Revert of Pass time constant to bwe smoothing filter.

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index a296513d9dfc4b1add42931a85c09e100e0028ef..dbc49662e6be7c4ebcf25d2d478fcee9091ec00f 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -105,8 +105,6 @@
     // Use ISAC as default codec so as to prevent unnecessary |voice_engine_|
     // calls from the default ctor behavior.
     stream_config_.send_codec_spec.codec_inst = kIsacCodec;
-    stream_config_.min_bitrate_bps = 10000;
-    stream_config_.max_bitrate_bps = 65000;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
@@ -402,27 +400,5 @@
       helper.event_log());
 }
 
-TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
-  ConfigHelper helper;
-  internal::AudioSendStream send_stream(
-      helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
-  EXPECT_CALL(*helper.channel_proxy(),
-              SetBitrate(helper.config().max_bitrate_bps, _));
-  send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
-                               6000);
-}
-
-TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
-  ConfigHelper helper;
-  internal::AudioSendStream send_stream(
-      helper.config(), helper.audio_state(), helper.worker_queue(),
-      helper.congestion_controller(), helper.bitrate_allocator(),
-      helper.event_log());
-  EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
-  send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
-}
-
 }  // namespace test
 }  // namespace webrtc