Pass time constant to bwe smoothing filter.

webrtc/audio/audio_send_stream_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 87 matching lines @@
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
     // 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_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   CongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
@@ skipping 275 matching lines @@
   stream_config.send_codec_spec.cng_plfreq = 8000;
   stream_config.send_codec_spec.cng_payload_type = 105;
   EXPECT_CALL(*helper.voice_engine(), SetVADStatus(kChannelId, true, _, _))
       .WillOnce(Return(0));
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       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