Chromium Code Reviews| Index: webrtc/voice_engine/channel.cc |
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
| index 100832de3adeb2dc986cd255dc271a109ea6bbb7..49ed483148e22564c630171dfe0d7ddd1e917d10 100644 |
| --- a/webrtc/voice_engine/channel.cc |
| +++ b/webrtc/voice_engine/channel.cc |
| @@ -924,10 +924,7 @@ Channel::Channel(int32_t channelId, |
| rtp_packet_sender_proxy_(new RtpPacketSenderProxy()), |
| retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(), |
| kMaxRetransmissionWindowMs)), |
| - decoder_factory_(config.acm_config.decoder_factory), |
| - // Bitrate smoother can be initialized with arbitrary time constant |
| - // (0 used here). The actual time constant will be set in SetBitRate. |
| - bitrate_smoother_(0, Clock::GetRealTimeClock()) { |
| + decoder_factory_(config.acm_config.decoder_factory) { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId), |
| "Channel::Channel() - ctor"); |
| AudioCodingModule::Config acm_config(config.acm_config); |
| @@ -1332,28 +1329,10 @@ void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) { |
| "Channel::SetBitRate(bitrate_bps=%d)", bitrate_bps); |
| audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { |
| if (*encoder) |
|
minyue-webrtc
2016/12/22 14:51:27
{
...
}
michaelt
2016/12/22 15:10:10
Done.
|
| - (*encoder)->OnReceivedTargetAudioBitrate(bitrate_bps); |
| + (*encoder)->OnReceivedTargetAudioBitrate( |
| + bitrate_bps, rtc::Optional<int64_t>(probing_interval_ms)); |
| }); |
| retransmission_rate_limiter_->SetMaxRate(bitrate_bps); |
| - |
| - // We give smoothed bitrate allocation to audio network adaptor as |
| - // the uplink bandwidth. |
| - // 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) { |
| - (*encoder)->OnReceivedUplinkBandwidth( |
| - static_cast<int>(*bitrate_smoother_.GetAverage())); |
| - } |
| - }); |
| } |
| void Channel::OnIncomingFractionLoss(int fraction_lost) { |