aboutsummaryrefslogtreecommitdiffstats
path: root/src (follow)
Commit message (Collapse)AuthorAgeFilesLines
* output/oss: use pcm_export to pack 24 bit samplesMax Kellermann2012-03-221-10/+15
|
* 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
|
* 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.
* output/oss: move code to oss_probe_sample_format()Max Kellermann2012-03-211-34/+59
|
* pcm_export: use the byte_reverse library directlyMax Kellermann2012-03-214-152/+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_exportMax Kellermann2012-03-214-61/+167
|
* pcm_pack: fix regression in unpack_sample()Max Kellermann2012-03-211-1/+1
| | | | Should have been "==", not "!=".
* audio_format: DSD_OVER_USB is padded to 32 bitMax Kellermann2012-03-211-3/+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 attributeMax Kellermann2012-03-2113-110/+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).
* output/oss: always receive host byte order samplesMax Kellermann2012-03-211-7/+68
| | | | Don't use audio_format.reverse_endian.
* output/alsa: always receive host byte order samplesMax Kellermann2012-03-211-3/+61
| | | | Don't use audio_format.reverse_endian.
* decoder/pcm: always supply host byte order samplesMax Kellermann2012-03-211-15/+12
| | | | Don't use audio_format.reverse_endian.
* pcm_byteswap: move code to libutilMax Kellermann2012-03-213-38/+199
|
* output/alsa: merge alsa_data_free() into destructorMax Kellermann2012-03-211-8/+3
|
* audio_format: hack for DSD to USB conversionMax Kellermann2012-03-211-0/+9
| | | | Halve the sample rate for *:dsdusb:*.
* Fix the build on OSXRich Healey2012-03-211-0/+1
|
* audio_format: remove the format SAMPLE_FORMAT_DSD_LSBFIRSTMax Kellermann2012-03-2111-36/+4
| | | | | This format is unused since the DSDIFF decoder plugin now reverses the bit order.
* decoder/dsdiff: reverse bits to most significant bit firstMax Kellermann2012-03-211-6/+15
| | | | Allow to remove this complexity from the MPD core.
* dsd2pcm: move the bit reversing code to a generic libraryMax Kellermann2012-03-213-8/+70
| | | | Instead of doing run-time initialisation, use a constant lookup table.
* audio_format: basic support for DSD-over-USBMax Kellermann2012-03-1911-0/+30
|
* playlist/soundcloud: support libyajl2Robert Vollmert2012-03-191-5/+21
| | | | [mk: backwars compatibility and autoconf check]
* 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).
* util/list: allow typeof() with clangMax Kellermann2012-03-191-0/+5
|
* 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.
* command: read arbitrary local files with "lsinfo"Max Kellermann2012-03-061-1/+22
| | | | Requires UNIX domain socket connection.
* client_file: always allow access if client uid equals mpd uidMax Kellermann2012-03-061-0/+5
|
* playlist_edit: move UID check to client_allow_file()Max Kellermann2012-03-065-39/+129
|
* command, ack: add ack_quark()Max Kellermann2012-03-062-0/+16
| | | | To pass ack values around.
* use g_strerror() instead of strerror()Max Kellermann2012-03-0610-26/+27
| | | | Make sure we get a UTF-8 encoded string.
* command: fix the "DENIED" ACK codeMax Kellermann2012-03-061-1/+1
| | | | Use ACK_ERROR_PERMISSION instead of ACK_ERROR_NO_EXIST.
* playlist/soundcloud: use config_dup_block_string()Max Kellermann2012-03-011-5/+3
|
* A soundcloud playlist plugin.Robert Vollmert2012-03-013-0/+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/ .
* song_update, udp_server: workarounds for gcc 4.1 warningsMax Kellermann2012-03-012-0/+8
| | | | Annoying false positives.
* raop_output: fix raop_session inbalanceKurt Van Dijck2012-03-011-2/+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 PCMMax Kellermann2012-03-011-38/+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 formatMax Kellermann2012-03-014-0/+167
|
* audio_format: add DSD sample formatMax Kellermann2012-03-0111-0/+61
| | | | | Basic support for Direct Stream Digital. No conversion yet, and no decoder/output plugin support.
* pcm_convert: add method _reset()Max Kellermann2012-03-016-0/+39
| | | | Resets the libsamplerate state. Not being used yet.
* win32: Add a Windows OS resource file and iconAvuton Olrich2012-02-232-0/+34
|
* Merge remote branches 'jn/ffmpeg' and 'jn/wsp'Max Kellermann2012-02-151-2/+2
|\
| * input/cdio_paranoia: whitespace-fix a commentJonathan Neuschäfer2012-02-151-2/+2
| |
* | decoder/ffmpeg: always use AV_VERSION_INTJonathan Neuschäfer2012-02-151-1/+1
|/
* Merge branch 'af' of git://git.musicpd.org/jn/mpdMax Kellermann2012-02-151-4/+4
|\
| * decoder/audiofile: fix compiler warnings with libaudiofile 0.3.3Jonathan Neuschäfer2012-02-151-4/+4
| | | | | | | | This might break older versions, I didn't test.
* | use audio_output_plugins_for_each's plugin iteratorJonathan Neuschäfer2012-02-151-2/+2
| |
* | rtsp_client: strncat -> g_strlcatJonathan Neuschäfer2012-02-151-7/+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.
* | main: handle negative strtol return valueJonathan Neuschäfer2012-02-151-2/+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.
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-03 06:27:37 +0000