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| * cmdline: change --version formattingMax Kellermann2012-06-121-5/+5
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| * decoder_list: add _for_each() macrosMax Kellermann2012-06-124-16/+17
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| * archive_list: add _for_each() macrosMax Kellermann2012-06-122-25/+25
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| * cmdline: don't initalise archive plugins for --versionMax Kellermann2012-06-122-3/+0
| | | | | | | | Dump all archive plugins, even those that fail to initialise.
| * encoder_list: add macro _for_each()Max Kellermann2012-06-122-6/+14
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| * encoder_list: make the list truly "const"Max Kellermann2012-06-121-1/+1
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| * output_list: declare variables inside _plugins_for_each()Max Kellermann2012-06-123-14/+7
| | | | | | | | Don't require the caller to provide them.
| * output_list: make the list truly "const"Max Kellermann2012-06-122-2/+2
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| * output_list: simplify audio_output_plugins_for_each() callMax Kellermann2012-06-121-2/+2
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| * cmdline: update copyright yearMax Kellermann2012-06-121-1/+1
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| * Work around incorrect g_file_test() behavior on Win32Denis Krjuchkov2012-06-125-0/+32
| | | | | | | | | | | | | | 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.
| * audio-parser, output_thread: work around -Wmaybe-uninitializedMax Kellermann2012-06-122-0/+16
| | | | | | | | False positives in gcc 4.7.
| * gcc.h: backport GCC_CHECK_VERSION() from v0.17.xMax Kellermann2012-06-121-1/+6
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* | Merge branch 'v0.16.x'Max Kellermann2012-05-292-2/+8
|\| | | | | | | | | Conflicts: NEWS
| * decoder_api: log the decoder plugin nameMax Kellermann2012-05-291-0/+4
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| * 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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* | 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.
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