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* | output/raop: delete the RAOP pluginMax Kellermann2012-05-299-2196/+0
| | | | | | | | | | | | | | | | | | This plugin is horrible code, I mean it. Last year, I tried hard to fix it, but I figured would take less time to do a full rewrite. Given that I don't even have any device that supports RAOP, I can't do that properly. After 16 months, nobody volunteered for fixing it. Hereby, I delete it, because having no RAOP plugin is better than having this mess. Sorry.
* | 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
* | tag_rva2: parse multiple ID3 "RVA2" tagsJonathan Dieter2012-04-231-2/+12
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* | tag_rva2: support separate album/track replay gainJonathan Dieter2012-04-231-4/+11
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* | tag_rva2: move code to rva2_apply_frame()Max Kellermann2012-04-231-16/+13
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* | tag_id3: export tag_id3_load()Max Kellermann2012-04-232-19/+41
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* | output/alsa: multiply writei() result with out_frame_sizeMax Kellermann2012-04-231-1/+3
| | | | | | | | | | | | .. and not in_frame_size, because this relates to the frame size being sent to ALSA. pcm_export_source_size() will then turn it back into the in_frame_size scale.
* | pcm_export: consider the pack24 flag in _source_size()Max Kellermann2012-04-231-0/+4
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* | pcm_export: add _frame_size()Max Kellermann2012-04-233-3/+30
| | | | | | | | Move code from the ALSA output plugin.
* | output/alsa: fix out_frame_size formula, multiply with channelsMax Kellermann2012-04-231-1/+3
| | | | | | | | | | The hard-coded "3 bytes" was wrong because it ignored the number of channels.
* | Merge branch 'v0.16.x'Max Kellermann2012-04-0512-68/+171
|\| | | | | | | | | | | Conflicts: src/output/osx_plugin.c src/text_input_stream.c
| * encoder/vorbis: generate end-of-stream packet when playback endsMax Kellermann2012-04-056-4/+42
| | | | | | | | | | Add the encoder_plugin method end(). This is important for the recorder plugin.
| * encoder_plugin: add state assertionsMax Kellermann2012-04-051-2/+61
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| * encoder/vorbis: generate end-of-stream packet before tagMax Kellermann2012-04-041-2/+0
| | | | | | | | | | Don't reset the ogg_stream_state object, because this discards the end-of-stream packet that was just added.
| * output/jack: check for connection failure before starting playbackMax Kellermann2012-04-041-0/+3
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| * output/jack: workaround for libjack1 crash bugMax Kellermann2012-04-041-0/+13
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| * directory: use strrchr() instead of g_basename()Max Kellermann2012-04-041-1/+9
| | | | | | | | g_basename() is deprecated in GLib 2.32.
| * uri: remove g_basename() call from uri_get_suffix()Max Kellermann2012-04-041-2/+2
| | | | | | | | | | g_basename() is deprecated in GLib 2.32. Instead, verify that the suffix does not have a backslash, to catch Windows path names.
| * update: properly skip symlinks in path that is to be updated.Anton Khirnov2012-04-041-1/+5
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| * output/osx: use the fifo_buffer library instead of rolling ownMax Kellermann2012-03-281-56/+37
| | | | | | | | | | | | | | | | 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.
| * Use g_message and not g_debug when removing songDan McGee2012-03-261-1/+1
| | | | | | | | | | | | | | | | | | When adding or updating a song, we get a log message even if debug is not enabled. It seems odd that removing a song shouldn't be done at the same log level; otherwise looking at the log leads you to believe songs are never removed from the library on update. Signed-off-by: Dan McGee <dan@archlinux.org>
| * event_pipe, test: explicitly ignore write() return valueMax Kellermann2012-03-191-1/+2
| | | | | | | | | | Some compilers are very picky, but we really aren't interested in the return value.
| * decoder/audiofile: fix compiler warnings with libaudiofile 0.3.3Jonathan Neuschäfer2012-03-191-4/+4
| | | | | | | | This might break older versions, I didn't test.
| * text_input_stream: detect end-of-fileMax Kellermann2012-03-191-2/+17
| | | | | | | | | | Fixes endless loop when the last line of a text file was not terminated (bug 3470).
