Index: webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc |
index 167f29ee956ed5000c0da40007aa3a1867e954d4..b5bdd9af8c0a563ce0fe3840d45b236860a21570 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc |
@@ -196,4 +196,29 @@ bool PlayoutDelayLimits::Write(uint8_t* data, |
return true; |
} |
+// Video Content Type. |
+// |
+// E.g. default video or screenshare. |
+// |
+// 0 1 |
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+// | ID | len=0 | Content type | |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+constexpr RTPExtensionType VideoContentTypeExtension::kId; |
+constexpr uint8_t VideoContentTypeExtension::kValueSizeBytes; |
+constexpr const char* VideoContentTypeExtension::kUri; |
+ |
+bool VideoContentTypeExtension::Parse(const uint8_t* data, |
+ VideoContentType* content_type) { |
+ *content_type = static_cast<VideoContentType>(data[0]); |
nisse-webrtc
2017/04/06 09:23:22
Maybe return false if the value on the wire is inv
ilnik
2017/04/06 10:04:49
Done.
|
+ return true; |
+} |
+ |
+bool VideoContentTypeExtension::Write(uint8_t* data, |
+ VideoContentType content_type) { |
+ data[0] = static_cast<uint8_t>(content_type); |
+ return true; |
+} |
+ |
} // namespace webrtc |