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Add field to counters for when first rtp/rtcp packet is sent/received.

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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
This commit is contained in:
asapersson@webrtc.org 2014-12-16 12:03:11 +00:00
parent b395a5ea65
commit d08d389ce8
10 changed files with 154 additions and 67 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

56
webrtc/common_types.h
View File

@ -200,7 +200,8 @@ class RtcpStatisticsCallback {
// 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 @@ struct RtcpPacketTypeCounter {
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 @@ struct RtcpPacketTypeCounter {
(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 @@ struct StreamDataCounters {
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 @@ struct StreamDataCounters {
}
// 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 @@ class StreamDataCountersCallback {
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 @@ class BitrateStatisticsObserver {
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() {}

1
webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
View File

@ -96,6 +96,7 @@ void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
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

8
webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
View File

@ -140,12 +140,18 @@ TEST_F(ReceiveStatisticsTest, GetReceiveStreamDataCounters) {
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);
}

4
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -316,6 +316,10 @@ RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
_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)
{

4
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -1632,6 +1632,10 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
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;

45
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -325,16 +325,21 @@ TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
}
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 @@ TEST_F(RtpRtcpImplTest, RtcpPacketTypeCounter_FirAndPli) {
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;

3
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -907,6 +907,9 @@ void RTPSender::UpdateRtpStats(const uint8_t* buffer,
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;
}

1
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -1162,6 +1162,7 @@ TEST_F(RtpSenderTest, BytesReportedCorrectly) {
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);

97
webrtc/video_engine/vie_channel.cc
View File

@ -147,9 +147,7 @@ ViEChannel::ViEChannel(int32_t channel_id,
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 @@ ViEChannel::~ViEChannel() {
}
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 @@ int32_t ViEChannel::StartSend() {
rtp_rtcp->SetSendingMediaStatus(true);
rtp_rtcp->SetSendingStatus(true);
}
start_send_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds();
return 0;
}

2
webrtc/video_engine/vie_channel.h
View File

@ -502,8 +502,6 @@ class ViEChannel
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_;
};