| Commit message (Collapse) | Author | Files | Lines | ||
|---|---|---|---|---|---|
| 2012-03-22 | output/oss: remember the real OSS format | Max Kellermann | 1 | -5/+13 | |
| Improving oss_reopen() by using the very same value that was used initially. | |||||
| 2012-03-22 | output/alsa: simplify setup_format() | Max Kellermann | 1 | -7/+4 | |
| 2012-03-22 | output/alsa: don't pass audio_format to _try_format() | Max Kellermann | 1 | -16/+13 | |
| Let the caller configure the audio_format object. | |||||
| 2012-03-22 | output/alsa: simplify alsa_output_try_format_both() | Max Kellermann | 1 | -45/+18 | |
| Merge three functions into one and call get_bitformat() only once. | |||||
| 2012-03-21 | output/oss: move code to oss_probe_sample_format() | Max Kellermann | 1 | -34/+59 | |
| 2012-03-21 | pcm_export: use the byte_reverse library directly | Max Kellermann | 4 | -152/+24 | |
| Delete the now-unused pcm_byteswap library, and optimize the pcm_export_state object. | |||||
| 2012-03-21 | output/{alsa,oss}: move endian code to new library pcm_export | Max Kellermann | 4 | -61/+167 | |
| 2012-03-21 | pcm_pack: fix regression in unpack_sample() | Max Kellermann | 1 | -1/+1 | |
| Should have been "==", not "!=". | |||||
| 2012-03-21 | audio_format: DSD_OVER_USB is padded to 32 bit | Max Kellermann | 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. | |||||
| 2012-03-21 | audio_format: remove the reverse_endian attribute | Max Kellermann | 13 | -110/+20 | |
| 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). | |||||
| 2012-03-21 | output/oss: always receive host byte order samples | Max Kellermann | 1 | -7/+68 | |
| Don't use audio_format.reverse_endian. | |||||
| 2012-03-21 | output/alsa: always receive host byte order samples | Max Kellermann | 1 | -3/+61 | |
| Don't use audio_format.reverse_endian. | |||||
| 2012-03-21 | decoder/pcm: always supply host byte order samples | Max Kellermann | 1 | -15/+12 | |
| Don't use audio_format.reverse_endian. | |||||
| 2012-03-21 | pcm_byteswap: move code to libutil | Max Kellermann | 3 | -38/+199 | |
| 2012-03-21 | output/alsa: merge alsa_data_free() into destructor | Max Kellermann | 1 | -8/+3 | |
| 2012-03-21 | audio_format: hack for DSD to USB conversion | Max Kellermann | 1 | -0/+9 | |
| Halve the sample rate for *:dsdusb:*. | |||||
| 2012-03-21 | Fix the build on OSX | Rich Healey | 1 | -0/+1 | |
| 2012-03-21 | audio_format: remove the format SAMPLE_FORMAT_DSD_LSBFIRST | Max Kellermann | 11 | -36/+4 | |
| This format is unused since the DSDIFF decoder plugin now reverses the bit order. | |||||
| 2012-03-21 | decoder/dsdiff: reverse bits to most significant bit first | Max Kellermann | 1 | -6/+15 | |
| Allow to remove this complexity from the MPD core. | |||||
| 2012-03-21 | dsd2pcm: move the bit reversing code to a generic library | Max Kellermann | 3 | -8/+70 | |
| Instead of doing run-time initialisation, use a constant lookup table. | |||||
| 2012-03-19 | audio_format: basic support for DSD-over-USB | Max Kellermann | 11 | -0/+30 | |
| 2012-03-19 | playlist/soundcloud: support libyajl2 | Robert Vollmert | 1 | -5/+21 | |
| [mk: backwars compatibility and autoconf check] | |||||
| 2012-03-19 | text_input_stream: detect end-of-file | Max Kellermann | 1 | -2/+17 | |
| Fixes endless loop when the last line of a text file was not terminated (bug 3470). | |||||
| 2012-03-19 | util/list: allow typeof() with clang | Max Kellermann | 1 | -0/+5 | |
| 2012-03-19 | event_pipe, test: explicitly ignore write() return value | Max Kellermann | 1 | -1/+2 | |
