In remote bitrate estimator pass packet using RtpPacketReceived class
Bug: webrtc:15054
Change-Id: I23c9008e1979a56bba9421a00e4f0f8ff937d122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40485}
diff --git a/modules/congestion_controller/receive_side_congestion_controller.cc b/modules/congestion_controller/receive_side_congestion_controller.cc
index d4c745b..e042678 100644
--- a/modules/congestion_controller/receive_side_congestion_controller.cc
+++ b/modules/congestion_controller/receive_side_congestion_controller.cc
@@ -98,11 +98,8 @@
} else {
// Receive-side BWE.
MutexLock lock(&mutex_);
- RTPHeader header;
- packet.GetHeader(&header);
PickEstimator(packet.HasExtension<AbsoluteSendTime>());
- rbe_->IncomingPacket(packet.arrival_time().ms(),
- packet.payload_size() + packet.padding_size(), header);
+ rbe_->IncomingPacket(packet);
}
}
diff --git a/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
index 0d4e15e..93a0554 100644
--- a/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
+++ b/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
@@ -13,15 +13,13 @@
#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
#define MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_REMOTE_BITRATE_ESTIMATOR_H_
-#include <map>
-#include <memory>
+#include <cstdint>
#include <vector>
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "modules/include/module_common_types.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
namespace webrtc {
@@ -45,12 +43,8 @@
// Called for each incoming packet. Updates the incoming payload bitrate
// estimate and the over-use detector. If an over-use is detected the
- // remote bitrate estimate will be updated. Note that `payload_size` is the
- // packet size excluding headers.
- // Note that `arrival_time_ms` can be of an arbitrary time base.
- virtual void IncomingPacket(int64_t arrival_time_ms,
- size_t payload_size,
- const RTPHeader& header) = 0;
+ // remote bitrate estimate will be updated.
+ virtual void IncomingPacket(const RtpPacketReceived& rtp_packet) = 0;
// Removes all data for `ssrc`.
virtual void RemoveStream(uint32_t ssrc) = 0;
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index 135f5f3..b4df1d7 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -23,6 +23,8 @@
#include "api/units/timestamp.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/metrics.h"
@@ -208,26 +210,19 @@
}
void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(
- int64_t arrival_time_ms,
- size_t payload_size,
- const RTPHeader& header) {
- if (!header.extension.hasAbsoluteSendTime) {
+ const RtpPacketReceived& rtp_packet) {
+ uint32_t send_time_24bits;
+ if (!rtp_packet.GetExtension<AbsoluteSendTime>(&send_time_24bits)) {
RTC_LOG(LS_WARNING)
<< "RemoteBitrateEstimatorAbsSendTimeImpl: Incoming packet "
"is missing absolute send time extension!";
return;
}
- IncomingPacketInfo(Timestamp::Millis(arrival_time_ms),
- header.extension.absoluteSendTime,
- DataSize::Bytes(payload_size), header.ssrc);
-}
-void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
- Timestamp arrival_time,
- uint32_t send_time_24bits,
- DataSize payload_size,
- uint32_t ssrc) {
- RTC_CHECK(send_time_24bits < (1ul << 24));
+ Timestamp arrival_time = rtp_packet.arrival_time();
+ DataSize payload_size =
+ DataSize::Bytes(rtp_packet.payload_size() + rtp_packet.padding_size());
+
if (!uma_recorded_) {
RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, BweNames::kReceiverAbsSendTime,
BweNames::kBweNamesMax);
@@ -270,7 +265,7 @@
TimeoutStreams(now);
RTC_DCHECK(inter_arrival_);
RTC_DCHECK(estimator_);
- ssrcs_.insert_or_assign(ssrc, now);
+ ssrcs_.insert_or_assign(rtp_packet.Ssrc(), now);
// For now only try to detect probes while we don't have a valid estimate.
