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Add periodic logging of received RTP headers and estimated clock offsets for e2e delay.

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R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2014-10-14 11:40:13 +00:00
parent 81a78930ee
commit eb24b04f16
4 changed files with 53 additions and 13 deletions
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1
webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
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@ -42,6 +42,7 @@ class RemoteNtpTimeEstimator {
Clock* clock_;
scoped_ptr<TimestampExtrapolator> ts_extrapolator_;
RtcpList rtcp_list_;
int64_t last_timing_log_ms_;
DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
};

18
webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
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@ -11,16 +11,19 @@
#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/timestamp_extrapolator.h"
namespace webrtc {
static const int kTimingLogIntervalMs = 10000;
// TODO(wu): Refactor this class so that it can be shared with
// vie_sync_module.cc.
RemoteNtpTimeEstimator::RemoteNtpTimeEstimator(Clock* clock)
: clock_(clock),
ts_extrapolator_(
new TimestampExtrapolator(clock_->TimeInMilliseconds())) {
ts_extrapolator_(new TimestampExtrapolator(clock_->TimeInMilliseconds())),
last_timing_log_ms_(-1) {
}
RemoteNtpTimeEstimator::~RemoteNtpTimeEstimator() {}
@ -61,7 +64,16 @@ int64_t RemoteNtpTimeEstimator::Estimate(uint32_t rtp_timestamp) {
ts_extrapolator_->ExtrapolateLocalTime(timestamp);
int64_t ntp_offset =
clock_->CurrentNtpInMilliseconds() - clock_->TimeInMilliseconds();
return receiver_capture_ms + ntp_offset;
int64_t receiver_capture_ntp_ms = receiver_capture_ms + ntp_offset;
int64_t now_ms = clock_->TimeInMilliseconds();
if (now_ms - last_timing_log_ms_ > kTimingLogIntervalMs) {
LOG(LS_INFO) << "RTP timestamp: " << rtp_timestamp
<< " in NTP clock: " << sender_capture_ntp_ms
<< " estimated time in receiver clock: " << receiver_capture_ms
<< " converted to NTP clock: " << receiver_capture_ntp_ms;
last_timing_log_ms_ = now_ms;
}
return receiver_capture_ntp_ms;
}
} // namespace webrtc

45
webrtc/video_engine/vie_receiver.cc
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@ -30,28 +30,34 @@
namespace webrtc {
static const int kPacketLogIntervalMs = 10000;
ViEReceiver::ViEReceiver(const int32_t channel_id,
VideoCodingModule* module_vcm,
RemoteBitrateEstimator* remote_bitrate_estimator,
RtpFeedback* rtp_feedback)
: receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
clock_(Clock::GetRealTimeClock()),
rtp_header_parser_(RtpHeaderParser::Create()),
rtp_payload_registry_(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(false))),
rtp_receiver_(RtpReceiver::CreateVideoReceiver(
channel_id, Clock::GetRealTimeClock(), this, rtp_feedback,
rtp_payload_registry_.get())),
rtp_receive_statistics_(ReceiveStatistics::Create(
Clock::GetRealTimeClock())),
rtp_payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
rtp_receiver_(
RtpReceiver::CreateVideoReceiver(channel_id,
clock_,
this,
rtp_feedback,
rtp_payload_registry_.get())),
rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
fec_receiver_(FecReceiver::Create(this)),
rtp_rtcp_(NULL),
vcm_(module_vcm),
remote_bitrate_estimator_(remote_bitrate_estimator),
ntp_estimator_(new RemoteNtpTimeEstimator(Clock::GetRealTimeClock())),
ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
rtp_dump_(NULL),
receiving_(false),
restored_packet_in_use_(false),
receiving_ast_enabled_(false) {
receiving_ast_enabled_(false),
last_packet_log_ms_(-1) {
assert(remote_bitrate_estimator);
}
@ -229,10 +235,29 @@ int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet,
}
int payload_length = rtp_packet_length - header.headerLength;
int64_t arrival_time_ms;
int64_t now_ms = clock_->TimeInMilliseconds();
if (packet_time.timestamp != -1)
arrival_time_ms = (packet_time.timestamp + 500) / 1000;
else
arrival_time_ms = TickTime::MillisecondTimestamp();
arrival_time_ms = now_ms;
{
// Periodically log the RTP header of incoming packets.
CriticalSectionScoped cs(receive_cs_.get());
if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
std::stringstream ss;
ss << "Packet received on SSRC: " << header.ssrc << " with payload type: "
<< static_cast<int>(header.payloadType) << ", timestamp: "
<< header.timestamp << ", sequence number: " << header.sequenceNumber
<< ", arrival time: " << arrival_time_ms;
if (header.extension.hasTransmissionTimeOffset)
ss << ", toffset: " << header.extension.transmissionTimeOffset;
if (header.extension.hasAbsoluteSendTime)
ss << ", abs send time: " << header.extension.absoluteSendTime;
LOG(LS_INFO) << ss.str();
last_packet_log_ms_ = now_ms;
}
}
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms,
payload_length, header);

2
webrtc/video_engine/vie_receiver.h
View File

@ -105,6 +105,7 @@ class ViEReceiver : public RtpData {
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
scoped_ptr<CriticalSectionWrapper> receive_cs_;
Clock* clock_;
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
scoped_ptr<RtpReceiver> rtp_receiver_;
@ -122,6 +123,7 @@ class ViEReceiver : public RtpData {
uint8_t restored_packet_[kViEMaxMtu];
bool restored_packet_in_use_;
bool receiving_ast_enabled_;
int64_t last_packet_log_ms_;
};
} // namespace webrt