Commit message (Collapse) | Author | Age | Files | Lines | |
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* | 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 | 8 | -226/+24 |
| | | | | | Delete the now-unused pcm_byteswap library, and optimize the pcm_export_state object. | ||||
* | output/{alsa,oss}: move endian code to new library pcm_export | Max Kellermann | 2012-03-21 | 5 | -61/+168 |
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* | pcm_pack: fix regression in unpack_sample() | Max Kellermann | 2012-03-21 | 1 | -1/+1 |
| | | | | Should have been "==", not "!=". | ||||
* | audio_format: DSD_OVER_USB is padded to 32 bit | Max Kellermann | 2012-03-21 | 1 | -3/+3 |
| | | | | | | For simplicity, pad the dCS samples to 32 bit. Packed 24 bit samples are rarely used. This patch does not include a real code change, because there is no user of DSD_OVER_USB yet. | ||||
* | audio_format: remove the reverse_endian attribute | Max Kellermann | 2012-03-21 | 14 | -112/+22 |
| | | | | | | Eliminate support for reverse endian samples from the MPD core. This moves a lot of complexity to the plugins that really need it (only ALSA and CDIO currently). | ||||
* | output/oss: always receive host byte order samples | Max Kellermann | 2012-03-21 | 1 | -7/+68 |
| | | | | Don't use audio_format.reverse_endian. | ||||
* | output/alsa: always receive host byte order samples | Max Kellermann | 2012-03-21 | 2 | -3/+62 |
| | | | | Don't use audio_format.reverse_endian. | ||||
* | decoder/pcm: always supply host byte order samples | Max Kellermann | 2012-03-21 | 1 | -15/+12 |
| | | | | Don't use audio_format.reverse_endian. | ||||
* | test: add unit test for util/byte_reverse.c | Max Kellermann | 2012-03-21 | 3 | -0/+94 |
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* | pcm_byteswap: move code to libutil | Max Kellermann | 2012-03-21 | 4 | -38/+201 |
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* | Makefile.am: link src/util/*.c into libutil.a | Max Kellermann | 2012-03-21 | 1 | -8/+14 |
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* | output/alsa: merge alsa_data_free() into destructor | Max Kellermann | 2012-03-21 | 1 | -8/+3 |
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* | audio_format: hack for DSD to USB conversion | Max Kellermann | 2012-03-21 | 1 | -0/+9 |
| | | | | Halve the sample rate for *:dsdusb:*. | ||||
* | test/run_convert: allow the out format to be a mask | Max Kellermann | 2012-03-21 | 1 | -2/+6 |
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* | Fix the build on OSX | Rich Healey | 2012-03-21 | 1 | -0/+1 |
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* | audio_format: remove the format SAMPLE_FORMAT_DSD_LSBFIRST | Max Kellermann | 2012-03-21 | 11 | -36/+4 |
| | | | | | This format is unused since the DSDIFF decoder plugin now reverses the bit order. | ||||
* | decoder/dsdiff: reverse bits to most significant bit first | Max Kellermann | 2012-03-21 | 2 | -6/+18 |
| | | | | Allow to remove this complexity from the MPD core. | ||||
* | dsd2pcm: move the bit reversing code to a generic library | Max Kellermann | 2012-03-21 | 4 | -8/+73 |
| | | | | Instead of doing run-time initialisation, use a constant lookup table. | ||||
* | audio_format: basic support for DSD-over-USB | Max Kellermann | 2012-03-19 | 11 | -0/+30 |
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* | playlist/soundcloud: support libyajl2 | Robert Vollmert | 2012-03-19 | 3 | -8/+27 |
| | | | | [mk: backwars compatibility and autoconf check] | ||||
* | 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). | ||||
* | test/dump_text_file: debug program for text_input_stream.c | Max Kellermann | 2012-03-19 | 2 | -0/+185 |
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* | test/run_input: use input_stream_*lock() | Max Kellermann | 2012-03-19 | 1 | -4/+4 |
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* | util/list: allow typeof() with clang | Max Kellermann | 2012-03-19 | 1 | -0/+5 |
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* | event_pipe, test: explicitly ignore write() return value | Max Kellermann | 2012-03-19 | 8 | -9/+12 |
| | | | | | Some compilers are very picky, but we really aren't interested in the return value. | ||||
* | command: read arbitrary local files with "lsinfo" | Max Kellermann | 2012-03-06 | 2 | -1/+27 |
| | | | | Requires UNIX domain socket connection. | ||||
* | client_file: always allow access if client uid equals mpd uid | Max Kellermann | 2012-03-06 | 1 | -0/+5 |
