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}

call/BUILD.gn

@@ -98,6 +98,8 @@ rtc_source_set("rtp_receiver") {
 
 rtc_source_set("rtp_sender") {
   sources = [
+    "payload_router.cc",
+    "payload_router.h",
     "rtp_transport_controller_send.cc",
     "rtp_transport_controller_send.h",
   ]
@@ -106,10 +108,16 @@ rtc_source_set("rtp_sender") {
     ":rtp_interfaces",
     "..:webrtc_common",
     "../api/transport:network_control",
+    "../api/video_codecs:video_codecs_api",
     "../modules/congestion_controller",
     "../modules/congestion_controller/rtp:congestion_controller",
     "../modules/pacing",
+    "../modules/rtp_rtcp:rtp_rtcp",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
+    "../modules/rtp_rtcp:rtp_video_header",
     "../modules/utility",
+    "../modules/video_coding:video_codec_interface",
+    "../rtc_base:checks",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:rtc_task_queue",
@@ -268,6 +276,7 @@ if (rtc_include_tests) {
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
+      "payload_router_unittest.cc",
       "receive_time_calculator_unittest.cc",
       "rtcp_demuxer_unittest.cc",
       "rtp_bitrate_configurator_unittest.cc",
@@ -304,11 +313,13 @@ if (rtc_include_tests) {
       "../modules/rtp_rtcp:mock_rtp_rtcp",
       "../modules/rtp_rtcp:rtp_rtcp_format",
       "../modules/utility:mock_process_thread",
+      "../modules/video_coding:video_codec_interface",
       "../rtc_base:checks",
       "../rtc_base:rtc_base_approved",
       "../system_wrappers",
       "../test:audio_codec_mocks",
       "../test:direct_transport",
+      "../test:field_trial",
       "../test:test_common",
       "../test:test_support",
       "../test:video_test_common",

call/payload_router.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "video/payload_router.h"
+#include "call/payload_router.h"
 
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"

call/payload_router.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef VIDEO_PAYLOAD_ROUTER_H_
-#define VIDEO_PAYLOAD_ROUTER_H_
+#ifndef CALL_PAYLOAD_ROUTER_H_
+#define CALL_PAYLOAD_ROUTER_H_
 
 #include <map>
 #include <vector>
@@ -41,7 +41,7 @@ class PayloadRouter : public EncodedImageCallback {
                 const std::vector<uint32_t>& ssrcs,
                 int payload_type,
                 const std::map<uint32_t, RtpPayloadState>& states);
-  ~PayloadRouter();
+  ~PayloadRouter() override;
 
   // PayloadRouter will only route packets if being active, all packets will be
   // dropped otherwise.
@@ -81,4 +81,4 @@ class PayloadRouter : public EncodedImageCallback {
 
 }  // namespace webrtc
 
-#endif  // VIDEO_PAYLOAD_ROUTER_H_
+#endif  // CALL_PAYLOAD_ROUTER_H_

call/payload_router_unittest.cc

@@ -11,13 +11,13 @@
 #include <memory>
 #include <string>
 
+#include "call/payload_router.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "modules/video_coding/include/video_codec_interface.h"
 #include "test/field_trial.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
-#include "video/payload_router.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;

video/BUILD.gn

@@ -16,8 +16,6 @@ rtc_static_library("video") {
     "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 @@ rtc_static_library("video") {
     "../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 @@ if (rtc_include_tests) {
       "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",

video/video_send_stream.h

@@ -17,13 +17,13 @@
 
 #include "api/fec_controller.h"
 #include "call/bitrate_allocator.h"
+#include "call/payload_router.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
 #include "common_video/libyuv/include/webrtc_libyuv.h"
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/event.h"
 #include "rtc_base/task_queue.h"
-#include "video/payload_router.h"
 #include "video/send_delay_stats.h"
 #include "video/send_statistics_proxy.h"
 #include "video/video_stream_encoder.h"

video/video_send_stream_impl.h

@@ -16,6 +16,7 @@
 #include <vector>
 
 #include "call/bitrate_allocator.h"
+#include "call/payload_router.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "common_video/include/video_bitrate_allocator.h"
 #include "modules/rtp_rtcp/include/flexfec_sender.h"
@@ -24,7 +25,6 @@
 #include "rtc_base/weak_ptr.h"
 #include "video/call_stats.h"
 #include "video/encoder_rtcp_feedback.h"
-#include "video/payload_router.h"
 #include "video/send_delay_stats.h"
 #include "video/send_statistics_proxy.h"
 #include "video/video_send_stream.h"

video/video_stream_decoder.cc

@@ -20,7 +20,6 @@
 #include "rtc_base/logging.h"
 #include "system_wrappers/include/metrics.h"
 #include "video/call_stats.h"
-#include "video/payload_router.h"
 #include "video/receive_statistics_proxy.h"
 
 namespace webrtc {