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* Proper error reporting from filter_configAlbin Eldstål-Damlin2009-12-143-21/+45
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* Fixed memory leak on incorrect route configurationAlbin Eldstål-Damlin2009-12-141-0/+4
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* Split filter_config into its own moduleAlbin Eldstål-Damlin2009-12-146-109/+176
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* test/run_{decoder,filter}: implemented GLib log callbackMax Kellermann2009-12-142-0/+25
| | | | Log to stderr, not to stdout (which broke PCM output).
* Error reporting, pcm_buffer, performance tweaksAlbin Eldstål-Damlin2009-12-141-52/+48
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* Initial filter chain and filter configuration for outputs.Albin Eldstål-Damlin2009-12-144-0/+118
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* Initial (statically configured) route filter pluginAlbin Eldstål-Damlin2009-12-144-0/+348
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* Minor documentation fixAlbin Eldstål-Damlin2009-12-141-2/+2
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* Merge branch 'v0.15.x'Max Kellermann2009-12-149-17/+53
|\ | | | | | | | | Conflicts: src/decoder/ffmpeg_plugin.c
| * decoder/wavpack: don't use the nonstandard "uchar" typeMax Kellermann2009-12-111-1/+1
| | | | | | | | Use the signed C99 type int8_t instead.
| * mixer: explicitly close all mixers on shutdownMax Kellermann2009-12-082-0/+5
| | | | | | | | | | | | Mixers with the "global" flag set aren't closed automatically when the output device is closed. Thus, they might still be open when MPD shuts down.
| * mapper: apply filesystem_charset to playlistsMax Kellermann2009-12-085-11/+37
| | | | | | | | | | | | | | | | This fixes an inconsistency in the stored playlist subsystem: when obtaining the list of playlists (listplaylist, listplaylistinfo), the file names in the playlist directory are converted to UTF-8 (according to filesystem_charset), but when saving or loading playlists, the filesystem_charset setting was ignored.
| * command: verify playlist name in the "rm" commandMax Kellermann2009-12-082-0/+4
| | | | | | | | Call spl_valid_name() in spl_delete().
| * mapper: fix memory leak when playlist_directory is not setMax Kellermann2009-12-082-2/+3
| | | | | | | | Don't allocate the file name before the playlist_dir==NULL check.
| * tag_id3: fix ID3v1 charset conversionsvitoos2009-11-302-1/+3
| | | | | | | | | | If we define id3v1_encoding, then the tags are not added to the database.
| * ffmpeg: don't try to force stereoMax Kellermann2009-11-302-4/+2
| | | | | | | | | | | | The plugin code tried to force libavcodec to supply stereo samples. That however has never actually worked. By removing this code, we are able to play surround files for the first time.
* | decoder_api: prefer stream_tag over decoder_tagThomas Jansen2009-12-071-2/+2
| | | | | | | | | | | | If both tags (stream and decoder) are present, we prefer the stream tag. Fixes #2698, where ICY tag contained useful information, but was overwritten with bogus decoder tag data.
* | pcm_volume: change old code to use format instead of bitsViliam Mateicka2009-12-031-1/+1
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* | encoders: remove unnessesary pointers to const stringsViliam Mateicka2009-12-035-15/+5
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* | httpd: use get_mime_type to determine encoder contentViliam Mateicka2009-12-031-7/+6
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* | encoders: implement new get_mime_types methodViliam Mateicka2009-12-035-0/+45
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* | encoder: add get_mime_type() method to determine content type by httpd ↵Viliam Mateicka2009-12-031-0/+17
| | | | | | | | output plugin
* | pcm_mix: change old code to use format instead of bitsViliam Mateicka2009-12-031-1/+1
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* | null_encoder: use pcm_bufferViliam Mateicka2009-12-031-10/+18
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* | flac_encoder: add support for libFLAC < 1.1.3Viliam Mateicka2009-12-031-18/+47
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* | compress: add config.hJeffrey Middleton2009-12-021-0/+19
| | | | | | | | | | This includes some default values of #defined constants used in the code; it won't compile without it.
* | audio_format: changed "bits" to "enum sample_format"Max Kellermann2009-12-0250-214/+511
| | | | | | | | | | | | This patch prepares support for floating point samples (and probably other formats). It changes the meaning of the "bits" attribute from a bit count to a symbolic value.
* | test: added normalize test programMax Kellermann2009-12-023-0/+89
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* | compress: upgraded to AudioCompress 2.0J. Shagam2009-12-028-467/+236
| | | | | | | | | | | | | | | | 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]
* | decoder/mpcdec: set 24 bit sample formatMax Kellermann2009-11-251-1/+1
| | | | | | | | | | This fixes a regression due to a typo caused by "decoder: use audio_format_init_checked()".
* | pcm_mix: implemented 32 bit supportMax Kellermann2009-11-192-1/+24
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* | pcm_volume: implemented 32 bit supportMax Kellermann2009-11-193-0/+43
| | | | | | | | Support 32 bit samples with software mixer.
* | test: added program to test pcm_convert.cMax Kellermann2009-11-193-0/+107
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* | test/software_volume: check for errorsMax Kellermann2009-11-191-1/+6
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* | test/software_volume: fixed audio_format parserMax Kellermann2009-11-191-3/+3
| | | | | | | | Assign default value only if none was given on the command line.
* | Merged release 0.15.6 from branch 'v0.15.x'Max Kellermann2009-11-194-10/+36
|\| | | | | | | | | | | | | Conflicts: NEWS configure.ac
| * decoder/flac: fixed compiler warningMax Kellermann2009-11-191-3/+1
| | | | | | | | | | | | Removed the "vtrack" local variable (which triggered a gcc warning because it was after the newly introduced NULL check), and run strtol() on the original parameter.
| * Modify version string to post-release version 0.15.7~gitAvuton Olrich2009-11-182-1/+4
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| * mpd version 0.15.6release-0.15.6Avuton Olrich2009-11-182-2/+2
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| * decoder/flac: fixed NULL pointer dereference in CUE codeMax Kellermann2009-11-182-0/+3
| | | | | | | | The function flac_vtrack_tnum() was missing a strrchr()==NULL check.
| * id3: allow 4 MB RIFF/AIFF tagsMax Kellermann2009-11-152-1/+3
| | | | | | | | | | | | | | Allow RIFF/AIFF ID3 tags up to 4 MB (old limit was 256 kB). This might still be too small for some users, and when somebody complains, we might do something more clever (like streaming the data into libid3tag?).
| * decoder/ffmpeg: align the output bufferMax Kellermann2009-11-152-5/+25
| | | | | | | | | | | | 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.
* | cmdline: print out list of encoders in --version infoViliam Mateicka2009-11-174-0/+30
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* | encoder: let wave encoder to use pcm_buffer, pcm conversion code cleanupViliam Mateicka2009-11-171-29/+27
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* | encoder: introducing flac encoder pluginViliam Mateicka2009-11-175-0/+324
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* | output/openal: use audio_format_to_string()Max Kellermann2009-11-151-3/+3
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* | crossfade: use audio_format_valid() in assertionMax Kellermann2009-11-151-3/+1
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* | valgrind.suppressions: added entry for g_main_context_default()Max Kellermann2009-11-141-0/+11
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* | decoder/audio: eliminate the "bits" variableMax Kellermann2009-11-141-4/+1
| | | | | | | | | | Pass the audiofile_setup_sample_format() result to audio_format_init_checked().
* | decoder/audiofile: moved code to audiofile_setup_sample_format()Max Kellermann2009-11-141-10/+20
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