
R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
432 lines
15 KiB
C++
432 lines
15 KiB
C++
/*
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* Copyright (c) 2012 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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/*
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* This file includes unit tests for the RTCPReceiver.
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*/
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#include "testing/gmock/include/gmock/gmock.h"
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#include "testing/gtest/include/gtest/gtest.h"
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// Note: This file has no directory. Lint warning must be ignored.
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#include "webrtc/common_types.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
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#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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namespace webrtc {
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namespace { // Anonymous namespace; hide utility functions and classes.
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// A very simple packet builder class for building RTCP packets.
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class PacketBuilder {
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public:
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static const int kMaxPacketSize = 1024;
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PacketBuilder()
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: pos_(0),
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pos_of_len_(0) {
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}
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void Add8(uint8_t byte) {
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EXPECT_LT(pos_, kMaxPacketSize - 1);
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buffer_[pos_] = byte;
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++ pos_;
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}
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void Add16(uint16_t word) {
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Add8(word >> 8);
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Add8(word & 0xFF);
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}
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void Add32(uint32_t word) {
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Add8(word >> 24);
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Add8((word >> 16) & 0xFF);
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Add8((word >> 8) & 0xFF);
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Add8(word & 0xFF);
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}
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void Add64(uint32_t upper_half, uint32_t lower_half) {
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Add32(upper_half);
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Add32(lower_half);
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}
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// Set the 5-bit value in the 1st byte of the header
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// and the payload type. Set aside room for the length field,
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// and make provision for backpatching it.
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// Note: No way to set the padding bit.
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void AddRtcpHeader(int payload, int format_or_count) {
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PatchLengthField();
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Add8(0x80 | (format_or_count & 0x1F));
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Add8(payload);
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pos_of_len_ = pos_;
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Add16(0xDEAD); // Initialize length to "clearly illegal".
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}
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void AddTmmbrBandwidth(int mantissa, int exponent, int overhead) {
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// 6 bits exponent, 17 bits mantissa, 9 bits overhead.
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uint32_t word = 0;
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word |= (exponent << 26);
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word |= ((mantissa & 0x1FFFF) << 9);
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word |= (overhead & 0x1FF);
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Add32(word);
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}
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void AddSrPacket(uint32_t sender_ssrc) {
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AddRtcpHeader(200, 0);
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Add32(sender_ssrc);
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Add64(0x10203, 0x4050607); // NTP timestamp
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Add32(0x10203); // RTP timestamp
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Add32(0); // Sender's packet count
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Add32(0); // Sender's octet count
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}
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void AddRrPacket(uint32_t sender_ssrc, uint32_t rtp_ssrc,
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uint32_t extended_max) {
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AddRtcpHeader(201, 1);
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Add32(sender_ssrc);
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Add32(rtp_ssrc);
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Add32(0); // No loss.
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Add32(extended_max);
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Add32(0); // Jitter.
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Add32(0); // Last SR.
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Add32(0); // Delay since last SR.
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}
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const uint8_t* packet() {
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PatchLengthField();
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return buffer_;
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}
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unsigned int length() {
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return pos_;
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}
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private:
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void PatchLengthField() {
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if (pos_of_len_ > 0) {
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// Backpatch the packet length. The client must have taken
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// care of proper padding to 32-bit words.
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int this_packet_length = (pos_ - pos_of_len_ - 2);
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ASSERT_EQ(0, this_packet_length % 4)
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<< "Packets must be a multiple of 32 bits long"
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<< " pos " << pos_ << " pos_of_len " << pos_of_len_;
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buffer_[pos_of_len_] = this_packet_length >> 10;
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buffer_[pos_of_len_+1] = (this_packet_length >> 2) & 0xFF;
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pos_of_len_ = 0;
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}
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}
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int pos_;
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// Where the length field of the current packet is.
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// Note that 0 is not a legal value, so is used for "uninitialized".
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int pos_of_len_;
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uint8_t buffer_[kMaxPacketSize];
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};
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// This test transport verifies that no functions get called.
