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 2cb12750211842bda5111e67dcba460fb6570800..2525bad40200ef176460335473e964a7e9363b4c 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -91,6 +91,7 @@ struct ConfigHelper {
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
+    stream_config_.rtp.ssrc = kSsrc;

[the sun, 2016/11/23 09:12:34]

This is dupe from 3 lines above.

[michaelt, 2016/11/23 12:28:29]

Removed the duplicated line.

     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
@@ -98,6 +99,8 @@ struct ConfigHelper {
     // 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_; }
@@ -391,5 +394,30 @@ TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
       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);
+
+  EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 6000));
+  send_stream.OnBitrateUpdated(50000, 0.0, 50, 6000);
+}
+
 }  // namespace test
 }  // namespace webrtc