Add sender controlled playout delay limits

webrtc/common_types.h

Index: webrtc/common_types.h
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 587b90d753ce3f3eb092dfd01edb8b1b68303719..02633ba87a481370e3fcb4dcfa8204b63fe35f8c 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -751,6 +751,24 @@ struct PacketTime {
                        // If unknown, this value will be set to zero.
 };
 
+// Minimum and maximum playout delay values from capture to render.
+// These are best effort values.
+//
+// A value < 0 indicates no change from previous valid value.
+//
+// min = max = 0 indicates that the receiver should try and render
+// frame as soon as possible.
+//
+// min = x, max = y indicates that the receiver is free to adapt
+// in the range (x, y) based on network jitter.
+//
+// Note: Given that this gets embedded in a union, it is up-to the owner to
+// initialize these values.
+struct PlayoutDelay {
+  int min_ms;
+  int max_ms;
+};
+
 struct RTPHeaderExtension {
   RTPHeaderExtension();
 
@@ -772,6 +790,8 @@ struct RTPHeaderExtension {
   // ts_126114v120700p.pdf
   bool hasVideoRotation;
   uint8_t videoRotation;
+
+  PlayoutDelay playout_delay = {-1, -1};
 };
 
 struct RTPHeader {