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Fix the final few occurences of polyp.

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git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1042 fefdeb5f-60dc-0310-8127-8f9354f1896f
This commit is contained in:
Pierre Ossman 2006-06-20 13:02:34 +00:00
parent 230f97a4a4
commit 3ff68bd7a5
6 changed files with 16 additions and 16 deletions
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20
doc/modules.html.in
View file

@ -312,12 +312,12 @@ as <tt>module-cli-protocol-tcp</tt>.</p>
<h3>module-x11-publish</h3>
<p>Publishes the access credentials to the Polypaudio server in the
<p>Publishes the access credentials to the PulseAudio server in the
X11 root window. The following properties are used:
<tt>PULSE_SERVER</tt>, <tt>POYLP_SINK</tt>, <tt>PULSE_SOURCE</tt>,
<tt>PULSE_COOKIE</tt>. This is very useful when using SSH or any other
remote login tool for logging into other machines and getting audio
playback to your local speakers. The Polypaudio client libraries make
playback to your local speakers. The PulseAudio client libraries make
use of this data automatically. Instead of using this module you may
use the tool <tt>pax11publish</tt> which may be used to access, modify
and import credential data from/to the X11 display.</p>
@ -354,7 +354,7 @@ already loaded protocol module is used.</td></tr> </table>
<a name="rtp"/>
<h2>RTP/SDP/SAP Transport</h2>
<p>Polypaudio can stream audio data to an IP multicast group via the
<p>PulseAudio can stream audio data to an IP multicast group via the
standard protocols <a
href="http://en.wikipedia.org/wiki/Real-time_Transport_Protocol">RTP</a>,
<a
@ -373,7 +373,7 @@ supported. Only uncompressed audio is supported, hence you won't be
able to multicast more than a few streams at the same time over a
standard LAN.</p>
<p>Polypaudio implements both a sender and a reciever for RTP
<p>PulseAudio implements both a sender and a reciever for RTP
traffic. The sender announces itself via SAP/SDP on the same multicast
group as it sends the RTP data to. The reciever picks up the SAP/SDP
announcements and creates a playback stream for each
@ -418,26 +418,26 @@ module to create an RTP source.</p>
<h2>JACK Connectivity</h2>
<p>Polypaudio can be hooked up to a <a
<p>PulseAudio can be hooked up to a <a
href="http://jackit.sourceforge.net/">JACK Audio Connection Kit</a> server which is a specialized sound server used for professional audio production on Unix/Linux. Both a
Polypaudio sink and a source are available. For each channel a port is
PulseAudio sink and a source are available. For each channel a port is
created in the JACK server.</p>
<h3>module-jack-sink</h3>
<p>This module implements a Polypaudio sink that connects to JACK and registers as many output ports as requested.</p>
<p>This module implements a PulseAudio sink that connects to JACK and registers as many output ports as requested.</p>
<table>
<tr><td><tt>sink_name=</tt></td><td>The name for the Polypaudio sink. If ommited defaults to <tt>jack_out</tt>.</td></tr>
<tr><td><tt>sink_name=</tt></td><td>The name for the PulseAudio sink. If ommited defaults to <tt>jack_out</tt>.</td></tr>
<tr><td><tt>server_name=</tt></td><td>The JACK server to connect to. If ommited defaults to the default server.</td></tr>
<tr><td><tt>client_name=</tt></td><td>The client name to tell the JACK server. If ommited defaults to <tt>PulseAudio</tt>.</td></tr>
<tr><td><tt>channels=</tt></td><td>Number of channels to register. If ommited defaults to the number of physical playback ports of the JACK server.</td></tr>
<tr><td><tt>connect=</tt></td><td>Takes a boolean value. If enabled (the default) Polypaudio will try to connect its ports to the physicial playback ports of the JACK server</td></tr>
<tr><td><tt>connect=</tt></td><td>Takes a boolean value. If enabled (the default) PulseAudio will try to connect its ports to the physicial playback ports of the JACK server</td></tr>
</table>
<h3>module-jack-source</h3>
<p>This module implements a Polypaudio source that connects to JACK
<p>This module implements a PulseAudio source that connects to JACK
and registers as many input ports as requested. Takes the same
arguments as <tt>module-jack-sink</tt>, except for <tt>sink_name</tt>
which is replaced by <tt>source_name</tt> (with a default of <tt>jack_in</tt>) for obvious reasons.</p>

2
src/daemon/default.pa.in
View file

@ -1,4 +1,4 @@
#!@POLYPAUDIO_BINARY@ -nF
#!@PULSEAUDIO_BINARY@ -nF
#
# This file is part of PulseAudio.

2
src/daemon/esdcompat.in
View file

@ -95,4 +95,4 @@ EOF
shift
done
eval "exec '@POLYPAUDIO_BINARY@'$ARGS"
eval "exec '@PULSEAUDIO_BINARY@'$ARGS"

2
src/modules/rtp/module-rtp-send.c
View file

@ -298,7 +298,7 @@ int pa__init(pa_core *c, pa_module*m) {
r = getsockname(fd, (struct sockaddr*) &sa_dst, &k);
assert(r >= 0);
n = pa_sprintf_malloc("Polypaudio RTP Stream on %s", pa_get_fqdn(hn, sizeof(hn)));
n = pa_sprintf_malloc("PulseAudio RTP Stream on %s", pa_get_fqdn(hn, sizeof(hn)));
p = pa_sdp_build(af,
af == AF_INET ? (void*) &((struct sockaddr_in*) &sa_dst)->sin_addr : (void*) &((struct sockaddr_in6*) &sa_dst)->sin6_addr,

2
src/pulse/client.conf.in
View file

@ -21,7 +21,7 @@
## commented out. Use either ; or # for commenting
## Path to the pulseaudio daemon to run when autospawning.
; daemon-binary = @POLYPAUDIO_BINARY@
; daemon-binary = @PULSEAUDIO_BINARY@
## Extra arguments to pass to the pulseaudio daemon
; extra-arguments = --log-target=syslog --exit-idle-time=5

4
src/utils/padsp
View file

@ -22,8 +22,8 @@
while getopts 'hs:n:m:MSDd' param ; do
case $param in
s)
POLYP_SERVER="$OPTARG"
export POLYP_SERVER
PULSE_SERVER="$OPTARG"
export PULSE_SERVER
;;
n)
PADSP_CLIENT_NAME="$OPTARG"