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60 Commits

Author SHA1 Message Date
Max Kellermann
2784d65618 release v0.18.9 2014-03-02 11:25:01 +01:00
Max Kellermann
47ea69233b output/alsa: remove the obsolete Raspberry Pi workaround
Has been superseded by the previous commit.
2014-03-02 11:22:04 +01:00
Max Kellermann
a884e37de1 output/alsa: call snd_pcm_prepare() after snd_pcm_drop()
Don't wait for an optimistic write to fail.  This is an improved
workaround for the infamous Raspberry Pi bug (see commit af991765).
It works much better and comes without the negative side effects.  The
old workaround is now obsolete.
2014-03-02 11:12:25 +01:00
Max Kellermann
0102a8665a event/SignalMonitor: fix build failure due to missing signal.h include 2014-03-02 10:21:31 +01:00
Max Kellermann
d34ae0850c AllCommands: "findadd" requires the "add" permission 2014-02-27 23:08:22 +01:00
Max Kellermann
6526de024a output/pulse: remove bogus g_free() call 2014-02-24 21:23:49 +01:00
Max Kellermann
5e1e92626c event/SignalMonitor: unblock signals after fork
Fixes hanging child process in the "pipe" output plugin.
2014-02-18 19:13:50 +01:00
Max Kellermann
7fee85c80a configure.ac: fix linker failure when libvorbis/libogg are static
Link libvorbisfile first, followed to libvorbis and finally libogg.
This order is necessary because libvorbisfile depends on libvorbis.
2014-02-18 18:39:19 +01:00
Max Kellermann
5d87a274a5 configure.ac: link the Vorbis encoder with libogg
Fixes another linker failure.  Similar to commit ea406875
2014-02-17 19:42:38 +01:00
Max Kellermann
57e862712a configure.ac: prepare for 0.18.9 2014-02-09 22:58:14 +01:00
Max Kellermann
ddb5390d88 release v0.18.8 2014-02-07 00:06:31 +01:00
Max Kellermann
fce20e514e NEWS: fix 0.18.7 release year 2014-02-07 00:06:31 +01:00
Max Kellermann
af66ed2505 doc/user: document the RoarAudio output plugin 2014-02-06 21:46:29 +01:00
Max Kellermann
ea4068757d configure.ac: link the Vorbis encoder with libvorbis
Since the encoder plugin uses a libvorbis function (and not only
libvorbisenc functions), we need to link with libvorbis explicitly.
2014-02-06 21:32:50 +01:00
Max Kellermann
2b10ecfa37 IcyMetadataParser: more robust tag parser
Allow semicolons and single quotes in the stream title.  This is not
part of any specification, but found in real life.
2014-01-27 10:08:21 +01:00
Max Kellermann
f7eb2b697e test/test_icy_parser: unit test for IcyMetaDataParser.cxx 2014-01-27 09:51:31 +01:00
Max Kellermann
da67260c95 new developer mailing list 2014-01-20 17:20:57 +01:00
Max Kellermann
ab9c9068d4 Queue: rename struct queue to Queue
Works around a build failure on Solaris because annoyingly, Solaris
reserves the name "queue".  This rename was pending anyway.
2014-01-20 08:57:46 +01:00
Max Kellermann
6b4d7d7315 Queue: make the constructor "explicit" 2014-01-20 08:57:41 +01:00
Max Kellermann
313d1d5d83 decoder/ffmpeg: support libav v10_alpha1 2014-01-15 11:33:18 +01:00
Max Kellermann
b7d6133593 decoder/ffmpeg: include cleanup 2014-01-15 11:31:51 +01:00
Max Kellermann
5b6bb114ad decoder/ffmpeg: check for av_samples_get_buffer_size() errors
Fixes potential nullptr dereference.
2014-01-15 11:25:58 +01:00
Max Kellermann
56f082c9d4 util/PeakBuffer: fix nullptr dereference when peak_size==0 2014-01-15 11:24:29 +01:00
Max Kellermann
a1b798e555 SongFilter, TagConfig: cast TAG_NUM_OF_ITEM_TYPES to integer
Fixes clang warning.
2014-01-15 11:23:41 +01:00
Max Kellermann
c91e08fbfd OutputAPI: fix typo in include guard 2014-01-15 11:22:59 +01:00
Max Kellermann
f882434547 configure.ac: prepare for 0.18.8 2014-01-15 11:22:06 +01:00
Max Kellermann
05ad335ae9 release v0.18.7 2014-01-13 11:39:27 +01:00
Max Kellermann
7faeb2ff2b configure.ac: reject libmpcdec SV7 in configure script
Look for symbol "mpc_demux_init" which does not exist in SV7.  This
avoids build failures when SV7 was found by configure.ac.
2014-01-11 21:02:12 +01:00
Max Kellermann
fdd76b3461 decoder/faad: fix memory leak 2014-01-08 22:11:00 +01:00
Max Kellermann
e490e5d0ab playlist/pls: don't free stack buffer 2014-01-08 19:50:44 +01:00
Max Kellermann
3a05c421e0 doc/user: fix typo 2014-01-07 18:06:58 +01:00
Max Kellermann
afc70c120e util/UriUtil: uri_get_suffix() fails if name begins with dot
A file called ".jpg" is not a JPEG file with an empty name; it is
merely a hidden file.
2013-12-29 17:40:51 +01:00
Max Kellermann
d7f80eab68 configure.ac: improved check for libyajl 1.0
If we have libyajl 2.0.1 (without a pkg-config file), our configure.ac
would assume this is the libyajl 1.0 API, because the function
yajl_alloc() exists in both.  This commit changes the library check to
the function yajl_parse_complete() which was removed in the 2.0 API.
This fixes build failure with libyajl 2.0.1.
2013-12-29 14:12:33 +01:00
Max Kellermann
e30b356eb0 daemon: no initgroups() when already running as the configured user
We can assume that initgroups() would be a no-op in that case, however
initgroups() is not allowed for unprivileged users anyway.
2013-12-29 13:59:05 +01:00
Max Kellermann
09a0803116 Daemon: fix typo in comment 2013-12-29 13:59:05 +01:00
Max Kellermann
20ffedc745 Daemon: simplify nested "if" 2013-12-29 13:57:12 +01:00
Max Kellermann
0b1ad27ba8 Daemon: fix typo in cast 2013-12-29 13:47:29 +01:00
Max Kellermann
6a1b2f0387 configure.ac: prepare for 0.18.7 2013-12-29 10:40:59 +01:00
Max Kellermann
fb34519b96 release v0.18.6 2013-12-24 12:01:01 +01:00
Max Kellermann
91fed47648 PlayerThread: log the last song that was played 2013-12-24 11:58:10 +01:00
Max Kellermann
c05691b546 OutputControl: update both ReplayGainFilters
The "mode" of the second ReplayGainFilter was never set, and thus
replay gain was never applied to the new song during cross-fade.
2013-12-24 11:53:21 +01:00
Steven O'Brien
6b3b8c6f2e fix FfmpegDecoderPlugin to use relative timestamps 2013-12-20 22:28:33 +01:00
Max Kellermann
a191db84f2 util/Error: add missing <algorithm> include
For std::move().
2013-12-19 10:58:20 +01:00
Michal Smucr
e4d69f38b0 riff: recognize upper-case "ID3" chunk name
Some tagging libraries (eg. TagLib) produce that variant.
2013-12-19 09:35:54 +01:00
Max Kellermann
97fc001180 input/cdio: fix typo in #include path
Broken by commit 3b0fea5f
2013-12-17 08:58:00 +01:00
Max Kellermann
42a09ff17a mixer/alsa: fix deadlock
This deadlock was a regression by commit 8e38b4f8.  Since we currently
can't resolve this, let's revert the commit, and add a GLib specific
workaround for the build failure.
2013-12-15 19:07:25 +01:00
Max Kellermann
c170fed6f9 .gitignore: ignore "test-driver"
File generated by automake version 1.14.
2013-12-14 12:33:20 +01:00
Max Kellermann
8e38b4f83c mixer/alsa: use BlockingCall() instead of EventLoop::AddCall()
This is safer, and works without epoll().  Fixes a build failure with
uClibc, which does not support epoll().
2013-12-13 14:35:36 +01:00
Max Kellermann
db4ae19246 doc/mpd.conf.5: remove redundant documentation
The real and detailed documentation is in the user manual.
2013-12-11 21:04:28 +01:00
Max Kellermann
82a89c6bfe doc/user: document the "ao" output
Move from doc/mpd.conf.5.
2013-12-11 21:03:11 +01:00
Max Kellermann
166c70cab3 doc/user: document the "fifo" output
Move from doc/mpd.conf.5.
2013-12-11 20:58:06 +01:00
Max Kellermann
de78fe38c8 doc/user: document shout option "protocol" 2013-12-11 20:54:42 +01:00
mobidyc
96fa69ff6b SongUpdate: accept files without metadata
If the file was recognized by a decoder plugin, accept it - don't
require metadata.
2013-12-10 19:34:35 +01:00
Max Kellermann
39d94bd3ea TagFile: add return value API documentation 2013-12-10 19:32:26 +01:00
Lukas Stabe
695ca29274 output/osx: fix build failure 2013-12-10 19:19:31 +01:00
Max Kellermann
02e8da6c98 NEWS: add openal line 2013-12-10 19:19:27 +01:00
Lukas Stabe
0ea5f4ac3a output/openal: check __APPLE__ instead of HAVE_OSX
On OSX, the configure-flag --enable-osx is used to enable the
unsupported osx output. It sets the HAVE_OSX preprocessor define.

