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Ensure that sequence numbers are initialized in DelayBasedBwe unittests

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Bug: b/299667054
Change-Id: I6bcc4ec9e3588842e6da7d9265c145680de0c52b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332260
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41431}
This commit is contained in:
Björn Terelius 2023-12-21 14:55:29 +01:00 committed by WebRTC LUCI CQ
parent 771b524606
commit 51563cc36c
2 changed files with 14 additions and 9 deletions
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14
modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
View file

@ -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;

9
modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
View file

@ -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.