code The function flac_vtrack_tnum() was missing a strrchr()==NULL check.
2009-11-18 19:55:38 +01:00
- flac: fixed NULL pointer dereference in CUE code
output_thread: check again if output is open on PAUSE Basically the same as the 0.15.5 patch "check again if output is open on CANCEL". Same race condition, same fix.
2009-10-21 08:07:07 +02:00
* output_thread: check again if output is open on PAUSE
update: delete ignored symlinks from database 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.
2009-10-27 22:06:28 +01:00
* update: delete ignored symlinks from database
song_save: increased maximum line length to 32 kB 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.
2009-11-01 15:27:55 +01:00
* database: increased maximum line length to 32 kB
sticker: added fallback for sqlite3_prepare_v2() This function was not present in SQLite < 3.4.
2009-11-10 20:55:29 +01:00
* sticker: added fallback for sqlite3_prepare_v2()
Modify version string to post-release version 0.15.6~git
2009-10-18 18:50:17 +02:00
mpd version 0.15.5
2009-10-18 18:50:16 +02:00
ver 0.15.5 (2009/10/18)
input/curl: don't abort if a packet has only metadata When a received chunk of data has only icy-metadata, there was no usable data left for input_curl_read() to return, and thus it returned 0 bytes. "0" however is a special value for "end of file" or "error". This patch makes input_curl_read() read more data from the socket, until the read request can be fulfilled (or until there's really EOF).
2009-10-11 23:13:49 +02:00
* input:
- curl: don't abort if a packet has only metadata
input/curl: fixed endless loop during buffering When the connection is lost while buffering, the CURL input plugin may enter an endless loop, because it does not check the EOF condition. This patch makes fill_buffer() return success only if there's at least one buffer, which is enough of a check.x
2009-10-13 16:24:28 +02:00
- curl: fixed endless loop during buffering
riff, aiff: fixed "limited range" gcc warning On 32 bit systems with large file support enabled (i.e. "sizeof(off_t) > sizeof(size_t)") gcc emits a warning because a size_t cast to off_t can never become negative.
2009-10-11 23:15:38 +02:00
* tags:
- riff, aiff: fixed "limited range" gcc warning
decoder/flac: fixed two memory leaks in the CUE tag loader Don't initialize "vc" and "cs" with FLAC__metadata_object_new(); that value is overwritten by FLAC__metadata_get_tags() and FLAC__metadata_get_cuesheet().
2009-10-16 17:39:17 +02:00
* decoders:
- flac: fixed two memory leaks in the CUE tag loader
decoder_thread: change the fallback decoder name to "mad" When there is no Content-Type response header, try the "mad" decoder plugin. It uesd to be named "mp3", and we forgot to change the fallback name in decoder_thread.c.
2009-10-11 23:14:16 +02:00
* decoder_thread: change the fallback decoder name to "mad"
output_thread: check again if output is open on CANCEL When the player thread unpauses, it sends CANCEL to the output thread, after having checked that the output is still open. Problem is when the output thread closes the device before it can process the CANCEL command - race condition. This patch adds another "open" check inside the output thread.
2009-10-16 15:09:22 +02:00
* output_thread: check again if output is open on CANCEL
update: fixed memory leak during container scan The return value of map_directory_child_fs() must be freed.
2009-10-16 17:38:15 +02:00
* update: fixed memory leak during container scan
Modify version string to post-release version 0.15.5~git
2009-10-03 14:44:26 +02:00
mpd version 0.15.4
2009-10-03 14:44:26 +02:00
ver 0.15.4 (2009/10/03)
decoder/vorbis: revert "faster tag scanning with ov_test_callback()" This patch made ov_time_total() unusable, and MPD did not know the duration of songs.
2009-09-10 23:04:01 +02:00
* decoders:
- vorbis: revert "faster tag scanning with ov_test_callback()"
decoder/faad: skip assertion failure on large ID3 tags When the ID3 tag in an AAC file is larger than the current buffer, the function decoder_buffer_consume() aborts. By using the new function decoder_buffer_skip() instead, we can safely skip the ID3 tag.
2009-09-30 15:22:47 +02:00
- faad: skip assertion failure on large ID3 tags
decoder/ffmpeg: use the "artist" tag if "author" is not present Usually, we read our "artist" tag from ffmpeg's "author" tag. In some cases however (e.g. APE), this tag is named "artist". This patch implements a fallback: if no "author" is found, MPD tries to use "artist".
2009-09-30 15:41:43 +02:00
- ffmpeg: use the "artist" tag if "author" is not present
output/osx: fix the OS X 10.6 build Include CoreServices/CoreServices.h.
2009-09-20 23:30:37 +02:00
* output:
- osx: fix the OS X 10.6 build
Modify version string to post-release version 0.15.4~git
2009-08-30 07:59:24 +02:00
mpd version 0.15.3
2009-08-30 07:59:24 +02:00
ver 0.15.3 (2009/08/29)
decoder/vorbis: faster tag scanning with ov_test_callback() using ov_test_callback with function CALLBACKS_STREAMONLY will cause scanning to stop after the comment field. ov_open (and ov_test) default to CALLBACKS_DEFAULT which scans the file structure causing a huge slowdown. The speed improvement is huge: It scanned my files around 10x faster This procedure has been recommended by monthy (main vorbis developer) and was said to be safe for scanning files.
2009-08-24 22:14:22 +02:00
* decoders:
- vorbis: faster tag scanning with ov_test_callback()
output_init: initialize the "pause" flag Fix stuttering due to uninitialized variable.
2009-08-18 11:32:54 +02:00
* output:
- fix stuttering due to uninitialized variable
update: don't re-read unchanged container files MPD checks if every flac (possibly other types as well) file contains cuesheet on every update, which produces unneeded I/O. My music collection is on NFS share, so it's quite noticeable. IMHO, it shouldn't re-read unchanged files, so I wrote simple patch to fix it.
2009-08-19 21:21:29 +02:00
* update: don't re-read unchanged container files
Modify version string to post-release version 0.15.3~git
2009-08-15 20:57:50 +02:00
mpd version 0.15.2
2009-08-15 20:57:50 +02:00
ver 0.15.2 (2009/08/15)
ape: check the tag size (fixes integer underflow) The expression "tagLen - size > 0" may result in an integer underflow and a buffer overflow, when "size" is larger than "tagLen". "size" is read from the input file, and must not be trusted. This patch changes the expression to "tagLen > size", which is a lot safer.
2009-07-18 22:45:56 +02:00
* tags:
- ape: check the tag size (fixes integer underflow)
ape: added protection against large memory allocations The function tag_ape_load() retrieves a 32 bit unsigned integer from the input file, and passes it to g_malloc(). This is dangerous, and may be used for a denial of service attack on MPD.
2009-07-19 17:38:46 +02:00
- ape: added protection against large memory allocations
mad: skip ID3 frames when libid3tag is disabled When libid3tag is disabled, the libmad decoder plugin is unable to identify ID3 frames. If the file starts with an (unidentified) ID3 frame, it assumes that the file is not a valid MP3 song. This patch solves this by adding minimal stubs for the ID3 functions.
2009-07-22 12:57:03 +02:00
* decoders:
- mad: skip ID3 frames when libid3tag is disabled
decoder/flac: parse all replaygain tags The FLAC replaygain parser used the "||" operator. This made the code stop after the first value which was found.