src/output/OpenALOutputPlugin.cxx uses this define to determine wether
it is building on OSX, and imports different headers (which have
nothing to do with the osx output) depending on wether or not it is
set.
2013-12-02 13:08:52 +01:00
Max Kellermann
47c50c079d decoder/ffmpeg: use IgnoreError instead of local Error instance 2013-11-28 00:05:26 +01:00
Gaetan Bisson
3b0fea5fae input/cdio_paranoia: support libcdio-paranoia 0.90 2013-11-27 08:25:17 +01:00
Max Kellermann
443516cdda configure.ac: prepare for 0.18.6 2013-11-27 08:25:17 +01:00
45 changed files with 550 additions and 287 deletions

1
.gitignore vendored

@@ -31,6 +31,7 @@ libtool
ltmain.sh
missing
mkinstalldirs
/test-driver
mpd
mpd.service
stamp-h1

@@ -1061,6 +1061,7 @@ C_TESTS = \
test/test_util \
test/test_byte_reverse \
test/test_mixramp \
test/test_icy_parser \
test/test_pcm \
test/test_queue_priority
@@ -1496,6 +1497,16 @@ test_test_mixramp_LDADD = \
$(GLIB_LIBS) \
$(CPPUNIT_LIBS)
test_test_icy_parser_SOURCES = \
src/Log.cxx \
test/test_icy_parser.cxx
test_test_icy_parser_CPPFLAGS = $(AM_CPPFLAGS) $(CPPUNIT_CFLAGS) -DCPPUNIT_HAVE_RTTI=0
test_test_icy_parser_CXXFLAGS = $(AM_CXXFLAGS) -Wno-error=deprecated-declarations
test_test_icy_parser_LDADD = \
libtag.a \
$(GLIB_LIBS) \
$(CPPUNIT_LIBS)
test_test_pcm_SOURCES = \
test/test_pcm_util.hxx \
test/test_pcm_dither.cxx \

48
NEWS

@@ -1,3 +1,51 @@
ver 0.18.9 (2014/03/02)
* protocol
- "findadd" requires the "add" permission
* output
- alsa: improved workaround for noise after manual song change
* decoder
- vorbis: fix linker failure when libvorbis/libogg are static
* encoder
- vorbis: fix another linker failure
* output
- pipe: fix hanging child process due to blocked signals
* fix build failure due to missing signal.h include
ver 0.18.8 (2014/02/07)
* decoder
- ffmpeg: support libav v10_alpha1
* encoder
- vorbis: fix linker failure
* output
- roar: documentation
* more robust Icy-Metadata parser
* fix Solaris build failure
ver 0.18.7 (2014/01/13)
* playlist
- pls: fix crash after parser error
- soundcloud: fix build failure with libyajl 2.0.1
* decoder
- faad: fix memory leak
- mpcdec: reject libmpcdec SV7 in configure script
* daemon: don't initialize supplementary groups when already running
as the configured user
ver 0.18.6 (2013/12/24)
* input
- cdio_paranoia: support libcdio-paranoia 0.90
* tags
- riff: recognize upper-case "ID3" chunk name
* decoder
- ffmpeg: use relative timestamps
* output
- openal: fix build failure on Mac OS X
- osx: fix build failure
* mixer
- alsa: fix build failure with uClibc
* fix replay gain during cross-fade
* accept files without metadata
ver 0.18.5 (2013/11/23)
* configuration
- fix crash when db_file is configured without music_directory

@@ -1,6 +1,6 @@
AC_PREREQ(2.60)
AC_INIT(mpd, 0.18.5, musicpd-dev-team@lists.sourceforge.net)
AC_INIT(mpd, 0.18.9, mpd-devel@musicpd.org)
VERSION_MAJOR=0
VERSION_MINOR=18
@@ -732,7 +732,7 @@ dnl --------------------------------- Soundcloud ------------------------------
if test x$enable_soundcloud != xno; then
PKG_CHECK_MODULES([YAJL], [yajl >= 2.0],
[found_soundcloud=yes],
AC_CHECK_LIB([yajl], [yajl_alloc],
AC_CHECK_LIB([yajl], [yajl_parse_complete],
[found_soundcloud=yes YAJL_CFLAGS=-DHAVE_YAJL1 YAJL_LIBS=-lyajl],
[found_soundcloud=no]))
fi
@@ -749,6 +749,7 @@ MPD_AUTO_PKG(cdio_paranoia, CDIO_PARANOIA, [libcdio_paranoia],
if test x$enable_cdio_paranoia = xyes; then
AC_DEFINE([ENABLE_CDIO_PARANOIA], 1,
[Define to enable libcdio_paranoia support])
AC_CHECK_HEADERS(cdio/paranoia/paranoia.h)
fi
AM_CONDITIONAL(ENABLE_CDIO_PARANOIA, test x$enable_cdio_paranoia = xyes)
@@ -959,7 +960,7 @@ AM_CONDITIONAL(ENABLE_SNDFILE, test x$enable_sndfile = xyes)
dnl --------------------------------- musepack --------------------------------
MPD_AUTO_LIB(mpc, MPCDEC, mpcdec, main, [-lmpcdec], [],
MPD_AUTO_LIB(mpc, MPCDEC, mpcdec, mpc_demux_init, [-lmpcdec], [],
[mpcdec], [libmpcdec not found])
if test x$enable_mpc = xyes; then
AC_DEFINE(HAVE_MPCDEC, 1, [Define to use libmpcdec for MPC decoding])
@@ -1022,7 +1023,7 @@ if test x$enable_tremor = xyes; then
fi
fi
MPD_AUTO_PKG(vorbis, VORBIS, [vorbis vorbisfile ogg],
MPD_AUTO_PKG(vorbis, VORBIS, [vorbisfile vorbis ogg],
[Ogg Vorbis decoder], [libvorbis not found])
if test x$enable_vorbis = xyes; then
AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support])
@@ -1138,7 +1139,7 @@ fi
AM_CONDITIONAL(ENABLE_FLAC_ENCODER, test x$enable_flac_encoder = xyes)
dnl ---------------------------- Ogg Vorbis Encoder ---------------------------
MPD_AUTO_PKG(vorbis_encoder, VORBISENC, [vorbisenc],
MPD_AUTO_PKG(vorbis_encoder, VORBISENC, [vorbisenc vorbis ogg],
[Ogg Vorbis encoder], [libvorbisenc not found])
if test x$enable_vorbis_encoder = xyes; then

