[gst-cvs] gst-plugins-good: rtpsirenpay: Put the bitrate in the RTP caps

[Olivier Crête]

Module: gst-plugins-good
Branch: master
Commit: dc61a71e5d8052fc5f270fd8478c44fd3796cf99
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=dc61a71e5d8052fc5f270fd8478c44fd3796cf99

Author: Olivier Crête <olivier.crete at collabora.co.uk>
Date:   Tue Jun  9 14:13:31 2009 -0400

rtpsirenpay: Put the bitrate in the RTP caps

The MS code seems to require the bitrate to interoperate and
draft-ietf-avt-rtp-g7221-00 also has it.

---

 gst/rtp/gstrtpsirenpay.c |    3 ++-
 1 files changed, 2 insertions(+), 1 deletions(-)

diff --git a/gst/rtp/gstrtpsirenpay.c b/gst/rtp/gstrtpsirenpay.c
index b03281b..211663e 100644
--- a/gst/rtp/gstrtpsirenpay.c
+++ b/gst/rtp/gstrtpsirenpay.c
@@ -52,7 +52,8 @@ GST_STATIC_PAD_TEMPLATE ("src",
         "media = (string) \"audio\", "
         "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
         "clock-rate = (int) 16000, "
-        "encoding-name = (string) \"SIREN\", " "dct-length = (int) 320")
+        "encoding-name = (string) \"SIREN\", "
+        "bitrate = (string) \"16000\", " "dct-length = (int) 320")
     );
 
 static gboolean gst_rtpsirenpay_setcaps (GstBaseRTPPayload * payload,