| Commit message (Collapse) | Author | Age | Files | Lines | ||
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| * | | tag_rva2: parse multiple ID3 "RVA2" tags | Jonathan Dieter | 2012-04-23 | 1 | -2/+12 | |
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| * | | tag_rva2: support separate album/track replay gain | Jonathan Dieter | 2012-04-23 | 1 | -4/+11 | |
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| * | | tag_rva2: move code to rva2_apply_frame() | Max Kellermann | 2012-04-23 | 1 | -16/+13 | |
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| * | | tag_id3: export tag_id3_load() | Max Kellermann | 2012-04-23 | 2 | -19/+41 | |
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| * | | output/alsa: multiply writei() result with out_frame_size | Max Kellermann | 2012-04-23 | 1 | -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 Kellermann | 2012-04-23 | 1 | -0/+4 | |
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| * | | pcm_export: add _frame_size() | Max Kellermann | 2012-04-23 | 3 | -3/+30 | |
| | | | | | | | | | Move code from the ALSA output plugin. | |||||
| * | | output/alsa: fix out_frame_size formula, multiply with channels | Max Kellermann | 2012-04-23 | 1 | -1/+3 | |
| | | | | | | | | | | | The hard-coded "3 bytes" was wrong because it ignored the number of channels. | |||||
| * | | Merge branch 'v0.16.x' | Max Kellermann | 2012-04-05 | 12 | -68/+171 | |
| |\| | | | | | | | | | | | Conflicts: src/output/osx_plugin.c src/text_input_stream.c | |||||
| | * | encoder/vorbis: generate end-of-stream packet when playback ends | Max Kellermann | 2012-04-05 | 6 | -4/+42 | |
| | | | | | | | | | | | Add the encoder_plugin method end(). This is important for the recorder plugin. | |||||
| | * | encoder_plugin: add state assertions | Max Kellermann | 2012-04-05 | 1 | -2/+61 | |
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| | * | encoder/vorbis: generate end-of-stream packet before tag | Max Kellermann | 2012-04-04 | 1 | -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 playback | Max Kellermann | 2012-04-04 | 1 | -0/+3 | |
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| | * | output/jack: workaround for libjack1 crash bug | Max Kellermann | 2012-04-04 | 1 | -0/+13 | |
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| | * | directory: use strrchr() instead of g_basename() | Max Kellermann | 2012-04-04 | 1 | -1/+9 | |
| | | | | | | | | | g_basename() is deprecated in GLib 2.32. | |||||
| | * | uri: remove g_basename() call from uri_get_suffix() | Max Kellermann | 2012-04-04 | 1 | -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 Khirnov | 2012-04-04 | 1 | -1/+5 | |
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| | * | output/osx: use the fifo_buffer library instead of rolling own | Max Kellermann | 2012-03-28 | 1 | -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 song | Dan McGee | 2012-03-26 | 1 | -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 value | Max Kellermann | 2012-03-19 | 1 | -1/+2 | |
| | | | | | | | | | | | Some compilers are very picky, but we really aren't interested in the return value. | |||||
| | * | decoder/audiofile: fix compiler warnings with libaudiofile 0.3.3 | Jonathan Neuschäfer | 2012-03-19 | 1 | -4/+4 | |
| | | | | | | | | | This might break older versions, I didn't test. | |||||
| | * | text_input_stream: detect end-of-file | Max Kellermann | 2012-03-19 | 1 | -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 support | Jurgen Kramer | 2012-04-04 | 2 | -7/+31 | |
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| * | | db_lock, archive/bz2, ...: workaround for G_STATIC_MUTEX_INIT warning | Max Kellermann | 2012-04-04 | 2 | -0/+11 | |
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| * | | input/curl: use g_source_get_time() | Max Kellermann | 2012-04-04 | 2 | -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_USB | Max Kellermann | 2012-03-27 | 11 | -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-USB | Max Kellermann | 2012-03-27 | 1 | -4/+15 | |
