[asterisk-scf-commits] asterisk-scf/release/slice.git branch "master" updated.
Commits to the Asterisk SCF project code repositories
asterisk-scf-commits at lists.digium.com
Wed Sep 14 17:43:48 CDT 2011
branch "master" has been updated
via f7a6cfa0ce860cca6407d11ddf57699f117467fe (commit)
via 66bcc079b7ec83fb484756c9da3d09d160ab1cbd (commit)
via 08dce549dc9525de99a212081275b33e23d60b0b (commit)
via 2f28b8f7396b52bfd3a4298f65d798df599671a9 (commit)
via fde12f1f0f75904a355fe1db2c591da2fc6210cd (commit)
via e449cfaa1657a1299e3312a6ad5a5a4e2b6e1b0f (commit)
via 8c7018ca46f1c84877a451dacadcb8c3fee5ec01 (commit)
from ff44d0e1ff76a086e4851c15604cf9d1ba9ad3fc (commit)
Summary of changes:
.../HookIf.ice => Media/Formats/OtherFormats.ice} | 43 ++++++++++----------
slice/AsteriskSCF/Media/MediaIf.ice | 8 ++++
slice/AsteriskSCF/Media/RTP/MediaRTPIf.ice | 2 +-
.../SessionCommunications/TelephonyEventsIf.ice | 17 +++++++-
4 files changed, 47 insertions(+), 23 deletions(-)
copy slice/AsteriskSCF/{System/Hook/HookIf.ice => Media/Formats/OtherFormats.ice} (53%)
- Log -----------------------------------------------------------------
commit f7a6cfa0ce860cca6407d11ddf57699f117467fe
Author: Mark Michelson <mmichelson at digium.com>
Date: Wed Sep 14 16:51:01 2011 -0500
Merging changes to MediaRTPIf.ice
diff --git a/slice/AsteriskSCF/Media/RTP/MediaRTPIf.ice b/slice/AsteriskSCF/Media/RTP/MediaRTPIf.ice
index 5794b31..5b3584a 100644
--- a/slice/AsteriskSCF/Media/RTP/MediaRTPIf.ice
+++ b/slice/AsteriskSCF/Media/RTP/MediaRTPIf.ice
@@ -279,7 +279,7 @@ module V1
* @throws SessionAllocationFailure if the media service is unable to allocate a session
* to match the provided parameters.
*/
- RTPSession* allocate(RTPServiceLocatorParams params) throws SessionAllocationFailure;
+ RTPSession* allocate(RTPServiceLocatorParams params, RTPOptions options, out RTPAllocationOutputs outputs) throws SessionAllocationFailure;
};
}; /* end module V1 */
commit 66bcc079b7ec83fb484756c9da3d09d160ab1cbd
Merge: 08dce54 fde12f1
Author: Mark Michelson <mmichelson at digium.com>
Date: Wed Sep 14 16:50:47 2011 -0500
Merge branch 'dtmf'
Conflicts:
slice/AsteriskSCF/SessionCommunications/SessionCommunicationsIf.ice
commit 08dce549dc9525de99a212081275b33e23d60b0b
Merge: ff44d0e 2f28b8f
Author: Mark Michelson <mmichelson at digium.com>
Date: Wed Sep 14 11:02:18 2011 -0500
Merge branch 'media-operation'
commit 2f28b8f7396b52bfd3a4298f65d798df599671a9
Merge: 91f56f1 72ed2a9
Author: Mark Michelson <mmichelson at digium.com>
Date: Tue Sep 13 12:01:56 2011 -0500
Merge branch 'master' into media-operation
commit fde12f1f0f75904a355fe1db2c591da2fc6210cd
Author: Mark Michelson <mmichelson at digium.com>
Date: Wed Aug 10 09:49:26 2011 -0500
Define a constant for the name of the RFC4733 format.
diff --git a/slice/AsteriskSCF/Media/Formats/OtherFormats.ice b/slice/AsteriskSCF/Media/Formats/OtherFormats.ice
index bc10658..65a76fe 100644
--- a/slice/AsteriskSCF/Media/Formats/OtherFormats.ice
+++ b/slice/AsteriskSCF/Media/Formats/OtherFormats.ice
@@ -32,6 +32,8 @@ module Other
module V1
{
+const string RFC4733Name = "rfc4733";
+
/**
* RFC 4733 telephone event audio format
*/
commit e449cfaa1657a1299e3312a6ad5a5a4e2b6e1b0f
Author: Mark Michelson <mmichelson at digium.com>
Date: Mon Aug 8 17:07:22 2011 -0500
Add DTMF continuation event.
diff --git a/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice b/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice
index 11acb78..ee7bc68 100644
--- a/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice
+++ b/slice/AsteriskSCF/SessionCommunications/TelephonyEventsIf.ice
@@ -40,6 +40,21 @@ module V1
};
/**
+ * Indicates the continuation of a DTMF press.