@@ -155,7 +155,7 @@ foo(const char *abc, int xyz)
<para>
Send your patches to the mailing list:
musicpd-dev-team@lists.sourceforge.net
mpd-devel@musicpd.org
</para>
</chapter>
</book>

@@ -363,137 +363,6 @@ errors on bandwidth-limited devices. Some users have reported good results
with this set to 50000, but not all devices support values this high. Most
users do not need to change this. The default is 256000000 / sample_rate(kHz),
or 5804 microseconds for CD-quality audio.
.SH OPTIONAL OSS OUTPUT PARAMETERS
.TP
.B device <dev>
This specifies the device to use for audio output. The default is "/dev/dsp".
.TP
.B mixer_device <mixer dev>
This specifies which mixer to use. The default is "/dev/mixer".
.TP
.B mixer_control <mixer ctrl>
This specifies which mixer control to use (sometimes referred to as the
"device"). The default is to use the main PCM mixer. An example is "Pcm".
.SH OPTIONAL PULSE OUTPUT PARAMETERS
.TP
.B server <server list>
A space separated list of servers to try to connect to. See
<\fBhttp://www.pulseaudio.org/wiki/ServerStrings\fP> for more details. The
default is to let PulseAudio choose a server.
If you specify more than one server name, MPD tries to connect to one
after another until it successfully establishes a connection.
.TP
.B sink <sink>
The sink to output to. The default is to let PulseAudio choose a sink.
.SH OPTIONAL JACK OUTPUT PARAMETERS
.TP
.B client_name <name>
The client name to use when connecting to JACK. The output ports <name>:left
and <name>:right will also be created for the left and right channels,
respectively.
.TP
.B ports <left_port,right_port>
This specifies the left and right ports to connect to for the left and right
channels, respectively. The default is to let JACK choose a pair of ports.
.TP
.B ringbuffer_size <size in bytes>
This specifies the size of the ringbuffer in bytes. The default is 32768.
.SH OPTIONAL AO OUTPUT PARAMETERS
.TP
.B driver <driver>
This specifies the libao driver to use for audio output. Possible values
depend on what libao drivers are available. See
<\fBhttp://www.xiph.org/ao/doc/drivers.html\fP> for information on some
commonly used drivers. Typical values for Linux include "oss" and "alsa09".
The default is "default", which causes libao to select an appropriate plugin.
.TP
.B options <opts>
This specifies the options to use for the selected libao driver. For oss, the
only option available is "dsp". For alsa09, the available options are: "dev",
"buf_size", and "periods". See <\fBhttp://www.xiph.org/ao/doc/drivers.html\fP>
for available options for some commonly used drivers. Options are assigned
using "=", and ";" is used to separate options. An example for oss:
"dsp=/dev/dsp". An example for alsa09: "dev=hw:0,0;buf_size=4096". The
default is "".
.TP
.B write_size <size in bytes>
This specifies how many bytes to write to the audio device at once. This
parameter is to work around a bug in older versions of libao on sound cards
with very small buffers. The default is 1024.
.SH REQUIRED FIFO OUTPUT PARAMETERS
.TP
.B path <path>
This specifies the path of the FIFO to output to. Must be an absolute path.
If the path does not exist it will be created when mpd is started, and removed
when mpd is stopped. The FIFO will be created with the same user and group as
mpd is running as. Default permissions can be modified by using the builtin
shell command "umask". If a FIFO already exists at the specified path it will
be reused, and will \fBnot\fP be removed when mpd is stopped. You can use the
"mkfifo" command to create this, and then you may modify the permissions to
your liking.
.SH REQUIRED SHOUT OUTPUT PARAMETERS
.TP
.B name <name>
This specifies not only the unique audio output name, but also the stream
title.
.TP
.B host <hostname>
This specifies the hostname of the icecast server to connect to.
.TP
.B port <port>
This specifies the port of the icecast server to connect to.
.TP
.B mount <mountpoint>
This specifies the icecast mountpoint to use.
.TP
.B password <password>
This specifies the password to use when logging in to the icecast server.
.TP
.B quality <quality>
This specifies the encoding quality to use. The value must be between 0
and 10. Fractional values, such as 2.5, are permitted. Either the quality or
the bitrate parameter must be specified, but not both. For Ogg, a
higher quality number produces higher quality output. For MP3, it's
just the opposite, with lower numbers producing higher quality output.
.TP
.B bitrate <kbps>
This specifies the bitrate to use for encoding. Either the quality or the
bitrate parameter must be specified, but not both.
.TP
.B format <sample_rate:bits:channels>
This specifies the sample rate, bits per sample, and number of channels to use
for encoding.
.SH OPTIONAL SHOUT OUTPUT PARAMETERS
.TP
.B encoding <encoding>
This specifies which output encoding to use. Should be either "ogg"
or "mp3", "mp3" is needed for shoutcast streaming. The default is "ogg".
.TP
.B protocol <protocol>
This specifies the protocol that wil be used to connect to the
icecast/shoutcast server. The options are "shoutcast", "icecast1" and
"icecast2". The default is "icecast2".
.TP
.B user <username>
This specifies the username to use when logging in to the icecast server. The
default is "source".
.TP
.B public <yes or no>
This specifies whether to request that the stream be listed in all public
stream directories that the icecast server knows about. The default is no.
.TP
.B timeout <seconds>
This specifies the number of seconds to wait before giving up on trying to
connect to the icecast server. The default is 2 seconds.
.TP
.B description <description>
This specifies a description of the stream.
.TP
.B url <url>
This specifies a URL associated with the stream.
.TP
.B genre <genre>
This specifies the genre(s) of the stream.
.SH FILES
.TP
.BI ~/.mpdconf