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| * | | pcm_export: implement 24 to 32 bit conversion | Max Kellermann | 2012-03-27 | 4 | -4/+26 | |
| | | | | | | | | | For 32 bit DSD-over-USB support. | |||||
| * | | output/alsa: use pcm_export for the DSD-over-USB conversion | Max Kellermann | 2012-03-27 | 1 | -11/+10 | |
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| * | | pcm_export: support DSD to DSD-over-USB conversion | Max Kellermann | 2012-03-27 | 4 | -10/+74 | |
| | | | | | | | | | Prepare for removing SAMPLE_FORMAT_DSD_OVER_USB. | |||||
| * | | output/alsa: move pcm_export_open() to caller | Max Kellermann | 2012-03-27 | 1 | -11/+16 | |
| | | | | | | | | | Give the caller more control, prepare for DSD-over-USB improvements. | |||||
| * | | pcm_export: support packing SAMPLE_FORMAT_DSD_OVER_USB | Max Kellermann | 2012-03-27 | 1 | -1/+1 | |
| | | | | | | | | | It's a padded 24 bit format. | |||||
| * | | pcm_export: initialize the "pack" buffer | Max Kellermann | 2012-03-27 | 1 | -0/+2 | |
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| * | | pcm_export: fix API documentation | Max Kellermann | 2012-03-27 | 1 | -3/+3 | |
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| * | | output/alsa: more debug output | Max Kellermann | 2012-03-27 | 1 | -0/+8 | |
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| * | | Fix processing of sticker database path | Dan McGee | 2012-03-26 | 1 | -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 USB | Max Kellermann | 2012-03-22 | 1 | -1/+54 | |
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| * | | pcm_dsd: implement DSD to 24 bit USB conversion | Max Kellermann | 2012-03-22 | 3 | -0/+150 | |
| | | | | | | | | | | | | | Implements the dCS suggested standard: http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf | |||||
| * | | playlist/soundcloud: libyajl2 uses size_t for string lengths | Max Kellermann | 2012-03-22 | 1 | -2/+14 | |
| | | | | | | | | | Fixes build failure on 64 bit. | |||||
| * | | output/alsa: split the frame_size attribute | Max Kellermann | 2012-03-22 | 1 | -6/+18 | |
| | | | | | | | | | Make it in_frame_size and out_frame_size, to account for packing. | |||||
| * | | audio_format: remove the packed S24 format | Max Kellermann | 2012-03-22 | 14 | -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 samples | Max Kellermann | 2012-03-22 | 1 | -15/+48 | |
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| * | | output/oss: use pcm_export to pack 24 bit samples | Max Kellermann | 2012-03-22 | 1 | -10/+15 | |
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| * | | pcm_export: add option "pack" | Max Kellermann | 2012-03-22 | 4 | -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 format | Max Kellermann | 2012-03-22 | 1 | -5/+13 | |
| | | | | | | | | | | | Improving oss_reopen() by using the very same value that was used initially. | |||||
| * | | output/alsa: simplify setup_format() | Max Kellermann | 2012-03-22 | 1 | -7/+4 | |
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| * | | output/alsa: don't pass audio_format to _try_format() | Max Kellermann | 2012-03-22 | 1 | -16/+13 | |
| | | | | | | | | | Let the caller configure the audio_format object. | |||||
| * | | output/alsa: simplify alsa_output_try_format_both() | Max Kellermann | 2012-03-22 | 1 | -45/+18 | |
| | | | | | | | | | Merge three functions into one and call get_bitformat() only once. | |||||
| * | | output/oss: move code to oss_probe_sample_format() | Max Kellermann | 2012-03-21 | 1 | -34/+59 | |
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| * | | pcm_export: use the byte_reverse library directly | Max Kellermann | 2012-03-21 | 4 | -152/+24 | |
| | | | | | | | | | | | Delete the now-unused pcm_byteswap library, and optimize the pcm_export_state object. | |||||
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index : mpd | |
| clone of git://git.musicpd.org/master/mpd.git | Alexander Sulfrian |
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