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f782492948
configuration time. Bug: webrtc:8995 Change-Id: I3d63a76e472a8948c98c98450e96d3301fa2688b Reviewed-on: https://webrtc-review.googlesource.com/78701 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23409}
471 lines
19 KiB
C++
471 lines
19 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
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#include "modules/rtp_rtcp/include/rtp_receiver.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/rtp_receiver_impl.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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using ::testing::NiceMock;
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using ::testing::UnorderedElementsAre;
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const uint32_t kTestRate = 64000u;
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const uint8_t kTestPayload[] = {'t', 'e', 's', 't'};
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const uint8_t kPcmuPayloadType = 96;
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const int64_t kGetSourcesTimeoutMs = 10000;
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const uint32_t kSsrc1 = 123;
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const uint32_t kSsrc2 = 124;
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const uint32_t kCsrc1 = 111;
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const uint32_t kCsrc2 = 222;
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static uint32_t rtp_timestamp(int64_t time_ms) {
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return static_cast<uint32_t>(time_ms * kTestRate / 1000);
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}
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} // namespace
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class RtpReceiverTest : public ::testing::Test {
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protected:
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RtpReceiverTest()
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: fake_clock_(123456),
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rtp_receiver_(
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RtpReceiver::CreateAudioReceiver(&fake_clock_,
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&mock_rtp_data_,
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&rtp_payload_registry_)) {
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rtp_receiver_->RegisterReceivePayload(kPcmuPayloadType,
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SdpAudioFormat("PCMU", 8000, 1));
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}
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~RtpReceiverTest() {}
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bool FindSourceByIdAndType(const std::vector<RtpSource>& sources,
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uint32_t source_id,
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RtpSourceType type,
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RtpSource* source) {
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for (size_t i = 0; i < sources.size(); ++i) {
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if (sources[i].source_id() == source_id &&
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sources[i].source_type() == type) {
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(*source) = sources[i];
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return true;
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}
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}
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return false;
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}
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SimulatedClock fake_clock_;
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NiceMock<MockRtpData> mock_rtp_data_;
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RTPPayloadRegistry rtp_payload_registry_;
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std::unique_ptr<RtpReceiver> rtp_receiver_;
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};
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TEST_F(RtpReceiverTest, GetSources) {
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int64_t now_ms = fake_clock_.TimeInMilliseconds();
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RTPHeader header;
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header.payloadType = kPcmuPayloadType;
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header.ssrc = kSsrc1;
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header.timestamp = rtp_timestamp(now_ms);
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header.numCSRCs = 2;
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header.arrOfCSRCs[0] = kCsrc1;
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header.arrOfCSRCs[1] = kCsrc2;
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const PayloadUnion payload_specific{
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AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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auto sources = rtp_receiver_->GetSources();
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// One SSRC source and two CSRC sources.
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EXPECT_THAT(sources, UnorderedElementsAre(
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RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC),
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RtpSource(now_ms, kCsrc1, RtpSourceType::CSRC),
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RtpSource(now_ms, kCsrc2, RtpSourceType::CSRC)));
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// Advance the fake clock and the method is expected to return the
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// contributing source object with same source id and updated timestamp.
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fake_clock_.AdvanceTimeMilliseconds(1);
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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sources = rtp_receiver_->GetSources();
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now_ms = fake_clock_.TimeInMilliseconds();
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EXPECT_THAT(sources, UnorderedElementsAre(
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RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC),
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RtpSource(now_ms, kCsrc1, RtpSourceType::CSRC),
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RtpSource(now_ms, kCsrc2, RtpSourceType::CSRC)));
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// Test the edge case that the sources are still there just before the
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// timeout.
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int64_t prev_time_ms = fake_clock_.TimeInMilliseconds();
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fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
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sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources,
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UnorderedElementsAre(
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RtpSource(prev_time_ms, kSsrc1, RtpSourceType::SSRC),
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RtpSource(prev_time_ms, kCsrc1, RtpSourceType::CSRC),
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RtpSource(prev_time_ms, kCsrc2, RtpSourceType::CSRC)));
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// Time out.
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fake_clock_.AdvanceTimeMilliseconds(1);
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sources = rtp_receiver_->GetSources();
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// All the sources should be out of date.
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ASSERT_EQ(0u, sources.size());
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}
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// Test the case that the SSRC is changed.