2009-07-22 13:31:48 +02:00
- flac: parse all replaygain tags
decoder/flac: don't allocate cuesheet twice (memleak) The function flac_cue_track() first calls FLAC__metadata_object_new(), then overwrites this pointer with FLAC__metadata_get_cuesheet(). This allocate two FLAC__StreamMetadata objects, but the first pointer is lost, and never freed.
2009-08-14 11:51:42 +02:00
- flac: don't allocate cuesheet twice (memleak)
output: fixed shout stuck pause bug Explicitly make the output thread leave the ao_pause() loop. This patch is a workaround, and the "pause" flag is not managed in a thread-safe way, but that's good enough for now.
2009-08-14 11:52:12 +02:00
* output:
- shout: fixed stuck pause bug
output/shout: minimize the unpause latency During the pause loop, manually sleep for 500ms if shout_delay() returns a value greater than that. Don't exhaust libshout's buffer.
2009-08-14 11:52:36 +02:00
- shout: minimize the unpause latency
update: free empty path string (memleak) When you pass an empty string to directory_update_init(), it was not freed by update_task().
2009-08-14 11:51:35 +02:00
* update: free empty path string (memleak)
update: free temporary string in container scan (memleak) The return value of map_directory_child_fs() must be freed.
2009-08-14 11:51:51 +02:00
* update: free temporary string in container scan (memleak)
directory: free empty directories after removing them (memleak) dirvec_delete() does not free the object, we have to call directory_free() afterwards.
2009-08-14 11:52:00 +02:00
* directory: free empty directories after removing them (memleak)
Modify version string to post-release version 0.15.2~git
2009-07-15 22:36:41 +02:00
mpd version 0.15.1
2009-07-15 22:36:41 +02:00
ver 0.15.1 (2009/07/15)
NEWS: fixed typo
2009-07-14 14:24:02 +02:00
* decoders:
decoder/flac: fix assertion failure in tag_free() call Initialize flac_data.tag right after flac_data_init(). This way, the "goto fail" won't jump to the point where tag_free(NULL) can be called.
2009-07-06 22:09:30 +02:00
- flac: fix assertion failure in tag_free() call
output/httpd: include sys/types.h On Mac OS X, the httpd plugin cannot be compiled, because OS X's system headers do nto include sys/types.h, although they use u_int32_t.
2009-07-06 14:40:06 +02:00
* output:
- httpd: include sys/types.h (fixes Mac OS X)
output_all: don't resume playback when stopping during pause When MPD was paused, and the client sent the "stop" command (or "clear"), a glitch caused MPD to continue playback for a split second. This was because audio_output_all_cancel() calls audio_output_all_update(), which reopens all output devices, and re-ignites the playback loop.
2009-06-29 22:20:36 +02:00
* commands:
- don't resume playback when stopping during pause
database: fixed NULL pointer dereference after charset change When the filesystem_charset is changed in mpd.conf, MPD should discard the old database. In this error branch, MPD did not fill the GError object properly, and logged a warning message instead, which caused a segmentation fault.
2009-06-30 16:29:40 +02:00
* database: fixed NULL pointer dereference after charset change
log: fix double free() bug during shutdown Don't free an internal configuration value in log_init(). Call config_get_path() instead of manually calling parsePath().
2009-07-05 07:14:24 +02:00
* log: fix double free() bug during shutdown
NEWS: updated NEWS file for 0.15.1
2009-06-25 11:29:08 +02:00
mpd version 0.15
2009-06-24 03:43:57 +02:00
ver 0.15 (2009/06/23)
NEWS: update change log
2009-01-22 16:06:39 +01:00
* input:
- parse Icy-Metadata
added support for the MMS protocol This patch implements the MMS protocol, by using libmms. It is quite experimental: it does not support seeking yet, and it is currently using synchronous I/O, which causes MPD to hang while waiting for the server.
2009-01-29 21:42:10 +01:00
- added support for the MMS protocol
song_print: hide HTTP password in playlist 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.
2009-02-27 19:20:11 +01:00
- hide HTTP password in playlist
NEWS: flagged some new features "experimental" 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
2009-04-02 07:12:38 +02:00
- lastfm: new input plugin for last.fm radio (experimental and incomplete!)
curl: moved proxy settings to "input" block The old global settings "http_proxy_host", "http_proxy_port", "http_proxy_user" and "http_proxy_password" continue to work.
2009-04-25 13:35:04 +02:00
- curl: moved proxy settings to "input" block
NEWS: update change log
2009-01-22 16:06:39 +01:00
* tags:
- support the "album artist" tag
tag: added support for MusicBrainz tags Added all important id tags from the MusicBrainz wiki: http://musicbrainz.org/doc/MusicBrainzTag This should automatically enable its suport in the vorbis and flac decoder plugins.
2009-01-24 20:02:55 +01:00
- support MusicBrainz tags
NEWS: update change log
2009-01-22 16:06:39 +01:00
- parse RVA2 tags in mp3 files
news: add more miscellaneous missing news
2009-04-01 14:50:22 +02:00
- parse ID3 tags in AIFF/RIFF/WAV files
- ffmpeg: support new metadata API
- ffmpeg: added support for the tags comment, genre, year
NEWS: update change log
2009-01-22 16:06:39 +01:00
* decoders:
- audiofile: streaming support added
audiofile: added 24 bit support Don't hard code the "bits" parameter to 16. Try to use the input's sample format, if possible.
2009-03-01 10:53:46 +01:00
- audiofile: added 24 bit support
NEWS: update change log
2009-01-22 16:06:39 +01:00
- modplug: another MOD plugin, based on libmodplug
- mikmod disabled by default, due to severe security issues in libmikmod
sidplay: new decoder plugin for playing C64 SID files
2009-02-11 20:31:17 +01:00
- sidplay: new decoder plugin for C64 SID (using libsidplay2)
NEWS: flagged some new features "experimental" 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
2009-04-02 07:12:38 +02:00
- fluidsynth: new decoder plugin for MIDI files (using libfluidsynth,
experimental due to shortcomings in libfluidsynth)
wildmidi: new decoder plugin for MIDI files
2009-02-12 16:43:18 +01:00
- wildmidi: another decoder plugin for MIDI files (using libwildmidi)
flac: parse stream tags Parse the vorbis comments in libflac's metadata_callback and pass them as tag struct to the decoder API.
2009-03-01 14:07:23 +01:00
- flac: parse stream tags
mpcdec: support the new libmpcdec SV8 API
2009-03-27 19:51:59 +01:00
- mpcdec: support the new libmpcdec SV8 API
decoder_list: added configuration option to disable decoder plugins
2009-02-15 18:40:47 +01:00
- added configuration option to disable decoder plugins
news: add more miscellaneous missing news
2009-04-01 14:50:22 +02:00
- flac: support embedded cuesheets
- ffmpeg: updated list of supported formats
NEWS: update change log
2009-01-22 16:06:39 +01:00
* audio outputs:
output: added option to disable audio outputs by default The option "enabled" is on by default. If you specify "enabled no" in an audio_output section, then this device is disabled by default.
2009-02-28 19:40:39 +01:00
- added option to disable audio outputs by default
output_thread: wait 10 seconds before reopening after play failure 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().
