[Asterisk-code-review] core: Add VP9 passthrough support. (asterisk[master])

[Joshua Colp]

Joshua Colp has uploaded this change for review. ( https://gerrit.asterisk.org/6080


Change subject: core: Add VP9 passthrough support.
......................................................................

core: Add VP9 passthrough support.

This change adds VP9 as a known codec and creates a cached
"vp9" media format for use.

Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc
---
M channels/chan_pjsip.c
M include/asterisk/format_cache.h
M main/codec_builtin.c
M main/format_cache.c
M main/rtp_engine.c
5 files changed, 26 insertions(+), 2 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/80/6080/1

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index f009943..51b5dab 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1596,6 +1596,7 @@
 				 * fully support other video codecs */
 
 				if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL ||
+					ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp9) != AST_FORMAT_CMP_NOT_EQUAL ||
 					(channel->session->endpoint->media.webrtc &&
 					 ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h264) != AST_FORMAT_CMP_NOT_EQUAL)) {
 					/* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
diff --git a/include/asterisk/format_cache.h b/include/asterisk/format_cache.h
index 92272e8..0675463 100644
--- a/include/asterisk/format_cache.h
+++ b/include/asterisk/format_cache.h
@@ -184,6 +184,11 @@
 extern struct ast_format *ast_format_vp8;
 
 /*!
+ * \brief Built-in cached vp9 format.
+ */
+extern struct ast_format *ast_format_vp9;
+
+/*!
  * \brief Built-in cached jpeg format.
  */
 extern struct ast_format *ast_format_jpeg;
diff --git a/main/codec_builtin.c b/main/codec_builtin.c
index 32ec12d..25aa513 100644
--- a/main/codec_builtin.c
+++ b/main/codec_builtin.c
@@ -820,6 +820,13 @@
 	.sample_rate = 1000,
 };
 
+static struct ast_codec vp9 = {
+	.name = "vp9",
+	.description = "VP9 video",
+	.type = AST_MEDIA_TYPE_VIDEO,
+	.sample_rate = 1000,
+};
+
 static struct ast_codec t140red = {
 	.name = "red",
 	.description = "T.140 Realtime Text with redundancy",
@@ -966,6 +973,7 @@
 	res |= CODEC_REGISTER_AND_CACHE(h264);
 	res |= CODEC_REGISTER_AND_CACHE(mpeg4);
 	res |= CODEC_REGISTER_AND_CACHE(vp8);
+	res |= CODEC_REGISTER_AND_CACHE(vp9);
 	res |= CODEC_REGISTER_AND_CACHE(t140red);
 	res |= CODEC_REGISTER_AND_CACHE(t140);
 	res |= CODEC_REGISTER_AND_CACHE(t38);
diff --git a/main/format_cache.c b/main/format_cache.c
index 302bbf8..1a67ebe 100644
--- a/main/format_cache.c
+++ b/main/format_cache.c
@@ -191,6 +191,11 @@
 struct ast_format *ast_format_vp8;
 
 /*!
+ * \brief Built-in cached vp9 format.
+ */
+struct ast_format *ast_format_vp9;
+
+/*!
  * \brief Built-in cached jpeg format.
  */
 struct ast_format *ast_format_jpeg;
@@ -345,6 +350,7 @@
 	ao2_replace(ast_format_h264, NULL);
 	ao2_replace(ast_format_mp4, NULL);
 	ao2_replace(ast_format_vp8, NULL);
+	ao2_replace(ast_format_vp9, NULL);
 	ao2_replace(ast_format_t140_red, NULL);
 	ao2_replace(ast_format_t140, NULL);
 	ao2_replace(ast_format_t38, NULL);
@@ -444,6 +450,8 @@
 		ao2_replace(ast_format_mp4, format);
 	} else if (!strcmp(name, "vp8")) {
 		ao2_replace(ast_format_vp8, format);
+	} else if (!strcmp(name, "vp9")) {
+		ao2_replace(ast_format_vp9, format);
 	} else if (!strcmp(name, "red")) {
 		ao2_replace(ast_format_t140_red, format);
 	} else if (!strcmp(name, "t140")) {
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index abd4b1f..fe60c4e 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -3268,9 +3268,10 @@
 	set_next_mime_type(ast_format_siren7, 0, "audio", "G7221", 16000);
 	set_next_mime_type(ast_format_siren14, 0, "audio", "G7221", 32000);
 	set_next_mime_type(ast_format_g719, 0, "audio", "G719", 48000);
-	/* Opus and VP8 */
+	/* Opus, VP8, and VP9 */
 	set_next_mime_type(ast_format_opus, 0,  "audio", "opus", 48000);
 	set_next_mime_type(ast_format_vp8, 0,  "video", "VP8", 90000);
+	set_next_mime_type(ast_format_vp9, 0, "video", "VP9", 90000);
 
 	/* Define the static rtp payload mappings */
 	add_static_payload(0, ast_format_ulaw, 0);
@@ -3311,6 +3312,7 @@
 	add_static_payload(105, ast_format_t140_red, 0);   /* Real time text chat (with redundancy encoding) */
 	add_static_payload(106, ast_format_t140, 0);     /* Real time text chat */
 	add_static_payload(107, ast_format_opus, 0);
+	add_static_payload(108, ast_format_vp9, 0);
 
 	add_static_payload(110, ast_format_speex, 0);
 	add_static_payload(111, ast_format_g726, 0);
@@ -3413,4 +3415,4 @@
 		rtp->engine->set_stream_num(rtp, stream_num);
 	}
 	ao2_unlock(rtp);
-}
\ No newline at end of file
+}

-- 
To view, visit https://gerrit.asterisk.org/6080
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc
Gerrit-Change-Number: 6080
Gerrit-PatchSet: 1
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20170724/bd31bcf3/attachment-0001.html>