Disable pacer disabling.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 252ffb2a3e59737e125e10c605b137ec9d67aeca..438dd18e7e40e3fdba1665a4abc73ba2e595400a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -712,13 +712,11 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
     // Convert from TickTime to Clock since capture_time_ms is based on
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
-    if (!paced_sender_->SendPacket(
-            RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
-            corrected_capture_tims_ms, length - header.headerLength, true)) {
-      // We can't send the packet right now.
-      // We will be called when it is time.
-      return length;
-    }
+    paced_sender_->InsertPacket(
+        RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
+        corrected_capture_tims_ms, length - header.headerLength, true);
+
+    return length;
   }
   int rtx = kRtxOff;
   {
@@ -1041,20 +1039,17 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
-    if (!paced_sender_->SendPacket(priority, rtp_header.ssrc,
-                                   rtp_header.sequenceNumber, corrected_time_ms,
-                                   payload_length, false)) {
-      if (last_capture_time_ms_sent_ == 0 ||
-          corrected_time_ms > last_capture_time_ms_sent_) {
-        last_capture_time_ms_sent_ = corrected_time_ms;
-        TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                                 "PacedSend", corrected_time_ms,
-                                 "capture_time_ms", corrected_time_ms);
-      }
-      // We can't send the packet right now.
-      // We will be called when it is time.
-      return 0;
+    paced_sender_->InsertPacket(priority, rtp_header.ssrc,
+                                rtp_header.sequenceNumber, corrected_time_ms,
+                                payload_length, false);
+    if (last_capture_time_ms_sent_ == 0 ||
+        corrected_time_ms > last_capture_time_ms_sent_) {
+      last_capture_time_ms_sent_ = corrected_time_ms;
+      TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                               "PacedSend", corrected_time_ms,
+                               "capture_time_ms", corrected_time_ms);
     }
+    return 0;
   }
   if (capture_time_ms > 0) {
     UpdateDelayStatistics(capture_time_ms, now_ms);