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Author SHA1 Message Date
Avuton Olrich
7cd9dbd574 configure: make all tests the same, when possible. 2009-03-05 23:17:00 -08:00
Avuton Olrich
cd52258211 configure: Add 'Streaming Support' section to the configure display. 
Add 'Streaming Support section to the configure display, clarify
the text and unify with the rest of the display.
 2009-03-06 06:51:00 +01:00
Avuton Olrich
6ab53cf62a configure: display the status of lastfm protocol support 2009-03-06 06:50:50 +01:00
Avuton Olrich
f6413d2d5f configure: Fix configure warning presented by lastfm radio. 
The configure problem was:
./configure: line 6934: ,: command not found
 2009-03-05 16:46:19 -08:00
Max Kellermann
01cf7feac7 pipe: added music_buffer, rewrite music_pipe 
Turn the music_pipe into a simple music_chunk queue.  The music_chunk
allocation code is moved to music_buffer, and is now managed with a
linked list instead of a ring buffer.  Two separate music_pipe objects
are used by the decoder for the "current" and the "next" song, which
greatly simplifies the cross-fading code.
 2009-03-06 00:42:03 +01:00
Max Kellermann
000b2d4f3a music_pipe: added music_pipe_push() 
Added music_pipe_allocate(), music_pipe_push() and
music_pipe_cancel().  Those functions allow the caller (decoder thread
in this case) to do its own chunk management.  The functions
music_pipe_flush() and music_pipe_tag() can now be removed.
 2009-03-06 00:42:01 +01:00
Max Kellermann
10be8a8714 playlist_control: fix requeue after seek 
The queue update after a seek was wrong: the queued song is cleared by
a successful seek.  This caused queue/cross-fading problems after a
seek.
 2009-03-06 00:41:59 +01:00
Max Kellermann
b0fcce65d8 flac: explicitly check for STOP command 
After the decoder command was obtained, don't wait until libflac
detects EOF (as a side effect), quit the decoder immediately.  This
check was missing completely.
 2009-03-05 18:20:43 +01:00
Max Kellermann
efd606337e flac: check command after flac_process_single() failure 
When the MPD core sends the decoder a command while
flac_process_single() is executed, this function fails.  Abort the
decoder only if not seeking.  This fixes a seeking bug.
 2009-03-05 18:20:41 +01:00
Max Kellermann
74a2813d78 music_chunk: added music_chunk_write(), music_chunk_expand() 
Moved some code from music_pipe_write() and music_pipe_expand().  Only
music_chunk.c should access the music_chunk internals.
 2009-03-05 17:37:11 +01:00
Max Kellermann
c655f804a9 music_pipe: moved struct music_chunk to chunk.h 2009-03-03 22:23:25 +01:00
Max Kellermann
1063c1f2e3 alsa: log period and buffer size 
Log the real period and buffer size.  This might be useful when
debugging xruns.  Note that the same information is available in
/proc/asound/card*/pcm*p/sub*/hw_params
 2009-03-03 22:19:37 +01:00
Avuton Olrich
3e5a445467 ls: Print output of supported uri to fp rather than stdout. 
Since there are no other callers than stdout, this wouldn't be a
problem, but since there maybe in the future go ahead and fix it.
 2009-03-03 13:12:39 -08:00
Viliam Mateicka
3b76ca7186 ffmpeg: fix version comparision for av_get_bits_per_sample_format() implemetation 
function was implemented in the version we are comparing to so there must be higher or equal
 2009-03-03 21:30:55 +01:00
Viliam Mateicka
c89482de65 ffmpeg: support for new metadata api 2009-03-03 21:30:46 +01:00
Viliam Mateicka
39fb8bebc2 test: init input streams cause decoders are using it 2009-03-03 21:30:26 +01:00
Avuton Olrich
e7f034dcef cmdline: Print available protocols when --version is run. 2009-03-03 21:25:19 +01:00
Max Kellermann
0f64e658fd alsa: fall back to 32 bit samples if 16 is not supported 
There are a few high-end devices (e.g. ICE1724) which cannot even play
16 bit audio.  Try the 32 bit fallback, which we already implemented
for 24 bit.
 2009-03-03 09:38:20 +01:00
Max Kellermann
f82793f741 updated valgrind.suppressions 2009-03-03 07:56:06 +01:00
Eric Wollesen
b8ebb748c9 Add sticker list command. 
[mk: merged memory leak patch; fixed indentation (tabs); fixed
documentation typo]
 2009-03-03 07:49:23 +01:00
Max Kellermann
95b53281a4 updated valgrind.suppressions 2009-03-02 23:11:45 +01:00
Max Kellermann
4220e6b0ad input_lastfm: new input plugin for last.fm radio 
The lastfm input plugin enables MPD to play lastfm:// URLs.  This
plugin is not complete yet: it plays only the first song in the
last.fm playlist, and the playlist parser isn't even implemented
properly.
 2009-03-02 23:11:31 +01:00
Max Kellermann
9cd5b0af74 test/run_decoder: initialize decoder.initialized 
Keep valgrind happy.
 2009-03-02 23:09:58 +01:00
Max Kellermann
a2e47d55eb test/run_decoder: initialize the input plugins 2009-03-02 23:08:24 +01:00
Max Kellermann
cfb350f4f0 input: pass config_param to input_plugin.init() 
Allow input plugins to configure with an "input" block in mpd.conf.
Also allow the user to disable a plugin completely.