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class TestTransport : public Transport,
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public RtpData {
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public:
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explicit TestTransport()
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: rtcp_receiver_(NULL) {
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}
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void SetRTCPReceiver(RTCPReceiver* rtcp_receiver) {
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rtcp_receiver_ = rtcp_receiver;
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}
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virtual int SendPacket(int /*ch*/, const void* /*data*/, int /*len*/) {
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ADD_FAILURE(); // FAIL() gives a compile error.
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return -1;
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}
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// Injects an RTCP packet into the receiver.
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virtual int SendRTCPPacket(int /* ch */, const void *packet, int packet_len) {
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ADD_FAILURE();
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return 0;
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}
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virtual int OnReceivedPayloadData(const uint8_t* payloadData,
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const uint16_t payloadSize,
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const WebRtcRTPHeader* rtpHeader) {
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ADD_FAILURE();
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return 0;
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}
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RTCPReceiver* rtcp_receiver_;
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};
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class RtcpReceiverTest : public ::testing::Test {
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protected:
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RtcpReceiverTest()
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: over_use_detector_options_(),
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system_clock_(1335900000),
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remote_bitrate_observer_(),
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remote_bitrate_estimator_(
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RemoteBitrateEstimatorFactory().Create(
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&remote_bitrate_observer_,
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&system_clock_)) {
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test_transport_ = new TestTransport();
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RtpRtcp::Configuration configuration;
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configuration.id = 0;
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configuration.audio = false;
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configuration.clock = &system_clock_;
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configuration.outgoing_transport = test_transport_;
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configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get();
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rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
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rtcp_receiver_ = new RTCPReceiver(0, &system_clock_, rtp_rtcp_impl_);
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test_transport_->SetRTCPReceiver(rtcp_receiver_);
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}
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~RtcpReceiverTest() {
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delete rtcp_receiver_;
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delete rtp_rtcp_impl_;
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delete test_transport_;
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}
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// Injects an RTCP packet into the receiver.
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// Returns 0 for OK, non-0 for failure.
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int InjectRtcpPacket(const uint8_t* packet,
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uint16_t packet_len) {
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RTCPUtility::RTCPParserV2 rtcpParser(packet,
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packet_len,
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true); // Allow non-compound RTCP
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RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
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int result = rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
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&rtcpParser);
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// The NACK list is on purpose not copied below as it isn't needed by the
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// test.
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rtcp_packet_info_.rtcpPacketTypeFlags =
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rtcpPacketInformation.rtcpPacketTypeFlags;
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rtcp_packet_info_.remoteSSRC = rtcpPacketInformation.remoteSSRC;
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rtcp_packet_info_.applicationSubType =
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rtcpPacketInformation.applicationSubType;
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rtcp_packet_info_.applicationName = rtcpPacketInformation.applicationName;
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rtcp_packet_info_.reportBlock = rtcpPacketInformation.reportBlock;
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rtcp_packet_info_.fractionLost = rtcpPacketInformation.fractionLost;
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rtcp_packet_info_.roundTripTime = rtcpPacketInformation.roundTripTime;
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rtcp_packet_info_.lastReceivedExtendedHighSeqNum =
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rtcpPacketInformation.lastReceivedExtendedHighSeqNum;
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rtcp_packet_info_.jitter = rtcpPacketInformation.jitter;
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rtcp_packet_info_.interArrivalJitter =
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rtcpPacketInformation.interArrivalJitter;
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rtcp_packet_info_.sliPictureId = rtcpPacketInformation.sliPictureId;
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rtcp_packet_info_.rpsiPictureId = rtcpPacketInformation.rpsiPictureId;
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rtcp_packet_info_.receiverEstimatedMaxBitrate =
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rtcpPacketInformation.receiverEstimatedMaxBitrate;
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rtcp_packet_info_.ntp_secs = rtcpPacketInformation.ntp_secs;
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rtcp_packet_info_.ntp_frac = rtcpPacketInformation.ntp_frac;
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rtcp_packet_info_.rtp_timestamp = rtcpPacketInformation.rtp_timestamp;
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return result;
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}
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OverUseDetectorOptions over_use_detector_options_;
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SimulatedClock system_clock_;
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ModuleRtpRtcpImpl* rtp_rtcp_impl_;
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RTCPReceiver* rtcp_receiver_;
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TestTransport* test_transport_;
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RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
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MockRemoteBitrateObserver remote_bitrate_observer_;
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scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
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};
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TEST_F(RtcpReceiverTest, BrokenPacketIsIgnored) {
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const uint8_t bad_packet[] = {0, 0, 0, 0};
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EXPECT_EQ(0, InjectRtcpPacket(bad_packet, sizeof(bad_packet)));
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EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
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}
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TEST_F(RtcpReceiverTest, InjectSrPacket) {
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const uint32_t kSenderSsrc = 0x10203;
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PacketBuilder p;
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p.AddSrPacket(kSenderSsrc);
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EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
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// The parser will note the remote SSRC on a SR from other than his
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// expected peer, but will not flag that he's gotten a packet.