@@ -829,7 +829,7 @@ systemctl start mpd.socket</programlisting>
<tbody>
<row>
<entry>
<varname>default_bute_order</varname>
<varname>default_byte_order</varname>
<parameter>little_endian|big_endian</parameter>
</entry>
<entry>
@@ -1398,6 +1398,59 @@ systemctl start mpd.socket</programlisting>
The <varname>ao</varname> plugin uses the portable
<filename>libao</filename> library.
</para>
<informaltable>
<tgroup cols="2">
<thead>
<row>
<entry>Setting</entry>
<entry>Description</entry>
</row>
</thead>
<tbody>
<row>
<entry>
<varname>driver</varname>
<parameter>D</parameter>
</entry>
<entry>
The <filename>libao</filename> driver to use for
audio output. Possible values depend on what libao
drivers are available. See <ulink
url="http://www.xiph.org/ao/doc/drivers.html">http://www.xiph.org/ao/doc/drivers.html</ulink>
for information on some commonly used drivers.
Typical values for Linux include "oss" and "alsa09".
The default is "default", which causes libao to
select an appropriate plugin.
</entry>
</row>
<row>
<entry>
<varname>options</varname>
<parameter>O</parameter>
</entry>
<entry>
Options to pass to the selected
<filename>libao</filename> driver.
</entry>
</row>
<row>
<entry>
<varname>write_size</varname>
<parameter>O</parameter>
</entry>
<entry>
This specifies how many bytes to write to the audio
device at once. This parameter is to work around a
bug in older versions of libao on sound cards with
very small buffers. The default is 1024.
</entry>
</row>
</tbody>
</tgroup>
</informaltable>
</section>
<section>
@@ -1408,6 +1461,38 @@ systemctl start mpd.socket</programlisting>
FIFO (First In, First Out) file. The data can be read by
another program.
</para>
<informaltable>
<tgroup cols="2">
<thead>
<row>
<entry>Setting</entry>
<entry>Description</entry>
</row>
</thead>
<tbody>
<row>
<entry>
<varname>path</varname>
<parameter>P</parameter>
</entry>
<entry>
This specifies the path of the FIFO to write to.
Must be an absolute path. If the path does not
exist, it will be created when MPD is started, and
removed when MPD is stopped. The FIFO will be
created with the same user and group as MPD is
running as. Default permissions can be modified by
using the builtin shell command "umask". If a FIFO
already exists at the specified path it will be
reused, and will not be removed when MPD is stopped.
You can use the "mkfifo" command to create this, and
then you may modify the permissions to your liking.
</entry>
</row>
</tbody>
</tgroup>
</informaltable>
</section>
<section>
@@ -1767,6 +1852,51 @@ systemctl start mpd.socket</programlisting>
</informaltable>
</section>
<section>
<title><varname>roar</varname></title>
<para>
The <varname>roar</varname> plugin connects to a <ulink
url="http://roaraudio.keep-cool.org/">RoarAudio</ulink>
server.
</para>
<informaltable>
<tgroup cols="2">
<thead>
<row>
<entry>Setting</entry>
<entry>Description</entry>
</row>
</thead>
<tbody>
<row>
<entry>
<varname>server</varname>
<parameter>HOSTNAME</parameter>
</entry>
<entry>
The host name of the RoarAudio server. If not
specified, then MPD will connect to the default
locations.
</entry>
</row>
<row>
<entry>
<varname>role</varname>
<parameter>ROLE</parameter>
</entry>
<entry>
The "role" that MPD registers itself as in the
RoarAudio server. The default is "music".
</entry>
</row>
</tbody>
</tgroup>
</informaltable>
</section>
<section>
<title><varname>recorder</varname></title>
@@ -1883,6 +2013,18 @@ systemctl start mpd.socket</programlisting>
Defaults to 2 seconds.
</entry>
</row>
<row>
<entry>
<varname>protocol</varname>
<parameter>icecast2|icecast1|shoutcast</parameter>
</entry>
<entry>
Specifies the protocol that wil be used to connect
to the icecast/shoutcast server. The default
is "<parameter>icecast2</parameter>".
</entry>
</row>
<row>
<entry>
<varname>mount</varname>

@@ -52,7 +52,7 @@ static char *user_name;
static uid_t user_uid = (uid_t)-1;
/** the Unix group id which MPD runs as */
static gid_t user_gid = (pid_t)-1;
static gid_t user_gid = (gid_t)-1;
/** the absolute path of the pidfile */
static AllocatedPath pidfile = AllocatedPath::Null();
@@ -106,18 +106,21 @@ daemonize_set_user(void)
return;
/* set gid */
if (user_gid != (gid_t)-1 && user_gid != getgid()) {
if (setgid(user_gid) == -1) {
FormatFatalSystemError("Failed to set group %d",
(int)user_gid);
}
if (user_gid != (gid_t)-1 && user_gid != getgid() &&
setgid(user_gid) == -1) {
FormatFatalSystemError("Failed to set group %d",
(int)user_gid);
}
#ifdef _BSD_SOURCE
/* init suplementary groups
/* init supplementary groups
* (must be done before we change our uid)
*/
if (!had_group && initgroups(user_name, user_gid) == -1) {
if (!had_group &&
/* no need to set the new user's supplementary groups if
we are already this user */
user_uid != getuid() &&
initgroups(user_name, user_gid) == -1) {
FormatFatalSystemError("Failed to set supplementary groups "
"of user \"%s\"",
user_name);

@@ -81,31 +81,85 @@ icy_add_item(Tag &tag, TagType type, const char *value)
}
static void
icy_parse_tag_item(Tag &tag, const char *item)
icy_parse_tag_item(Tag &tag, const char *name, const char *value)
{
gchar **p = g_strsplit(item, "=", 0);
if (strcmp(name, "StreamTitle") == 0)
icy_add_item(tag, TAG_TITLE, value);
else
FormatDebug(icy_metadata_domain,
"unknown icy-tag: '%s'", name);
}
if (p[0] != nullptr && p[1] != nullptr) {
if (strcmp(p[0], "StreamTitle") == 0)
icy_add_item(tag, TAG_TITLE, p[1]);
else
FormatDebug(icy_metadata_domain,
"unknown icy-tag: '%s'", p[0]);
/**
* Find a single quote that is followed by a semicolon (or by the end
* of the string). If that fails, return the first single quote. If
* that also fails, return #end.
*/
static char *
find_end_quote(char *p, char *const end)
{
char *fallback = std::find(p, end, '\'');
if (fallback >= end - 1 || fallback[1] == ';')
return fallback;
p = fallback + 1;
while (true) {
p = std::find(p, end, '\'');
if (p == end)
return fallback;
if (p == end - 1 || p[1] == ';')
return p;
++p;
}
g_strfreev(p);
}
static Tag *
icy_parse_tag(const char *p)
icy_parse_tag(char *p, char *const end)
{
assert(p != nullptr);
assert(end != nullptr);
assert(p <= end);
Tag *tag = new Tag();
gchar **items = g_strsplit(p, ";", 0);
for (unsigned i = 0; items[i] != nullptr; ++i)
icy_parse_tag_item(*tag, items[i]);
while (p != end) {
const char *const name = p;
char *eq = std::find(p, end, '=');
if (eq == end)
break;
g_strfreev(items);
*eq = 0;
p = eq + 1;
if (*p != '\'') {
/* syntax error; skip to the next semicolon,
try to recover */
char *semicolon = std::find(p, end, ';');
if (semicolon == end)
break;
p = semicolon + 1;
continue;
}
++p;
const char *const value = p;
char *quote = find_end_quote(p, end);
if (quote == end)
break;
*quote = 0;
p = quote + 1;
icy_parse_tag_item(*tag, name, value);
char *semicolon = std::find(p, end, ';');
if (semicolon == end)
break;
p = semicolon + 1;
}
return tag;
}
@@ -152,15 +206,11 @@ IcyMetaDataParser::Meta(const void *data, size_t length)
++length;
if (meta_position == meta_size) {
/* null-terminate the string */
meta_data[meta_size] = 0;
/* parse */
delete tag;
tag = icy_parse_tag(meta_data);
tag = icy_parse_tag(meta_data, meta_data + meta_size);
g_free(meta_data);
/* change back to normal data mode */

@@ -18,7 +18,7 @@
*/
#ifndef MPD_OUTPUT_API_HXX
#define MPD_OUTPUT_API_HxX
#define MPD_OUTPUT_API_HXX
#include "OutputPlugin.hxx"
#include "OutputInternal.hxx"

@@ -101,6 +101,8 @@ audio_output_set_replay_gain_mode(struct audio_output *ao,
{
if (ao->replay_gain_filter != nullptr)
replay_gain_filter_set_mode(ao->replay_gain_filter, mode);
if (ao->other_replay_gain_filter != nullptr)
replay_gain_filter_set_mode(ao->other_replay_gain_filter, mode);
}
void

@@ -1081,8 +1081,11 @@ Player::Run()
delete cross_fade_tag;
if (song != nullptr)
if (song != nullptr) {
const auto uri = song->GetURI();
FormatDefault(player_domain, "played \"%s\"", uri.c_str());
song->Free();
}
pc.Lock();

@@ -30,7 +30,7 @@ struct playlist {
/**
* The song queue - it contains the "real" playlist.
*/
struct queue queue;
struct Queue queue;
/**
* This value is true if the player is currently playing (or

