| Commit message (Collapse) | Author | Files | Lines |
|---|
|
|
|
g_file_test is redefined to be g_file_test_utf8 and thus can't handle
non-ASCII characters. This fix adds simple wrapper (taken from glib)
that fixes encoding and calls g_file_test_utf8. All required inclusions
of glib_compat.h are added as well.
|
|
This plugin is horrible code, I mean it. Last year, I tried hard to
fix it, but I figured would take less time to do a full rewrite.
Given that I don't even have any device that supports RAOP, I can't do
that properly. After 16 months, nobody volunteered for fixing it.
Hereby, I delete it, because having no RAOP plugin is better than
having this mess. Sorry.
|
|
|
|
|
|
|
|
|
|
Add the encoder_plugin method end(). This is important for the
recorder plugin.
|
|
Don't reset the ogg_stream_state object, because this discards the
end-of-stream packet that was just added.
|
|
|
|
|
|
|
|
The existing buffer implementation has a major flaw: it is unable to
re-fill the buffer until it has been consumed completely, leading to
many occasions where the render callback needs to generate silence,
just because the play() implementation was unable to append more
data. The fifo_buffer library handles that well.
|
|
|
|
Fixes endless loop when the last line of a text file was not
terminated (bug 3470).
|
|
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/ .
|
|
|
|
|
|
This was disabled when compiled with a new ffmpeg version. Older
ffmpeg versions used it explicitly, while newer ones may pass it
through from the codec.
|
|
This fixes seeking in the vorbis decoder during MPD startup.
|
|
This fixes a bug when libsamplerate returns an empty buffer for a very
small input buffer. The caller thinks this is an error, bug there is
no GError object.
|
|
This finally enables the new embedded CUE sheet code: when a song file
contains a playlist, it is printed in the "lsinfo" output, so clients
get to know about this.
|
|
To replace libcue, the unmaintained and crashy library.
|
|
|
|
|
|
|
|
Implements support for libavcodec 0.9, which removes the compatibility
macros SAMPLE_FMT_*
|
|
Use the tag_item_names table to look up the names of all MPD tags, and
remove the duplicate entries from ffmpeg_tag_maps.
|
|
Use libasound's polling functions, implement a bridge to GSource /
GPollFD and send idle events to clients when an external program
changes the ALSA mixer volume.
|
|
|
|
|
|
|
|
URLContext is deprecated.
|
|
|
|
When we don't have enough data, generate some silence, hoping the
input buffer will fill soon. Reducing the render buffer size is not
legal.
|
|
Blocking inside the render callback is forbidden, and this sleep call
didn't make any sense.
|
|
For simpler seeking within current song.
|
|
Moving songs using either 'move' or 'moveid' to position -1 (after the
current song) would fail for a song which is just before the current
song.
This patch corrects the check to see if the current song is in the range
to be moved. Since the range is from `start` up to `end` (exclusive) the
check was incorrect, but is now fixed.
|
|
The implementation of cancel() did not work well: you cannot use
alSourceUnqueueBuffers() to unqueue queued buffers, and our function
openal_unqueue_buffers() left the OpenAL library in a rather undefined
state; nothing was supposed to be queued, but the "filled" variable
was not reset.
|
|
The OpenAL specification says that AL_FORMAT_MONO8 and
AL_FORMAT_STEREO8 expect unsigned 8 bit samples, but MPD uses unsigned
samples.
|
|
The local variable was already divided by 1000, and the return value
was being divided by 1000 again - doh! This caused delays in the
httpd output plugin that were too small by three orders of magnitude,
and the buffer was filled too quickly.
|
|
Silently choosing default is misleading and can cause hours of investigation.
It's better to fail immediately telling user what is wrong with config.
|
|
|
|
|
|
Initialize the audio_format before calling avcodec_open(), because
avcodec_open() will fill bogus values.
|
|
Yet another common support case.
|
|
This is a common support case, and hopefully, the new error message
will allow the user to understand the error without requiring support.
|
|
Use stat() instead of g_file_test() to detect other types of errors,
such as "permission denied".
|
|
|
|
Convert to padded 24 bit samples, instead of falling back to 16 bit.
|
