Commit c7b1038a authored by Max Kellermann's avatar Max Kellermann

Merge branch 'v0.18.x'

parents d43aa129 c170fed6
......@@ -31,6 +31,7 @@ libtool
ltmain.sh
missing
mkinstalldirs
/test-driver
mpd
mpd.service
stamp-h1
......
......@@ -3,6 +3,16 @@ ver 0.19 (not yet released)
- new commands "addtagid", "cleartagid"
* new resampler option using libsoxr
ver 0.18.6 (not yet released)
* input
- cdio_paranoia: support libcdio-paranoia 0.90
* output
- openal: fix build failure on Mac OS X
- osx: fix build failure
* mixer
- alsa: fix build failure with uClibc
* accept files without metadata
ver 0.18.5 (2013/11/23)
* configuration
- fix crash when db_file is configured without music_directory
......
......@@ -827,6 +827,7 @@ 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)
......
......@@ -316,137 +316,6 @@ 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
......
......@@ -1555,6 +1555,59 @@ 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>
......@@ -1565,6 +1618,38 @@ 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>
......@@ -2042,6 +2127,18 @@ systemctl start mpd.socket</programlisting>
</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>
<parameter>URI</parameter>
</entry>
......
......@@ -89,8 +89,7 @@ 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,
......
......@@ -119,8 +119,7 @@ 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;
......@@ -341,11 +340,9 @@ 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
......
......@@ -42,7 +42,12 @@
#include <glib.h>
#include <assert.h>
#ifdef HAVE_CDIO_PARANOIA_PARANOIA_H
#include <cdio/parannoia/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"
......@@ -46,7 +47,7 @@ class AlsaMixerMonitor final : private MultiSocketMonitor {
public:
AlsaMixerMonitor(EventLoop &_loop, snd_mixer_t *_mixer)
:MultiSocketMonitor(_loop), mixer(_mixer) {
_loop.AddCall([this](){ InvalidateSockets(); });
BlockingCall(_loop, [this](){ InvalidateSockets(); });
}
private:
......
......@@ -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
......
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