@@ -40,7 +40,7 @@
void
playlist_print_uris(Client &client, const playlist &playlist)
{
const queue &queue = playlist.queue;
const Queue &queue = playlist.queue;
queue_print_uris(client, queue, 0, queue.GetLength());
}
@@ -49,7 +49,7 @@ bool
playlist_print_info(Client &client, const playlist &playlist,
unsigned start, unsigned end)
{
const queue &queue = playlist.queue;
const Queue &queue = playlist.queue;
if (end > queue.GetLength())
/* correct the "end" offset */

@@ -65,7 +65,7 @@ playlist_print_uri(FILE *file, const char *uri)
}
PlaylistResult
spl_save_queue(const char *name_utf8, const queue &queue)
spl_save_queue(const char *name_utf8, const Queue &queue)
{
if (map_spl_path().IsNull())
return PlaylistResult::DISABLED;

@@ -25,7 +25,7 @@
#include <stdio.h>
struct Song;
struct queue;
struct Queue;
struct playlist;
struct PlayerControl;
class Error;
@@ -40,7 +40,7 @@ playlist_print_uri(FILE *fp, const char *uri);
* Saves a queue object into a stored playlist file.
*/
PlaylistResult
spl_save_queue(const char *name_utf8, const queue &queue);
spl_save_queue(const char *name_utf8, const Queue &queue);
/**
* Saves a playlist object into a stored playlist file.

@@ -23,7 +23,7 @@
#include <stdlib.h>
queue::queue(unsigned _max_length)
Queue::Queue(unsigned _max_length)
:max_length(_max_length), length(0),
version(1),
items(new Item[max_length]),
@@ -36,7 +36,7 @@ queue::queue(unsigned _max_length)
{
}
queue::~queue()
Queue::~Queue()
{
Clear();
@@ -45,7 +45,7 @@ queue::~queue()
}
int
queue::GetNextOrder(unsigned _order) const
Queue::GetNextOrder(unsigned _order) const
{
assert(_order < length);
@@ -62,7 +62,7 @@ queue::GetNextOrder(unsigned _order) const
}
void
queue::IncrementVersion()
Queue::IncrementVersion()
{
static unsigned long max = ((uint32_t) 1 << 31) - 1;
@@ -77,7 +77,7 @@ queue::IncrementVersion()
}
void
queue::ModifyAtOrder(unsigned _order)
Queue::ModifyAtOrder(unsigned _order)
{
assert(_order < length);
@@ -86,7 +86,7 @@ queue::ModifyAtOrder(unsigned _order)
}
unsigned
queue::Append(Song *song, uint8_t priority)
Queue::Append(Song *song, uint8_t priority)
{
assert(!IsFull());
@@ -105,7 +105,7 @@ queue::Append(Song *song, uint8_t priority)
}
void
queue::SwapPositions(unsigned position1, unsigned position2)
Queue::SwapPositions(unsigned position1, unsigned position2)
{
unsigned id1 = items[position1].id;
unsigned id2 = items[position2].id;
@@ -120,7 +120,7 @@ queue::SwapPositions(unsigned position1, unsigned position2)
}
void
queue::MovePostion(unsigned from, unsigned to)
Queue::MovePostion(unsigned from, unsigned to)
{
const Item tmp = items[from];
@@ -156,7 +156,7 @@ queue::MovePostion(unsigned from, unsigned to)
}
void
queue::MoveRange(unsigned start, unsigned end, unsigned to)
Queue::MoveRange(unsigned start, unsigned end, unsigned to)
{
Item tmp[end - start];
// Copy the original block [start,end-1]
@@ -198,7 +198,7 @@ queue::MoveRange(unsigned start, unsigned end, unsigned to)
}
void
queue::MoveOrder(unsigned from_order, unsigned to_order)
Queue::MoveOrder(unsigned from_order, unsigned to_order)
{
assert(from_order < length);
assert(to_order <= length);
@@ -217,7 +217,7 @@ queue::MoveOrder(unsigned from_order, unsigned to_order)
}
void
queue::DeletePosition(unsigned position)
Queue::DeletePosition(unsigned position)
{
assert(position < length);
@@ -254,7 +254,7 @@ queue::DeletePosition(unsigned position)
}
void
queue::Clear()
Queue::Clear()
{
for (unsigned i = 0; i < length; i++) {
Item *item = &items[i];
@@ -270,7 +270,7 @@ queue::Clear()
}
static void
queue_sort_order_by_priority(struct queue *queue, unsigned start, unsigned end)
queue_sort_order_by_priority(Queue *queue, unsigned start, unsigned end)
{
assert(queue != nullptr);
assert(queue->random);
@@ -278,8 +278,8 @@ queue_sort_order_by_priority(struct queue *queue, unsigned start, unsigned end)
assert(end <= queue->length);
auto cmp = [queue](unsigned a_pos, unsigned b_pos){
const queue::Item &a = queue->items[a_pos];
const queue::Item &b = queue->items[b_pos];
const Queue::Item &a = queue->items[a_pos];
const Queue::Item &b = queue->items[b_pos];
return a.priority > b.priority;
};
@@ -288,7 +288,7 @@ queue_sort_order_by_priority(struct queue *queue, unsigned start, unsigned end)
}
void
queue::ShuffleOrderRange(unsigned start, unsigned end)
Queue::ShuffleOrderRange(unsigned start, unsigned end)
{
assert(random);
assert(start <= end);
@@ -303,7 +303,7 @@ queue::ShuffleOrderRange(unsigned start, unsigned end)
* priority group.
*/
void
queue::ShuffleOrderRangeWithPriority(unsigned start, unsigned end)
Queue::ShuffleOrderRangeWithPriority(unsigned start, unsigned end)
{
assert(random);
assert(start <= end);
@@ -337,13 +337,13 @@ queue::ShuffleOrderRangeWithPriority(unsigned start, unsigned end)
}
void
queue::ShuffleOrder()
Queue::ShuffleOrder()
{
ShuffleOrderRangeWithPriority(0, length);
}
void
queue::ShuffleOrderFirst(unsigned start, unsigned end)
Queue::ShuffleOrderFirst(unsigned start, unsigned end)
{
rand.AutoCreate();
@@ -352,7 +352,7 @@ queue::ShuffleOrderFirst(unsigned start, unsigned end)
}
void
queue::ShuffleOrderLast(unsigned start, unsigned end)
Queue::ShuffleOrderLast(unsigned start, unsigned end)
{
rand.AutoCreate();
@@ -361,7 +361,7 @@ queue::ShuffleOrderLast(unsigned start, unsigned end)
}
void
queue::ShuffleRange(unsigned start, unsigned end)
Queue::ShuffleRange(unsigned start, unsigned end)
{
assert(start <= end);
assert(end <= length);
@@ -377,7 +377,7 @@ queue::ShuffleRange(unsigned start, unsigned end)
}
unsigned
queue::FindPriorityOrder(unsigned start_order, uint8_t priority,
Queue::FindPriorityOrder(unsigned start_order, uint8_t priority,
unsigned exclude_order) const
{
assert(random);
@@ -394,7 +394,7 @@ queue::FindPriorityOrder(unsigned start_order, uint8_t priority,
}
unsigned
queue::CountSamePriority(unsigned start_order, uint8_t priority) const
Queue::CountSamePriority(unsigned start_order, uint8_t priority) const
{
assert(random);
assert(start_order <= length);
@@ -410,7 +410,7 @@ queue::CountSamePriority(unsigned start_order, uint8_t priority) const
}
bool
queue::SetPriority(unsigned position, uint8_t priority, int after_order)
Queue::SetPriority(unsigned position, uint8_t priority, int after_order)
{
assert(position < length);
@@ -468,7 +468,7 @@ queue::SetPriority(unsigned position, uint8_t priority, int after_order)
}
bool
queue::SetPriorityRange(unsigned start_position, unsigned end_position,
Queue::SetPriorityRange(unsigned start_position, unsigned end_position,
uint8_t priority, int after_order)
{
assert(start_position <= end_position);

