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* | decoder/flac: eliminate the obsolete "track number" codeMax Kellermann2012-10-024-41/+18
| | | | | | | | This has been deprecated by the "embcue" playlist plugin.
* | decoder/flac: remove unused function flac_tag_load()Max Kellermann2012-10-022-17/+0
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* | decoder/flac: use C++ compilerMax Kellermann2012-10-028-60/+109
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* | {decoder,encoder}/flac: drop support for libFLAC 1.1Max Kellermann2012-10-022-168/+5
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* | decoder/adplug: new decoder pluginMax Kellermann2012-09-252-0/+173
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* | decoder/vorbis: skip 16 bit quantisation, provide float samplesSimon Hosie2012-09-251-0/+44
| | | | | | | | | | | | | | | | | | | | 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.
* | decoder/vorbis: rename local variablesMax Kellermann2012-09-251-14/+13
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* | decoder/vorbis: improved support for initial seekMax Kellermann2012-09-251-1/+1
| | | | | | | | Call decoder_get_command() before doing anything else.
* | decoder/vorbis: make variables more localMax Kellermann2012-09-251-26/+20
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* | Merge branch 'v0.17.x'Max Kellermann2012-09-253-22/+16
|\| | | | | | | | | Conflicts: src/locate.c
| * decoder_control: remove MixRamp debug messagesMax Kellermann2012-09-252-6/+4
| | | | | | | | | | These are confusing, and since MixRamp development has ceased, not useful to anybody.
| * decoder/wavpack: support all APEv2 tagsMax Kellermann2012-09-251-16/+12
| | | | | | | | | | WavPack tags are always APEv2, by definition. Reuse the tag_table from tag_ape.c, instead of rolling our own.
* | src/decoder/opus: new decoder plugin for the Opus codecMax Kellermann2012-09-0511-0/+778
| | | | | | | | Using libopus and libogg.
* | decoder/{flac,vorbis}: move tag table to XiphTags.cMax Kellermann2012-09-054-16/+60
| | | | | | | | Merge duplicate data.
* | decoder/ogg_codec: return UNKNOWN on errorMax Kellermann2012-09-042-1/+2
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* | decoder/ogg_common: rename to ogg_codec.cMax Kellermann2012-09-044-19/+19
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* | decoder/ogg_common: pass decoder to _type_detect()Max Kellermann2012-09-044-5/+5
| | | | | | | | Allow the function to be cancelled.
* | decoder/ogg_common: apply coding styleMax Kellermann2012-09-042-6/+9
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* | decoder/_ogg_common: rename to ogg_common.cMax Kellermann2012-09-046-5/+5
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* | Merge branch 'v0.17.x'Max Kellermann2012-09-041-6/+8
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| * decoder/_ogg_common: fix buffer size checkMax Kellermann2012-09-041-1/+1
| | | | | | | | Fixes potential access to uninitialised memory.
| * decoder/_ogg_common: simplify the large "if" expressionMax Kellermann2012-09-041-6/+8
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* | decoder/mad, output_thread: add gcc_unlikely()Max Kellermann2012-08-291-2/+2
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* | Merge branch 'v0.17.x'Max Kellermann2012-08-151-53/+28
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| * decoder/fluidsynth: add "sample_rate" settingMax Kellermann2012-08-151-6/+14
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| * decoder/fluidsynth: add "soundfont" settingMax Kellermann2012-08-151-6/+7
| | | | | | | | Replaces the old global "soundfont" which never worked.
| * configure.ac: auto-detect libfluidsynthMax Kellermann2012-08-151-9/+0
| | | | | | | | | | Now that the libfluidsynth API was sanitized, we can enable the plugin automatically if libfluidsynth is installed.
| * decoder/fluidsynth: stop playback at end of fileMax Kellermann2012-08-151-5/+4
| | | | | | | | Use libfluidsynth's new function fluid_player_get_status().
| * decoder/fluidsynth: don't duplicate pathMax Kellermann2012-08-151-6/+1
| | | | | | | | The libfluidsynth now accepts const strings.
| * decoder/fluidsynth: check if file is really a MIDIMax Kellermann2012-08-151-4/+1
| | | | | | | | Use fluid_is_midifile() to verify the file format.
| * decoder/fluidsynth: remove throttle (requires libfluidsynth 1.1)Max Kellermann2012-08-151-17/+1
| | | | | | | | | | The libfluidsynth API is now sane, and does not require real-time decoding.
