Merging upstream version 126.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

3 files changed, 16 insertions, 10 deletions

diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
index 16bd4153a6..e56aa4a09c 100644
--- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
@@ -26,6 +26,7 @@
 #include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
 #include "rtc_base/checks.h"
 #include "test/field_trial.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 constexpr size_t kMtu = 1200;
@@ -52,6 +53,7 @@ RtpStream::RtpStream(int fps, int bitrate_bps)
 // previous frame, no frame will be generated. The frame is split into
 // packets.
 int64_t RtpStream::GenerateFrame(int64_t time_now_us,
+                                 int64_t* next_sequence_number,
                                  std::vector<PacketResult>* packets) {
   if (time_now_us < next_rtp_time_) {
     return next_rtp_time_;
@@ -66,6 +68,7 @@ int64_t RtpStream::GenerateFrame(int64_t time_now_us,
     packet.sent_packet.send_time =
         Timestamp::Micros(time_now_us + kSendSideOffsetUs);
     packet.sent_packet.size = DataSize::Bytes(payload_size);
+    packet.sent_packet.sequence_number = (*next_sequence_number)++;
     packets->push_back(packet);
   }
   next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
@@ -131,14 +134,15 @@ void StreamGenerator::SetBitrateBps(int bitrate_bps) {
 
 // TODO(holmer): Break out the channel simulation part from this class to make
 // it possible to simulate different types of channels.
-int64_t StreamGenerator::GenerateFrame(std::vector<PacketResult>* packets,
-                                       int64_t time_now_us) {
+int64_t StreamGenerator::GenerateFrame(int64_t time_now_us,
+                                       int64_t* next_sequence_number,
+                                       std::vector<PacketResult>* packets) {
   RTC_CHECK(packets != NULL);
   RTC_CHECK(packets->empty());
   RTC_CHECK_GT(capacity_, 0);
   auto it =
       std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
-  (*it)->GenerateFrame(time_now_us, packets);
+  (*it)->GenerateFrame(time_now_us, next_sequence_number, packets);
   for (PacketResult& packet : *packets) {
     int capacity_bpus = capacity_ / 1000;
     int64_t required_network_time_us =
@@ -233,8 +237,8 @@ bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
   stream_generator_->SetBitrateBps(bitrate_bps);
   std::vector<PacketResult> packets;
 
-  int64_t next_time_us =
-      stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
+  int64_t next_time_us = stream_generator_->GenerateFrame(
+      clock_.TimeInMicroseconds(), &next_sequence_number_, &packets);
   if (packets.empty())
     return false;
 
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
index 89eb1a353f..634e6a4d82 100644
--- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
@@ -15,12 +15,11 @@
 #include <stdint.h>
 
 #include <memory>
-#include <string>
 #include <vector>
 
-#include "absl/strings/string_view.h"
 #include "api/transport/field_trial_based_config.h"
 #include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
 #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
 #include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
 #include "system_wrappers/include/clock.h"
@@ -61,6 +60,7 @@ class RtpStream {
   // previous frame, no frame will be generated. The frame is split into
   // packets.
   int64_t GenerateFrame(int64_t time_now_us,
+                        int64_t* next_sequence_number,
                         std::vector<PacketResult>* packets);
 
   // The send-side time when the next frame can be generated.
@@ -102,8 +102,9 @@ class StreamGenerator {
 
   // TODO(holmer): Break out the channel simulation part from this class to make
   // it possible to simulate different types of channels.
-  int64_t GenerateFrame(std::vector<PacketResult>* packets,
-                        int64_t time_now_us);
+  int64_t GenerateFrame(int64_t time_now_us,
+                        int64_t* next_sequence_number,
+                        std::vector<PacketResult>* packets);
 
  private:
   // Capacity of the simulated channel in bits per second.
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
index 22693d67e9..211d86c95d 100644
--- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
@@ -281,7 +281,8 @@ void SendSideBandwidthEstimation::OnRouteChange() {
   uma_update_state_ = kNoUpdate;
   uma_rtt_state_ = kNoUpdate;
   last_rtc_event_log_ = Timestamp::MinusInfinity();
-  if (loss_based_bandwidth_estimator_v2_->UseInStartPhase()) {
+  if (LossBasedBandwidthEstimatorV2Enabled() &&
+      loss_based_bandwidth_estimator_v2_->UseInStartPhase()) {
     loss_based_bandwidth_estimator_v2_.reset(
         new LossBasedBweV2(key_value_config_));
   }