Removed unneeded inheritence for SendDelayStats class

Bug: None
Change-Id: Ida0f086702c7168d51e9e31f9d95a795e326593b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319583
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40726}

call/rtp_video_sender_unittest.cc

@@ -32,7 +32,6 @@
 #include "test/scenario/scenario.h"
 #include "test/scoped_key_value_config.h"
 #include "test/time_controller/simulated_time_controller.h"
-#include "video/send_delay_stats.h"
 #include "video/send_statistics_proxy.h"
 
 namespace webrtc {
@@ -62,17 +61,15 @@ class MockRtcpIntraFrameObserver : public RtcpIntraFrameObserver {
 };
 
 RtpSenderObservers CreateObservers(
-    RtcpRttStats* rtcp_rtt_stats,
     RtcpIntraFrameObserver* intra_frame_callback,
     ReportBlockDataObserver* report_block_data_observer,
     StreamDataCountersCallback* rtp_stats,
     BitrateStatisticsObserver* bitrate_observer,
     FrameCountObserver* frame_count_observer,
     RtcpPacketTypeCounterObserver* rtcp_type_observer,
-    SendSideDelayObserver* send_delay_observer,
+    SendSideDelayObserver* send_delay_observer) {
-    SendPacketObserver* send_packet_observer) {
   RtpSenderObservers observers;
-  observers.rtcp_rtt_stats = rtcp_rtt_stats;
+  observers.rtcp_rtt_stats = nullptr;
   observers.intra_frame_callback = intra_frame_callback;
   observers.rtcp_loss_notification_observer = nullptr;
   observers.report_block_data_observer = report_block_data_observer;
@@ -81,7 +78,7 @@ RtpSenderObservers CreateObservers(
   observers.frame_count_observer = frame_count_observer;
   observers.rtcp_type_observer = rtcp_type_observer;
   observers.send_delay_observer = send_delay_observer;
-  observers.send_packet_observer = send_packet_observer;
+  observers.send_packet_observer = nullptr;
   return observers;
 }
 
@@ -127,7 +124,6 @@ class RtpVideoSenderTestFixture {
                                             ssrcs,
                                             rtx_ssrcs,
                                             payload_type)),
-        send_delay_stats_(time_controller_.GetClock()),
         bitrate_config_(GetBitrateConfig()),
         transport_controller_(
             time_controller_.GetClock(),
@@ -148,9 +144,9 @@ class RtpVideoSenderTestFixture {
     router_ = std::make_unique<RtpVideoSender>(
         time_controller_.GetClock(), suspended_ssrcs, suspended_payload_states,
         config_.rtp, config_.rtcp_report_interval_ms, &transport_,
-        CreateObservers(nullptr, &encoder_feedback_, &stats_proxy_,
+        CreateObservers(&encoder_feedback_, &stats_proxy_, &stats_proxy_,
-                        &stats_proxy_, &stats_proxy_, frame_count_observer,
+                        &stats_proxy_, frame_count_observer, &stats_proxy_,
-                        &stats_proxy_, &stats_proxy_, &send_delay_stats_),
+                        &stats_proxy_),
         &transport_controller_, &event_log_, &retransmission_rate_limiter_,
         std::make_unique<FecControllerDefault>(time_controller_.GetClock()),
         nullptr, CryptoOptions{}, frame_transformer,
@@ -206,7 +202,6 @@ class RtpVideoSenderTestFixture {
   GlobalSimulatedTimeController time_controller_;
   RtcEventLogNull event_log_;
   VideoSendStream::Config config_;
-  SendDelayStats send_delay_stats_;
   BitrateConstraints bitrate_config_;
   RtpTransportControllerSend transport_controller_;
   SendStatisticsProxy stats_proxy_;

video/send_delay_stats.h

@@ -32,10 +32,10 @@ namespace webrtc {
 // TODO(bugs.webrtc.org/11993): OnSendPacket and OnSentPacket will eventually
 // be called consistently on the same thread. Once we're there, we should be
 // able to avoid locking (at least for the fast path).
-class SendDelayStats : public SendPacketObserver {
+class SendDelayStats {
  public:
   explicit SendDelayStats(Clock* clock);
-  ~SendDelayStats() override;
+  ~SendDelayStats();
 
   // Adds the configured ssrcs for the rtp streams.
   // Stats will be calculated for these streams.
@@ -44,11 +44,8 @@ class SendDelayStats : public SendPacketObserver {
   // Called when a packet is sent (leaving socket).
   bool OnSentPacket(int packet_id, Timestamp time);
 
-  // From SendPacketObserver.
   // Called when a packet is sent to the transport.
-  void OnSendPacket(uint16_t packet_id,
+  void OnSendPacket(uint16_t packet_id, Timestamp capture_time, uint32_t ssrc);
-                    Timestamp capture_time,
-                    uint32_t ssrc) override;
 
  private:
   // Map holding sent packets (mapped by sequence number).

video/send_delay_stats_unittest.cc

@@ -11,6 +11,7 @@
 #include "video/send_delay_stats.h"
 
 #include <cstdint>
+#include <memory>
 #include <vector>
 
 #include "call/rtp_config.h"
@@ -54,8 +55,7 @@ class SendDelayStatsTest : public ::testing::Test {
   }
 
   void OnSendPacket(uint16_t id, uint32_t ssrc, Timestamp capture) {
-    SendPacketObserver* observer = stats_.get();
+    stats_->OnSendPacket(id, capture, ssrc);
-    observer->OnSendPacket(id, capture, ssrc);
   }
 
   bool OnSentPacket(uint16_t id) {