* | gcc.h: re-add gcc_const and gcc_pureMax Kellermann2012-08-021-2/+0
| | | | | | | | Remove GLib dependency from some headers.
* | decoder/sidplay: fix C++ compiler warningsMax Kellermann2012-08-011-4/+4
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* Add song duration to DSF and DSDIFF DSD decoders.Jurgen Kramer2012-07-132-3/+24
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* patch to split DSD decoder into separate decoders for DSF en DFF. Move commonJurgen Kramer2012-06-276-323/+565
| | | | functions to new dsdlib. Update user doc.
* Merge branch 'v0.16.x'Max Kellermann2012-06-121-0/+1
|\ | | | | | | | | | | | | | | | | | | | | | | Conflicts: src/cmdline.c src/decoder/wildmidi_decoder_plugin.c src/gcc.h src/glib_compat.h src/input_stream.c src/output_list.c src/output_thread.c valgrind.suppressions
| * Work around incorrect g_file_test() behavior on Win32Denis Krjuchkov2012-06-121-0/+1
| | | | | | | | | | | | | | 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.
* | Merge branch 'v0.16.x'Max Kellermann2012-05-291-2/+4
|\| | | | | | | | | Conflicts: NEWS
| * decoder/ffmpeg: improve "decoding failed" messageJonathan Neuschäfer2012-05-291-1/+1
| | | | | | | | | | "Frame skipped" might cause the impression that the decoding of a whole song failed.
| * decoder/ffmpeg: add webm as a supported formatJonathan Neuschäfer2012-05-291-1/+3
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| * decoder/audiofile: fix compiler warnings with libaudiofile 0.3.3Jonathan Neuschäfer2012-03-191-4/+4
| | | | | | | | This might break older versions, I didn't test.
* | Add support for DSF files to DSDIFF decoder - v4Jurgen Kramer2012-05-021-36/+229
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Version 4 of my patch to add DSF support to the DSDIFF decoder plugin. This time I have taken a different approach and created a new read_metadata function specific for reading DSF files. This saves an indent (and for me a lot of indent nightmares) and also useful for splitting the DSF and DFF decoders later on. There are still a few lines which exceed the 80 character width limit by a few chars. I was not able to stay within the limit and create (for me) readable code. Jurgen
* | audio_format: remove SAMPLE_FORMAT_DSD_OVER_USBMax Kellermann2012-03-271-1/+0
| | | | | | | | | | | | | | DSD-over-USB should not be a MPD core format, because it is not a "natural" format; it is just a temnporary over-the-wire format. This format has been implemented in pcm_export, and does not need to be supported by pcm_convert.
* | audio_format: remove the packed S24 formatMax Kellermann2012-03-221-1/+0
| | | | | | | | | | | | For simplicity, the MPD core should not have to deal with packing. It is rarely used, and those plugins that need it should use the pcm_export library instead.
* | decoder/pcm: always supply host byte order samplesMax Kellermann2012-03-211-15/+12
| | | | | | | | Don't use audio_format.reverse_endian.
* | audio_format: remove the format SAMPLE_FORMAT_DSD_LSBFIRSTMax Kellermann2012-03-211-1/+0
| | | | | | | | | | This format is unused since the DSDIFF decoder plugin now reverses the bit order.
* | decoder/dsdiff: reverse bits to most significant bit firstMax Kellermann2012-03-211-6/+15
| | | | | | | | Allow to remove this complexity from the MPD core.
* | audio_format: basic support for DSD-over-USBMax Kellermann2012-03-191-0/+1
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* | decoder/dsdiff: don't convert to PCMMax Kellermann2012-03-011-38/+9
| | | | | | | | | | Move the responsibility for the conversion to the PCM library. This will allow passing the verbatim DSD samples to an output plugin.
* | audio_format: add DSD sample formatMax Kellermann2012-03-011-0/+2
| | | | | | | | | | Basic support for Direct Stream Digital. No conversion yet, and no decoder/output plugin support.
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