commit a837030f8fdb541ab16bc75fb06477316e85fe2f
author Niels Möller <nisse@webrtc.org> Mon Sep 02 15:16:49 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Mon Sep 02 14:04:47 2019 +0000
tree 1c09ebccb860e95512965a02a6d3f35a0a5a47be
parent d112c758015c095895489d21bed198352f689515

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}

media/base/media_channel.h

diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index c237874..0f502d3 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -25,7 +25,7 @@
 #include "api/media_transport_config.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
-#include "api/rtp_receiver_interface.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
@@ -42,6 +42,7 @@
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/critical_section.h"
 #include "rtc_base/dscp.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/network_route.h"