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 1b311e64190cd80eb146ddba3628ac67a503d30d..98e6ae79b8e38543eea9afb2dba9e0ece30ca03f 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc |
@@ -215,4 +215,34 @@ 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(rtc::ArrayView<const uint8_t> data, |
+ VideoContentType* content_type) { |
+ if (data.size() == 1 && |
+ data[0] < static_cast<uint8_t>(VideoContentType::kTotalContentTypes)) { |
+ *content_type = static_cast<VideoContentType>(data[0]); |
+ return true; |
+ } else { |
tommi
2017/04/10 10:59:18
no need for |else|
ilnik
2017/04/10 12:47:42
Done.
|
+ return false; |
+ } |
+} |
+ |
+bool VideoContentTypeExtension::Write(uint8_t* data, |
+ VideoContentType content_type) { |
+ data[0] = static_cast<uint8_t>(content_type); |
tommi
2017/04/10 10:59:18
is there a safe cast method that we can use instea
ilnik
2017/04/10 12:47:42
I don't want to write one, as it will introduce a
|
+ return true; |
+} |
+ |
} // namespace webrtc |