Chromium Code Reviews| Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
| diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
| index ae9dae2fbea4bc9e66d6a8b6826236258cc2ecab..7a19f2c3de7ed58f558829982b4401dd28b1f4ed 100644 |
| --- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
| +++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc |
| @@ -13,6 +13,7 @@ |
| #include <algorithm> |
| #include "webrtc/base/checks.h" |
| +#include "webrtc/base/exp_filter.h" |
| #include "webrtc/base/safe_conversions.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" |
| @@ -80,6 +81,37 @@ double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) { |
| } |
| } |
| +class PacketLossFractionSmoother { |
| + public: |
| + explicit PacketLossFractionSmoother(const Clock* clock) |
| + : clock_(clock), |
| + last_sample_time_ms_(clock_->TimeInMilliseconds()), |
| + smoother_(new rtc::ExpFilter(0.9999f)) {} |
| + |
| + ~PacketLossFractionSmoother() = default; |
|
kwiberg-webrtc
2016/10/11 14:56:19
I don't think this line does anything. This is wha
minyue-webrtc
2016/10/11 15:31:09
Done.
|
| + |
| + // Gets the smoothed packet loss fraction. |
| + float GetAverage() const { |
| + float value = smoother_->filtered(); |
| + return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value; |
| + } |
| + |
| + // Add new observation to the packet loss fraction smoother. |
| + void AddSample(float packet_loss_fraction) { |
| + int64_t now_ms = clock_->TimeInMilliseconds(); |
| + smoother_->Apply(static_cast<float>(now_ms - last_sample_time_ms_), |
| + packet_loss_fraction); |
| + last_sample_time_ms_ = now_ms; |
| + } |
| + |
| + private: |
| + const Clock* const clock_; |
| + int64_t last_sample_time_ms_; |
| + |
| + // An exponential filter is used to smooth the packet loss fraction. |
| + std::unique_ptr<rtc::ExpFilter> smoother_; |
|
kwiberg-webrtc
2016/10/11 14:56:19
Do you need to point to the ExpFilter? Can't you e
minyue-webrtc
2016/10/11 15:31:09
no. I simply copied from the old implementation. B
|
| +}; |
| + |
| } // namespace |
| AudioEncoderOpus::Config::Config() = default; |
| @@ -113,9 +145,11 @@ AudioEncoderOpus::AudioEncoderOpus( |
| AudioNetworkAdaptorCreator&& audio_network_adaptor_creator) |
| : packet_loss_rate_(0.0), |
| inst_(nullptr), |
| + packet_loss_fraction_smoother_(new PacketLossFractionSmoother( |
| + config.clock ? config.clock : Clock::GetRealTimeClock())), |
| audio_network_adaptor_creator_( |
| audio_network_adaptor_creator |
| - ? audio_network_adaptor_creator |
| + ? std::move(audio_network_adaptor_creator) |
| : [this](const std::string& config_string, const Clock* clock) { |
| return DefaultAudioNetworkAdaptorCreator(config_string, |
| clock); |
| @@ -234,8 +268,11 @@ void AudioEncoderOpus::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) { |
| void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction( |
| float uplink_packet_loss_fraction) { |
| - if (!audio_network_adaptor_) |
| - return; |
| + if (!audio_network_adaptor_) { |
| + packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction); |
| + float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage(); |
| + return SetProjectedPacketLossRate(average_fraction_loss); |
| + } |
| audio_network_adaptor_->SetUplinkPacketLossFraction( |
| uplink_packet_loss_fraction); |
| ApplyAudioNetworkAdaptor(); |
| @@ -244,7 +281,7 @@ void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction( |
| void AudioEncoderOpus::OnReceivedTargetAudioBitrate( |
| int target_audio_bitrate_bps) { |
| if (!audio_network_adaptor_) |
| - return; |
| + return SetTargetBitrate(target_audio_bitrate_bps); |
| audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps); |
| ApplyAudioNetworkAdaptor(); |
| } |