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/audio/BUILD.gn b/audio/BUILD.gn
index 4bd6f57..abf4c67 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -54,6 +54,7 @@
"../api/audio_codecs:audio_codecs_api",
"../api/rtc_event_log",
"../api/task_queue",
+ "../api/transport/rtp:rtp_source",
"../call:bitrate_allocator",
"../call:call_interfaces",
"../call:rtp_interfaces",