+ */
+ unsliceable class ContinueDTMFEvent extends TelephonyEvent
+ {
+ /**
+ * The ASCII value of the DTMF that is being pressed.
+ */
+ byte signal;
+ /**
+ * The cumulative duration of the DTMF press in milliseconds.
+ */
+ int duration;
+ };
+
+ /**
* Indicates the end of a DTMF press.
*
* With certain methods of conveying DTMF, such as SIP INFO,
@@ -53,7 +68,7 @@ module V1
*/
byte signal;
/**
- * The duration of the key press in milliseconds.
+ * The total duration of the key press in milliseconds.
*/
int duration;
};
commit 8c7018ca46f1c84877a451dacadcb8c3fee5ec01
Author: Mark Michelson <mmichelson at digium.com>
Date: Fri Aug 5 09:33:17 2011 -0500
Add an RFC 4733 format.
diff --git a/slice/AsteriskSCF/Media/Formats/OtherFormats.ice b/slice/AsteriskSCF/Media/Formats/OtherFormats.ice
new file mode 100644
index 0000000..bc10658
--- /dev/null
+++ b/slice/AsteriskSCF/Media/Formats/OtherFormats.ice
@@ -0,0 +1,52 @@
+/*
+ * Asterisk SCF -- An open-source communications framework.
+ *
+ * Copyright (C) 2011, Digium, Inc.
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk SCF project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE.txt file
+ * at the top of the source tree.
+ */
+
+#include <AsteriskSCF/Media/MediaIf.ice>
+
+module AsteriskSCF
+{
+
+module Media
+{
+
+module Formats
+{
+
+module Other
+{
+
+["suppress"]
+module V1
+{
+
+/**
+ * RFC 4733 telephone event audio format
+ */
+unsliceable class RFC4733 extends AsteriskSCF::Media::V1::TelephoneEventFormat
+{
+ //XXX Are there any parameters we'd want to put here? Like the events
+ //we support?
+};
+
+}; /* end module V1 */
+
+}; /* end module Other */
+
+}; /* end module Formats */
+
+}; /* end module Media */
+
+}; /* end module AsteriskSCF */
diff --git a/slice/AsteriskSCF/Media/MediaIf.ice b/slice/AsteriskSCF/Media/MediaIf.ice
index f9bb522..8acc4dc 100644
--- a/slice/AsteriskSCF/Media/MediaIf.ice
+++ b/slice/AsteriskSCF/Media/MediaIf.ice
@@ -464,6 +464,14 @@ module V1
int maximumBitrate;
};
+ /**
+ * An additional format class which is provided for telephone events
+ */
+ class TelephoneEventFormat extends Format
+ {
+ //XXX Anything we need to define for this??
