commit a2f1533e27ef96d96700f2d2194c37e84c64bfdc
author Stefan Holmer <stefan@webrtc.org> Wed Jul 11 17:11:31 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Wed Jul 11 15:38:39 2018 +0000
tree 644c44c4af388374ece51b97e4a058dc6f03f595
parent 28bb391918e594fd828628b938f97a0a87b220b8

Moved PayloadRouter to call/.

This is done in preparation for moving ownership of PayloadRouter to RtpTransportControllerSend.

Bug: webrtc:9517
Change-Id: I4a5b449cbcfc23db594dc5bb68ca322dd8fa33b7
Reviewed-on: https://webrtc-review.googlesource.com/88241
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23936}

video/BUILD.gn

diff --git a/video/BUILD.gn b/video/BUILD.gn
index 5f5f12d..f60b772 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -16,8 +16,6 @@
     "encoder_rtcp_feedback.h",
     "overuse_frame_detector.cc",
     "overuse_frame_detector.h",
-    "payload_router.cc",
-    "payload_router.h",
     "quality_threshold.cc",
     "quality_threshold.h",
     "receive_statistics_proxy.cc",
@@ -70,6 +68,7 @@
     "../call:bitrate_allocator",
     "../call:call_interfaces",
     "../call:rtp_interfaces",
+    "../call:rtp_sender",
     "../call:video_stream_api",
     "../modules/rtp_rtcp:rtp_rtcp_format",
     "../modules/rtp_rtcp:rtp_video_header",
@@ -359,7 +358,6 @@
       "end_to_end_tests/stats_tests.cc",
       "end_to_end_tests/transport_feedback_tests.cc",
       "overuse_frame_detector_unittest.cc",
-      "payload_router_unittest.cc",
       "picture_id_tests.cc",
       "quality_scaling_tests.cc",
       "quality_threshold_unittest.cc",