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TEST_F(RtpReceiverTest, GetSourcesChangeSSRC) {
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int64_t prev_time_ms = -1;
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int64_t now_ms = fake_clock_.TimeInMilliseconds();
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RTPHeader header;
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header.payloadType = kPcmuPayloadType;
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header.ssrc = kSsrc1;
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header.timestamp = rtp_timestamp(now_ms);
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const PayloadUnion payload_specific{
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AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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auto sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources, UnorderedElementsAre(
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RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
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// The SSRC is changed and the old SSRC is expected to be returned.
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fake_clock_.AdvanceTimeMilliseconds(100);
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prev_time_ms = now_ms;
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now_ms = fake_clock_.TimeInMilliseconds();
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header.ssrc = kSsrc2;
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header.timestamp = rtp_timestamp(now_ms);
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources, UnorderedElementsAre(
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RtpSource(prev_time_ms, kSsrc1, RtpSourceType::SSRC),
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RtpSource(now_ms, kSsrc2, RtpSourceType::SSRC)));
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// The SSRC is changed again and happen to be changed back to 1. No
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// duplication is expected.
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fake_clock_.AdvanceTimeMilliseconds(100);
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header.ssrc = kSsrc1;
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header.timestamp = rtp_timestamp(now_ms);
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prev_time_ms = now_ms;
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now_ms = fake_clock_.TimeInMilliseconds();
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources, UnorderedElementsAre(
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RtpSource(prev_time_ms, kSsrc2, RtpSourceType::SSRC),
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RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
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// Old SSRC source timeout.
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fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
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now_ms = fake_clock_.TimeInMilliseconds();
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources, UnorderedElementsAre(
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RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
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}
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TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
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int64_t now_ms = fake_clock_.TimeInMilliseconds();
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RTPHeader header;
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header.payloadType = kPcmuPayloadType;
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header.timestamp = rtp_timestamp(now_ms);
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const PayloadUnion payload_specific{
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AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
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header.numCSRCs = 1;
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size_t kSourceListSize = 20;
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for (size_t i = 0; i < kSourceListSize; ++i) {
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header.ssrc = i;
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header.arrOfCSRCs[0] = (i + 1);
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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}
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RtpSource source(0, 0, RtpSourceType::SSRC);
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auto sources = rtp_receiver_->GetSources();
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// Expect |kSourceListSize| SSRC sources and |kSourceListSize| CSRC sources.
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ASSERT_EQ(2 * kSourceListSize, sources.size());
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for (size_t i = 0; i < kSourceListSize; ++i) {
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// The SSRC source IDs are expected to be 19, 18, 17 ... 0
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
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EXPECT_EQ(now_ms, source.timestamp_ms());
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// The CSRC source IDs are expected to be 20, 19, 18 ... 1
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, (i + 1), RtpSourceType::CSRC, &source));
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EXPECT_EQ(now_ms, source.timestamp_ms());
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}
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fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
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for (size_t i = 0; i < kSourceListSize; ++i) {
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// The SSRC source IDs are expected to be 19, 18, 17 ... 0
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
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EXPECT_EQ(now_ms, source.timestamp_ms());
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// The CSRC source IDs are expected to be 20, 19, 18 ... 1
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, (i + 1), RtpSourceType::CSRC, &source));
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EXPECT_EQ(now_ms, source.timestamp_ms());
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}
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// Timeout. All the existing objects are out of date and are expected to be
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// removed.
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fake_clock_.AdvanceTimeMilliseconds(1);
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header.ssrc = kSsrc1;
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header.arrOfCSRCs[0] = kCsrc1;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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auto* rtp_receiver_impl = static_cast<RtpReceiverImpl*>(rtp_receiver_.get());
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auto ssrc_sources = rtp_receiver_impl->ssrc_sources_for_testing();
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ASSERT_EQ(1u, ssrc_sources.size());
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EXPECT_EQ(kSsrc1, ssrc_sources.begin()->source_id());
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EXPECT_EQ(RtpSourceType::SSRC, ssrc_sources.begin()->source_type());
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
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ssrc_sources.begin()->timestamp_ms());
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auto csrc_sources = rtp_receiver_impl->csrc_sources_for_testing();
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ASSERT_EQ(1u, csrc_sources.size());
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EXPECT_EQ(kCsrc1, csrc_sources.begin()->source_id());
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EXPECT_EQ(RtpSourceType::CSRC, csrc_sources.begin()->source_type());
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
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csrc_sources.begin()->timestamp_ms());
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}
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// The audio level from the RTPHeader extension should be stored in the
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// RtpSource with the matching SSRC.