2009-03-09 19:08:35 +01:00
- wait 10 seconds before reopening after play failure
NEWS: update change log
2009-01-22 16:06:39 +01:00
- shout: enlarged buffer size to 32 kB
null: added option to disable timer synchronization The null plugin synchronizes the playback so it will happen in real time. This patch adds a configuration option which disables this: the playback will then be as fast as possible. This can be useful to profile MPD.
2009-01-22 16:06:47 +01:00
- null: allow disabling synchronization
mvp: fall back to stereo When the channel count is greater than 2, fall back to stereo sound.
2009-02-25 21:51:39 +01:00
- mvp: fall back to stereo
mvp: fall back to 16 bit audio samples 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.
2009-02-25 21:52:11 +01:00
- mvp: fall back to 16 bit audio samples
mvp: check for reopen errors When the MVP device has been closed in the cancel() method, and the play() method attempts to reopen it, check for errors.
2009-02-25 21:57:02 +01:00
- mvp: check for reopen errors
mvp: fixed default device detection The check "open()!=0" is wrong, you have to write "open()>=0", because -1 means error, and 0 is a valid file handle.
2009-02-26 22:10:58 +01:00
- mvp: fixed default device detection
pipe: new audio output plugin which runs a command [mk: adapted to new output plugin API]
2009-02-28 16:11:59 +01:00
- pipe: new audio output plugin which runs a command
alsa: better period_time default value for high sample rates 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.
2009-03-08 03:55:01 +01:00
- alsa: better period_time default value for high sample rates
solaris: new audio output plugin for Solaris /dev/audio
2009-03-16 09:55:10 +01:00
- solaris: new audio output plugin for Solaris /dev/audio
news: add more miscellaneous missing news
2009-04-01 14:50:22 +02:00
- httpd: new audio output plugin for web based streaming, similar to icecast
but built in.
NEWS: update change log
2009-01-22 16:06:39 +01:00
* commands:
news: move supports a range
2009-04-01 14:29:11 +02:00
- "playlistinfo" and "move" supports a range now
NEWS: update change log
2009-01-22 16:06:39 +01:00
- added "sticker database", command "sticker", which allows clients
news: Add consume and single commands
2009-04-01 15:05:16 +02:00
to implement features like "song rating"
- added "consume" command which removes a song after play
- added "single" command, if activated, stops playback after current song or
repeats the song if "repeat" is active.
news: add pulseaudio mixer
2009-04-01 14:27:43 +02:00
* mixers:
- rewritten mixer code to support multiple mixers
- new pulseaudio mixer
alsa_mixer: add mixer_index option This allows you to select controls with duplicate names.
2009-04-28 09:15:48 +02:00
- alsa: new mixer_index option supports choosing between multiple
identically-named controls on a device.
Update news with RVA2 and Audio archive support.
2008-12-28 13:02:41 +01:00
* Add audio archive extraction support:
- bzip2
- iso9660
- zip
log: deprecated "error_file" option Removed the "error_file" option. There is only one log file now. If a user wants to see only the errors, he should configure a log_level.
2008-12-28 19:48:52 +01:00
* the option "error_file" was removed, all messages are logged into
news: Add consume and single commands
2009-04-01 15:05:16 +02:00
"log_file"
log: support syslog() Allow logging to syslog if log_file is configured to "syslog".
2008-12-28 19:48:53 +01:00
* support logging to syslog
mapper: fall back to XDG music directory If no music_directory is configured, fall back to the XDG music directory, determined by g_get_user_special_dir(G_USER_DIRECTORY_MUSIC).
2009-01-01 19:07:38 +01:00
* fall back to XDG music directory if no music_directory is configured
state_file: errors are non-fatal in read_state_file() If the state file cannot be read, for whatever reason, don't abort MPD. The state file isn't _that_ important.
2009-01-03 14:53:23 +01:00
* failure to read the state file is non-fatal
don't exit after --create-db Start the daemon after --create-db. This makes --create-db a flag which discards the old database and starts with a fresh one.
2009-01-04 20:31:23 +01:00
* --create-db starts the MPD daemon instead of exiting
mapper: make the music_directory optional Without a music_directory, MPD is an excellent streaming client.
2009-01-18 16:56:07 +01:00
* playlist_directory and music_directory are optional
playlist: recalculate the queued song after random is toggled When the random mode is toggled, MPD did not clear the queue. Because of this, MPD continued with the next (random or non-random) song according to the previous mode. Clear the queued song to fix that.
2009-01-23 00:07:10 +01:00
* playlist: recalculate the queued song after random is toggled
playlist: don't unpause on delete When you delete a song from the playlist which was paused, MPD forgot that it was paused and started playing the next song.
2009-01-23 16:17:21 +01:00
* playlist: don't unpause on delete
player_thread: pause when all audio outputs fail to play 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.
2009-04-25 11:55:36 +02:00
* pause when all audio outputs fail to play
daemon: ignore "user" setting if already running as that user 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.
2009-02-15 16:47:21 +01:00
* daemon: ignore "user" setting if already running as that user
listen: allocate sockaddr_storage struct for accept() 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.
2009-02-28 15:20:35 +01:00
* listen: fix broken client IP addresses in log
listen: bind() failure on secondary address is non-fatal Several users had problems with binding MPD to "localhost". The cause was duplicate /etc/hosts entries: the resolver library returns 127.0.0.1 twice, and of course, MPD attempts to bind to "both" of them. This patch makes failures non-fatal, given that at least one address was bound successfully. This is a workaround; users should rather fix their /etc/hosts file.
2009-06-10 08:01:07 +02:00
* listen: bind failure on secondary address is non-fatal
news: add more miscellaneous missing news
2009-04-01 14:50:22 +02:00
* 24/32 bit audio support
* print available protocols in --version
player_thread: fill buffer after seeking 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.
2009-03-10 21:19:51 +01:00
* fill buffer after seeking
pcm_resample: choose the fallback resampler at runtime 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.
2009-03-14 15:26:36 +01:00
* choose the fallback resampler at runtime
news: add more miscellaneous missing news
2009-04-01 14:50:22 +02:00
* steps taken towards win32 compatibility
* require glib 2.6 or greater
* built-in documentation using doxygen and docbook
NEWS: merge from v0.14.x Copy changelog items from the upcoming 0.14.1 release.
2009-01-15 16:16:37 +01:00
mpd version 0.14.2
2009-02-13 17:13:06 +01:00
ver 0.14.2 (2009/02/13)
configure.ac: define HAVE_FFMPEG after all checks 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.
2009-02-09 22:22:06 +01:00
* configure.ac:
- define HAVE_FFMPEG after all checks
ffmeg: added support for the tags comment, genre, year ffmpeg_tag_internal() does not look for a few tags that mpd supports. Most noteably: comment -> TAG_ITEM_COMMENT -> Description genre -> TAG_ITEM_GENRE -> WM/Genre (not WM/GenreID) year -> TAG_ITEM_DATE -> WM/Year I *think* that this is the last of the tags that AVFormatContext() in ffmpeg supports that mpd also uses.
2009-01-30 09:02:09 +01:00
* decoders:
- ffmpeg: added support for the tags comment, genre, year
ffmpeg: don't warn of empty packet output 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.
2009-02-03 21:44:14 +01:00
- ffmpeg: don't warn of empty packet output
ffmpeg: check if the time stamp is valid 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.