@@ -41,7 +41,7 @@ struct Song;
* - the unique id (which stays the same, regardless of moves)
* - the order number (which only differs from "position" in random mode)
*/
struct queue {
struct Queue {
/**
* reserve max_length * HASH_MULT elements in the id
* number space
@@ -103,16 +103,16 @@ struct queue {
/** random number generator for shuffle and random mode */
LazyRandomEngine rand;
queue(unsigned max_length);
explicit Queue(unsigned max_length);
/**
* Deinitializes a queue object. It does not free the queue
* pointer itself.
*/
~queue();
~Queue();
queue(const queue &other) = delete;
queue &operator=(const queue &other) = delete;
Queue(const Queue &) = delete;
Queue &operator=(const Queue &) = delete;
unsigned GetLength() const {
assert(length <= max_length);

@@ -38,7 +38,7 @@ extern "C" {
* @param end the index of the last song (excluding)
*/
static void
queue_print_song_info(Client &client, const queue &queue,
queue_print_song_info(Client &client, const Queue &queue,
unsigned position)
{
song_print_info(client, queue.Get(position));
@@ -51,7 +51,7 @@ queue_print_song_info(Client &client, const queue &queue,
}
void
queue_print_info(Client &client, const queue &queue,
queue_print_info(Client &client, const Queue &queue,
unsigned start, unsigned end)
{
assert(start <= end);
@@ -62,7 +62,7 @@ queue_print_info(Client &client, const queue &queue,
}
void
queue_print_uris(Client &client, const queue &queue,
queue_print_uris(Client &client, const Queue &queue,
unsigned start, unsigned end)
{
assert(start <= end);
@@ -75,7 +75,7 @@ queue_print_uris(Client &client, const queue &queue,
}
void
queue_print_changes_info(Client &client, const queue &queue,
queue_print_changes_info(Client &client, const Queue &queue,
uint32_t version)
{
for (unsigned i = 0; i < queue.GetLength(); i++) {
@@ -85,7 +85,7 @@ queue_print_changes_info(Client &client, const queue &queue,
}
void
queue_print_changes_position(Client &client, const queue &queue,
queue_print_changes_position(Client &client, const Queue &queue,
uint32_t version)
{
for (unsigned i = 0; i < queue.GetLength(); i++)
@@ -95,7 +95,7 @@ queue_print_changes_position(Client &client, const queue &queue,
}
void
queue_find(Client &client, const queue &queue,
queue_find(Client &client, const Queue &queue,
const SongFilter &filter)
{
for (unsigned i = 0; i < queue.GetLength(); i++) {

@@ -27,28 +27,28 @@
#include <stdint.h>
struct queue;
struct Queue;
class SongFilter;
class Client;
void
queue_print_info(Client &client, const queue &queue,
queue_print_info(Client &client, const Queue &queue,
unsigned start, unsigned end);
void
queue_print_uris(Client &client, const queue &queue,
queue_print_uris(Client &client, const Queue &queue,
unsigned start, unsigned end);
void
queue_print_changes_info(Client &client, const queue &queue,
queue_print_changes_info(Client &client, const Queue &queue,
uint32_t version);
void
queue_print_changes_position(Client &client, const queue &queue,
queue_print_changes_position(Client &client, const Queue &queue,
uint32_t version);
void
queue_find(Client &client, const queue &queue,
queue_find(Client &client, const Queue &queue,
const SongFilter &filter);
#endif

@@ -60,7 +60,7 @@ queue_save_song(FILE *fp, int idx, const Song &song)
}
void
queue_save(FILE *fp, const queue &queue)
queue_save(FILE *fp, const Queue &queue)
{
for (unsigned i = 0; i < queue.GetLength(); i++) {
uint8_t prio = queue.GetPriorityAtPosition(i);
@@ -72,7 +72,7 @@ queue_save(FILE *fp, const queue &queue)
}
void
queue_load_song(TextFile &file, const char *line, queue &queue)
queue_load_song(TextFile &file, const char *line, Queue &queue)
{
if (queue.IsFull())
return;

@@ -27,16 +27,16 @@
#include <stdio.h>
struct queue;
struct Queue;
class TextFile;
void
queue_save(FILE *fp, const queue &queue);
queue_save(FILE *fp, const Queue &queue);
/**
* Loads one song from the state file and appends it to the queue.
*/
void
queue_load_song(TextFile &file, const char *line, queue &queue);
queue_load_song(TextFile &file, const char *line, Queue &queue);
#endif

@@ -101,7 +101,7 @@ bool
SongFilter::Item::Match(const Tag &_tag) const
{
bool visited_types[TAG_NUM_OF_ITEM_TYPES];
std::fill_n(visited_types, TAG_NUM_OF_ITEM_TYPES, false);
std::fill_n(visited_types, size_t(TAG_NUM_OF_ITEM_TYPES), false);
for (unsigned i = 0; i < _tag.num_items; i++) {
visited_types[_tag.items[i]->type] = true;

@@ -95,8 +95,7 @@ Song::UpdateFile()
TagBuilder tag_builder;
if (!tag_file_scan(path_fs,
&full_tag_handler, &tag_builder) ||
!tag_builder.IsDefined())
&full_tag_handler, &tag_builder))
return false;
if (tag_builder.IsEmpty())

@@ -28,6 +28,9 @@ struct tag_handler;
/**
* Scan the tags of a song file. Invokes matching decoder plugins,
* but does not invoke the special "APE" and "ID3" scanners.
*
* @return true if the file was recognized (even if no metadata was
* found)
*/
bool
tag_file_scan(Path path,

@@ -90,7 +90,7 @@ static const struct command commands[] = {
{ "disableoutput", PERMISSION_ADMIN, 1, 1, handle_disableoutput },
{ "enableoutput", PERMISSION_ADMIN, 1, 1, handle_enableoutput },
{ "find", PERMISSION_READ, 2, -1, handle_find },
{ "findadd", PERMISSION_READ, 2, -1, handle_findadd},
{ "findadd", PERMISSION_ADD, 2, -1, handle_findadd},
{ "idle", PERMISSION_READ, 0, -1, handle_idle },
{ "kill", PERMISSION_ADMIN, -1, -1, handle_kill },
{ "list", PERMISSION_READ, 1, -1, handle_list },

@@ -393,6 +393,7 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
if (!ret) {
LogError(error);
NeAACDecClose(decoder);
decoder_buffer_free(buffer);
return;
}
@@ -461,6 +462,7 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
/* cleanup */
NeAACDecClose(decoder);
decoder_buffer_free(buffer);
}
static bool

