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