Split out RtpSource from libjingle_peerconnection_api
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 925f32f..1777648 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -177,6 +177,7 @@
"transport:network_control",
"transport/media:audio_interfaces",
"transport/media:video_interfaces",
+ "transport/rtp:rtp_source",
"units:data_rate",
"units:timestamp",
"video:encoded_image",