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TEST_F(RtpReceiverTest, GetSourcesContainsAudioLevelExtension) {
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RTPHeader header;
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int64_t time1_ms = fake_clock_.TimeInMilliseconds();
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header.payloadType = kPcmuPayloadType;
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header.ssrc = kSsrc1;
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header.timestamp = rtp_timestamp(time1_ms);
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header.extension.hasAudioLevel = true;
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header.extension.audioLevel = 10;
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const PayloadUnion payload_specific{
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AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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auto sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
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time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
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// Receive a packet from a different SSRC with a different level and check
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// that they are both remembered.
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fake_clock_.AdvanceTimeMilliseconds(1);
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int64_t time2_ms = fake_clock_.TimeInMilliseconds();
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header.ssrc = kSsrc2;
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header.timestamp = rtp_timestamp(time2_ms);
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header.extension.hasAudioLevel = true;
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header.extension.audioLevel = 20;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources,
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UnorderedElementsAre(
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RtpSource(time1_ms, kSsrc1, RtpSourceType::SSRC, 10),
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RtpSource(time2_ms, kSsrc2, RtpSourceType::SSRC, 20)));
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// Receive a packet from the first SSRC again and check that the level is
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// updated.
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fake_clock_.AdvanceTimeMilliseconds(1);
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int64_t time3_ms = fake_clock_.TimeInMilliseconds();
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header.ssrc = kSsrc1;
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header.timestamp = rtp_timestamp(time3_ms);
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header.extension.hasAudioLevel = true;
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header.extension.audioLevel = 30;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources,
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UnorderedElementsAre(
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RtpSource(time3_ms, kSsrc1, RtpSourceType::SSRC, 30),
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RtpSource(time2_ms, kSsrc2, RtpSourceType::SSRC, 20)));
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}
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TEST_F(RtpReceiverTest,
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MissingAudioLevelHeaderExtensionClearsRtpSourceAudioLevel) {
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RTPHeader header;
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int64_t time1_ms = fake_clock_.TimeInMilliseconds();
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header.payloadType = kPcmuPayloadType;
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header.ssrc = kSsrc1;
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header.timestamp = rtp_timestamp(time1_ms);
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header.extension.hasAudioLevel = true;
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header.extension.audioLevel = 10;
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const PayloadUnion payload_specific{
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AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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auto sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
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time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
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// Receive a second packet without the audio level header extension and check
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// that the audio level is cleared.
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fake_clock_.AdvanceTimeMilliseconds(1);
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int64_t time2_ms = fake_clock_.TimeInMilliseconds();
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header.timestamp = rtp_timestamp(time2_ms);
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header.extension.hasAudioLevel = false;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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sources = rtp_receiver_->GetSources();
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EXPECT_THAT(sources, UnorderedElementsAre(
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RtpSource(time2_ms, kSsrc1, RtpSourceType::SSRC)));
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}
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TEST_F(RtpReceiverTest, UpdatesTimestampsIfAndOnlyIfPacketArrivesInOrder) {
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RTPHeader header;
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int64_t time1_ms = fake_clock_.TimeInMilliseconds();
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header.payloadType = kPcmuPayloadType;
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header.ssrc = kSsrc1;
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header.timestamp = rtp_timestamp(time1_ms);
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header.extension.hasAudioLevel = true;
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header.extension.audioLevel = 10;
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header.sequenceNumber = 0xfff0;
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const PayloadUnion payload_specific{
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AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
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uint32_t latest_timestamp;
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int64_t latest_receive_time_ms;
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// No packet received yet.
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EXPECT_FALSE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
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&latest_receive_time_ms));
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// Initial packet
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const uint32_t timestamp_1 = header.timestamp;
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const int64_t receive_time_1 = fake_clock_.TimeInMilliseconds();
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
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&latest_receive_time_ms));
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EXPECT_EQ(latest_timestamp, timestamp_1);
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EXPECT_EQ(latest_receive_time_ms, receive_time_1);
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// Late packet, timestamp not recorded.
