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Move BWE_TEST_LOGGING_COMPILE_TIME_ENABLE code to the worker thread.

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This is part of moving calls to GetSendRates() to the worker.

Change-Id: Ifb93096a863ddf2669237e7f44af296d0e086b20
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177661
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31582}
This commit is contained in:
Tomas Gunnarsson
2020-06-29 17:41:22 +02:00
committed by Commit Bot
parent 096c0b0921
commit a1163749fd
3 changed files with 47 additions and 24 deletions
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6
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
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@ -289,6 +289,10 @@ void ModuleRtpRtcpImpl2::SetCsrcs(const std::vector<uint32_t>& csrcs) {
// TODO(pbos): Handle media and RTX streams separately (separate RTCP
// feedbacks).
RTCPSender::FeedbackState ModuleRtpRtcpImpl2::GetFeedbackState() {
// TODO(bugs.webrtc.org/11581): Called by potentially multiple threads.
// "Send*" methods and on the ProcessThread. Make sure it's only called on the
// construction thread.
RTCPSender::FeedbackState state;
// This is called also when receiver_only is true. Hence below
// checks that rtp_sender_ exists.
@ -653,6 +657,7 @@ void ModuleRtpRtcpImpl2::BitrateSent(uint32_t* total_rate,
uint32_t* video_rate,
uint32_t* fec_rate,
uint32_t* nack_rate) const {
RTC_DCHECK_RUN_ON(worker_queue_);
RtpSendRates send_rates = rtp_sender_->packet_sender.GetSendRates();
*total_rate = send_rates.Sum().bps<uint32_t>();
if (video_rate)
@ -663,6 +668,7 @@ void ModuleRtpRtcpImpl2::BitrateSent(uint32_t* total_rate,
}
RtpSendRates ModuleRtpRtcpImpl2::GetSendRates() const {
RTC_DCHECK_RUN_ON(worker_queue_);
return rtp_sender_->packet_sender.GetSendRates();
}

57
modules/rtp_rtcp/source/rtp_sender_egress.cc
View File

@ -17,7 +17,6 @@
#include "absl/strings/match.h"
#include "api/transport/field_trial_based_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_utils/to_queued_task.h"
@ -70,7 +69,9 @@ RtpSenderEgress::RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
packet_history_(packet_history),
transport_(config.outgoing_transport),
event_log_(config.event_log),
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
is_audio_(config.audio),
#endif
need_rtp_packet_infos_(config.need_rtp_packet_infos),
transport_feedback_observer_(config.transport_feedback_callback),
send_side_delay_observer_(config.send_side_delay_observer),
@ -110,30 +111,18 @@ void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
RTC_DCHECK_RUN_ON(&pacer_checker_);
RTC_DCHECK(packet);
const uint32_t packet_ssrc = packet->Ssrc();
RTC_DCHECK(packet->packet_type().has_value());
RTC_DCHECK(HasCorrectSsrc(*packet));
int64_t now_ms = clock_->TimeInMilliseconds();
if (is_audio_) {
const uint32_t packet_ssrc = packet->Ssrc();
const int64_t now_ms = clock_->TimeInMilliseconds();
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
GetSendRates().Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "AudioNackBitrate_kbps", now_ms,
GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
packet_ssrc);
worker_queue_->PostTask(
ToQueuedTask(task_safety_, [this, now_ms, packet_ssrc]() {
BweTestLoggingPlot(now_ms, packet_ssrc);
}));
#endif
} else {
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
GetSendRates().Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "VideoNackBitrate_kbps", now_ms,
GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
packet_ssrc);
#endif
}
if (need_rtp_packet_infos_ &&
packet->packet_type() == RtpPacketToSend::Type::kVideo) {
@ -226,11 +215,12 @@ void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
RtpPacketMediaType packet_type = *packet->packet_type();
RtpPacketCounter counter(*packet);
size_t size = packet->size();
worker_queue_->PostTask(ToQueuedTask(
task_safety_, [this, now_ms, ssrc = packet->Ssrc(), packet_type,
worker_queue_->PostTask(
ToQueuedTask(task_safety_, [this, now_ms, packet_ssrc, packet_type,
counter = std::move(counter), size]() {
RTC_DCHECK_RUN_ON(worker_queue_);
UpdateRtpStats(now_ms, ssrc, packet_type, std::move(counter), size);
UpdateRtpStats(now_ms, packet_ssrc, packet_type, std::move(counter),
size);
}));
}
}
@ -516,4 +506,25 @@ void RtpSenderEgress::PeriodicUpdate() {
send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
}
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
void RtpSenderEgress::BweTestLoggingPlot(int64_t now_ms, uint32_t packet_ssrc) {
RTC_DCHECK_RUN_ON(worker_queue_);
const auto rates = GetSendRates();
if (is_audio_) {
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
rates.Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "AudioNackBitrate_kbps", now_ms,
rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
} else {
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
rates.Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "VideoNackBitrate_kbps", now_ms,
rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
}
}
#endif // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
} // namespace webrtc

6
modules/rtp_rtcp/source/rtp_sender_egress.h
View File

@ -20,6 +20,7 @@
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/data_rate.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
@ -110,6 +111,9 @@ class RtpSenderEgress {
RtpPacketMediaType packet_type,
RtpPacketCounter counter,
size_t packet_size);
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
void BweTestLoggingPlot(int64_t now_ms, uint32_t packet_ssrc);
#endif
// Called on a timer, once a second, on the worker_queue_.
void PeriodicUpdate();
@ -125,7 +129,9 @@ class RtpSenderEgress {
RtpPacketHistory* const packet_history_;
Transport* const transport_;
RtcEventLog* const event_log_;
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
const bool is_audio_;
#endif
const bool need_rtp_packet_infos_;
TransportFeedbackObserver* const transport_feedback_observer_;