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* | playlist_edit: move UID check to client_allow_file() | Max Kellermann | 2012-03-06 | 6 | -39/+130 |
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* | command, ack: add ack_quark() | Max Kellermann | 2012-03-06 | 2 | -0/+16 |
| | | | | To pass ack values around. | ||||
* | use g_strerror() instead of strerror() | Max Kellermann | 2012-03-06 | 12 | -28/+29 |
| | | | | Make sure we get a UTF-8 encoded string. | ||||
* | command: fix the "DENIED" ACK code | Max Kellermann | 2012-03-06 | 1 | -1/+1 |
| | | | | Use ACK_ERROR_PERMISSION instead of ACK_ERROR_NO_EXIST. | ||||
* | configure.ac: detect libyajl for playlist/soundcloud | Max Kellermann | 2012-03-01 | 2 | -3/+10 |
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* | playlist/soundcloud: use config_dup_block_string() | Max Kellermann | 2012-03-01 | 1 | -5/+3 |
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* | A soundcloud playlist plugin. | Robert Vollmert | 2012-03-01 | 6 | -0/+471 |
| | | | | | | | | | | | | | | | | | | | | | | | Requires YAJL to build, and this doesn't include the necessary automake changes. Can be built using ./configure CFLAGS="-I/usr/include/yajl" LIBS="-lyajl" --enable-soundcloud Add the following to your config: playlist_plugin { name "soundcloud" enabled "true" apikey "c4c979fd6f241b5b30431d722af212e8" } Then you can stream from soundcloud using calls like: mpc load soundcloud://track/<track-id> mpc load soundcloud://playlist/<playlist-id> mpc load soundcloud://url/http://soundcloud.com/some/track/or/playlist For the last case, you can leave off the http:// or http://soundcloud.com/ . | ||||
* | song_update, udp_server: workarounds for gcc 4.1 warnings | Max Kellermann | 2012-03-01 | 2 | -0/+8 |
| | | | | Annoying false positives. | ||||
* | raop_output: fix raop_session inbalance | Kurt Van Dijck | 2012-03-01 | 1 | -2/+8 |
| | | | | | | | | raop_session_free must be called from raop_output_finish, not from raop_output_remove. In raop_output_remove, do close the ntp_server & control port. Signed-off-by: Kurt Van Dijck <kurt.van.dijck@skynet.be> | ||||
* | decoder/dsdiff: don't convert to PCM | Max Kellermann | 2012-03-01 | 2 | -39/+9 |
| | | | | | Move the responsibility for the conversion to the PCM library. This will allow passing the verbatim DSD samples to an output plugin. | ||||
* | pcm_convert: support the DSD format | Max Kellermann | 2012-03-01 | 5 | -0/+170 |
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* | audio_format: add DSD sample format | Max Kellermann | 2012-03-01 | 11 | -0/+61 |
| | | | | | Basic support for Direct Stream Digital. No conversion yet, and no decoder/output plugin support. | ||||
* | pcm_convert: add method _reset() | Max Kellermann | 2012-03-01 | 6 | -0/+39 |
| | | | | Resets the libsamplerate state. Not being used yet. | ||||
* | win32: Add a Windows OS resource file and icon | Avuton Olrich | 2012-02-23 | 4 | -1/+69 |
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* | Merge remote branches 'jn/ffmpeg' and 'jn/wsp' | Max Kellermann | 2012-02-15 | 1 | -2/+2 |
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| * | input/cdio_paranoia: whitespace-fix a comment | Jonathan Neuschäfer | 2012-02-15 | 1 | -2/+2 |
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* | | decoder/ffmpeg: always use AV_VERSION_INT | Jonathan Neuschäfer | 2012-02-15 | 1 | -1/+1 |
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* | doc/protocol: fix some grammar | Jonathan Neuschäfer | 2012-02-15 | 1 | -2/+2 |
| | | | | (Maybe it should be "A new message is indicated...", I don't know.) ( ^^ ) | ||||
* | Merge branch 'af' of git://git.musicpd.org/jn/mpd | Max Kellermann | 2012-02-15 | 1 | -4/+4 |
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| * | decoder/audiofile: fix compiler warnings with libaudiofile 0.3.3 | Jonathan Neuschäfer | 2012-02-15 | 1 | -4/+4 |
| | | | | | | | | This might break older versions, I didn't test. | ||||
* | | use audio_output_plugins_for_each's plugin iterator | Jonathan Neuschäfer | 2012-02-15 | 1 | -2/+2 |
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* | | rtsp_client: strncat -> g_strlcat | Jonathan Neuschäfer | 2012-02-15 | 1 | -7/+7 |
| | | | | | | | | | | | | | | | | | | The main difference is that strncat takes the maximum number of characters to copy as its third argument, while g_strlcat takes the size of the buffer, which is how the code was using strncat. Incomplete requests may still be constructed as a result of the reqest buffer filling up. |