2009-02-03 21:55:28 +01:00
- ffmpeg: check if the time stamp is valid
ffmpeg: fixed seek integer overflow 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.
2009-02-03 22:46:00 +01:00
- ffmpeg: fixed seek integer overflow
ffmpeg: added TTA support The ffmpeg library supports the "True Audio Codec". The entry in ffmpeg_suffixes was missing.
2009-02-11 18:14:59 +01:00
- ffmpeg: enable WAV streaming
- ffmpeg: added TTA support
wavpack: pass NULL if the .wvc file fails to open The wavpack library seems to use the .wvc stream even if the OPEN_WVC flag is not set. In this case, pass NULL to be sure libwavpack won't use it.
2009-01-30 15:57:51 +01:00
- wavpack: pass NULL if the .wvc file fails to open
mikmod: call MikMod_Exit() only in the finish() method 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]
2009-01-28 19:13:44 +01:00
- mikmod: call MikMod_Exit() only in the finish() method
aac: fix stream metadata Pass the input_stream object to decoder_data(). Without it, the MPD core does not see stream tags.
2009-02-12 18:39:04 +01:00
- aac: fix stream metadata
jack: allocate ring buffers before connecting If the ring buffers are allocated after jack_activate(), mpd_jack_process() might segfault because it attempts to access them.
2009-01-29 18:11:30 +01:00
* audio outputs:
- jack: allocate ring buffers before connecting
jack: clear "shutdown" flag on reconnect When the connection failed once, you had to restart MPD, because it never cleared the jack_data.shutdown flag. Instead, it refused to play anything "because there is no client thread" (which is wrong at that point).
2009-01-29 17:08:44 +01:00
- jack: clear "shutdown" flag on reconnect
jack: reduced sleep time to 1ms When waiting for free space in the ring buffer, the JACK plugin sleeped 10ms until there is enough space. This delay was too large for low-latency setups (<10ms), and created a lot of xruns. Work around that by reducing the sleep time to 1ms. A proper solution for this would be to use an event based approach, and we will do it, just not now.
2009-01-29 17:08:51 +01:00
- jack: reduced sleep time to 1ms
shout_mp3: call lame_close() in clear_encoder() method 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.
2009-02-02 18:19:11 +01:00
- shout: fixed memory leak in the mp3 encoder
shout: switch to blocking mode 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.
2009-02-09 16:35:54 +01:00
- shout: switch to blocking mode
shout: use libshout's synchronization Removed the manual timer synchronization from the shout plugin. libshout's shout_sync() function does it for us.
2009-02-09 16:35:59 +01:00
- shout: use libshout's synchronization
shout: don't postpone metadata Don't duplicate the tag received by the send_metadata() method - send it to the shout server directly.
2009-02-09 16:36:06 +01:00
- shout: don't postpone metadata
shout: clear buffer before calling the encoder 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.
2009-02-09 16:36:11 +01:00
- shout: clear buffer before calling the encoder
mapper: remove trailing slashes from music_directory When the user configures a music_directory with a trailing slash, it may break playlist loading, because MPD expects a double slash. Chop off the trailing slash.
2009-01-30 13:47:45 +01:00
* mapper: remove trailing slashes from music_directory
player_thread: set player error when output device fails 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.
2009-02-02 18:02:52 +01:00
* player: set player error when output device fails
update: recursively purge deleted directories 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().
2009-02-12 18:57:12 +01:00
* update: recursively purge deleted directories
update: free deleted subdirectories Use delete_directory() for removing sub directories instead of dirvec_clear(). This ensures that all memory occupied by subdirectories of deleted directories is freed.
2009-02-12 19:11:26 +01:00
* update: free deleted subdirectories
Modify version string to post-release version 0.14.2~git
2009-01-17 20:41:52 +01:00
mpd version 0.14.1
2009-01-17 20:41:52 +01:00
ver 0.14.1 (2009/01/17)
NEWS: merge from v0.14.x Copy changelog items from the upcoming 0.14.1 release.
2009-01-15 16:16:37 +01:00
* decoders:
- mp4: support the writer/composer tag
- id3: strip leading and trailing whitespace from ID3 tags
oggvorbis: define HAVE_OGGVORBIS when tremor is enabled When tremor (libvorbisidec) is used, HAVE_OGGVORBIS was not defined.
2009-01-15 22:54:07 +01:00
- oggvorbis: fix tremor support
oggvorbis: disable seeking on remote songs When libvorbis knows that a song is seekable, it seeks around like crazy in the file before starting to decode it. This is very expensive on remote HTTP resources, and delays MPD for 10 or 20 seconds. This patch disables seeking on remote songs, because the advantages of quickly playing a song seem to weigh more than the theoretical ability of seeking for most MPD users. If users feel this feature is needed, we will make a configuration option for that.
2009-01-17 11:43:13 +01:00
- oggvorbis: disable seeking on remote files
NEWS: merge from v0.14.x Copy changelog items from the upcoming 0.14.1 release.
2009-01-15 16:16:37 +01:00
* audio outputs:
- jack: allocate default port names (fixes a crash)
* update:
- refresh stats after update
- save the database even if it is empty
* input_curl:
- use select() to eliminate busy loop during connect
- honour http_proxy_* config directives
input_curl: call curl_multi_info_read() in constructor To check for early connect failures, call curl_multi_info_read() in the constructor input_curl_open(). This fixes an assertion failure.
2009-01-15 16:12:11 +01:00
- fix assertion failure on "connection refused"
input_curl: set "ready" flag on EOF Set the "ready" flag for empty resources.
2009-01-15 16:12:22 +01:00
- fix assertion failure with empty HTTP responses
pcm_resample_fallback: corrected the sample calculation Due to rounding errors, it was possible that the fallback resampler returned partial frames.
2009-01-16 18:52:01 +01:00
* corrected the sample calculation in the fallback resampler
NEWS: merge from v0.14.x Copy changelog items from the upcoming 0.14.1 release.
2009-01-15 16:16:37 +01:00
* log: automatically append newline
* fix setenv() conflict on Solaris
* configure.ac: check for pkg-config before using it
* fix minor memory leak in decoder_tag()
* fix cross-fading bug: it used to play some chunks of the new song twice
playlist: safely search the playlist for deleted song When a song file is deleted during database update, all pointers to it must be removed from the playlist. The "for" loop in deleteASongFromPlaylist() did not deal with multiple copies of the deleted song properly, and left instances of the (to-be-invalidated) pointer in. Fix this by reversing the loop.
2009-01-14 11:41:22 +01:00
* playlist
- fix assertion failure during playlist load
- implement Fisher-Yates shuffle properly
- safely search the playlist for deleted song
* use custom PRNG for volume dithering (speedup)
* detect libid3tag without pkg-config
MPD version 0.14
2008-12-25 22:18:46 +01:00
ver 0.14 (2008/12/25)
NEWS: reformatted 0.14 items Trying to bring a little bit of structure into that list.
2008-11-05 21:29:25 +01:00
* audio outputs:
NEWS: added new features
2008-12-02 19:15:41 +01:00
- wait 10 seconds before reopening a failed device
NEWS: reformatted 0.14 items Trying to bring a little bit of structure into that list.
2008-11-05 21:29:25 +01:00
- fifo: new plugin
- null: new plugin
- shout: block while trying to connect instead of failing
- shout: new timeout parameter
NEWS: updated 0.14 changelog Adding some of the most important changes since 0.13.
