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}

call/video_receive_stream.h

diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index fa37fe8..3869c81 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -19,12 +19,14 @@
 
 #include "api/call/transport.h"
 #include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
 #include "api/media_transport_config.h"
 #include "api/media_transport_interface.h"
 #include "api/rtp_headers.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_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_timing.h"
 #include "api/video_codecs/sdp_video_format.h"
@@ -34,7 +36,6 @@
 
 namespace webrtc {
 
-class FrameDecryptorInterface;
 class RtpPacketSinkInterface;
 class VideoDecoderFactory;