Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
index 7f33bc2ee57c1dc2b44aeade3feaaaf87f8072a2..46e5bcf00e4adf3bc2e2b7209166a60ba9d2520b 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
@@ -17,6 +17,7 @@ |
#include "webrtc/base/checks.h" |
#include "webrtc/base/logging.h" |
#include "webrtc/common_types.h" |
+#include "webrtc/config.h" |
#include "webrtc/system_wrappers/include/trace.h" |
#ifdef _WIN32 |
@@ -26,6 +27,21 @@ |
namespace webrtc { |
+RTPExtensionType StringToRtpExtensionType(const std::string& extension) { |
+ if (extension == RtpExtension::kTOffset) |
+ return kRtpExtensionTransmissionTimeOffset; |
+ if (extension == RtpExtension::kAudioLevel) |
+ return kRtpExtensionAudioLevel; |
+ if (extension == RtpExtension::kAbsSendTime) |
+ return kRtpExtensionAbsoluteSendTime; |
+ if (extension == RtpExtension::kVideoRotation) |
+ return kRtpExtensionVideoRotation; |
+ if (extension == RtpExtension::kTransportSequenceNumber) |
+ return kRtpExtensionTransportSequenceNumber; |
+ RTC_NOTREACHED() << "Looking up unsupported RTP extension."; |
+ return kRtpExtensionNone; |
+} |
+ |
RtpRtcp::Configuration::Configuration() |
: audio(false), |
receiver_only(false), |