Index: webrtc/video/video_send_stream.cc |
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc |
index 93ff1f1f951dabea1efe8968f5f6fd5c180c2ecd..d68d947c815e8336ebc7949d3e13fc24afd0b0a4 100644 |
--- a/webrtc/video/video_send_stream.cc |
+++ b/webrtc/video/video_send_stream.cc |
@@ -22,6 +22,7 @@ |
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
#include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
#include "webrtc/modules/pacing/packet_router.h" |
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
#include "webrtc/modules/utility/include/process_thread.h" |
#include "webrtc/video/call_stats.h" |
#include "webrtc/video/video_capture_input.h" |
@@ -228,16 +229,9 @@ VideoSendStream::VideoSendStream( |
// One-byte-extension local identifiers are in the range 1-14 inclusive. |
RTC_DCHECK_GE(id, 1); |
RTC_DCHECK_LE(id, 14); |
- if (extension == RtpExtension::kTOffset) { |
- RTC_CHECK_EQ(0, vie_channel_.EnableSendTimestampOffset(id)); |
- } else if (extension == RtpExtension::kAbsSendTime) { |
- RTC_CHECK_EQ(0, vie_channel_.EnableSendAbsoluteSendTime(id)); |
- } else if (extension == RtpExtension::kVideoRotation) { |
- RTC_CHECK_EQ(0, vie_channel_.EnableSendVideoRotation(id)); |
- } else if (extension == RtpExtension::kTransportSequenceNumber) { |
- RTC_CHECK_EQ(0, vie_channel_.EnableSendTransportSequenceNumber(id)); |
- } else { |
- RTC_NOTREACHED() << "Registering unsupported RTP extension."; |
+ for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
+ RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( |
+ StringToRtpExtensionType(extension), id)); |
stefan-webrtc
2016/02/26 14:54:11
Should we no longer fail on trying to set audio ex
pbos-webrtc
2016/02/26 15:01:04
Done.
|
} |
} |