Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index a7df1f2176ce5ade86669c8295c461758f9a9b2b..d946676fd6861ce63ba3657af9a629ad75c30acc 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -1301,7 +1301,7 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) { |
return 0; |
} |
-void Channel::SetBitRate(int bitrate_bps) { |
+void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) { |
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
"Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps); |
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
@@ -1312,10 +1312,15 @@ void Channel::SetBitRate(int bitrate_bps) { |
// We give smoothed bitrate allocation to audio network adaptor as |
// the uplink bandwidth. |
- // TODO(michaelt) : Remove kDefaultBitrateSmoothingTimeConstantMs as soon as |
- // we pass the probing interval to this function. |
- constexpr int64_t kDefaultBitrateSmoothingTimeConstantMs = 20000; |
- bitrate_smoother_.SetTimeConstantMs(kDefaultBitrateSmoothingTimeConstantMs); |
+ // The probing spikes should not affect the bitrate smoother more than 25%. |
+ // To simplify the calculations we use a step response as input signal. |
+ // The step response of an exponential filter is |
+ // u(t) = 1 - e^(-t / time_constant). |
+ // In order to limit the affect of a BWE spike within 25% of its value before |
+ // the next probing, we would choose a time constant that fulfills |
+ // 1 - e^(-probing_interval_ms / time_constant) < 0.25 |
+ // Then 4 * probing_interval_ms is a good choice. |
+ bitrate_smoother_.SetTimeConstantMs(probing_interval_ms * 4); |
bitrate_smoother_.AddSample(bitrate_bps); |
audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
if (*encoder) { |