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Revert "Add trace of enqueued and sent RTP packets"

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This reverts commit 45b9192ad981dcdc12ad4aef087fff2195bd030c.

Reason for revert: When tracing is disabled, this results in a clang warning (unused variable), which results in a build error since Werror is enabled by default.

Original change's description:
> 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}

TBR=sprang@webrtc.org,kron@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11617
Change-Id: I854c17e587c624691a0e5e3ec9fd38c2607eda84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176380
Commit-Queue: Casey Fischer <caseyfischer@google.com>
Reviewed-by: Adam Nathan <adamnathan@google.com>
Cr-Commit-Position: refs/heads/master@{#31399}
This commit is contained in:
Casey Fischer 2020-06-01 18:38:55 +00:00 committed by Commit Bot
parent fce28fa091
commit 45bb717a28
3 changed files with 13 additions and 45 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
modules/pacing/paced_sender.cc
View File

@ -22,7 +22,6 @@
#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 {
@ -115,15 +114,8 @@ void PacedSender::SetPacingRates(DataRate pacing_rate, DataRate padding_rate) {
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));
}
}

40
modules/pacing/packet_router.cc
View File

@ -24,7 +24,6 @@
#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 {
@ -137,10 +136,6 @@ void PacketRouter::RemoveReceiveRtpModule(
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.
@ -173,9 +168,6 @@ void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
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
@ -187,28 +179,22 @@ std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
if (last_send_module_ != nullptr &&
last_send_module_->SupportsRtxPayloadPadding()) {
padding_packets = last_send_module_->GeneratePadding(size.bytes());
}
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;
}
}
if (!padding_packets.empty()) {
return padding_packets;
}
}
for (auto& packet : padding_packets) {
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
"PacketRouter::GeneratePadding::Loop", "sequence_number",
packet->SequenceNumber(), "rtp_timestamp",
packet->Timestamp());
// 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;
}
}
}
return padding_packets;

10
modules/pacing/task_queue_paced_sender.cc
View File

@ -17,7 +17,6 @@
#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 {
@ -122,15 +121,6 @@ void TaskQueuePacedSender::SetPacingRates(DataRate pacing_rate,
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_) {