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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:35:49 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:35:49 +0000 |
commit | d8bbc7858622b6d9c278469aab701ca0b609cddf (patch) | |
tree | eff41dc61d9f714852212739e6b3738b82a2af87 /third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc | |
parent | Releasing progress-linux version 125.0.3-1~progress7.99u1. (diff) | |
download | firefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.tar.xz firefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.zip |
Merging upstream version 126.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc | 38 |
1 files changed, 38 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc index 9e6ede6dc0..2c3a71b369 100644 --- a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc +++ b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc @@ -2348,5 +2348,43 @@ TEST_F(PacingControllerTest, FlushesPacketsOnKeyFrames) { pacer->ProcessPackets(); } +TEST_F(PacingControllerTest, CanControlQueueSizeUsingTtl) { + const uint32_t kSsrc = 12345; + const uint32_t kAudioSsrc = 2345; + uint16_t sequence_number = 1234; + + PacingController::Configuration config; + config.drain_large_queues = false; + config.packet_queue_ttl.video = TimeDelta::Millis(500); + auto pacer = + std::make_unique<PacingController>(&clock_, &callback_, trials_, config); + pacer->SetPacingRates(DataRate::BitsPerSec(100'000), DataRate::Zero()); + + Timestamp send_time = Timestamp::Zero(); + for (int i = 0; i < 100; ++i) { + // Enqueue a new audio and video frame every 33ms. + if (clock_.CurrentTime() - send_time > TimeDelta::Millis(33)) { + for (int j = 0; j < 3; ++j) { + auto packet = BuildPacket(RtpPacketMediaType::kVideo, kSsrc, + /*sequence_number=*/++sequence_number, + /*capture_time_ms=*/2, + /*size_bytes=*/1000); + pacer->EnqueuePacket(std::move(packet)); + } + auto packet = BuildPacket(RtpPacketMediaType::kAudio, kAudioSsrc, + /*sequence_number=*/++sequence_number, + /*capture_time_ms=*/2, + /*size_bytes=*/100); + pacer->EnqueuePacket(std::move(packet)); + send_time = clock_.CurrentTime(); + } + + EXPECT_LE(clock_.CurrentTime() - pacer->OldestPacketEnqueueTime(), + TimeDelta::Millis(500)); + clock_.AdvanceTime(pacer->NextSendTime() - clock_.CurrentTime()); + pacer->ProcessPackets(); + } +} + } // namespace } // namespace webrtc |