Add field to counters for when first rtp/rtcp packet is sent/received.
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 43d1bd7..e647aad 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -200,7 +200,8 @@
// Statistics for RTCP packet types.
struct RtcpPacketTypeCounter {
RtcpPacketTypeCounter()
- : nack_packets(0),
+ : first_packet_time_ms(-1),
+ nack_packets(0),
fir_packets(0),
pli_packets(0),
nack_requests(0),
@@ -212,6 +213,16 @@
pli_packets += other.pli_packets;
nack_requests += other.nack_requests;
unique_nack_requests += other.unique_nack_requests;
+ if (other.first_packet_time_ms != -1 &&
+ (other.first_packet_time_ms < first_packet_time_ms ||
+ first_packet_time_ms == -1)) {
+ // Use oldest time.
+ first_packet_time_ms = other.first_packet_time_ms;
+ }
+ }
+
+ int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
+ return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
}
int UniqueNackRequestsInPercent() const {
@@ -222,25 +233,27 @@
(unique_nack_requests * 100.0f / nack_requests) + 0.5f);
}
- uint32_t nack_packets; // Number of RTCP NACK packets.
- uint32_t fir_packets; // Number of RTCP FIR packets.
- uint32_t pli_packets; // Number of RTCP PLI packets.
- uint32_t nack_requests; // Number of NACKed RTP packets.
+ int64_t first_packet_time_ms; // Time when first packet is sent/received.
+ uint32_t nack_packets; // Number of RTCP NACK packets.
+ uint32_t fir_packets; // Number of RTCP FIR packets.
+ uint32_t pli_packets; // Number of RTCP PLI packets.
+ uint32_t nack_requests; // Number of NACKed RTP packets.
uint32_t unique_nack_requests; // Number of unique NACKed RTP packets.
};
// Data usage statistics for a (rtp) stream.
struct StreamDataCounters {
StreamDataCounters()
- : bytes(0),
- header_bytes(0),
- padding_bytes(0),
- packets(0),
- retransmitted_bytes(0),
- retransmitted_header_bytes(0),
- retransmitted_padding_bytes(0),
- retransmitted_packets(0),
- fec_packets(0) {}
+ : first_packet_time_ms(-1),
+ bytes(0),
+ header_bytes(0),
+ padding_bytes(0),
+ packets(0),
+ retransmitted_bytes(0),
+ retransmitted_header_bytes(0),
+ retransmitted_padding_bytes(0),
+ retransmitted_packets(0),
+ fec_packets(0) {}
void Add(const StreamDataCounters& other) {
bytes += other.bytes;
@@ -252,6 +265,16 @@
retransmitted_padding_bytes += other.retransmitted_padding_bytes;
retransmitted_packets += other.retransmitted_packets;
fec_packets += other.fec_packets;
+ if (other.first_packet_time_ms != -1 &&
+ (other.first_packet_time_ms < first_packet_time_ms ||
+ first_packet_time_ms == -1)) {
+ // Use oldest time.
+ first_packet_time_ms = other.first_packet_time_ms;
+ }
+ }
+
+ int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
+ return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
}
size_t TotalBytes() const {
@@ -268,6 +291,7 @@
}
// TODO(pbos): Rename bytes -> media_bytes.
+ int64_t first_packet_time_ms; // Time when first packet is sent/received.
size_t bytes; // Payload bytes, excluding RTP headers and padding.
size_t header_bytes; // Number of bytes used by RTP headers.
size_t padding_bytes; // Number of padding bytes.