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EXPECT_EQ(kSenderSsrc, rtcp_packet_info_.remoteSSRC);
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EXPECT_EQ(0U,
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kRtcpSr & rtcp_packet_info_.rtcpPacketTypeFlags);
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}
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TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
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const uint32_t kSenderSsrc = 0x10203;
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const uint32_t kSourceSsrc = 0x40506;
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const int64_t kRtcpIntervalMs = 1000;
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rtcp_receiver_->SetSSRC(kSourceSsrc);
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uint32_t sequence_number = 1234;
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system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
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// No RR received, shouldn't trigger a timeout.
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EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
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// Add a RR and advance the clock just enough to not trigger a timeout.
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PacketBuilder p1;
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p1.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
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EXPECT_EQ(0, InjectRtcpPacket(p1.packet(), p1.length()));
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system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs - 1);
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EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
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// Add a RR with the same extended max as the previous RR to trigger a
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// sequence number timeout, but not a RR timeout.
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PacketBuilder p2;
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p2.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
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EXPECT_EQ(0, InjectRtcpPacket(p2.packet(), p2.length()));
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system_clock_.AdvanceTimeMilliseconds(2);
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EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
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// Advance clock enough to trigger an RR timeout too.
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system_clock_.AdvanceTimeMilliseconds(3 * kRtcpIntervalMs);
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EXPECT_TRUE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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// We should only get one timeout even though we still haven't received a new
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// RR.
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EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
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// Add a new RR with increase sequence number to reset timers.
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PacketBuilder p3;
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sequence_number++;
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p2.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
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EXPECT_EQ(0, InjectRtcpPacket(p2.packet(), p2.length()));
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EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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EXPECT_FALSE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
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// Verify we can get a timeout again once we've received new RR.
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system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
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PacketBuilder p4;
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p4.AddRrPacket(kSenderSsrc, kSourceSsrc, sequence_number);
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EXPECT_EQ(0, InjectRtcpPacket(p4.packet(), p4.length()));
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system_clock_.AdvanceTimeMilliseconds(kRtcpIntervalMs + 1);
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EXPECT_FALSE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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EXPECT_TRUE(rtcp_receiver_->RtcpRrSequenceNumberTimeout(kRtcpIntervalMs));
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system_clock_.AdvanceTimeMilliseconds(2 * kRtcpIntervalMs);
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EXPECT_TRUE(rtcp_receiver_->RtcpRrTimeout(kRtcpIntervalMs));
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}
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TEST_F(RtcpReceiverTest, TmmbrReceivedWithNoIncomingPacket) {
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// This call is expected to fail because no data has arrived.
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EXPECT_EQ(-1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
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}
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TEST_F(RtcpReceiverTest, TmmbrPacketAccepted) {
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const uint32_t kMediaFlowSsrc = 0x2040608;
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const uint32_t kSenderSsrc = 0x10203;
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const uint32_t kMediaRecipientSsrc = 0x101;
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rtcp_receiver_->SetSSRC(kMediaFlowSsrc); // Matches "media source" above.
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PacketBuilder p;
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p.AddSrPacket(kSenderSsrc);
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// TMMBR packet.