| Some compilers are very picky, but we really aren't interested in the return value. | |||||
| 2012-03-06 | command: read arbitrary local files with "lsinfo" | Max Kellermann | 1 | -1/+22 | |
| Requires UNIX domain socket connection. | |||||
| 2012-03-06 | client_file: always allow access if client uid equals mpd uid | Max Kellermann | 1 | -0/+5 | |
| 2012-03-06 | playlist_edit: move UID check to client_allow_file() | Max Kellermann | 5 | -39/+129 | |
| 2012-03-06 | command, ack: add ack_quark() | Max Kellermann | 2 | -0/+16 | |
| To pass ack values around. | |||||
| 2012-03-06 | use g_strerror() instead of strerror() | Max Kellermann | 10 | -26/+27 | |
| Make sure we get a UTF-8 encoded string. | |||||
| 2012-03-06 | command: fix the "DENIED" ACK code | Max Kellermann | 1 | -1/+1 | |
| Use ACK_ERROR_PERMISSION instead of ACK_ERROR_NO_EXIST. | |||||
| 2012-03-01 | playlist/soundcloud: use config_dup_block_string() | Max Kellermann | 1 | -5/+3 | |
| 2012-03-01 | A soundcloud playlist plugin. | Robert Vollmert | 3 | -0/+451 | |
| 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/ . | |||||
| 2012-03-01 | song_update, udp_server: workarounds for gcc 4.1 warnings | Max Kellermann | 2 | -0/+8 | |
| Annoying false positives. | |||||
| 2012-03-01 | raop_output: fix raop_session inbalance | Kurt Van Dijck | 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> | |||||
| 2012-03-01 | decoder/dsdiff: don't convert to PCM | Max Kellermann | 1 | -38/+9 | |
| Move the responsibility for the conversion to the PCM library. This will allow passing the verbatim DSD samples to an output plugin. | |||||
| 2012-03-01 | pcm_convert: support the DSD format | Max Kellermann | 4 | -0/+167 | |
| 2012-03-01 | audio_format: add DSD sample format | Max Kellermann | 11 | -0/+61 | |
| Basic support for Direct Stream Digital. No conversion yet, and no decoder/output plugin support. | |||||
| 2012-03-01 | pcm_convert: add method _reset() | Max Kellermann | 6 | -0/+39 | |
| Resets the libsamplerate state. Not being used yet. | |||||
| 2012-02-23 | win32: Add a Windows OS resource file and icon | Avuton Olrich | 2 | -0/+34 | |
| 2012-02-15 | decoder/ffmpeg: always use AV_VERSION_INT | Jonathan Neuschäfer | 1 | -1/+1 | |
| 2012-02-15 | input/cdio_paranoia: whitespace-fix a comment | Jonathan Neuschäfer | 1 | -2/+2 | |
| 2012-02-15 | decoder/audiofile: fix compiler warnings with libaudiofile 0.3.3 | Jonathan Neuschäfer | 1 | -4/+4 | |
| This might break older versions, I didn't test. | |||||
| 2012-02-15 | use audio_output_plugins_for_each's plugin iterator | Jonathan Neuschäfer | 1 | -2/+2 | |
| 2012-02-15 | rtsp_client: strncat -> g_strlcat | Jonathan Neuschäfer | 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. | |||||
| 2012-02-15 | main: handle negative strtol return value | Jonathan Neuschäfer | 1 | -2/+3 | |
| size_t is unsigned most of the time, so we can't really use it to check for negative values. Also handle strtol overflow. | |||||
| 2012-02-15 | decoder/sidplay: remove unused variable 'ret' | Jonathan Neuschäfer | 1 | -1/+0 | |
| 2012-02-14 | command: move code to protocol/argparser.c | Max Kellermann | 3 | -166/+241 | |
| 2012-02-14 | command: use standard error messages in check_int() | Max Kellermann | 1 | -49/+27 | |
| 2012-02-14 | command: check for empty string after strtol() | Max Kellermann | 1 | -7/+7 | |
| An empty string is obviously not a valid integer. | |||||
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index : mpd | |
| clone of git://git.musicpd.org/master/mpd.git | Alexander Sulfrian |
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