@@ -38,7 +38,10 @@ extern "C" {
#include <libavutil/avutil.h>
#include <libavutil/log.h>
#include <libavutil/mathematics.h>
#include <libavutil/dict.h>
#if LIBAVUTIL_VERSION_MAJOR >= 53
#include <libavutil/frame.h>
#endif
}
#include <assert.h>
@@ -120,8 +123,7 @@ mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence)
if (whence == AVSEEK_SIZE)
return stream->input.size;
Error error;
if (!stream->input.LockSeek(pos, whence, error))
if (!stream->input.LockSeek(pos, whence, IgnoreError()))
return -1;
return stream->input.offset;
@@ -224,6 +226,9 @@ copy_interleave_frame(const AVCodecContext *codec_context,
codec_context->channels,
frame->nb_samples,
codec_context->sample_fmt, 1);
if (data_size <= 0)
return data_size;
if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
codec_context->channels > 1) {
if(*global_buffer_size < data_size) {
@@ -252,13 +257,14 @@ static DecoderCommand
ffmpeg_send_packet(Decoder &decoder, InputStream &is,
const AVPacket *packet,
AVCodecContext *codec_context,
const AVRational *time_base,
const AVStream *stream,
AVFrame *frame,
uint8_t **buffer, int *buffer_size)
{
if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE)
decoder_timestamp(decoder,
time_from_ffmpeg(packet->pts, *time_base));
time_from_ffmpeg(packet->pts - stream->start_time,
stream->time_base));
AVPacket packet2 = *packet;
@@ -342,11 +348,9 @@ ffmpeg_probe(Decoder *decoder, InputStream &is)
PADDING = 16,
};
Error error;
unsigned char buffer[BUFFER_SIZE];
size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE);
if (nbytes <= PADDING || !is.LockRewind(error))
if (nbytes <= PADDING || !is.LockRewind(IgnoreError()))
return nullptr;
/* some ffmpeg parsers (e.g. ac3_parser.c) read a few bytes
@@ -453,7 +457,11 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
decoder_initialized(decoder, audio_format,
input.seekable, total_time);
#if LIBAVUTIL_VERSION_MAJOR >= 53
AVFrame *frame = av_frame_alloc();
#else
AVFrame *frame = avcodec_alloc_frame();
#endif
if (!frame) {
LogError(ffmpeg_domain, "Could not allocate frame");
avformat_close_input(&format_context);
@@ -473,7 +481,7 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
if (packet.stream_index == audio_stream)
cmd = ffmpeg_send_packet(decoder, input,
&packet, codec_context,
&av_stream->time_base,
av_stream,
frame,
&interleaved_buffer, &interleaved_buffer_size);
else
@@ -484,7 +492,8 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
if (cmd == DecoderCommand::SEEK) {
int64_t where =
time_to_ffmpeg(decoder_seek_where(decoder),
av_stream->time_base);
av_stream->time_base) +
av_stream->start_time;
if (av_seek_frame(format_context, audio_stream, where,
AV_TIME_BASE) < 0)
@@ -496,7 +505,9 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
}
} while (cmd != DecoderCommand::STOP);
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0)
#if LIBAVUTIL_VERSION_MAJOR >= 53
av_frame_free(&frame);
#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0)
avcodec_free_frame(&frame);
#else
av_freep(&frame);

@@ -21,8 +21,6 @@
#define MPD_FFMPEG_METADATA_HXX
extern "C" {
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#include <libavutil/dict.h>
}
@@ -35,6 +33,6 @@ struct tag_handler;
void
ffmpeg_scan_dictionary(AVDictionary *dict,
const struct tag_handler *handler, void *handler_ctx);
const tag_handler *handler, void *handler_ctx);
#endif

@@ -39,6 +39,12 @@
#include <algorithm>
#ifdef USE_SIGNALFD
#include <pthread.h>
#endif
#include <signal.h>
class SignalMonitor final : private SocketMonitor {
#ifdef USE_SIGNALFD
SignalFD fd;
@@ -99,7 +105,21 @@ static std::atomic_bool signal_pending[MAX_SIGNAL];
static Manual<SignalMonitor> monitor;
#ifndef USE_SIGNALFD
#ifdef USE_SIGNALFD
/**
* This is a pthread_atfork() callback that unblocks the signals that
* were blocked for our signalfd(). Without this, our child processes
* would inherit the blocked signals.
*/
static void
at_fork_child()
{
sigprocmask(SIG_UNBLOCK, &signal_mask, nullptr);
}
#else
static void
SignalCallback(int signo)
{
@@ -108,6 +128,7 @@ SignalCallback(int signo)
if (!signal_pending[signo].exchange(true))
monitor->WakeUp();
}
#endif
void
@@ -115,6 +136,8 @@ SignalMonitorInit(EventLoop &loop)
{
#ifdef USE_SIGNALFD
sigemptyset(&signal_mask);
pthread_atfork(nullptr, nullptr, at_fork_child);
#endif
monitor.Construct(loop);

@@ -41,7 +41,12 @@
#include <glib.h>
#include <assert.h>
#ifdef HAVE_CDIO_PARANOIA_PARANOIA_H
#include <cdio/paranoia/paranoia.h>
#else
#include <cdio/paranoia.h>
#endif
#include <cdio/cd_types.h>
struct CdioParanoiaInputStream {

@@ -24,6 +24,7 @@
#include "Main.hxx"
#include "event/MultiSocketMonitor.hxx"
#include "event/Loop.hxx"
#include "event/Call.hxx"
#include "util/ASCII.hxx"
#include "util/ReusableArray.hxx"
#include "util/Error.hxx"
@@ -46,10 +47,22 @@ class AlsaMixerMonitor final : private MultiSocketMonitor {
public:
AlsaMixerMonitor(EventLoop &_loop, snd_mixer_t *_mixer)
:MultiSocketMonitor(_loop), mixer(_mixer) {
#ifdef USE_EPOLL
_loop.AddCall([this](){ InvalidateSockets(); });
#else
_loop.AddIdle(InitAlsaMixerMonitor, this);
#endif
}
private:
#ifndef USE_EPOLL
static gboolean InitAlsaMixerMonitor(gpointer data) {
AlsaMixerMonitor &amm = *(AlsaMixerMonitor *)data;
amm.InvalidateSockets();
return false;
}
#endif
virtual int PrepareSockets() override;
virtual void DispatchSockets() override;
};

@@ -106,12 +106,16 @@ struct AlsaOutput {
snd_pcm_uframes_t period_position;
/**
* Set to non-zero when the Raspberry Pi workaround has been
* activated in alsa_recover(); decremented by each write.
* This will avoid activating it again, leading to an endless
* loop. This problem was observed with a "RME Digi9636/52".
* Do we need to call snd_pcm_prepare() before the next write?
* It means that we put the device to SND_PCM_STATE_SETUP by
* calling snd_pcm_drop().
*
* Without this flag, we could easily recover after a failed
* optimistic write (returning -EBADFD), but the Raspberry Pi
* audio driver is infamous for generating ugly artefacts from
* this.
*/
unsigned pi_workaround;
bool must_prepare;
/**
* This buffer gets allocated after opening the ALSA device.
@@ -676,8 +680,6 @@ alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
{
AlsaOutput *ad = (AlsaOutput *)ao;
ad->pi_workaround = 0;
int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) {
@@ -699,6 +701,8 @@ alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
ad->in_frame_size = audio_format.GetFrameSize();
ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format);
ad->must_prepare = false;
return true;
}
@@ -736,29 +740,6 @@ alsa_recover(AlsaOutput *ad, int err)
case SND_PCM_STATE_XRUN:
ad->period_position = 0;
err = snd_pcm_prepare(ad->pcm);
if (err == 0 && ad->pi_workaround == 0) {
/* this works around a driver bug observed on
the Raspberry Pi: after snd_pcm_drop(), the
whole ring buffer must be invalidated, but
the snd_pcm_prepare() call above makes the
driver play random data that just happens
to be still in the buffer; by adding and
cancelling some silence, this bug does not
occur */
alsa_write_silence(ad, ad->period_frames);
/* cancel the silence data right away to avoid
increasing latency; even though this
function call invalidates the portion of
silence, the driver seems to avoid the
bug */
snd_pcm_reset(ad->pcm);
/* disable the workaround for some time */
ad->pi_workaround = 8;
}
break;
case SND_PCM_STATE_DISCONNECTED:
break;
@@ -801,6 +782,7 @@ alsa_cancel(struct audio_output *ao)
AlsaOutput *ad = (AlsaOutput *)ao;
ad->period_position = 0;
ad->must_prepare = true;
snd_pcm_drop(ad->pcm);
}
@@ -822,6 +804,16 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
assert(size % ad->in_frame_size == 0);
if (ad->must_prepare) {
ad->must_prepare = false;
int err = snd_pcm_prepare(ad->pcm);
if (err < 0) {
error.Set(alsa_output_domain, err, snd_strerror(-err));
return 0;
}
}
chunk = ad->pcm_export->Export(chunk, size, size);
assert(size % ad->out_frame_size == 0);
@@ -834,9 +826,6 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
ad->period_position = (ad->period_position + ret)
% ad->period_frames;
if (ad->pi_workaround > 0)
--ad->pi_workaround;
size_t bytes_written = ret * ad->out_frame_size;
return ad->pcm_export->CalcSourceSize(bytes_written);
}

