Add trace of enqueued and sent RTP packets
This is useful in debugging the latency from a packet
is enqueued until it's sent.
Bug: webrtc:11617
Change-Id: Ic2f194334a2e178de221df3a0838481035bb3505
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176231
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31381}
diff --git a/modules/pacing/paced_sender.cc b/modules/pacing/paced_sender.cc
index 88effe4..a0e7676 100644
--- a/modules/pacing/paced_sender.cc
+++ b/modules/pacing/paced_sender.cc
@@ -22,6 +22,7 @@
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
@@ -114,8 +115,15 @@
void PacedSender::EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
{
+ TRACE_EVENT0(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+ "PacedSender::EnqueuePackets");
rtc::CritScope cs(&critsect_);
for (auto& packet : packets) {
+ TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+ "PacedSender::EnqueuePackets::Loop", "sequence_number",
+ packet->SequenceNumber(), "rtp_timestamp",
+ packet->Timestamp());
+
pacing_controller_.EnqueuePacket(std::move(packet));
}
}
diff --git a/modules/pacing/packet_router.cc b/modules/pacing/packet_router.cc
index e9e8d4b..02befc9 100644
--- a/modules/pacing/packet_router.cc
+++ b/modules/pacing/packet_router.cc
@@ -24,6 +24,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
@@ -136,6 +137,10 @@
void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info) {
+ TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"), "PacketRouter::SendPacket",
+ "sequence_number", packet->SequenceNumber(), "rtp_timestamp",
+ packet->Timestamp());
+
rtc::CritScope cs(&modules_crit_);
// With the new pacer code path, transport sequence numbers are only set here,
// on the pacer thread. Therefore we don't need atomics/synchronization.
@@ -168,6 +173,9 @@
std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
DataSize size) {
+ TRACE_EVENT1(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+ "PacketRouter::GeneratePadding", "bytes", size.bytes());
+
rtc::CritScope cs(&modules_crit_);
// First try on the last rtp module to have sent media. This increases the
// the chance that any payload based padding will be useful as it will be
@@ -179,22 +187,28 @@
if (last_send_module_ != nullptr &&
last_send_module_->SupportsRtxPayloadPadding()) {
padding_packets = last_send_module_->GeneratePadding(size.bytes());
- if (!padding_packets.empty()) {
- return padding_packets;
+ }
+
+ if (padding_packets.empty()) {
+ // Iterate over all modules send module. Video modules will be at the front
+ // and so will be prioritized. This is important since audio packets may not
+ // be taken into account by the bandwidth estimator, e.g. in FF.
+ for (RtpRtcp* rtp_module : send_modules_list_) {
+ if (rtp_module->SupportsPadding()) {
+ padding_packets = rtp_module->GeneratePadding(size.bytes());
+ if (!padding_packets.empty()) {
+ last_send_module_ = rtp_module;
+ break;
+ }
+ }
}
}
- // Iterate over all modules send module. Video modules will be at the front
- // and so will be prioritized. This is important since audio packets may not
- // be taken into account by the bandwidth estimator, e.g. in FF.
- for (RtpRtcp* rtp_module : send_modules_list_) {
- if (rtp_module->SupportsPadding()) {
- padding_packets = rtp_module->GeneratePadding(size.bytes());
- if (!padding_packets.empty()) {
- last_send_module_ = rtp_module;
- break;
- }
- }
+ for (auto& packet : padding_packets) {
+ TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+ "PacketRouter::GeneratePadding::Loop", "sequence_number",
+ packet->SequenceNumber(), "rtp_timestamp",
+ packet->Timestamp());
}
return padding_packets;
diff --git a/modules/pacing/task_queue_paced_sender.cc b/modules/pacing/task_queue_paced_sender.cc
index 41eebea..d058e03 100644
--- a/modules/pacing/task_queue_paced_sender.cc
+++ b/modules/pacing/task_queue_paced_sender.cc
@@ -17,6 +17,7 @@
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_utils/to_queued_task.h"
+#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
@@ -121,6 +122,15 @@
void TaskQueuePacedSender::EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
+ TRACE_EVENT0(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+ "TaskQueuePacedSender::EnqueuePackets");
+ for (auto& packet : packets) {
+ TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+ "TaskQueuePacedSender::EnqueuePackets::Loop",
+ "sequence_number", packet->SequenceNumber(), "rtp_timestamp",
+ packet->Timestamp());
+ }
+
task_queue_.PostTask([this, packets_ = std::move(packets)]() mutable {
RTC_DCHECK_RUN_ON(&task_queue_);
for (auto& packet : packets_) {
