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7120742701
Now that we have moved WebRTC from src/webrtc to src/, common_types.h and typedefs.h are triggering a cpplint error. The cpplint complaint is: Include the directory when naming .h files [build/include] [4] This CL disables the error but we have to remove these two headers from the root directory. NOPRESUBMIT=true Bug: webrtc:5876 Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333 Reviewed-on: https://webrtc-review.googlesource.com/1577 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19859}
342 lines
13 KiB
C++
342 lines
13 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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const bool kInOrder = true;
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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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nullptr,
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&rtp_payload_registry_)) {
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CodecInst voice_codec = {};
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voice_codec.pltype = kPcmuPayloadType;
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voice_codec.plfreq = 8000;
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voice_codec.rate = kTestRate;
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memcpy(voice_codec.plname, "PCMU", 5);
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rtp_receiver_->RegisterReceivePayload(voice_codec);
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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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PayloadUnion payload_specific = {AudioPayload()};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
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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, !kInOrder));
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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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PayloadUnion payload_specific = {AudioPayload()};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
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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, !kInOrder));
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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, !kInOrder));
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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, !kInOrder));
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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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PayloadUnion payload_specific = {AudioPayload()};
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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(header, kTestPayload,
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sizeof(kTestPayload),
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payload_specific, !kInOrder));
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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, !kInOrder));
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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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PayloadUnion payload_specific = {AudioPayload()};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
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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, !kInOrder));
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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, !kInOrder));
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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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PayloadUnion payload_specific = {AudioPayload()};
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
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header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
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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, !kInOrder));
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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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} // namespace webrtc
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