Attachment 39238 Details for Bug 30662 – Updated spec patch

[patch] Updated spec patch

0001-StreamHandler-Add-signal-specifying-the-type-of-DTMF.patch (text/plain), 2.90 KB, created by Olivier Crête on 2010-10-06 09:46:08 UTC

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Creator: Olivier Crête

Created: 2010-10-06 09:46:08 UTC

Size: 2.90 KB

patch

obsolete

>From 61826e4feeb876785147d6e054728de25c1567a3 Mon Sep 17 00:00:00 2001
>From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= <tester@tester.ca>
>Date: Wed, 6 Oct 2010 12:36:45 -0400
>Subject: [PATCH] StreamHandler: Add signal specifying the type of DTMF event to send
>
>Some protocols specify the type of DTMF event send and the payload type
>just at the time when the are sent. This signal  should make these easier to implement.
>---
> Makefile                      |    2 +-
> spec/Media_Stream_Handler.xml |   34 ++++++++++++++++++++++++++++++++++
> 2 files changed, 35 insertions(+), 1 deletions(-)
>
>diff --git a/Makefile b/Makefile
>index c0b75df..ab9263e 100644
>--- a/Makefile
>+++ b/Makefile
>@@ -7,7 +7,7 @@ XSLTPROC = xsltproc --xinclude --nonet
> CANONXML = xmllint --nsclean --noblanks --c14n --nonet
> XML_LINEBREAKS = perl -pe 's/>/>\n/g'
> DROP_NAMESPACE = perl -pe '$$hash = chr(35); s{xmlns:tp="http://telepathy\.freedesktop\.org/wiki/DbusSpec$${hash}extensions-v0"}{}g'
>-RST2HTML = rst2html
>+RST2HTML = rst2html.py
> PYTHON = python
> 
> XMLS = $(wildcard spec/*.xml)
>diff --git a/spec/Media_Stream_Handler.xml b/spec/Media_Stream_Handler.xml
>index c9ae78f..2f08fa2 100644
>--- a/spec/Media_Stream_Handler.xml
>+++ b/spec/Media_Stream_Handler.xml
>@@ -523,6 +523,40 @@ Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.</
>         over this stream until StopTelephonyEvent is called.
>       </tp:docstring>
>     </signal>
>+    <tp:enum name="Media_Stream_Telephony_Event_Type" type="u">
>+      <tp:enumvalue suffix="Auto" value="0">
>+        <tp:docstring>Automatically chose the type of event to send</tp:docstring>
>+      </tp:enumvalue>
>+      <tp:enumvalue suffix="RTP_RFC4733" value="1">
>+        <tp:docstring>Send using RTP packets as defined in RFC 4733</tp:docstring>
>+      </tp:enumvalue>
>+      <tp:enumvalue suffix="Sound" value="2">
>+        <tp:docstring>Send as sound</tp:docstring>
>+      </tp:enumvalue>
>+    </tp:enum>
>+    <signal name="StartTelephonyEventFull"
>+      tp:name-for-bindings="Start_Telephony_Event_Full">
>+      <arg name="Event" type="y">
>+        <tp:docstring>
>+          A telephony event code as defined by RFC 4733.
>+        </tp:docstring>
>+      </arg>
>+      <arg name="Type" type="u" tp:type="Media_Stream_Telephony_Event_Type">
>+        <tp:docstring>
>+          The type of event to send
>+        </tp:docstring>
>+      </arg>
>+      <arg name="Codec_ID" type="u">
>+        <tp:docstring>
>+          The payload type to sent events of Type "RTP_RFC473
>+        </tp:docstring>
>+      </arg>
>+      <tp:docstring>
>+        Request that a telephony event (as defined by RFC 4733) is transmitted
>+        over this stream until StopTelephonyEvent is called. This differs from
>+        StartTelephonyEvent in that you can specify how to send the event.
>+      </tp:docstring>
>+    </signal>
>     <signal name="StopTelephonyEvent"
>       tp:name-for-bindings="Stop_Telephony_Event">
>       <tp:docstring>
>-- 
>1.7.2.2
>

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