 2009-03-02 23:08:17 +01:00
Max Kellermann
9a350acf04 input_plugin: added methods init(), finish() 
Instead of hard-coding the plugin global initialization in
input_stream_global_init(), make it walk the plugin list and
initialize all plugins.
 2009-03-02 20:45:50 +01:00
Max Kellermann
36d24fb7ea input: moved plugins to ./src/input/ 
Create a sub directory for input plugins.
 2009-03-02 20:40:31 +01:00
Max Kellermann
2e51365ea4 input_stream: moved struct input_plugin to input_plugin.h 
Start to separate private from public input_stream API.
 2009-03-02 20:13:08 +01:00
Viliam Mateicka
8694574f63 ffmpeg: use ffmpeg's sampleformat for output format 2009-03-02 20:12:36 +01:00
Viliam Mateicka
60a5b5562b fixing unused parameter warning 2009-03-02 19:00:21 +01:00
Viliam Mateicka
57d836da49 fixing unsigned to signed comparision 
[mk: cast off_t to uint32_t; same fix for aiff.c]
 2009-03-02 18:59:59 +01:00
Viliam Mateicka
406b0403a5 mixer: adding code to optionally disable all hw mixers 2009-03-02 18:57:49 +01:00
Max Kellermann
2f438e5d23 tag_id3: parse ID3 tags in AIFF files 
Added a small AIFF parser library, code copied from the RIFF parser
(big-endian integers).  Look for an "ID3" chunk, and let libid3tag
parse it.
 2009-03-02 18:12:44 +01:00
Max Kellermann
336f624277 tag_id3: parse ID3 tags in RIFF/WAV files 
Added a small RIFF parser library.  Look for an "id3" chunk, and let
libid3tag parse it.
 2009-03-02 18:00:46 +01:00
Max Kellermann
72176db429 alsa: fall back to 32 bit samples if 24 is not supported 
Some sound chips/drivers (e.g. Intel HDA) don't support 24 bit
samples, they want to get 32 bit instead.  Now that MPD's PCM library
supports 32 bit, add a 32 bit fallback when 24 bit is not supported.
 2009-03-02 16:41:38 +01:00
Max Kellermann
a5a15beac2 pcm_convert: added 32 bit support 
All PCM sub libraries have 32 bit support now.  Add support to the
glue function pcm_convert().
 2009-03-02 16:41:10 +01:00
Max Kellermann
3165e26f9a pcm_format: added conversion from 32 bit 
Support converting 32 bit samples to any other supported sample
format.
 2009-03-02 16:41:08 +01:00
Max Kellermann
d4e4c57b8d pcm_format: added pcm_convert_to_32() 
Added code to convert all other sample formats to 32 bit.
 2009-03-02 16:39:54 +01:00
Max Kellermann
d24f2ba5ee pcm_dither: added pcm_dither_32_to_16() 
For 32 bit dithering, reuse the 24 bit dithering code, but apply a 8
bit right shift first.
 2009-03-02 16:37:11 +01:00
Max Kellermann
78e08f655a pcm_dither: renamed struct pcm_dither_24 to struct pcm_dither 
There is nothing 24 bit specific in the pcm_dither_24 struct.  Since
we want to reuse the struct for 32 bit dithering, let's drop the "_24"
suffix from the struct name.
 2009-03-02 16:37:05 +01:00
Max Kellermann
d9c1434298 pcm_resample: use 24 bit resampling code for 32 bit samples 
Resampling 32 bit samples is the same as resampling 24 bit samples -
both are stored in the int32_t type.
 2009-03-02 16:37:00 +01:00
Max Kellermann
1b31f52285 pcm_channels: added implementation for 32 bit samples 
Some 24 bit code can be reused.  The 32 bit variant has to use 64 bit
integers, because 32 bit integers could overflow.  This may be a
performance hit on 32 bit CPUs.
 2009-03-02 16:36:49 +01:00
Max Kellermann
062f37071c audio_format: allow 32 bit samples 
This is the first patch in a series to enable 32 bit audio samples in
MPD.  32 bit samples are more tricky than 24 bit samples, because the
integer may overflow when you operate on a sample.
 2009-03-02 15:46:09 +01:00
Max Kellermann
8c0bce0b94 audio_format: allow up to 8 channels 
audio_valid_sample_format() verifies the number of channels.  Let's
just say up to 8 channels is allowed (which is possible with some
consumer sound chips).  I don't know if there are bigger cards, and
since I cannot test it, I'll limit it to 8 for now.
 2009-03-02 15:43:45 +01:00
Max Kellermann
a1561252d0 directory: directory_load() returns GError 
Do error reporting with GLib's GError library in this library, too.
 2009-03-02 15:42:42 +01:00
Max Kellermann
c0ffec2fd1 database: db_load() returns GError 
Do error reporting with GLib's GError library.
 2009-03-02 15:42:21 +01:00
Max Kellermann
eb5b3ce553 database: no CamelCase 
Renamed a bunch of variables.
 2009-03-02 15:41:44 +01:00
Max Kellermann
b7bfa24f22 pcm_volume: return bool 
Don't abort MPD when a sample format is not supported by pcm_volume().
 2009-03-02 09:42:16 +01:00
Max Kellermann
0579b6ed27 pcm_volume: no CamelCase 2009-03-01 20:11:41 +01:00
Max Kellermann
4194f4b18b audio_parser: added API documentation 2009-03-01 20:08:48 +01:00