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fake_clock_.AdvanceTimeMilliseconds(10);
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header.timestamp -= 900;
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header.sequenceNumber -= 2;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
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&latest_receive_time_ms));
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EXPECT_EQ(latest_timestamp, timestamp_1);
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EXPECT_EQ(latest_receive_time_ms, receive_time_1);
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// New packet, still late, no wraparound.
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fake_clock_.AdvanceTimeMilliseconds(10);
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header.timestamp += 1800;
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header.sequenceNumber += 1;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
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&latest_receive_time_ms));
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EXPECT_EQ(latest_timestamp, timestamp_1);
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EXPECT_EQ(latest_receive_time_ms, receive_time_1);
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// New packet, new timestamp recorded
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fake_clock_.AdvanceTimeMilliseconds(10);
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header.timestamp += 900;
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header.sequenceNumber += 2;
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const uint32_t timestamp_2 = header.timestamp;
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const int64_t receive_time_2 = fake_clock_.TimeInMilliseconds();
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const uint16_t seqno_2 = header.sequenceNumber;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
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&latest_receive_time_ms));
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EXPECT_EQ(latest_timestamp, timestamp_2);
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EXPECT_EQ(latest_receive_time_ms, receive_time_2);
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// New packet, timestamp wraps around
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fake_clock_.AdvanceTimeMilliseconds(10);
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header.timestamp += 900;
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header.sequenceNumber += 20;
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const uint32_t timestamp_3 = header.timestamp;
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const int64_t receive_time_3 = fake_clock_.TimeInMilliseconds();
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EXPECT_LT(header.sequenceNumber, seqno_2); // Wrap-around
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific));
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EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
|
|
&latest_receive_time_ms));
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|
EXPECT_EQ(latest_timestamp, timestamp_3);
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EXPECT_EQ(latest_receive_time_ms, receive_time_3);
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}
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|
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TEST_F(RtpReceiverTest, UpdatesTimestampsWhenStreamResets) {
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RTPHeader header;
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|
int64_t time1_ms = fake_clock_.TimeInMilliseconds();
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|
header.payloadType = kPcmuPayloadType;
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|
header.ssrc = kSsrc1;
|
|
header.timestamp = rtp_timestamp(time1_ms);
|
|
header.extension.hasAudioLevel = true;
|
|
header.extension.audioLevel = 10;
|
|
header.sequenceNumber = 0xfff0;
|
|
|
|
const PayloadUnion payload_specific{
|
|
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
|
|
uint32_t latest_timestamp;
|
|
int64_t latest_receive_time_ms;
|
|
|
|
// No packet received yet.
|
|
EXPECT_FALSE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
|
|
&latest_receive_time_ms));
|
|
// Initial packet
|
|
const uint32_t timestamp_1 = header.timestamp;
|
|
const int64_t receive_time_1 = fake_clock_.TimeInMilliseconds();
|
|
const uint16_t seqno_1 = header.sequenceNumber;
|
|
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
|
header, kTestPayload, sizeof(kTestPayload), payload_specific));
|
|
EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
|
|
&latest_receive_time_ms));
|
|
EXPECT_EQ(latest_timestamp, timestamp_1);
|
|
EXPECT_EQ(latest_receive_time_ms, receive_time_1);
|
|
|
|
// Packet with far in the past seqno, but unlikely to be a wrap-around.
|
|
// Treated as a seqno discontinuity, and timestamp is recorded.
|
|
fake_clock_.AdvanceTimeMilliseconds(10);
|
|
header.timestamp += 900;
|
|
header.sequenceNumber = 0x9000;
|
|
|
|
const uint32_t timestamp_2 = header.timestamp;
|
|
const int64_t receive_time_2 = fake_clock_.TimeInMilliseconds();
|
|
const uint16_t seqno_2 = header.sequenceNumber;
|
|
EXPECT_LT(seqno_1 - seqno_2, 0x8000); // In the past.
|
|
|
|
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
|
|
header, kTestPayload, sizeof(kTestPayload), payload_specific));
|
|
EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
|
|
&latest_receive_time_ms));
|
|
EXPECT_EQ(latest_timestamp, timestamp_2);
|
|
EXPECT_EQ(latest_receive_time_ms, receive_time_2);
|
|
}
|
|
|
|
} // namespace webrtc
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