// We currently assume that only packets larger than 200 bytes are paced by
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
index 7db2aea..5924f2b 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
@@ -49,9 +49,7 @@
~RemoteBitrateEstimatorAbsSendTime() override;
- void IncomingPacket(int64_t arrival_time_ms,
- size_t payload_size,
- const RTPHeader& header) override;
+ void IncomingPacket(const RtpPacketReceived& rtp_packet) override;
TimeDelta Process() override;
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
void RemoveStream(uint32_t ssrc) override;
@@ -89,11 +87,6 @@
static void MaybeAddCluster(const Cluster& cluster_aggregate,
std::list<Cluster>& clusters);
- void IncomingPacketInfo(Timestamp arrival_time,
- uint32_t send_time_24bits,
- DataSize payload_size,
- uint32_t ssrc);
-
std::list<Cluster> ComputeClusters() const;
const Cluster* FindBestProbe(const std::list<Cluster>& clusters) const;
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index 2032628..077315f 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -19,6 +19,8 @@
#include "modules/remote_bitrate_estimator/inter_arrival.h"
#include "modules/remote_bitrate_estimator/overuse_detector.h"
#include "modules/remote_bitrate_estimator/overuse_estimator.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"
@@ -73,19 +75,19 @@
}
void RemoteBitrateEstimatorSingleStream::IncomingPacket(
- int64_t arrival_time_ms,
- size_t payload_size,
- const RTPHeader& header) {
+ const RtpPacketReceived& rtp_packet) {
+ absl::optional<int32_t> transmission_time_offset =
+ rtp_packet.GetExtension<TransmissionOffset>();
if (!uma_recorded_) {
- BweNames type = BweNames::kReceiverTOffset;
- if (!header.extension.hasTransmissionTimeOffset)
- type = BweNames::kReceiverNoExtension;
+ BweNames type = transmission_time_offset.has_value()
+ ? BweNames::kReceiverTOffset
+ : BweNames::kReceiverNoExtension;
RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, type, BweNames::kBweNamesMax);
uma_recorded_ = true;
}
- uint32_t ssrc = header.ssrc;
+ uint32_t ssrc = rtp_packet.Ssrc();
uint32_t rtp_timestamp =
- header.timestamp + header.extension.transmissionTimeOffset;
+ rtp_packet.Timestamp() + transmission_time_offset.value_or(0);
int64_t now_ms = clock_->TimeInMilliseconds();
SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.find(ssrc);
if (it == overuse_detectors_.end()) {
@@ -113,6 +115,7 @@
incoming_bitrate_.Reset();
last_valid_incoming_bitrate_ = 0;
}
+ size_t payload_size = rtp_packet.payload_size() + rtp_packet.padding_size();
incoming_bitrate_.Update(payload_size, now_ms);
const BandwidthUsage prior_state = estimator->detector.State();
@@ -120,7 +123,7 @@
int64_t time_delta = 0;
int size_delta = 0;
if (estimator->inter_arrival.ComputeDeltas(
- rtp_timestamp, arrival_time_ms, now_ms, payload_size,
+ rtp_timestamp, rtp_packet.arrival_time().ms(), now_ms, payload_size,
×tamp_delta, &time_delta, &size_delta)) {
double timestamp_delta_ms = timestamp_delta * kTimestampToMs;
estimator->estimator.Update(time_delta, timestamp_delta_ms, size_delta,
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
index 5e2961d..05ec03d 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
@@ -43,9 +43,7 @@
~RemoteBitrateEstimatorSingleStream() override;
- void IncomingPacket(int64_t arrival_time_ms,
- size_t payload_size,
- const RTPHeader& header) override;
+ void IncomingPacket(const RtpPacketReceived& rtp_packet) override;
TimeDelta Process() override;
void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
void RemoveStream(uint32_t ssrc) override;
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
index 899037f..ee96445 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
@@ -13,6 +13,9 @@
#include <limits>
#include <utility>
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
namespace webrtc {
@@ -229,15 +232,17 @@
int64_t arrival_time,
uint32_t rtp_timestamp,
uint32_t absolute_send_time) {
- RTPHeader header;
- memset(&header, 0, sizeof(header));
- header.ssrc = ssrc;
- header.timestamp = rtp_timestamp;
- header.extension.hasAbsoluteSendTime = true;
- header.extension.absoluteSendTime = absolute_send_time;
- RTC_CHECK_GE(arrival_time + arrival_time_offset_ms_, 0);
- bitrate_estimator_->IncomingPacket(arrival_time + arrival_time_offset_ms_,
- payload_size, header);
+ RtpHeaderExtensionMap extensions;
+ extensions.Register<AbsoluteSendTime>(1);
+ RtpPacketReceived rtp_packet(&extensions);
+ rtp_packet.SetSsrc(ssrc);
+ rtp_packet.SetTimestamp(rtp_timestamp);
+ rtp_packet.SetExtension<AbsoluteSendTime>(absolute_send_time);
+ rtp_packet.SetPayloadSize(payload_size);
+ rtp_packet.set_arrival_time(
+ Timestamp::Millis(arrival_time + arrival_time_offset_ms_));
+
+ bitrate_estimator_->IncomingPacket(rtp_packet);
}
// Generates a frame of packets belonging to a stream at a given bitrate and
