Implement the "scan_stream" method that can read tags from any
input_stream object. This requires a FLAC__IOCallbacks implementation
based on the input_stream API.
Improves quality by not squeezing 32 bit samples down to 16 bit, and
then back to 32 bit floating point. Reduces CPU usage by skipping a
conversion step.
Internally the vorbis (non-Tremor) decoder is working in floating
point, and it's not really necessary to cut the output back to 16-bit
if the soundcard or OS supports higher resolution.
The decoder can be trivially modified to bypass its internal
quantisation and produce floating-point output, and a separate
quantisation can be used as appropriate to the platform.
libvorbisidec and libvorbis export the same symbols, which is a
dangerous thing. Since libvorbisenc depends on libvorbis, this can
get nasty, so let's disable the Vorbis encoder unless the user
explicitly wants it.
This commit reimplements the core of the "single" mode. Instead of
doing the detection in the playlist code from the outside, it is moved
to the player thread, which gets a new option called "border_pause".
It will now pause playback exactly at the beginning of the new song,
making the feature more reliable.
Now that the player thread knows what will happen, it can suppress
cross-fading.
Fixes mantis tickets 0003055 and 0003166.
That function is not pure, it writes to error.
When marked as pure, the compiler is allowed to assume it does not do
anything to error, so it can remain NULL, which would result in an
invalid read in print_error().
Ignore APE tags that have no usable tags, and use the ID3 tag instead.
This is useful when the APE tag only contains replay gain, and the
real tags are stored as ID3. This implements feature request Mantis
#0003521.
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.
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.
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 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.
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.
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 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.
WinAPI explicitly declares filesystem encoding.
It can be determined by GetACP().
Use that instead of Glib routine that always "detects" UTF-8 on Win32,
which is incorrect for MPD case.
Ensure that WINVER is defined early enough, so other system headers
won't fall back to their default value. Specifically, this solves a
build failure (-Werror) with mingw-w64 ("WINVER redefined").
When we have an absolute path that's not inside the music directory,
allow loading it anyway, if we're in "secure" mode (i.e. the client is
connected via UNIX socket).
Right now, a playlist with absolute pathnames can only add songs that
are in the same the directory of the playlist or under it.
If uri is an absolute pathname and base_uri is set,
playlist_check_translate_song() will check that base_uri is a prefix
of uri, excluding every other song in the music directory outside
base_uri.
I think in this case base_uri should be completely ignored (and made
NULL) and uri should just be checked against music root directory.
Previously, the condition "defined(play_audio_format)" was used to see
if an output device has been opened, but if the device had failed on
startup, an assertion failure could occur. This patch adds a separate
flag.
The Naim Uniti does not appear to support icecast-style streaming of FLAC
music but does support the codec from a DLNA server. This change looks for
"transferMode.dlna.org: Streaming" in the HTTP request header and responds
with something the Uniti (and hopefully other DLNA clients) accepts.
The only difference in the DLNA streaming mode is the reponse header and
that icecast metadata is disabled. If a client request indicates both modes
are supported, the DLNA mode is preferred (as the Uniti says it supports
both but then rejects a FLAC ICY stream).
Note: This change may be specific to Naim equipment (the only device it was
tested on). E.g. the hardcoding of Content-Length which works but is not a
logically correct value. The change should be backwards-compatible, so
only those clients requesting a DLNA stream will see any difference.
When playing a CUE track, the player thread waited for the decoder to
become ready, and then sent a SEEK command to the beginning of the CUE
track. If that is near the start of the song file, and the track is
short enough, the decoder could have finished decoding already at that
point, and seeking fails.
This commit makes this initial seek more robust: instead of letting
the player thread deal with the difficult timings, let the decoder API
emulate a SEEK command, and return it to the decoder plugin, as soon
as the plugin finishes its initialization.
Add GMutex, GCond attributes which will be used by callers to
conditionally wait on the stream.
Remove the (now-useless) plugin method buffer(), wait on GCond
instead. Lock the input_stream before each method call. Do the same
with the playlist plugins.
