Revert of Pass time constant to bwe smoothing filter.

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 840ae68d062da6dbca584258a11554c7e717c4c4..b031762b4cecd173e21766e4ab3f91c070a37fd9 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -229,8 +229,7 @@
 
 uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
-                                           int64_t rtt,
-                                           int64_t probing_interval_ms) {
+                                           int64_t rtt) {
   RTC_DCHECK_GE(bitrate_bps,
                 static_cast<uint32_t>(config_.min_bitrate_bps));
   // The bitrate allocator might allocate an higher than max configured bitrate
@@ -239,7 +238,7 @@
   if (bitrate_bps > max_bitrate_bps)
     bitrate_bps = max_bitrate_bps;
 
-  channel_proxy_->SetBitrate(bitrate_bps, probing_interval_ms);
+  channel_proxy_->SetBitrate(bitrate_bps);
 
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.