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| | * decoder/faad: remove workaround for ancient libfaad2 ABI bugMax Kellermann2014-11-021-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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!
| | * decoder/ffmpeg: recognize MIME type audio/aacpSteven OBrien2014-11-021-0/+2
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| | * configure.ac: prepare for 0.18.17Max Kellermann2014-11-021-0/+2
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| * | Decoder, Playlist: ignore URI query string for plugin detectionMax Kellermann2014-11-011-0/+1
| | | | | | | | | | | | Use the new uri_get_suffix() overload that removes the query string.
| * | PlaylistFile: don't allow empty playlist nameMax Kellermann2014-10-311-0/+1
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| * | decoder/mad: fix negative replay gain valuesMax Kellermann2014-10-281-0/+1
| | | | | | | | | | | | Negating an unsigned integer does not work.
| * | TextInputStream: don't ignore unterminated last lineMax Kellermann2014-10-281-0/+1
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| * | playlist/m3u: recognize the file suffix ".m3u8"Max Kellermann2014-10-261-0/+2
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| * | OutputThread: fall back to PCM if given DSD sample rate is not supportedMax Kellermann2014-10-251-0/+1
| | | | | | | | | | | | | | | Works around the "PCM conversion from f to dsd is not implemented" error message that prevents DSD playback.
| * | decoder/faad: remove workaround for ancient libfaad2 ABI bugMax Kellermann2014-10-251-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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!
* | | AllCommands: "commands" returns playlist commands only if playlist_directory ↵Max Kellermann2014-10-251-0/+2
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* | | Set pulseaudio channel map to WAVE-EXSteven Newbury2014-10-251-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* | | configure.ac: prepare for 0.20Max Kellermann2014-10-251-0/+2
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* | decoder/ffmpeg: recognize MIME type audio/aacpSteven OBrien2014-10-251-0/+2
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* | configure.ac: auto-disable plugins that require GLib when --disable-glib is usedMax Kellermann2014-10-241-0/+1
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* | OutputThread: close the output plugin after filter failureMax Kellermann2014-10-241-0/+2
| | | | | | | | Fixes memory leak because ao_plugin_close() never gets called.
* | pcm/PcmConvert: assign {src,dest}_format at the endMax Kellermann2014-10-231-0/+1
| | | | | | | | | | Fixes assertion failure in destructor by not assigning {src,dest}_format when an error occurs.
* | configure.ac: prepare for 0.19.2Max Kellermann2014-10-231-0/+2
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* | release v0.19.1Max Kellermann2014-10-191-1/+1
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* | TextInputStream: manually shift the buffer before readingMax Kellermann2014-10-191-0/+1
| | | | | | | | | | Fixes truncated lines in m3u and cue files (regression by commit 67958f7).
* | playlist/extm3u: strip first line for #EXTM3U detectionMax Kellermann2014-10-171-0/+2
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* | Makefile.am: distribute systemd/mpd.socketMax Kellermann2014-10-121-0/+1
| | | | | | | | | | The file systemd/mpd.service.in is being distributed implicitly, but systemd/mpd.socket is not and needs to be added to EXTRA_DIST.
* | ThreadInputStream: call ThreadRead() inside the thread instead of Read()Max Kellermann2014-10-111-0/+2
| | | | | | | | Fixes deadlock bug in the "mms" plugin.
* | NEWS: add missing lineMax Kellermann2014-10-111-0/+1
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* | NEWS: fix typo in version numberMax Kellermann2014-10-111-1/+1
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* | configure.ac: prepare for 0.19.1Max Kellermann2014-10-111-0/+2
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* | release v0.19Max Kellermann2014-10-101-1/+1
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* | doc: document the mount/neighbor commandsMax Kellermann2014-10-101-1/+2
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* | TagString: remove ISO-Latin-1 fallbackMax Kellermann2014-10-101-0/+1
| | | | | | | | | | | | | | | | 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.
* | Main: run the OS X native event loopnanotech2014-10-101-0/+1
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* | Tag: support "AlbumSort"Max Kellermann2014-09-291-1/+1
| | | | | | | | | | The new tag is supported by all decoders that use the tag name table, and the ID3v2 tag "TSOA" maps to it.
* | Add MusicBrainz' Release Track Id tagWieland Hoffmann2014-09-271-0/+1
| | | | | | | | | | | | | | | | | | | | The Release Track Id uniquely identifies a recording on a release - that is, even if a recording appears twice on a release (meaning that the combination of recording and release id are not enough to figure out which one it is), the release track id will allow differentiating the two. The tag names are taken from https://musicbrainz.org/doc/MusicBrainz_Picard/Tags/Mapping
* | Merge tag 'v0.18.16'Max Kellermann2014-09-261-1/+4
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| * release v0.18.16Max Kellermann2014-09-261-1/+1
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| * configure.ac: fix DSD breakage due to typoMax Kellermann2014-09-261-0/+1
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| * configure.ac: prepare for 0.18.16Max Kellermann2014-09-261-0/+2
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| * release v0.18.15Max Kellermann2014-09-261-1/+1
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* | decoder/mpg123: support ID3v2, ReplayGain and MixRampMax Kellermann2014-09-241-0/+1
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* | Merge branch 'v0.18.x'Max Kellermann2014-09-241-0/+6
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| * OutputThread: retain negative mix ratioMax Kellermann2014-09-181-0/+1
| | | | | | | | Fixes MixRamp breakage.
| * command/list: reset used size after the list has been processedAndrzej Rybczak2014-09-181-0/+2
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| * thread/Posix{Cond,Mutex}: don't ues PTHREAD_*_INITIALIZER on NetBSDMax Kellermann2014-09-131-0/+1
| | | | | | | | | | | | | | | | | | | | 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.
| * configure.ac: prepare for 0.18.15Max Kellermann2014-09-131-0/+2
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* | decoder/sndfile: support float and 16 bit samplesMax Kellermann2014-09-191-0/+2
| | | | | | | | | | Support these PCM formats natively, instead of letting libsndfile convert everything to 32 bit.
* | Merge tag 'v0.18.14'Max Kellermann2014-09-111-0/+8
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| * release v0.18.14Max Kellermann2014-09-111-1/+1
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| * decoder/ffmpeg: pass MIME type to ffmpeg/libav version 11Max Kellermann2014-09-071-0/+1
| | | | | | | | | | | | | | That attribute was uninitialized before, which could crash libavformat. See Debian bug 760669
| * DecoderThread: clear the pipe when handling late SEEKMax Kellermann2014-09-071-0/+1
| | | | | | | | | | See code comment. Fixes assertion failure in decoder_command_finished().
| * decoder/audiofile: fix crash after seekingMax Kellermann2014-09-061-0/+2
| | | | | | | | | | | | Log call was added to the wrong branch. Fixes regression by commit ca1a1149
| * protocol/ArgParser: fix integer overflow in parse_range()Max Kellermann2014-09-041-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | 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.
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