2008-11-05 21:30:05 +01:00
- shout: support mp3 encoding and the shoutcast protocol
- shout: send silence during pause, so clients don't get disconnected
NEWS: reformatted 0.14 items Trying to bring a little bit of structure into that list.
2008-11-05 21:29:25 +01:00
* decoders:
NEWS: updated 0.14 changelog Adding some of the most important changes since 0.13.
2008-11-05 21:30:05 +01:00
- ffmpeg: new plugin
NEWS: reformatted 0.14 items Trying to bring a little bit of structure into that list.
2008-11-05 21:29:25 +01:00
- wavpack: new plugin
NEWS: added new features
2008-12-02 19:15:41 +01:00
- aac: stream support added
configure.ac: disable the mikmod plugin by default libmikmod seems to be unmaintained, and has several critical bugs which make MPD crash. http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=461519 http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=476339 Disable the plugin by default to minimize harm. Users should explicitly enable the mikmod decoder (--enable-mod) if they wish to have it anyway.
2008-12-24 03:24:23 +01:00
- mod: disabled by default due to critical bugs in all libmikmod versions
NEWS: reformatted 0.14 items Trying to bring a little bit of structure into that list.
2008-11-05 21:29:25 +01:00
* commands:
- "addid" takes optional second argument to specify position
NEWS: updated 0.14 changelog Adding some of the most important changes since 0.13.
2008-11-05 21:30:05 +01:00
- "idle" notifies the client when a notable change occurs
ChangeLog entry for Bonjour support. git-svn-id: https://svn.musicpd.org/mpd/trunk@6469 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-06-03 21:27:24 +02:00
* Zeroconf support using Bonjour
Adding ChangeLog entry for zeroconf_enabled, adding Zeroconf section to mpdconf.example, and updating the zeroconf_* docs. git-svn-id: https://svn.musicpd.org/mpd/trunk@6474 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-06-03 22:09:12 +02:00
* New zeroconf_enabled option so that Zeroconf support can be disabled
Adding some ChangeLog entries. git-svn-id: https://svn.musicpd.org/mpd/trunk@6558 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-06-12 20:39:47 +02:00
* Stop the player/decode processes when not playing to allow the CPU to sleep
* Fix a bug where closing an ALSA dmix device could cause MPD to hang
ChangeLog: adding LAME tag ReplayGain git-svn-id: https://svn.musicpd.org/mpd/trunk@6810 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-08-25 19:02:41 +02:00
* Support for reading ReplayGain from LAME tags on MP3s
NEWS: updated 0.14 changelog Adding some of the most important changes since 0.13.
2008-11-05 21:30:05 +01:00
* MPD is now threaded, which greatly improves performance and stability
* memory usage reduced by merging duplicate tags in the database
* support connecting via unix domain socket
* allow authenticated local users to add any local file to the playlist
* 24 bit audio support
* optimized PCM conversions and dithering
* much code has been replaced by using GLib
* the HTTP client has been replaced with libcurl
NEWS: added new features
2008-12-02 19:15:41 +01:00
* symbolic links in the music directory can be disabled; the default
is to ignore symlinks pointing outside the music directory
Starting 0.14.0 ChangeLog entry. git-svn-id: https://svn.musicpd.org/mpd/trunk@6403 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-05-30 22:19:12 +02:00
Setting release date in ChangeLog. git-svn-id: https://svn.musicpd.org/mpd/trunk@6320 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-05-28 14:03:40 +02:00
ver 0.13.0 (2007/5/28)
Adding all of those missing ChangeLog entries. git-svn-id: https://svn.musicpd.org/mpd/trunk@5680 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-03-20 23:07:35 +01:00
* New JACK audio output
* Support for "file" as an alternative to "filename" in search, find, and list
Doc updates for 0.13.0 git-svn-id: https://svn.musicpd.org/mpd/trunk@5786 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-03-31 17:20:39 +02:00
* FLAC 1.1.3 API support
Adding all of those missing ChangeLog entries. git-svn-id: https://svn.musicpd.org/mpd/trunk@5680 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-03-20 23:07:35 +01:00
* New playlistadd command for adding to stored playlists
* New playlistclear command for clearing stored playlists
* Fix a bug where "find any" and "list <type> any" wouldn't return any results
* Make "list any" return an error instead of no results and an OK
* New gapless_mp3_playback option to disable gapless MP3 playback
* Support for seeking HTTP streams
* Zeroconf support using Avahi
* libsamplerate support for high quality audio resampling
* ID3v2 "Original Artist/Performer" tag support
* New playlistsearch command for searching the playlist (similar to "search")
* New playlistfind command for finding songs in the playlist (similar to "find")
* libmikmod 3.2.0 beta support
Adding ChangeLog entry for tagtypes command. git-svn-id: https://svn.musicpd.org/mpd/trunk@5797 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-03-31 23:06:39 +02:00
* New tagtypes command for retrieving a list of available tag types
Updating ChangeLog. git-svn-id: https://svn.musicpd.org/mpd/trunk@5892 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-04-08 17:52:41 +02:00
* Fix a bug where no ACK was returned if loading a playlist failed
Adding ChangeLog entry for previous commit. git-svn-id: https://svn.musicpd.org/mpd/trunk@5949 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-04-25 22:39:30 +02:00
* Fix a bug where db_update in stats would be 0 after initial database creation
Adding count command. It's usage is identical to find, but instead of returning a list of matching songs, the number of results and total play time of the results are returned. git-svn-id: https://svn.musicpd.org/mpd/trunk@5950 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-04-26 01:46:11 +02:00
* New count command for getting stats on found songs (similar to "find")
Committing pat's rewrite of the stored playlist code. This also adds two new commands: playlistmove and playlistdelete. git-svn-id: https://svn.musicpd.org/mpd/trunk@6116 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-05-16 14:02:10 +02:00
* New playlistmove command for moving songs in stored playlists
* New playlistdelete command for deleting songs from stored playlists
Adding rename command, for renaming stored playlists. git-svn-id: https://svn.musicpd.org/mpd/trunk@6246 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-05-24 20:07:19 +02:00
* New rename command for renaming stored playlists
Increasing default buffer_before_play from 0% to 10%. git-svn-id: https://svn.musicpd.org/mpd/trunk@6290 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-05-27 15:12:02 +02:00
* Increased default buffer_before_play from 0% to 10% to prevent skipping
Adding all of those missing ChangeLog entries. git-svn-id: https://svn.musicpd.org/mpd/trunk@5680 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-03-20 23:07:35 +01:00