@@ -288,7 +312,7 @@
uint32_t ssrc) = 0;
};
-// Rate statistics for a stream
+// Rate statistics for a stream.
struct BitrateStatistics {
BitrateStatistics() : bitrate_bps(0), packet_rate(0), timestamp_ms(0) {}
@@ -307,7 +331,7 @@
uint32_t ssrc) = 0;
};
-// Callback, used to notify an observer whenever frame counts have been updated
+// Callback, used to notify an observer whenever frame counts have been updated.
class FrameCountObserver {
public:
virtual ~FrameCountObserver() {}
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 321ea13..7ba83f7 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -96,6 +96,7 @@
if (receive_counters_.packets == 1) {
received_seq_first_ = header.sequenceNumber;
+ receive_counters_.first_packet_time_ms = clock_->TimeInMilliseconds();
}
// Count only the new packets received. That is, if packets 1, 2, 3, 5, 4, 6
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 6cc6008..93351e5 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -140,12 +140,18 @@
StreamDataCounters counters;
statistician->GetReceiveStreamDataCounters(&counters);
+ EXPECT_GT(counters.first_packet_time_ms, -1);
EXPECT_EQ(1u, counters.packets);
- // GetReceiveStreamDataCounters includes reset counter values.
statistician->ResetStatistics();
+ // GetReceiveStreamDataCounters includes reset counter values.
+ statistician->GetReceiveStreamDataCounters(&counters);
+ EXPECT_GT(counters.first_packet_time_ms, -1);
+ EXPECT_EQ(1u, counters.packets);
+
receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
statistician->GetReceiveStreamDataCounters(&counters);
+ EXPECT_GT(counters.first_packet_time_ms, -1);
EXPECT_EQ(2u, counters.packets);
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 8d04480..8149370 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -316,6 +316,10 @@
_lastReceived = _clock->TimeInMilliseconds();
+ if (packet_type_counter_.first_packet_time_ms == -1) {
+ packet_type_counter_.first_packet_time_ms = _lastReceived;
+ }
+
RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin();
while (pktType != RTCPUtility::kRtcpNotValidCode)
{
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 3386d9b..e060c8c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -1632,6 +1632,10 @@
CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ if (packet_type_counter_.first_packet_time_ms == -1) {
+ packet_type_counter_.first_packet_time_ms = _clock->TimeInMilliseconds();
+ }
+
if(_TMMBR ) // Attach TMMBR to send and receive reports.
{
rtcpPacketTypeFlags |= kRtcpTmmbr;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index d6e0252..f1d0604 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -325,16 +325,21 @@
}
TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_Nack) {
+ EXPECT_EQ(-1, receiver_.RtcpSent().first_packet_time_ms);
+ EXPECT_EQ(-1, sender_.RtcpReceived().first_packet_time_ms);
EXPECT_EQ(0U, sender_.RtcpReceived().nack_packets);
EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets);
+
// Receive module sends a NACK.
const uint16_t kNackLength = 1;
uint16_t nack_list[kNackLength] = {123};
EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength));
EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets);
+ EXPECT_GT(receiver_.RtcpSent().first_packet_time_ms, -1);
// Send module receives the NACK.
EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets);
+ EXPECT_GT(sender_.RtcpReceived().first_packet_time_ms, -1);
}
TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_FirAndPli) {
@@ -355,6 +360,46 @@
EXPECT_EQ(1U, sender_.RtcpReceived().pli_packets);
}
+TEST_F(RtpRtcpImplTest, AddStreamDataCounters) {
+ StreamDataCounters rtp;
+ const int64_t kStartTimeMs = 1;
+ rtp.first_packet_time_ms = kStartTimeMs;
+ rtp.packets = 1;
+ rtp.bytes = 1;
+ rtp.header_bytes = 2;
+ rtp.padding_bytes = 3;
+ EXPECT_EQ(rtp.TotalBytes(), rtp.bytes + rtp.header_bytes + rtp.padding_bytes);
+
+ StreamDataCounters rtp2;
+ rtp2.first_packet_time_ms = -1;
+ rtp2.packets = 10;
+ rtp2.bytes = 10;
+ rtp2.retransmitted_header_bytes = 4;
+ rtp2.retransmitted_bytes = 5;
+ rtp2.retransmitted_padding_bytes = 6;
+ rtp2.retransmitted_packets = 7;
+ rtp2.fec_packets = 8;
+
+ StreamDataCounters sum = rtp;
+ sum.Add(rtp2);
+ EXPECT_EQ(kStartTimeMs, sum.first_packet_time_ms);
+ EXPECT_EQ(11U, sum.packets);
+ EXPECT_EQ(11U, sum.bytes);
+ EXPECT_EQ(2U, sum.header_bytes);
+ EXPECT_EQ(3U, sum.padding_bytes);
+ EXPECT_EQ(4U, sum.retransmitted_header_bytes);
+ EXPECT_EQ(5U, sum.retransmitted_bytes);
+ EXPECT_EQ(6U, sum.retransmitted_padding_bytes);
+ EXPECT_EQ(7U, sum.retransmitted_packets);
+ EXPECT_EQ(8U, sum.fec_packets);
+ EXPECT_EQ(sum.TotalBytes(), rtp.TotalBytes() + rtp2.TotalBytes());
+
+ StreamDataCounters rtp3;
+ rtp3.first_packet_time_ms = kStartTimeMs + 10;
+ sum.Add(rtp3);
+ EXPECT_EQ(kStartTimeMs, sum.first_packet_time_ms); // Holds oldest time.