@@ -363,9 +363,9 @@ osx_output_open(struct audio_output *ao, AudioFormat &audio_format,
OSStatus status = AudioUnitInitialize(od->au);
if (status != noErr) {
error.Set(osx_output_domain, status,
"Unable to initialize OS X audio unit: %s",
GetMacOSStatusCommentString(status));
error.Format(osx_output_domain, status,
"Unable to initialize OS X audio unit: %s",
GetMacOSStatusCommentString(status));
return false;
}

@@ -25,7 +25,7 @@
#include <glib.h>
#ifndef HAVE_OSX
#ifndef __APPLE__
#include <AL/al.h>
#include <AL/alc.h>
#else

@@ -369,8 +369,6 @@ pulse_output_enable(struct audio_output *ao, Error &error)
po->mainloop = pa_threaded_mainloop_new();
if (po->mainloop == nullptr) {
g_free(po);
error.Set(pulse_output_domain,
"pa_threaded_mainloop_new() has failed");
return false;

@@ -68,7 +68,6 @@ pls_parser(GKeyFile *keyfile, std::forward_list<SongPointer> &songs)
FormatError(pls_domain, "Invalid PLS entry %s: '%s'",
key, error->message);
g_error_free(error);
g_free(key);
return;
}

@@ -87,7 +87,8 @@ riff_seek_id3(FILE *file)
/* pad byte */
++size;
if (memcmp(chunk.id, "id3 ", 4) == 0)
if (memcmp(chunk.id, "id3 ", 4) == 0 ||
memcmp(chunk.id, "ID3 ", 4) == 0)
/* found it! */
return size;

@@ -39,7 +39,7 @@ TagLoadConfig()
if (value == nullptr)
return;
std::fill_n(ignore_tag_items, TAG_NUM_OF_ITEM_TYPES, true);
std::fill_n(ignore_tag_items, size_t(TAG_NUM_OF_ITEM_TYPES), true);
if (StringEqualsCaseASCII(value, "none"))
return;

@@ -24,6 +24,7 @@
#include "Compiler.h"
#include <string>
#include <algorithm>
#include <assert.h>

@@ -130,8 +130,9 @@ PeakBuffer::Append(const void *data, size_t length)
return true;
}
if (peak_buffer == nullptr && peak_size > 0) {
peak_buffer = (fifo_buffer *)HugeAllocate(peak_size);
if (peak_buffer == nullptr) {
if (peak_size > 0)
peak_buffer = (fifo_buffer *)HugeAllocate(peak_size);
if (peak_buffer == nullptr)
return false;

@@ -32,7 +32,8 @@ const char *
uri_get_suffix(const char *uri)
{
const char *suffix = strrchr(uri, '.');
if (suffix == nullptr)
if (suffix == nullptr || suffix == uri ||
suffix[-1] == '/' || suffix[-1] == '\\')
return nullptr;
++suffix;

85
test/test_icy_parser.cxx Normal file

@@ -0,0 +1,85 @@
/*
* Unit tests for class IcyMetaDataParser.
*/
#include "config.h"
/* include the .cxx file to get access to internal functions */
#include "IcyMetaDataParser.cxx"
#include <cppunit/TestFixture.h>
#include <cppunit/extensions/TestFactoryRegistry.h>
#include <cppunit/ui/text/TestRunner.h>
#include <cppunit/extensions/HelperMacros.h>
#include <string>
#include <string.h>
static Tag *
icy_parse_tag(const char *p)
{
char *q = strdup(p);
Tag *tag = icy_parse_tag(q, q + strlen(q));
free(q);
return tag;
}
static void
CompareTagTitle(const Tag &tag, const std::string &title)
{
CPPUNIT_ASSERT_EQUAL(1u, tag.num_items);
const TagItem &item = *tag.items[0];
CPPUNIT_ASSERT_EQUAL(TAG_TITLE, item.type);
CPPUNIT_ASSERT_EQUAL(title, std::string(item.value));
}
static void
TestIcyParserTitle(const char *input, const char *title)
{
Tag *tag = icy_parse_tag(input);
CompareTagTitle(*tag, title);
delete tag;
}
static void
TestIcyParserEmpty(const char *input)
{
Tag *tag = icy_parse_tag(input);
CPPUNIT_ASSERT_EQUAL(0u, tag->num_items);
delete tag;
}
class IcyTest : public CppUnit::TestFixture {
CPPUNIT_TEST_SUITE(IcyTest);
CPPUNIT_TEST(TestIcyMetadataParser);
CPPUNIT_TEST_SUITE_END();
public:
void TestIcyMetadataParser() {
TestIcyParserEmpty("foo=bar;");
TestIcyParserTitle("StreamTitle='foo bar'", "foo bar");
TestIcyParserTitle("StreamTitle='foo bar';", "foo bar");
TestIcyParserTitle("StreamTitle='foo\"bar';", "foo\"bar");
TestIcyParserTitle("StreamTitle='foo=bar';", "foo=bar");
TestIcyParserTitle("a=b;StreamTitle='foo';", "foo");
TestIcyParserTitle("a=;StreamTitle='foo';", "foo");
TestIcyParserTitle("a=b;StreamTitle='foo';c=d", "foo");
TestIcyParserTitle("a=b;StreamTitle='foo'", "foo");
TestIcyParserTitle("a='b;c';StreamTitle='foo;bar'", "foo;bar");
TestIcyParserTitle("a='b'c';StreamTitle='foo'bar'", "foo'bar");
TestIcyParserTitle("StreamTitle='fo'o'b'ar';a='b'c'd'", "fo'o'b'ar");
}
};
CPPUNIT_TEST_SUITE_REGISTRATION(IcyTest);
int
main(gcc_unused int argc, gcc_unused char **argv)
{
CppUnit::TextUi::TestRunner runner;
auto &registry = CppUnit::TestFactoryRegistry::getRegistry();
runner.addTest(registry.makeTest());
return runner.run() ? EXIT_SUCCESS : EXIT_FAILURE;
}

@@ -26,7 +26,7 @@ Song::Free()
}
static void
check_descending_priority(const struct queue *queue,
check_descending_priority(const Queue *queue,
unsigned start_order)
{
assert(start_order < queue->GetLength());
@@ -55,7 +55,7 @@ QueuePriorityTest::TestPriority()
{
static Song songs[16];
struct queue queue(32);
Queue queue(32);
for (unsigned i = 0; i < ARRAY_SIZE(songs); ++i)
queue.Append(&songs[i], 0);

@@ -29,6 +29,10 @@ public:
"jpg"));
CPPUNIT_ASSERT_EQUAL(0, strcmp(uri_get_suffix("/foo.png/bar.jpg"),
"jpg"));
CPPUNIT_ASSERT_EQUAL((const char *)nullptr,
uri_get_suffix(".jpg"));
CPPUNIT_ASSERT_EQUAL((const char *)nullptr,
uri_get_suffix("/foo/.jpg"));
}
void TestRemoveAuth() {