* Lots of bug fixes, cleaned up code, and performance improvements
Importing ChangeLog entry from 0.12.2. git-svn-id: https://svn.musicpd.org/mpd/trunk@5679 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2007-03-20 22:43:19 +01:00
ver 0.12.2 (2007/3/20)
* Fix a bug where clients could cause MPD to segfault
Setting release date in ChangeLog. git-svn-id: https://svn.musicpd.org/mpd/trunk@4896 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-10-11 01:49:29 +02:00
ver 0.12.1 (2006/10/10)
Adding ChangeLog entries for 0.12.1 git-svn-id: https://svn.musicpd.org/mpd/trunk@4894 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-10-10 22:22:52 +02:00
* Fix segfault when scanning an MP3 that has a Xing tag with 0 frames
* Fix segfault when there's no audio output specified and one can't be detected
* Fix handling of escaping in quotes
* Allow a quality of -1 to be specified for shout outputs
* A few minor cleanups
Updating release date in ChangeLog. git-svn-id: https://svn.musicpd.org/mpd/trunk@4818 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-09-22 16:04:06 +02:00
ver 0.12.0 (2006/9/22)
Update ChangeLog. Please keep this up to date as you make changes from now on. git-svn-id: https://svn.musicpd.org/mpd/trunk@4710 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-08-28 20:12:18 +02:00
* New audio output code which supports:
* A plugin-like architecture
* Non-libao ("native") outputs:
* ALSA
* OSS
* OS X
* Media MVP
* PulseAudio
* Shout (Icecast or Shoutcast)
* Playing through multiple outputs at once
* Enabling/disabling outputs while MPD is running
* Saving output state (enabled/disabled) to the state_file
* OggFLAC support
* Musepack support
* Gapless MP3 playback
* MP3 ReplayGain support (using ID3v2 tags only)
* Support for MP2 files if MP3 support is enabled
* Composer, Performer, Comment, and Disc metadata support
* New outputs command for listing available audio outputs
* New enableoutput and disableoutput commands for enabling/disabling outputs
* New plchangesposid command for a stripped down version of plchanges
* New addid command for adding to the playlist and returning a song ID
* New commands and notcommands commands for checking available commands
Fixing up the new 0.12 ChangeLog entry some more. git-svn-id: https://svn.musicpd.org/mpd/trunk@4722 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-08-29 01:50:38 +02:00
* Can now specify any supported metadata type or "any" in search, find, and list
Update ChangeLog. Please keep this up to date as you make changes from now on. git-svn-id: https://svn.musicpd.org/mpd/trunk@4710 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-08-28 20:12:18 +02:00
* New volume_normalization parameter for enabling Audio Compress normalization
* New metadata_to_use parameter for choosing supported metadata types
* New pid_file parameter for saving the MPD process ID to the specified file
* The db_file parameter is now required
* The port parameter is now optional (defaults to 6600)
* Can specify bind_to_address multiple times
* New --kill argument for killing MPD if pid_file is specified
* Removed --update-db argument (use the update function in your client instead)
* New mpdconf.example
* New mpd.conf man page
* Removed bundled libmad and libid3tag
* Lots of bug fixes, cleaned up code, and performance improvements
"a" -> ) in ChangeLog git-svn-id: https://svn.musicpd.org/mpd/trunk@4479 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2006-07-27 23:37:22 +02:00
ver 0.11.5 (2004/11/1)
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
1) New id3v1_encoding config option to configure the id3v1 tag encoding (patch
from dottedmag)
update ChangeLog with 0.11.5 changes git-svn-id: https://svn.musicpd.org/mpd/trunk@2463 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-11-02 04:17:49 +01:00
2) Strip '\r' from m3u playlists (thank you windows)
3) Use random() instead of rand() for playlist randomizing
4) Fix a bug trying skipping some commented lines in m3u playlist files
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
5) Fix a bug when fetching metadata from streams that may cause certain
weirdnesses
update ChangeLog with 0.11.5 changes git-svn-id: https://svn.musicpd.org/mpd/trunk@2463 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-11-02 04:17:49 +01:00
6) Fix a bug where replaygain preamp was used on files w/o replaygain tags
7) Fix a busy loop when trying to prebuffer a nonexistant or missing stream
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
8) Fix a bug in forgetting to remove leading ' ' in content-type for http
streams
update ChangeLog with 0.11.5 changes git-svn-id: https://svn.musicpd.org/mpd/trunk@2463 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-11-02 04:17:49 +01:00
9) Check for ice-name in http headers
10) Be sure the strip all '\n' chars in tags
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
11) Set $HOME env variable when setuid'ing, this should fix the /root/.mcop
errors triggered by arts/libao
update ChangeLog with 0.11.5 changes git-svn-id: https://svn.musicpd.org/mpd/trunk@2463 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-11-02 04:17:49 +01:00
update changelog for 0.11.4 git-svn-id: https://svn.musicpd.org/mpd/trunk@1907 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-07-26 14:36:28 +02:00
ver 0.11.4 (2004/7/26)
1) Fixed a segfault when decoding mp3's with corrupt id3v2 tags
2) Fixed a memory leak when encountering id3v2 tags in mp3 decoder
1) bumb version to 0.11.3 2) Update changelog for 0.11.3 3) Merge avuton's mpdconf.example changes git-svn-id: https://svn.musicpd.org/mpd/trunk@1875 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-07-18 14:48:17 +02:00
ver 0.11.3 (2004/7/21)
1) Add support for http authentication for streams
2) Added replaygain pre-amp support
3) Better error handling for fread() in inputStream_file
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
4) Fixed a bug so that when a freeAllInterfaces is called, it sets
max_interface_connections to 0. This prevents potential segfaults and other
nastiness for forked processes, like the player and update-er (do to
interfacePrintWithFD()).
5) Allow blockingWrite() to handle errors more gracefully (for example, if the
disc is full, and thus the write() fails or can't be completed, we just skip
this write() and continue, instead of getting stuck in an infinite loop until
the write() becomes successful)
1) bumb version to 0.11.3 2) Update changelog for 0.11.3 3) Merge avuton's mpdconf.example changes git-svn-id: https://svn.musicpd.org/mpd/trunk@1875 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-07-18 14:48:17 +02:00
6) Updated mpdconf.example from sbh/avuton
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
7) If "user" is specified, then convert ~ in paths to the user's home path
specified by "user" config paramter (not the actual current user running mpd).
