blob: 482f58d1bbd8be52b630a6a8e2e3992c808611d9 [file] [log] [blame]
/*
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
using ::testing::AllOf;
using ::testing::ElementsAre;
using ::testing::Field;
using PacketInfo = StreamFeedbackObserver::StreamPacketInfo;
static constexpr uint32_t kSsrc = 8492;
class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver {
public:
MOCK_METHOD(void,
OnPacketFeedbackVector,
(std::vector<StreamPacketInfo> packet_feedback_vector),
(override));
};
RtpPacketSendInfo CreatePacket(uint32_t ssrc,
uint16_t rtp_sequence_number,
int64_t transport_sequence_number,
bool is_retransmission) {
RtpPacketSendInfo res;
res.media_ssrc = ssrc;
res.transport_sequence_number = transport_sequence_number;
res.rtp_sequence_number = rtp_sequence_number;
res.packet_type = is_retransmission ? RtpPacketMediaType::kRetransmission
: RtpPacketMediaType::kVideo;
return res;
}
} // namespace
TEST(TransportFeedbackDemuxerTest, ObserverSanity) {
TransportFeedbackDemuxer demuxer;
MockStreamFeedbackObserver mock;
demuxer.RegisterStreamFeedbackObserver({kSsrc}, &mock);
const uint16_t kRtpStartSeq = 55;
const int64_t kTransportStartSeq = 1;
demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq, kTransportStartSeq,
/*is_retransmit=*/false));
demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq + 1,
kTransportStartSeq + 1,
/*is_retransmit=*/false));
demuxer.AddPacket(CreatePacket(
kSsrc, kRtpStartSeq + 2, kTransportStartSeq + 2, /*is_retransmit=*/true));
rtcp::TransportFeedback feedback;
feedback.SetBase(kTransportStartSeq, 1000);
ASSERT_TRUE(feedback.AddReceivedPacket(kTransportStartSeq, 1000));
// Drop middle packet.
ASSERT_TRUE(feedback.AddReceivedPacket(kTransportStartSeq + 2, 3000));
EXPECT_CALL(
mock, OnPacketFeedbackVector(ElementsAre(
AllOf(Field(&PacketInfo::received, true),
Field(&PacketInfo::ssrc, kSsrc),
Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq),
Field(&PacketInfo::is_retransmission, false)),
AllOf(Field(&PacketInfo::received, false),
Field(&PacketInfo::ssrc, kSsrc),
Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 1),
Field(&PacketInfo::is_retransmission, false)),
AllOf(Field(&PacketInfo::received, true),
Field(&PacketInfo::ssrc, kSsrc),
Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 2),
Field(&PacketInfo::is_retransmission, true)))));
demuxer.OnTransportFeedback(feedback);
demuxer.DeRegisterStreamFeedbackObserver(&mock);
demuxer.AddPacket(
CreatePacket(kSsrc, kRtpStartSeq + 3, kTransportStartSeq + 3, false));
rtcp::TransportFeedback second_feedback;
second_feedback.SetBase(kTransportStartSeq + 3, 4000);
ASSERT_TRUE(second_feedback.AddReceivedPacket(kTransportStartSeq + 3, 4000));
EXPECT_CALL(mock, OnPacketFeedbackVector).Times(0);
demuxer.OnTransportFeedback(second_feedback);
}
} // namespace webrtc