Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index f3021fef6fec31f8e727fa3698158628dc1d89ee..81f4c13077be630d7d15022dad7130bd4655bde7 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -1304,7 +1304,8 @@ int32_t Channel::SetSendCodec(const CodecInst& codec) { |
return 0; |
} |
-void Channel::SetBitRate(int bitrate_bps) { |
+void Channel::SetBitRate(int bitrate_bps, |
+ rtc::Optional<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) { |
@@ -1315,17 +1316,22 @@ 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); |
- bitrate_smoother_.AddSample(bitrate_bps); |
- audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
- if (*encoder) { |
- (*encoder)->OnReceivedUplinkBandwidth( |
- static_cast<int>(*bitrate_smoother_.GetAverage())); |
- } |
- }); |
+ // 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 calculation show a step response for an |
+ // exponential filter (bitrate_smoother use an exponential filter) with the |
+ // length of probing_interval and an amplitude of 1. |
+ // 0.22 = 1 - e^(-t/probing_interval * 4) | t = probing_interval. |
+ if (probing_interval_ms) { |
+ bitrate_smoother_.SetTimeConstantMs(*probing_interval_ms * 4); |
+ bitrate_smoother_.AddSample(bitrate_bps); |
+ audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
+ if (*encoder) { |
+ (*encoder)->OnReceivedUplinkBandwidth( |
+ static_cast<int>(*bitrate_smoother_.GetAverage())); |
+ } |
+ }); |
+ } |
} |
void Channel::OnIncomingFractionLoss(int fraction_lost) { |