+}
+
TEST_F(RtpRtcpImplTest, SendsInitialNackList) {
// Send module sends a NACK.
const uint16_t kNackLength = 1;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 7c4307f..3d2dfbe 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -907,6 +907,9 @@
total_bitrate_sent_.Update(packet_length);
++counters->packets;
+ if (counters->packets == 1) {
+ counters->first_packet_time_ms = clock_->TimeInMilliseconds();
+ }
if (IsFecPacket(buffer, header)) {
++counters->fec_packets;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 866258b..253d1b9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1162,6 +1162,7 @@
rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
// Payload + 1-byte generic header.
+ EXPECT_GT(rtp_stats.first_packet_time_ms, -1);
EXPECT_EQ(rtp_stats.bytes, sizeof(payload) + 1);
EXPECT_EQ(rtp_stats.header_bytes, 12u);
EXPECT_EQ(rtp_stats.padding_bytes, 0u);
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index be6a2d1..fffb970 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -147,9 +147,7 @@
sender_(sender),
nack_history_size_sender_(kSendSidePacketHistorySize),
max_nack_reordering_threshold_(kMaxPacketAgeToNack),
- pre_render_callback_(NULL),
- start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()),
- start_send_ms_(0) {
+ pre_render_callback_(NULL) {
RtpRtcp::Configuration configuration = CreateRtpRtcpConfiguration();
configuration.remote_bitrate_estimator = remote_bitrate_estimator;
configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
@@ -237,76 +235,80 @@
}
void ViEChannel::UpdateHistograms() {
- // TODO(asapersson): Use time from first sent/received packet.
- float elapsed_minutes =
- (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
- if (elapsed_minutes < metrics::kMinRunTimeInSeconds / 60.0f) {
- return;
- }
+ int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
RtcpPacketTypeCounter rtcp_sent;
RtcpPacketTypeCounter rtcp_received;
GetRtcpPacketTypeCounters(&rtcp_sent, &rtcp_received);
if (sender_) {
- if (rtcp_received.nack_requests > 0) {
- RTC_HISTOGRAM_PERCENTAGE(
- "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
- rtcp_received.UniqueNackRequestsInPercent());
+ int64_t elapsed_sec = rtcp_received.TimeSinceFirstPacketInMs(now) / 1000;
+ if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
+ rtcp_received.nack_packets / elapsed_sec / 60);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
+ rtcp_received.fir_packets / elapsed_sec / 60);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
+ rtcp_received.pli_packets / elapsed_sec / 60);
+ if (rtcp_received.nack_requests > 0) {
+ RTC_HISTOGRAM_PERCENTAGE(
+ "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
+ rtcp_received.UniqueNackRequestsInPercent());
+ }
}
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
- rtcp_received.nack_packets / elapsed_minutes);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
- rtcp_received.fir_packets / elapsed_minutes);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
- rtcp_received.pli_packets / elapsed_minutes);
} else if (vie_receiver_.GetRemoteSsrc() > 0) {
// Get receive stats if we are receiving packets, i.e. there is a remote
// ssrc.
