Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index 5aa7228947ad7a92ab4b50b24bc37b7056872d64..4b5bd3a292b38435a235b96946a5c613c5d36ff9 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" |
@@ -176,12 +177,11 @@ class Call : public webrtc::Call, |
// The following members are only accessed (exclusively) from one thread and |
// from the destructor, and therefore doesn't need any explicit |
// synchronization. |
- int64_t received_video_bytes_; |
- int64_t received_audio_bytes_; |
- int64_t received_rtcp_bytes_; |
- int64_t first_rtp_packet_received_ms_; |
- int64_t last_rtp_packet_received_ms_; |
int64_t first_packet_sent_ms_; |
+ RateCounter received_bytes_per_second_counter_; |
+ RateCounter received_audio_bytes_per_second_counter_; |
+ RateCounter received_video_bytes_per_second_counter_; |
+ RateCounter received_rtcp_bytes_per_second_counter_; |
// TODO(holmer): Remove this lock once BitrateController no longer calls |
// OnNetworkChanged from multiple threads. |
@@ -239,12 +239,11 @@ Call::Call(const Call::Config& config) |
receive_crit_(RWLockWrapper::CreateRWLock()), |
send_crit_(RWLockWrapper::CreateRWLock()), |
event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())), |
- received_video_bytes_(0), |
- received_audio_bytes_(0), |
- received_rtcp_bytes_(0), |
- first_rtp_packet_received_ms_(-1), |
- last_rtp_packet_received_ms_(-1), |
first_packet_sent_ms_(-1), |
+ received_bytes_per_second_counter_(clock_, nullptr, true), |
+ received_audio_bytes_per_second_counter_(clock_, nullptr, true), |
+ received_video_bytes_per_second_counter_(clock_, nullptr, true), |
+ received_rtcp_bytes_per_second_counter_(clock_, nullptr, true), |
estimated_send_bitrate_sum_kbits_(0), |
pacer_bitrate_sum_kbits_(0), |
min_allocated_send_bitrate_bps_(0), |
@@ -341,30 +340,31 @@ void Call::UpdateSendHistograms() { |
} |
void Call::UpdateReceiveHistograms() { |
- if (first_rtp_packet_received_ms_ == -1) |
- return; |
- int64_t elapsed_sec = |
- (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000; |
- if (elapsed_sec < metrics::kMinRunTimeInSeconds) |
- return; |
- int audio_bitrate_kbps = received_audio_bytes_ * 8 / elapsed_sec / 1000; |
- int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000; |
- int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec; |
- if (video_bitrate_kbps > 0) { |
+ const int kMinRequiredPeriodicSamples = 5; |
+ AggregatedStats video_bytes_per_sec = |
+ received_video_bytes_per_second_counter_.GetStats(); |
+ if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps", |
- video_bitrate_kbps); |
+ video_bytes_per_sec.average * 8 / 1000); |
} |
- if (audio_bitrate_kbps > 0) { |
+ AggregatedStats audio_bytes_per_sec = |
+ received_audio_bytes_per_second_counter_.GetStats(); |
+ if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps", |
- audio_bitrate_kbps); |
+ audio_bytes_per_sec.average * 8 / 1000); |
} |
- if (rtcp_bitrate_bps > 0) { |
+ AggregatedStats rtcp_bytes_per_sec = |
+ received_rtcp_bytes_per_second_counter_.GetStats(); |
+ if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps", |
- rtcp_bitrate_bps); |
+ rtcp_bytes_per_sec.average * 8); |
+ } |
+ AggregatedStats recv_bytes_per_sec = |
+ received_bytes_per_second_counter_.GetStats(); |
+ if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps", |
+ recv_bytes_per_sec.average * 8 / 1000); |
} |
- RTC_LOGGED_HISTOGRAM_COUNTS_100000( |
- "WebRTC.Call.BitrateReceivedInKbps", |
- audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000); |
} |
PacketReceiver* Call::Receiver() { |
@@ -843,7 +843,11 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type, |
// TODO(pbos): Make sure it's a valid packet. |
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that |
// there's no receiver of the packet. |
- received_rtcp_bytes_ += length; |
+ if (received_bytes_per_second_counter_.HasSample()) { |
+ // First RTP packet has been received. |
+ received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
+ received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(length)); |
+ } |
bool rtcp_delivered = false; |
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) { |
ReadLockScoped read_lock(*receive_crit_); |
@@ -889,16 +893,13 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
if (length < 12) |
return DELIVERY_PACKET_ERROR; |
- last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds(); |
- if (first_rtp_packet_received_ms_ == -1) |
- first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_; |
- |
uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]); |
ReadLockScoped read_lock(*receive_crit_); |
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) { |
auto it = audio_receive_ssrcs_.find(ssrc); |
if (it != audio_receive_ssrcs_.end()) { |
- received_audio_bytes_ += length; |
+ received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
+ received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); |
auto status = it->second->DeliverRtp(packet, length, packet_time) |
? DELIVERY_OK |
: DELIVERY_PACKET_ERROR; |
@@ -910,7 +911,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) { |
auto it = video_receive_ssrcs_.find(ssrc); |
if (it != video_receive_ssrcs_.end()) { |
- received_video_bytes_ += length; |
+ received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
+ received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
auto status = it->second->DeliverRtp(packet, length, packet_time) |
? DELIVERY_OK |
: DELIVERY_PACKET_ERROR; |