* | Add support for DSD-over-USB version 1.0, remove pre-v1 supportJurgen Kramer2012-04-042-7/+31
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* | db_lock, archive/bz2, ...: workaround for G_STATIC_MUTEX_INIT warningMax Kellermann2012-04-042-0/+11
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* | input/curl: use g_source_get_time()Max Kellermann2012-04-042-12/+17
| | | | | | | | | | g_source_get_current_time() is deprecated since GLib 2.28. This patch adds a compatibility wrapper for older GLib versions to glib_compat.h.
* | audio_format: remove SAMPLE_FORMAT_DSD_OVER_USBMax Kellermann2012-03-2711-74/+1
| | | | | | | | | | | | | | 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.
* | output/alsa: support 32 bit DSD-over-USBMax Kellermann2012-03-271-4/+15
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* | pcm_export: implement 24 to 32 bit conversionMax Kellermann2012-03-274-4/+26
| | | | | | | | For 32 bit DSD-over-USB support.
* | output/alsa: use pcm_export for the DSD-over-USB conversionMax Kellermann2012-03-271-11/+10
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* | pcm_export: support DSD to DSD-over-USB conversionMax Kellermann2012-03-274-10/+74
| | | | | | | | Prepare for removing SAMPLE_FORMAT_DSD_OVER_USB.
* | output/alsa: move pcm_export_open() to callerMax Kellermann2012-03-271-11/+16
| | | | | | | | Give the caller more control, prepare for DSD-over-USB improvements.
* | pcm_export: support packing SAMPLE_FORMAT_DSD_OVER_USBMax Kellermann2012-03-271-1/+1
| | | | | | | | It's a padded 24 bit format.
* | pcm_export: initialize the "pack" bufferMax Kellermann2012-03-271-0/+2
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* | pcm_export: fix API documentationMax Kellermann2012-03-271-3/+3
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* | output/alsa: more debug outputMax Kellermann2012-03-271-0/+8
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* | Fix processing of sticker database pathDan McGee2012-03-261-2/+1
| | | | | | | | | | | | | | | | | | After a previous refactor, the current code fails on paths that need expansion (e.g, '~/.mpd/sticker.db'), because we are not passing the correct path to the sticker database code. Pass the expanded (and previously unused) string instead of the original string. Signed-off-by: Dan McGee <dan@archlinux.org>
* | output/alsa: add option to enable DSD over USBMax Kellermann2012-03-221-1/+54
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* | pcm_dsd: implement DSD to 24 bit USB conversionMax Kellermann2012-03-223-0/+150
| | | | | | | | | | | | Implements the dCS suggested standard: http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
* | playlist/soundcloud: libyajl2 uses size_t for string lengthsMax Kellermann2012-03-221-2/+14
| | | | | | | | Fixes build failure on 64 bit.
* | output/alsa: split the frame_size attributeMax Kellermann2012-03-221-6/+18
| | | | | | | | Make it in_frame_size and out_frame_size, to account for packing.
* | audio_format: remove the packed S24 formatMax Kellermann2012-03-2214-155/+4
| | | | | | | | | | | | 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.
* | output/alsa: use pcm_export to pack 24 bit samplesMax Kellermann2012-03-221-15/+48
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* | output/oss: use pcm_export to pack 24 bit samplesMax Kellermann2012-03-221-10/+15
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* | pcm_export: add option "pack"Max Kellermann2012-03-224-4/+39
| | | | | | | | | | Converts padded 24 bit samples to packed 24 bit samples. Will replace the packed S24 sample format, which is not used internally.
* | output/oss: remember the real OSS formatMax Kellermann2012-03-221-5/+13
| | | | | | | | | | Improving oss_reopen() by using the very same value that was used initially.
* | output/alsa: simplify setup_format()Max Kellermann2012-03-221-7/+4
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* | output/alsa: don't pass audio_format to _try_format()Max Kellermann2012-03-221-16/+13
| | | | | | | | Let the caller configure the audio_format object.
* | output/alsa: simplify alsa_output_try_format_both()Max Kellermann2012-03-221-45/+18
| | | | | | | | Merge three functions into one and call get_bitformat() only once.
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