Chromium Code Reviews| Index: webrtc/voice_engine/channel.cc |
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
| index 6e2e91c0a1bd0c0e62ccc4a753ef940d12ec2849..fbb8b9c96714bb2afd65605e538953e645ad14b3 100644 |
| --- a/webrtc/voice_engine/channel.cc |
| +++ b/webrtc/voice_engine/channel.cc |
| @@ -1351,14 +1351,9 @@ void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) { |
| // 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. |
| + rtc::CritScope lock(&bitrate_smoother_lock_); |
| 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())); |
| - } |
| - }); |
| + bitrate_bps_ = rtc::Optional<int>(bitrate_bps); |
| } |
| void Channel::OnIncomingFractionLoss(int fraction_lost) { |
| @@ -2755,6 +2750,19 @@ void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) { |
| audio_coding_->DisableNack(); |
| } |
| +void Channel::AdaptCodec() { |
| + audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| + rtc::CritScope lock(&bitrate_smoother_lock_); |
|
the sun
2016/12/13 16:43:24
Which threads can we be running on? Is it possible
michaelt
2016/12/13 17:01:26
No, since AdaptCodec() runs on the worker queue. A
|
| + if (bitrate_bps_) { |
| + bitrate_smoother_.AddSample(*bitrate_bps_); |
| + if (*encoder) { |
| + (*encoder)->OnReceivedUplinkBandwidth( |
| + static_cast<int>(*bitrate_smoother_.GetAverage())); |
| + } |
| + } |
| + }); |
| +} |
| + |
| // Called when we are missing one or more packets. |
| int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) { |
| return _rtpRtcpModule->SendNACK(sequence_numbers, length); |