- if (rtcp_sent.nack_requests > 0) {
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
- rtcp_sent.UniqueNackRequestsInPercent());
+ int64_t elapsed_sec = rtcp_sent.TimeSinceFirstPacketInMs(now) / 1000;
+ if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
+ rtcp_sent.nack_packets / elapsed_sec / 60);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
+ rtcp_sent.fir_packets / elapsed_sec / 60);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
+ rtcp_sent.pli_packets / elapsed_sec / 60);
+ if (rtcp_sent.nack_requests > 0) {
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
+ rtcp_sent.UniqueNackRequestsInPercent());
+ }
}
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
- rtcp_sent.nack_packets / elapsed_minutes);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
- rtcp_sent.fir_packets / elapsed_minutes);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
- rtcp_sent.pli_packets / elapsed_minutes);
StreamDataCounters rtp;
StreamDataCounters rtx;
GetReceiveStreamDataCounters(&rtp, &rtx);
StreamDataCounters rtp_rtx = rtp;
rtp_rtx.Add(rtx);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.BitrateReceivedInKbps",
- rtp_rtx.TotalBytes() * 8 / (elapsed_minutes * 60) / 1000);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
- rtp.MediaPayloadBytes() * 8 / (elapsed_minutes * 60) / 1000);
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateReceivedInKbps",
- rtp_rtx.padding_bytes * 8 / (elapsed_minutes * 60) / 1000);
- RTC_HISTOGRAM_COUNTS_10000(
- "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
- rtp_rtx.RetransmittedBytes() * 8 / (elapsed_minutes * 60) / 1000);
- uint32_t ssrc = 0;
- if (vie_receiver_.GetRtxSsrc(&ssrc)) {
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
- rtx.TotalBytes() * 8 / (elapsed_minutes * 60) / 1000);
+ elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000;
+ if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.BitrateReceivedInKbps",
+ rtp_rtx.TotalBytes() * 8 / elapsed_sec / 1000);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
+ rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PaddingBitrateReceivedInKbps",
+ rtp_rtx.padding_bytes * 8 / elapsed_sec / 1000);
+ RTC_HISTOGRAM_COUNTS_10000(
+ "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
+ rtp_rtx.RetransmittedBytes() * 8 / elapsed_sec / 1000);
+ uint32_t ssrc = 0;
+ if (vie_receiver_.GetRtxSsrc(&ssrc)) {
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
+ rtx.TotalBytes() * 8 / elapsed_sec / 1000);
+ }
}
}
}
void ViEChannel::UpdateHistogramsAtStopSend() {
- // TODO(asapersson): Use time from first sent packet.
- int64_t elapsed_sec =
- (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_send_ms_) / 1000;
- if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
- return;
- }
StreamDataCounters rtp;
StreamDataCounters rtx;
GetSendStreamDataCounters(&rtp, &rtx);
StreamDataCounters rtp_rtx = rtp;
rtp_rtx.Add(rtx);
+
+ int64_t elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(
+ Clock::GetRealTimeClock()->TimeInMilliseconds()) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
+ return;
+ }
RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.BitrateSentInKbps",
rtp_rtx.TotalBytes() * 8 / elapsed_sec / 1000);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateSentInKbps",
@@ -1393,7 +1395,6 @@
rtp_rtcp->SetSendingMediaStatus(true);
rtp_rtcp->SetSendingStatus(true);
}
- start_send_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds();
return 0;
}
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 88d4a94..b978353 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -502,8 +502,6 @@
int nack_history_size_sender_;
int max_nack_reordering_threshold_;
I420FrameCallback* pre_render_callback_;
- const int64_t start_ms_;
- int64_t start_send_ms_;
std::map<uint32_t, RTCPReportBlock> prev_report_blocks_;
};
