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Remove dependency from rtp_rtcp module to remote_bitrate_estimator

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This depenency is not needed and may lead to a circular dependency. The cl removes old unused functionaliy to log BWE related statistics using compile time flags.

Bug: webrtc:42225697
Change-Id: I6cc01b367c0c48ab30f34c12a10afc58d1e7822f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352142
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42386}
This commit is contained in:
Per K 2024-05-27 08:44:28 +00:00 committed by WebRTC LUCI CQ
parent d297c4c23f
commit 30f1cb318b
8 changed files with 0 additions and 68 deletions
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8
modules/rtp_rtcp/BUILD.gn
View File

@ -270,12 +270,6 @@ rtc_library("rtp_rtcp") {
]
}
if (rtc_enable_bwe_test_logging) {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
} else {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
}
deps = [
":leb128",
":rtp_rtcp_format",
@ -349,7 +343,6 @@ rtc_library("rtp_rtcp") {
"../../rtc_base/task_utils:repeating_task",
"../../system_wrappers",
"../../system_wrappers:metrics",
"../remote_bitrate_estimator",
"../video_coding:codec_globals_headers",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/base:core_headers",
@ -388,7 +381,6 @@ rtc_source_set("rtp_rtcp_legacy") {
"../../rtc_base:macromagic",
"../../rtc_base/synchronization:mutex",
"../../system_wrappers",
"../remote_bitrate_estimator",
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",

23
modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
View File

@ -17,7 +17,6 @@
#include "absl/strings/match.h"
#include "api/units/timestamp.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"
namespace webrtc {
@ -69,7 +68,6 @@ DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
packet_history_(packet_history),
transport_(config.outgoing_transport),
event_log_(config.event_log),
is_audio_(config.audio),
need_rtp_packet_infos_(config.need_rtp_packet_infos),
transport_feedback_observer_(config.transport_feedback_callback),
send_packet_observer_(config.send_packet_observer),
@ -94,27 +92,6 @@ void DEPRECATED_RtpSenderEgress::SendPacket(
RTC_DCHECK(HasCorrectSsrc(*packet));
Timestamp now = clock_->CurrentTime();
int64_t now_ms = now.ms();
if (is_audio_) {
#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);
#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
}
PacketOptions options;
{
MutexLock lock(&lock_);

1
modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
View File

@ -106,7 +106,6 @@ class DEPRECATED_RtpSenderEgress {
RtpPacketHistory* const packet_history_;
Transport* const transport_;
RtcEventLog* const event_log_;
const bool is_audio_;
const bool need_rtp_packet_infos_;
TransportFeedbackObserver* const transport_feedback_observer_;

6
modules/rtp_rtcp/source/receive_statistics_impl.cc
View File

@ -17,7 +17,6 @@
#include <vector>
#include "api/units/time_delta.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/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
@ -282,11 +281,6 @@ void StreamStatisticianImpl::MaybeAppendReportBlockAndReset(
// Only for report blocks in RTCP SR and RR.
last_report_cumulative_loss_ = cumulative_loss_;
last_report_seq_max_ = received_seq_max_;
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts", now.ms(),
cumulative_loss_, ssrc_);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max_pkts", now.ms(),
(received_seq_max_ - received_seq_first_),
ssrc_);
}
absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {

1
modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -24,7 +24,6 @@
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_nack_stats.h"

24
modules/rtp_rtcp/source/rtp_sender_egress.cc
View File

@ -161,9 +161,6 @@ void RtpSenderEgress::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
}
const Timestamp now = clock_->CurrentTime();
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
BweTestLoggingPlot(now, packet->Ssrc());
#endif
if (need_rtp_packet_infos_ &&
packet->packet_type() == RtpPacketToSend::Type::kVideo) {
// Last packet of a frame, add it to sequence number info map.
@ -501,25 +498,4 @@ void RtpSenderEgress::PeriodicUpdate() {
send_rates.Sum().bps(),
send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
}
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
void RtpSenderEgress::BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc) {
RTC_DCHECK_RUN_ON(worker_queue_);
const auto rates = GetSendRates(now);
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

4
modules/rtp_rtcp/source/rtp_sender_egress.h
View File

@ -25,7 +25,6 @@
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.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/packet_sequencer.h"
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
@ -120,9 +119,6 @@ class RtpSenderEgress {
RtpPacketMediaType packet_type,
RtpPacketCounter counter,
size_t packet_size);
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
void BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc);
#endif
// Called on a timer, once a second, on the worker_queue_.
void PeriodicUpdate();

1
modules/rtp_rtcp/source/rtp_sender_video.cc
View File

@ -27,7 +27,6 @@
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.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/absolute_capture_time_sender.h"
#include "modules/rtp_rtcp/source/byte_io.h"