1) bumb version to 0.11.3 2) Update changelog for 0.11.3 3) Merge avuton's mpdconf.example changes git-svn-id: https://svn.musicpd.org/mpd/trunk@1875 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-07-18 14:48:17 +02:00
update changelog for 0.11.2 git-svn-id: https://svn.musicpd.org/mpd/trunk@1792 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-07-04 18:26:59 +02:00
ver 0.11.2 (2004/7/5)
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
1) Work around in computing total time for mp3's whose first valid mpeg frame is
not layer III
update changelog for 0.11.2 git-svn-id: https://svn.musicpd.org/mpd/trunk@1792 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-07-04 18:26:59 +02:00
2) Fix mp3 and mp4 decoders when seeking past the end of the file
3) Fix replaygain for flac and vorbis
4) Fix memory leaks in flac decoder (from normalperson)
5) Fix Several other bugs in playlist.c and directory.c (from normalperson)
Update doc's and Changelog for 0.11.1 git-svn-id: https://svn.musicpd.org/mpd/trunk@1626 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-23 13:52:25 +02:00
ver 0.11.1 (2004/6/24)
1) Fix a bug that caused "popping" at the beginning of mp3's
2) Fix playlistid command
3) Fix move commands so they don't mess up the song id's
4) Added support for HTTP Proxy
5) Detect and skip recursive links in the music directory
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
6) Fix addPathToDB() so updating on a specific path doesn't exist correctly adds
the parent directories to the DB
Update doc's and Changelog for 0.11.1 git-svn-id: https://svn.musicpd.org/mpd/trunk@1626 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-23 13:52:25 +02:00
bump release date back a day git-svn-id: https://svn.musicpd.org/mpd/trunk@1518 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-17 06:52:49 +02:00
ver 0.11.0 (2004/6/18)
updated changelog for 0.11.0 git-svn-id: https://svn.musicpd.org/mpd/trunk@1466 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-13 20:48:18 +02:00
1) Support for playing mp3 and Ogg Vorbis streams
2) Non-blocking Update
3) Replaygain support for Ogg Vorbis and FLAC (by Eric Moore aka AliasMrJones)
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
4) audio_output_format option that allows for all audio output to be converted
to a format compatible with any sound card
updated changelog for 0.11.0 git-svn-id: https://svn.musicpd.org/mpd/trunk@1466 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-13 20:48:18 +02:00
5) Own routines for to always support UTF-8 <-> ISO-8859-1 conversion
6) Added "Id" and "Pos" metadata for songs in playlist
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
7) Added commands: plchanges, currentsong, playid, seekid, playlistid, moveid,
swapid, deleteid
updated changelog for 0.11.0 git-svn-id: https://svn.musicpd.org/mpd/trunk@1466 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-13 20:48:18 +02:00
8) UTF-8 validation of all tags
9) Update specific files/directories (for fast, incremental updating)
10) Added ACK error codes
11) Mod file support
12) Added command_list_ok_begin
13) Play after stop resumes from last position in the playlist
14) Play while pause resumes playback
15) Better signal handling by mackstann
16) Cleanup decoder interface (now called InputPlugins)
17) --create-db no long starts the daemon
18) --no-daemon outputs to log files
19) --stdout sends output to stdout/stderr
add default port change to ChangeLog git-svn-id: https://svn.musicpd.org/mpd/trunk@1482 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-14 21:01:06 +02:00
20) Default port is now 6600
21) Lots of other cleanups and Bugfixes
updated changelog for 0.11.0 git-svn-id: https://svn.musicpd.org/mpd/trunk@1466 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-13 20:48:18 +02:00
update changelog with 0.10.4 changes git-svn-id: https://svn.musicpd.org/mpd/trunk@1465 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-13 20:35:44 +02:00
ver 0.10.4 (2004/5/26)
1) Fix configure problems on OpenBSD with langinfo and iconv
2) Fix an infinte loop when writing to an interface and it has expired
3) Fix a segfault in decoding flac's
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
4) Ingore CRC stuff in mp3's since some encoders did not compute the CRC
correctly
update changelog with 0.10.4 changes git-svn-id: https://svn.musicpd.org/mpd/trunk@1465 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-06-13 20:35:44 +02:00
5) Fix a segfault in processing faulty mp4 metadata
update stuff for 0.10.3 release git-svn-id: https://svn.musicpd.org/mpd/trunk@561 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-31 19:14:26 +02:00
ver 0.10.3 (2004/4/2)
1) Fix a segfault when a blanck line is sent from a client
2) Fix for loading playlists on platforms where char is unsigned
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
3) When pausing, release audio device after we say pause is successful (this
makes pause appear to not lag)
4) When returning errors for unknown types by player, be sure to copy the
filename
update stuff for 0.10.3 release git-svn-id: https://svn.musicpd.org/mpd/trunk@561 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-31 19:14:26 +02:00
5) add --disable-alsa for disabling alsa mixer support
update changelog git-svn-id: https://svn.musicpd.org/mpd/trunk@580 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-04-01 06:18:58 +02:00
6) Use select() for a portable usleep()
oops, forgot to update Changelog git-svn-id: https://svn.musicpd.org/mpd/trunk@584 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-04-02 00:12:44 +02:00
7) For alsa mixer, default to "Master' element, not first element
update stuff for 0.10.3 release git-svn-id: https://svn.musicpd.org/mpd/trunk@561 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-31 19:14:26 +02:00
update for 0.10.2 git-svn-id: https://svn.musicpd.org/mpd/trunk@407 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-22 23:21:02 +01:00
ver 0.10.2 (2004/3/25)
1) Add suport for AAC
fix some changelog grammar git-svn-id: https://svn.musicpd.org/mpd/trunk@455 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-24 16:55:50 +01:00
2) Substitute '\n' with ' ' in tag info
update for 0.10.2 git-svn-id: https://svn.musicpd.org/mpd/trunk@407 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-22 23:21:02 +01:00
3) Remove empty directories from db
4) Resume from current position in song when using state file
5) Pause now closes the music device, and reopens it on resuming
6) Fix unnecessary big endian byte swapping
7) If locale is "C" or "POSIX", then use ISO-8859-1 as the fs charset
fix some changelog grammar git-svn-id: https://svn.musicpd.org/mpd/trunk@455 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-24 16:55:50 +01:00
8) Fix a bug where alsa mixer wasn't detecting volume changes
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
9) For alsa and software mixer, show volume to be the same as it was set (even
if its not the exact volume)
git-svn-id: https://svn.musicpd.org/mpd/trunk@445 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-24 03:32:32 +01:00
10) Report bitrate for wave files
fix some changelog grammar git-svn-id: https://svn.musicpd.org/mpd/trunk@455 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-24 16:55:50 +01:00
11) Compute song length of CBR mp3's more accurately
update for 0.10.2 git-svn-id: https://svn.musicpd.org/mpd/trunk@407 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-22 23:21:02 +01:00
change release date for 0.10.1 git-svn-id: https://svn.musicpd.org/mpd/trunk@222 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-07 17:59:52 +01:00
ver 0.10.1 (2004/3/7)
git-svn-id: https://svn.musicpd.org/mpd/trunk@210 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-05 20:43:16 +01:00
1) Check to see if we need to add "-lm" when linking mpd
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
2) Fix issues with skipping bad frames in an mp3 (this way we get the correct
samplerate and such)
git-svn-id: https://svn.musicpd.org/mpd/trunk@210 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-05 20:43:16 +01:00
3) Fix crossfading bug with ogg's
forgot note that libmad and libid3tag were upgraded git-svn-id: https://svn.musicpd.org/mpd/trunk@214 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-06 17:35:53 +01:00
4) Updated libmad and libid3tag included w/ source to 0.15.1b
git-svn-id: https://svn.musicpd.org/mpd/trunk@210 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-05 20:43:16 +01:00
update with additional oss volume bug fix git-svn-id: https://svn.musicpd.org/mpd/trunk@130 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-01 14:49:38 +01:00
ver 0.10.0 (2004/3/3)
update changelog for 0.10.0 update add command info to include dir support update mpd.1 and conf example to include ~ support git-svn-id: https://svn.musicpd.org/mpd/trunk@64 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-26 04:36:04 +01:00
1) Use UTF-8 for all client communications
2) Crossfading support
3) Password Authentication (all in plaintext)
4) Software mixer
5) Buffer Size is configurable
6) Reduced Memory consumption (use directory tree for search and find)
7) Bitrate support for Flac
8) setvol command (deprecates volume command)
9) add command takes directories
10) Path's in config file now work with ~
update git-svn-id: https://svn.musicpd.org/mpd/trunk@110 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-28 00:57:44 +01:00
11) Add samplerate,bits, and channels to status
12) Reenable playTime in stats display
13) Fix a segfault when doing: add ""
14) Fix a segfault with flac vorbis comments simply being "="
15) Fix a segfault/bug in queueNextSong with repeat+random
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
16) Fix a bug, where one process may segfault, and cause more processes to spawn
w/o killing ones that lost their parent.
