efd606337e
flac: check command after flac_process_single() failure
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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
74a2813d78
music_chunk: added music_chunk_write(), music_chunk_expand()
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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
c655f804a9
music_pipe: moved struct music_chunk to chunk.h
2009-03-03 22:23:25 +01:00
1063c1f2e3
alsa: log period and buffer size
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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
3e5a445467
ls: Print output of supported uri to fp rather than stdout.
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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
3b76ca7186
ffmpeg: fix version comparision for av_get_bits_per_sample_format() implemetation
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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
c89482de65
ffmpeg: support for new metadata api
2009-03-03 21:30:46 +01:00
39fb8bebc2
test: init input streams cause decoders are using it
2009-03-03 21:30:26 +01:00
e7f034dcef
cmdline: Print available protocols when --version is run.
2009-03-03 21:25:19 +01:00
0f64e658fd
alsa: fall back to 32 bit samples if 16 is not supported
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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
f82793f741
updated valgrind.suppressions
2009-03-03 07:56:06 +01:00
b8ebb748c9
Add sticker list command.
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[mk: merged memory leak patch; fixed indentation (tabs); fixed
documentation typo]
2009-03-03 07:49:23 +01:00
95b53281a4
updated valgrind.suppressions
2009-03-02 23:11:45 +01:00
4220e6b0ad
input_lastfm: new input plugin for last.fm radio
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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
9cd5b0af74
test/run_decoder: initialize decoder.initialized
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Keep valgrind happy.
2009-03-02 23:09:58 +01:00
a2e47d55eb
test/run_decoder: initialize the input plugins
2009-03-02 23:08:24 +01:00
cfb350f4f0
input: pass config_param to input_plugin.init()
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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
9a350acf04
input_plugin: added methods init(), finish()
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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
36d24fb7ea
input: moved plugins to ./src/input/
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Create a sub directory for input plugins.
2009-03-02 20:40:31 +01:00
2e51365ea4
input_stream: moved struct input_plugin to input_plugin.h
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Start to separate private from public input_stream API.
2009-03-02 20:13:08 +01:00
8694574f63
ffmpeg: use ffmpeg's sampleformat for output format
2009-03-02 20:12:36 +01:00
60a5b5562b
fixing unused parameter warning
2009-03-02 19:00:21 +01:00
57d836da49
fixing unsigned to signed comparision
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[mk: cast off_t to uint32_t; same fix for aiff.c]
2009-03-02 18:59:59 +01:00
406b0403a5
mixer: adding code to optionally disable all hw mixers
2009-03-02 18:57:49 +01:00
2f438e5d23
tag_id3: parse ID3 tags in AIFF files
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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
336f624277
tag_id3: parse ID3 tags in RIFF/WAV files
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Added a small RIFF parser library. Look for an "id3" chunk, and let
libid3tag parse it.
2009-03-02 18:00:46 +01:00
72176db429
alsa: fall back to 32 bit samples if 24 is not supported
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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
a5a15beac2
pcm_convert: added 32 bit support
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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
3165e26f9a
pcm_format: added conversion from 32 bit
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Support converting 32 bit samples to any other supported sample
format.
2009-03-02 16:41:08 +01:00
d4e4c57b8d
pcm_format: added pcm_convert_to_32()
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Added code to convert all other sample formats to 32 bit.
2009-03-02 16:39:54 +01:00
d24f2ba5ee
pcm_dither: added pcm_dither_32_to_16()
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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
78e08f655a
pcm_dither: renamed struct pcm_dither_24 to struct pcm_dither
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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
d9c1434298
pcm_resample: use 24 bit resampling code for 32 bit samples
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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
1b31f52285
pcm_channels: added implementation for 32 bit samples
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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
062f37071c
audio_format: allow 32 bit samples
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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
8c0bce0b94
audio_format: allow up to 8 channels
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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
a1561252d0
directory: directory_load() returns GError
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Do error reporting with GLib's GError library in this library, too.
2009-03-02 15:42:42 +01:00
c0ffec2fd1
database: db_load() returns GError
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Do error reporting with GLib's GError library.
2009-03-02 15:42:21 +01:00
eb5b3ce553
database: no CamelCase
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Renamed a bunch of variables.
2009-03-02 15:41:44 +01:00
b7bfa24f22
pcm_volume: return bool
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Don't abort MPD when a sample format is not supported by pcm_volume().
2009-03-02 09:42:16 +01:00
0579b6ed27
pcm_volume: no CamelCase
2009-03-01 20:11:41 +01:00
4194f4b18b
audio_parser: added API documentation
2009-03-01 20:08:48 +01:00
f48c58d17b
crossfade: fix doxygen tag
2009-03-01 20:05:27 +01:00
ba3a8474b6
flac: parse stream tags
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Parse the vorbis comments in libflac's metadata_callback and pass them
as tag struct to the decoder API.
2009-03-01 14:07:23 +01:00
92db09fdf8
listen: return GError on "unix path too long"
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When the unix domain socket path is too long, don't abort with
g_error().
2009-03-01 13:35:44 +01:00
f2664e329f
listen: removed unused macro BINDERROR()
2009-03-01 13:34:44 +01:00
d399d4b63b
output_init: removed getBlockParam()
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Use config_get_block_string() and manual GError handling instead.
2009-03-01 13:32:42 +01:00
f298fcf3a6
output_init: return GError on error
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Do error handling with GError instead of aborting with g_error().
2009-03-01 13:31:56 +01:00
cb942eeb45
output_init: moved code to audio_output_detect()
2009-03-01 13:31:32 +01:00
af66f666c6
audiofile: added 24 bit support
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Don't hard code the "bits" parameter to 16. Try to use the input's
sample format, if possible.
2009-03-01 10:53:46 +01:00
Max Kellermann