| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index ceec963ee29a04b2f2f9f8f11dfa446d0861db86..4bb58ec0f677b0d880c501e19092e3ff8f7bcff0 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -43,6 +43,7 @@
|
| #include "webrtc/system_wrappers/include/trace.h"
|
| #include "webrtc/video/call_stats.h"
|
| #include "webrtc/video/send_delay_stats.h"
|
| +#include "webrtc/video/stats_counter.h"
|
| #include "webrtc/video/video_receive_stream.h"
|
| #include "webrtc/video/video_send_stream.h"
|
| #include "webrtc/video/vie_remb.h"
|
| @@ -186,11 +187,10 @@ class Call : public webrtc::Call,
|
| // TODO(holmer): Remove this lock once BitrateController no longer calls
|
| // OnNetworkChanged from multiple threads.
|
| rtc::CriticalSection bitrate_crit_;
|
| - int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
|
| - int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
|
| uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
|
| - int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
|
| uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
|
| + AvgCounter estimated_send_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
|
| + AvgCounter pacer_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
|
|
|
| std::map<std::string, rtc::NetworkRoute> network_routes_;
|
|
|
| @@ -240,11 +240,10 @@ Call::Call(const Call::Config& config)
|
| first_rtp_packet_received_ms_(-1),
|
| last_rtp_packet_received_ms_(-1),
|
| first_packet_sent_ms_(-1),
|
| - estimated_send_bitrate_sum_kbits_(0),
|
| - pacer_bitrate_sum_kbits_(0),
|
| min_allocated_send_bitrate_bps_(0),
|
| - num_bitrate_updates_(0),
|
| configured_max_padding_bitrate_bps_(0),
|
| + estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),
|
| + pacer_bitrate_kbps_counter_(clock_, nullptr, true),
|
| remb_(clock_),
|
| congestion_controller_(
|
| new CongestionController(clock_, this, &remb_, event_log_.get())),
|
| @@ -311,22 +310,24 @@ void Call::UpdateHistograms() {
|
| }
|
|
|
| void Call::UpdateSendHistograms() {
|
| - if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
|
| + if (first_packet_sent_ms_ == -1)
|
| return;
|
| int64_t elapsed_sec =
|
| (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
|
| if (elapsed_sec < metrics::kMinRunTimeInSeconds)
|
| return;
|
| - int send_bitrate_kbps =
|
| - estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_;
|
| - int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_;
|
| - if (send_bitrate_kbps > 0) {
|
| + const int kMinRequiredPeriodicSamples = 5;
|
| + AggregatedStats send_bitrate_stats =
|
| + estimated_send_bitrate_kbps_counter_.ProcessAndGetStats();
|
| + if (send_bitrate_stats.num_samples >= kMinRequiredPeriodicSamples) {
|
| RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
|
| - send_bitrate_kbps);
|
| + send_bitrate_stats.average);
|
| }
|
| - if (pacer_bitrate_kbps > 0) {
|
| + AggregatedStats pacer_bitrate_stats =
|
| + pacer_bitrate_kbps_counter_.ProcessAndGetStats();
|
| + if (pacer_bitrate_stats.num_samples >= kMinRequiredPeriodicSamples) {
|
| RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
|
| - pacer_bitrate_kbps);
|
| + pacer_bitrate_stats.average);
|
| }
|
| }
|
|
|
| @@ -732,26 +733,32 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
|
| bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss,
|
| rtt_ms);
|
|
|
| - // Ignore updates where the bitrate is zero because the aggregate network
|
| - // state is down.
|
| - if (target_bitrate_bps > 0) {
|
| - {
|
| - ReadLockScoped read_lock(*send_crit_);
|
| - // Do not update the stats if we are not sending video.
|
| - if (video_send_streams_.empty())
|
| - return;
|
| - }
|
| + // Ignore updates if bitrate is zero (the aggregate network state is down).
|
| + if (target_bitrate_bps == 0) {
|
| rtc::CritScope lock(&bitrate_crit_);
|
| - // We only update these stats if we have send streams, and assume that
|
| - // OnNetworkChanged is called roughly with a fixed frequency.
|
| - estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
|
| - // Pacer bitrate might be higher than bitrate estimate if enforcing min
|
| - // bitrate.
|
| - uint32_t pacer_bitrate_bps =
|
| - std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_);
|
| - pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
|
| - ++num_bitrate_updates_;
|
| + estimated_send_bitrate_kbps_counter_.ProcessAndPause();
|
| + pacer_bitrate_kbps_counter_.ProcessAndPause();
|
| + return;
|
| + }
|
| +
|
| + bool sending_video;
|
| + {
|
| + ReadLockScoped read_lock(*send_crit_);
|
| + sending_video = !video_send_streams_.empty();
|
| + }
|
| +
|
| + rtc::CritScope lock(&bitrate_crit_);
|
| + if (!sending_video) {
|
| + // Do not update the stats if we are not sending video.
|
| + estimated_send_bitrate_kbps_counter_.ProcessAndPause();
|
| + pacer_bitrate_kbps_counter_.ProcessAndPause();
|
| + return;
|
| }
|
| + estimated_send_bitrate_kbps_counter_.Add(target_bitrate_bps / 1000);
|
| + // Pacer bitrate may be higher than bitrate estimate if enforcing min bitrate.
|
| + uint32_t pacer_bitrate_bps =
|
| + std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_);
|
| + pacer_bitrate_kbps_counter_.Add(pacer_bitrate_bps / 1000);
|
| }
|
|
|
| void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
|
|
|