webrtc发送端带宽预测

• 当接收侧感知到2-10%丢包率，发送端的预测值不变。
• 当实际丢包率超过预测值10%时，新的预测值可更新为As_hat(i)= As_hat(i-1)(1-0.5p)，其中p为丢包率。
• 当实际丢包率小于2%时预测时可更新为As_hat(i)= 1.08(As_hat(i-1))+0.5+1000，其中p为丢包率。
• 其中p是rtcp里的丢包率

void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {    uint32_t new_bitrate = current_bitrate_bps_;    // We trust the REMB and/or delay-based estimate during the first 2 seconds if    // we haven't had any packet loss reported, to allow startup bitrate probing.    if (last_fraction_loss_ == 0 && IsInStartPhase(now_ms)) {        new_bitrate = std::max(bwe_incoming_, new_bitrate);        new_bitrate = std::max(delay_based_bitrate_bps_, new_bitrate);        if (new_bitrate != current_bitrate_bps_) {            min_bitrate_history_.clear();            min_bitrate_history_.push_back(            std::make_pair(now_ms, current_bitrate_bps_));            CapBitrateToThresholds(now_ms, new_bitrate);            return;        }    }    UpdateMinHistory(now_ms);    if (last_packet_report_ms_ == -1) {        // No feedback received.        CapBitrateToThresholds(now_ms, current_bitrate_bps_);        return;    }    int64_t time_since_packet_report_ms = now_ms - last_packet_report_ms_;    int64_t time_since_feedback_ms = now_ms - last_feedback_ms_;    if (time_since_packet_report_ms < 1.2 * kFeedbackIntervalMs) {        // We only care about loss above a given bitrate threshold.        float loss = last_fraction_loss_ / 256.0f;        // We only make decisions based on loss when the bitrate is above a        // threshold. This is a crude way of handling loss which is uncorrelated        // to congestion.        if (current_bitrate_bps_ < bitrate_threshold_bps_ ||            loss <= low_loss_threshold_) {            // Loss < 2%: Increase rate by 8% of the min bitrate in the last            // kBweIncreaseIntervalMs.            // Note that by remembering the bitrate over the last second one can            // rampup up one second faster than if only allowed to start ramping            // at 8% per second rate now. E.g.:            // If sending a constant 100kbps it can rampup immediatly to 108kbps            // whenever a receiver report is received with lower packet loss.            // If instead one would do: current_bitrate_bps_ *= 1.08^(delta time),            // it would take over one second since the lower packet loss to achieve            // 108kbps.            new_bitrate = static_cast<uint32_t>(                min_bitrate_history_.front().second * 1.08 + 0.5);            // Add 1 kbps extra, just to make sure that we do not get stuck            // (gives a little extra increase at low rates, negligible at higher            // rates).            new_bitrate += 1000;    } else if (current_bitrate_bps_ > bitrate_threshold_bps_) {            if (loss <= high_loss_threshold_) {                // Loss between 2% - 10%: Do nothing.            } else {                // Loss > 10%: Limit the rate decreases to once a kBweDecreaseIntervalMs                // + rtt.                if (!has_decreased_since_last_fraction_loss_ &&                    (now_ms - time_last_decrease_ms_) >=                    (kBweDecreaseIntervalMs + last_round_trip_time_ms_)) {                    time_last_decrease_ms_ = now_ms;                    // Reduce rate:                    // newRate = rate * (1 - 0.5*lossRate);                    // where packetLoss = 256*lossRate;                    new_bitrate = static_cast<uint32_t>(                            (current_bitrate_bps_ *                                static_cast<double>(512 - last_fraction_loss_)) /                            512.0);                    has_decreased_since_last_fraction_loss_ = true;                }            }        }    } else if (time_since_feedback_ms >        kFeedbackTimeoutIntervals * kFeedbackIntervalMs &&            (last_timeout_ms_ == -1 ||                    now_ms - last_timeout_ms_ > kTimeoutIntervalMs)) {            if (in_timeout_experiment_) {                LOG(LS_WARNING) << "Feedback timed out (" << time_since_feedback_ms                << " ms), reducing bitrate.";                new_bitrate *= 0.8;                // Reset accumulators since we've already acted on missing feedback and                // shouldn't to act again on these old lost packets.                lost_packets_since_last_loss_update_Q8_ = 0;                expected_packets_since_last_loss_update_ = 0;                last_timeout_ms_ = now_ms;            }        }    //其实就是估算current_bitrate_bps_    CapBitrateToThresholds(now_ms, new_bitrate);}bitrate_threshold_bps_ 是怎么来的？SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)    : lost_packets_since_last_loss_update_Q8_(0),    expected_packets_since_last_loss_update_(0),    current_bitrate_bps_(0),    min_bitrate_configured_(congestion_controller::GetMinBitrateBps()),    max_bitrate_configured_(kDefaultMaxBitrateBps),    last_low_bitrate_log_ms_(-1),    has_decreased_since_last_fraction_loss_(false),    last_feedback_ms_(-1),    last_packet_report_ms_(-1),    last_timeout_ms_(-1),    last_fraction_loss_(0),    last_logged_fraction_loss_(0),    last_round_trip_time_ms_(0),    bwe_incoming_(0),    delay_based_bitrate_bps_(0),    time_last_decrease_ms_(0),    first_report_time_ms_(-1),    initially_lost_packets_(0),    bitrate_at_2_seconds_kbps_(0),    uma_update_state_(kNoUpdate),    rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),    event_log_(event_log),    last_rtc_event_log_ms_(-1),    in_timeout_experiment_(    webrtc::field_trial::IsEnabled("WebRTC-FeedbackTimeout")),    low_loss_threshold_(kDefaultLowLossThreshold),    high_loss_threshold_(kDefaultHighLossThreshold),    bitrate_threshold_bps_(1000 * kDefaultBitrateThresholdKbps) {    RTC_DCHECK(event_log);    if (BweLossExperimentIsEnabled()) {        uint32_t bitrate_threshold_kbps;        if (ReadBweLossExperimentParameters(&low_loss_threshold_,            &high_loss_threshold_,            &bitrate_threshold_kbps)) {            LOG(LS_INFO) << "Enabled BweLossExperiment with parameters "            << low_loss_threshold_ << ", " << high_loss_threshold_            << ", " << bitrate_threshold_kbps;            bitrate_threshold_bps_ = bitrate_threshold_kbps * 1000;        }    }}