From e5cf9ba1fdbccde85e680a803dc69cf4b539174c Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Thu, 6 Apr 2023 11:49:25 +0200
Subject: [PATCH] modules: update docs

---
 src/modules/module-rtp-session.c | 2 +-
 src/modules/module-rtp-sink.c    | 4 ++--
 src/modules/module-rtp-source.c  | 6 +++---
 3 files changed, 6 insertions(+), 6 deletions(-)

diff --git a/src/modules/module-rtp-session.c b/src/modules/module-rtp-session.c
index c78d3178f..4c50bec11 100644
--- a/src/modules/module-rtp-session.c
+++ b/src/modules/module-rtp-session.c
@@ -137,7 +137,7 @@ PW_LOG_TOPIC_STATIC(mod_topic, "mod." NAME);
 		"( sess.name=<a name for the session> ) "					\
 		"( sess.min-ptime=<minimum packet time in milliseconds, default:2> ) "		\
 		"( sess.max-ptime=<maximum packet time in milliseconds, default:20> ) "		\
-		"( sess.media=<string, the media type audio|midi|opus, default audio> ) "	\
+		"( sess.media=<string, the media type audio|midi|opus, default midi> ) "	\
 		"( audio.format=<format, default:"DEFAULT_FORMAT"> ) "				\
 		"( audio.rate=<sample rate, default:"SPA_STRINGIFY(DEFAULT_RATE)"> ) "		\
 		"( audio.channels=<number of channels, default:"SPA_STRINGIFY(DEFAULT_CHANNELS)"> ) "\
diff --git a/src/modules/module-rtp-sink.c b/src/modules/module-rtp-sink.c
index eea70aaae..7dd218369 100644
--- a/src/modules/module-rtp-sink.c
+++ b/src/modules/module-rtp-sink.c
@@ -53,7 +53,7 @@
  * - `sess.name = <str>`: a session name
  * - `sess.ts-offset = <int>`: an offset to apply to the timestamp, default -1 = random offset
  * - `sess.ts-refclk = <string>`: the name of a reference clock
- * - `sess.media = <string>`: the media type audio|midi, default audio
+ * - `sess.media = <string>`: the media type audio|midi|opus, default audio
  * - `stream.props = {}`: properties to be passed to the stream
  *
  * ## General options
@@ -129,7 +129,7 @@ PW_LOG_TOPIC_STATIC(mod_topic, "mod." NAME);
 		"( sess.name=<a name for the session> ) "						\
 		"( sess.min-ptime=<minimum packet time in milliseconds, default:2> ) "			\
 		"( sess.max-ptime=<maximum packet time in milliseconds, default:20> ) "			\
- 		"( sess.media=<string, the media type audio|midi, default audio> ) "			\
+ 		"( sess.media=<string, the media type audio|midi|opus, default audio> ) "		\
 		"( audio.format=<format, default:"DEFAULT_FORMAT"> ) "					\
 		"( audio.rate=<sample rate, default:"SPA_STRINGIFY(DEFAULT_RATE)"> ) "			\
 		"( audio.channels=<number of channels, default:"SPA_STRINGIFY(DEFAULT_CHANNELS)"> ) "	\
diff --git a/src/modules/module-rtp-source.c b/src/modules/module-rtp-source.c
index 56bc14290..28867805e 100644
--- a/src/modules/module-rtp-source.c
+++ b/src/modules/module-rtp-source.c
@@ -46,7 +46,7 @@
  * - `local.ifname = <str>`: interface name to use
  * - `node.always-process = <bool>`: true to receive even when not running
  * - `sess.latency.msec = <str>`: target network latency in milliseconds, default 100
- * - `sess.media = <string>`: the media type audio|midi, default audio
+ * - `sess.media = <string>`: the media type audio|midi|opus, default audio
  * - `stream.props = {}`: properties to be passed to the stream
  *
  * ## General options
@@ -104,8 +104,8 @@ PW_LOG_TOPIC_STATIC(mod_topic, "mod." NAME);
 #define USAGE   "( local.ifname=<local interface name to use> ) "						\
 		"( source.ip=<source IP address, default:"DEFAULT_SOURCE_IP"> ) "				\
  		"source.port=<int, source port> "								\
-		"( sess.latency.msec=<target network latency, default "SPA_STRINGIFY(DEFAULT_SESS_LATENCY)"> ) "	\
- 		"( sess.media=<string, the media type audio|midi, default audio> ) "				\
+		"( sess.latency.msec=<target network latency, default "SPA_STRINGIFY(DEFAULT_SESS_LATENCY)"> ) "\
+ 		"( sess.media=<string, the media type audio|midi|opus, default audio> ) "			\
 		"( audio.format=<format, default:"DEFAULT_FORMAT"> ) "						\
 		"( audio.rate=<sample rate, default:"SPA_STRINGIFY(DEFAULT_RATE)"> ) "				\
 		"( audio.channels=<number of channels, default:"SPA_STRINGIFY(DEFAULT_CHANNELS)"> ) "		\