Improve bitrate probing for the audio-only case.

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc

Index: webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 269a5d117d500d43a63c2d84de1b50f400796332..7d1b50bf66a2d72090be172d432803068bae9e0b 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -286,7 +286,8 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
     // For now only try to detect probes while we don't have a valid estimate.
     // We currently assume that only packets larger than 200 bytes are paced by
     // the sender.
-    if (payload_size > PacedSender::kMinProbePacketSize &&
+    const size_t kMinProbePacketSize = 200;
+    if (payload_size > kMinProbePacketSize &&
         (!remote_rate_.ValidEstimate() ||
          now_ms - first_packet_time_ms_ < kInitialProbingIntervalMs)) {
       // TODO(holmer): Use a map instead to get correct order?