| 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),
|
|
|