update with additional oss volume bug fix git-svn-id: https://svn.musicpd.org/mpd/trunk@130 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-01 14:49:38 +01:00
17) Fix a bug when the OSS device was unable to fetch the current volume,
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
it would close the device (when it maybe previously closed by the exact same
code)
git-svn-id: https://svn.musicpd.org/mpd/trunk@168 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-03-02 19:50:15 +01:00
18) command.c cleanup by mackstann
19) directory.c and command.c cleanup by tw-nym
update changelog for 0.10.0 update add command info to include dir support update mpd.1 and conf example to include ~ support git-svn-id: https://svn.musicpd.org/mpd/trunk@64 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-26 04:36:04 +01:00
import from SF CVS git-svn-id: https://svn.musicpd.org/mpd/trunk@1 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-24 00:41:20 +01:00
ver 0.9.4 (2004/1/21)
1) Fix a bug where updated tag info wasn't being detected
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
2) Set the default audio write size to 1024 bytes (should decrease cpu load a
bit on some machines).
import from SF CVS git-svn-id: https://svn.musicpd.org/mpd/trunk@1 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-24 00:41:20 +01:00
3) Make audio write size configurable via "audio_write_size" config option
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
4) Tweak output buffer size for connections by detecting the kernel output
buffer size.
import from SF CVS git-svn-id: https://svn.musicpd.org/mpd/trunk@1 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-24 00:41:20 +01:00
ver 0.9.3 (2003/10/31)
1) Store total time/length of songs in db and display in *info commands
2) Display instantaneous bitrate in status command
3) Add Wave Support using libaudiofile (Patch from normalperson)
4) Command code cleanup (Patch from tw-nym)
5) Optimize listing of playlists (10-100x faster)
6) Optimize interface output (write in 4kB chunks instead of on every '\n')
7) Fix bug that prevented rm command from working
8) Fix bug where deleting current song skips the next song
9) Use iconv to convert vorbis comments from UTF-8 to Latin1
ver 0.9.2 (2003/10/6)
1) Fix FreeBSD Compilation Problems
2) Fix bug in move command
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
3) Add mixer_control options to configure which mixer control/device mpd
controls
import from SF CVS git-svn-id: https://svn.musicpd.org/mpd/trunk@1 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-24 00:41:20 +01:00
4) Randomize on play -1
5) Fix a bug in toggling repeat off and at the end of the playlist
ver 0.9.1 (2003/9/30)
1) Fix a statement in the middle of declarations in listen.c, causes error for
gcc 2.7
ver 0.9.0 (2003/9/30)
1) Random play mode
2) Alsa Mixer Support
3) Save and Restore "state"
4) Default config file locations (.mpdconf and /etc/mpd.conf)
5) Make db file locations configurable
6) Move songs around in the playlist
7) Gapless playback
8) Use Xing tags for mp3's
9) Remove stop_on_error
10) Seeking support
11) Playlists can be loaded and deleted from subdirectories
12) Complete rewrite of player layer (fork()'s only once, opens and closes
audio device as needed).
13) Eliminate use and dependence of SIGIO
14) IPv6 support
15) Solaris compilations fixes
16) Support for different log levels
17) Timestamps for log entries
18) "user" config parameter for setuid (patch from Nagilum)
19) Other misc features and bug fixes
ver 0.8.7 (2003/9/3)
1) Fix a memory leak. When closing a interface, was called close() on the fd
instead of calling fclose() on the fp that was opened with fdopen().
ver 0.8.6 (2003/8/25)
Word wrap to 80 char lines
2008-12-21 04:27:02 +01:00
1) Fix a memory leak when a buffered existed, and a connection was unexpectedly
closed, and i wasn't free'ing the buffer apropriatly.
import from SF CVS git-svn-id: https://svn.musicpd.org/mpd/trunk@1 09075e82-0dd4-0310-85a5-a0d7c8717e4f
2004-02-24 00:41:20 +01:00
ver 0.8.5 (2003/8/17)
1) Fix a bug where an extra end of line is returned when attempting to play a
non existing file. This causes parsing errors for clients.
ver 0.8.4 (2003/8/13)
1) Fix a bug where garbage is returned with errors in "list" command
ver 0.8.3 (2003/8/12)
1) Fix a compilation error on older linux systems
2) Fix a bug in searching by title
3) Add "list" command
4) Add config options for specifying libao driver/plugin and options
5) Add config option to specify which address to bind to
6) Add support for loading and saving absolute pathnames in saved playlists
7) Playlist no longer creates duplicate entries for song data (more me
efficient)
8) Songs deleted from the db are now removed for the playlist as well
ver 0.8.2 (2003/7/22)
1) Increased the connection que for listen() from 0 to 5
2) Cleanup configure makefiles so that mpd uses MPD_LIBS and MPD_CFLAGS
rather than LIBS and CFLAGS
3) Put a cap on the number of commands per command list
4) Put a cap on the maximum number of buffered output lines
5) Get rid of TIME_WAIT/EADDRINUSE socket problem
6) Use asynchronious IO (i.e. trigger SIGIO instead so we can sleep in
select() calls longer)
ver 0.8.1 (2003/7/11)
1) FreeBSD fixes
2) Fix for rare segfault when updating
3) Fix bug where client was being hungup on when done playing current song
4) Fix bug when playing flac's where it incorrectly reports an error
5) Make stop playlist on error configurable
6) Configure checks for installed libmad and libid3tag and uses those if found
7) Use buffer->finished in *_decode's instead of depending on catching signals
ver 0.8.0 (2003/7/6)
1) Flac support
2) Make playlist max length configurable
3) New backward compatible status (backward compatible for 0.8.0 on)
4) listall command now can take a directory as an argument
5) Buffer rewritten to use shared memory instead of sockets
6) Playlist adding done using db
7) Add sort to list, and use binary search for finding
8) New "stats" command
9) Command list (for faster adding of large batches of files)
10) Add buffered chunks before play
11) Useful error reporting to clients (part of status command)
12) Use libid3tag for reading id3 tags (more stable)
13) Non-blocking output to clients
14) Fix bug when removing items from directory
15) Fix bug when playing mono mp3's
16) Fix bug when attempting to delete files when using samba
17) Lots of other bug fixes I can't remember
ver 0.7.0 (2003/6/20)
1) use mad instead of mpg123 for mp3 decoding
2) volume support
3) repeate playlist support
4) use autoconf/automake (i.e. "configure")
5) configurable max connections
ver 0.6.2 (2003/6/11)
1) Buffer support for ogg
2) new config file options: "connection_timeout" and "mpg123_ignore_junk"
3) new commands: "next", "previous", and "listall"
Thanks to Niklas Hofer for "next" and "previous" patches!
4) Search by filename
5) bug fix for pause when playing mp3's
ver 0.6.1 (2003/5/29)
1) Add conf file support
2) Fix a bug when doing mp3stop (do wait3(NULL,WNOHANG|WUNTRACED,NULL))
3) Fix a bug when fork'ing, fflush file buffers before forking so the
child doesn't print the same stuff in the buffer.
ver 0.6.0 (2003/5/25)
1) Add ogg vorbis support
2) Fix two bugs relating to tables, one for search by title, and one where we
freed the tables before directories, causing a segfault
3) The info command has been removed.
ver 0.5.0-0.5.2
Initial release(s). Support for MP3 via mpg123
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