+ };
+
}; /* end module V1 */
}; /* end module Media */
diff --git a/slice/AsteriskSCF/SessionCommunications/SessionCommunicationsIf.ice b/slice/AsteriskSCF/SessionCommunications/SessionCommunicationsIf.ice
index 07f2ead..1f16f64 100644
--- a/slice/AsteriskSCF/SessionCommunications/SessionCommunicationsIf.ice
+++ b/slice/AsteriskSCF/SessionCommunications/SessionCommunicationsIf.ice
@@ -567,192 +567,6 @@ module V1
*/
interface TelephonySession extends Session
{
- TelephonyEventsSourceSeq getSources();
- TelephonyEventsSinkSeq getSinks();
- };
-
- unsliceable class TelephonyEvent
- {
- };
-
- unsliceable class BeginDTMFEvent extends TelephonyEvent
- {
- byte digit;
- };
-
- unsliceable class EndDTMFEvent extends TelephonyEvent
- {
- byte digit;
- int duration;
- };
-
- interface TelephonyEventSink;
- sequence<TelephonyEventSink> TelephonyEventSinkSeq;
-
- /**
- * A source for telephony events
- */
- interface TelephonyEventSource
- {
- /**
- * Add a new sink to send telephony events to
- */
- idempotent void addSink(TelephonyEventSink* sink);
- /**
- * Get a list of the sinks to which this source sends events
- */
- idempotent TelephonyEventSinkSeq getSinks();
- };
-
- sequence<TelephonyEventSource> TelephonyEventSourceSeq;
-
- /**
- * A place to send telephony events
- */
- interface TelephonyEventSink
- {
- /**
- * Send a telephony event to this sink
- */
- void write(TelephonyEvent event);
- /**
- * Set the source for this sink
- */
- idempotent void setSource(TelephonyEventSource* source);
- /**
- * Get the source for this sink
- */
- idempotent TelephonyEventSource* getSource();
- };
-
- /**
- * A telephony session is a session that is with a telephone or telephone
- * network. It has events associated with it that do not occur with other types
- * of sessions.
- */
- interface TelephonySession extends Session
- {
- TelephonyEventSourceSeq getSources();
- TelephonyEventSinkSeq getSinks();
- };
-
- unsliceable class TelephonyEvent
- {
- };
-
- unsliceable class BeginDTMFEvent extends TelephonyEvent
- {
- byte digit;
- };
-
- unsliceable class EndDTMFEvent extends TelephonyEvent
- {
- byte digit;
- int duration;
- };
-
- interface TelephonyEventsSource
- {
- idempotent void addSink(TelephonyEventsSink* sink);
- idempotent TelephonyEventSink* getSinks();
- };
-
- interface TelephonyEventsSink
- {
- void write(TelephonyEvent event);
- idempotent void setSource(TelephonyEventsSource* source);
- idempotent TelephonyEventsSource* getSource();
- };
-
- /**
- * A telephony session is a session that is with a telephone or telephone
- * network. It has events associated with it that do not occur with other types
- * of sessions.
- */
- interface TelephonySession extends Session
- {
- /**
- * Get a sequence of the telephony event sources for this session
- */
- ["amd"] TelephonyEventSourceSeq getSources();
- /**
- * Get a sequence of the telephony event sinks for this session
- */
- ["amd"] TelephonyEventSinkSeq getSinks();
- };
-
- unsliceable class TelephonyEvent
- {
- };
-
- unsliceable class BeginDTMFEvent extends TelephonyEvent
- {
- byte digit;
- };
-
- unsliceable class EndDTMFEvent extends TelephonyEvent
- {
- byte digit;
- int duration;
- };
-
- interface TelephonyEventSink;
- sequence<TelephonyEventSink> TelephonyEventSinkSeq;
-
- /**
- * A source for telephony events
- */
- interface TelephonyEventSource
- {
- /**
- * Add a new sink to send telephony events to
- */
- idempotent void addSink(TelephonyEventSink* sink);
- /**
- * Get a list of the sinks to which this source sends events
- */
- idempotent TelephonyEventSinkSeq getSinks();
- };
-
- sequence<TelephonyEventSource> TelephonyEventSourceSeq;
-
- /**
- * A place to send telephony events
- */
- interface TelephonyEventSink
- {
- /**
- * Send a telephony event to this sink
- */
- void write(TelephonyEvent event);
- /**
- * Set the source for this sink
- */
- idempotent void setSource(TelephonyEventSource* source);
- /**
- * Get the source for this sink
- */
- idempotent TelephonyEventSource* getSource();
- };
-
- /**
- * A telephony session is a session that is with a telephone or telephone
- * network. It has events associated with it that do not occur with other types
- * of sessions.
- */
- interface TelephonySession extends Session
- {
- TelephonyEventSourceSeq getSources();
- TelephonyEventSinkSeq getSinks();
- };
-
- /**
- * A telephony session is a session that is with a telephone or telephone
- * network. It has events associated with it that do not occur with other types
- * of sessions.
- */
- interface TelephonySession extends Session
- {
/**
* Get a sequence of the telephony event sources for this session
*/
-----------------------------------------------------------------------
--
asterisk-scf/release/slice.git
More information about the asterisk-scf-commits
mailing list