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p.AddRtcpHeader(205, 3);
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p.Add32(kSenderSsrc);
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p.Add32(kMediaRecipientSsrc);
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p.Add32(kMediaFlowSsrc);
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p.AddTmmbrBandwidth(30000, 0, 0); // 30 Kbits/sec bandwidth, no overhead.
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EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
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EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
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TMMBRSet candidate_set;
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candidate_set.VerifyAndAllocateSet(1);
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EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(1, 0, &candidate_set));
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EXPECT_LT(0U, candidate_set.Tmmbr(0));
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EXPECT_EQ(kMediaRecipientSsrc, candidate_set.Ssrc(0));
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}
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TEST_F(RtcpReceiverTest, TmmbrPacketNotForUsIgnored) {
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const uint32_t kMediaFlowSsrc = 0x2040608;
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const uint32_t kSenderSsrc = 0x10203;
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const uint32_t kMediaRecipientSsrc = 0x101;
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const uint32_t kOtherMediaFlowSsrc = 0x9999;
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PacketBuilder p;
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p.AddSrPacket(kSenderSsrc);
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// TMMBR packet.
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p.AddRtcpHeader(205, 3);
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p.Add32(kSenderSsrc);
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p.Add32(kMediaRecipientSsrc);
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p.Add32(kOtherMediaFlowSsrc); // This SSRC is not what we're sending.
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p.AddTmmbrBandwidth(30000, 0, 0);
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rtcp_receiver_->SetSSRC(kMediaFlowSsrc);
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EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
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EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
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}
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TEST_F(RtcpReceiverTest, TmmbrPacketZeroRateIgnored) {
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const uint32_t kMediaFlowSsrc = 0x2040608;
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const uint32_t kSenderSsrc = 0x10203;
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const uint32_t kMediaRecipientSsrc = 0x101;
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rtcp_receiver_->SetSSRC(kMediaFlowSsrc); // Matches "media source" above.
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PacketBuilder p;
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p.AddSrPacket(kSenderSsrc);
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// TMMBR packet.
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p.AddRtcpHeader(205, 3);
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p.Add32(kSenderSsrc);
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p.Add32(kMediaRecipientSsrc);
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p.Add32(kMediaFlowSsrc);
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p.AddTmmbrBandwidth(0, 0, 0); // Rate zero.
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EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
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EXPECT_EQ(0, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
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}
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TEST_F(RtcpReceiverTest, TmmbrThreeConstraintsTimeOut) {
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const uint32_t kMediaFlowSsrc = 0x2040608;
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const uint32_t kSenderSsrc = 0x10203;
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const uint32_t kMediaRecipientSsrc = 0x101;
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rtcp_receiver_->SetSSRC(kMediaFlowSsrc); // Matches "media source" above.
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// Inject 3 packets "from" kMediaRecipientSsrc, Ssrc+1, Ssrc+2.
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// The times of arrival are starttime + 0, starttime + 5 and starttime + 10.
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for (uint32_t ssrc = kMediaRecipientSsrc;
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ssrc < kMediaRecipientSsrc+3; ++ssrc) {
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PacketBuilder p;
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p.AddSrPacket(kSenderSsrc);
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// TMMBR packet.
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p.AddRtcpHeader(205, 3);
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p.Add32(kSenderSsrc);
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p.Add32(ssrc);
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p.Add32(kMediaFlowSsrc);
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p.AddTmmbrBandwidth(30000, 0, 0); // 30 Kbits/sec bandwidth, no overhead.
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EXPECT_EQ(0, InjectRtcpPacket(p.packet(), p.length()));
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// 5 seconds between each packet.
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system_clock_.AdvanceTimeMilliseconds(5000);
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}
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// It is now starttime+15.
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EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
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TMMBRSet candidate_set;
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candidate_set.VerifyAndAllocateSet(3);
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EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(3, 0, &candidate_set));
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EXPECT_LT(0U, candidate_set.Tmmbr(0));
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// We expect the timeout to be 25 seconds. Advance the clock by 12
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// seconds, timing out the first packet.
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system_clock_.AdvanceTimeMilliseconds(12000);
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// Odd behaviour: Just counting them does not trigger the timeout.
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EXPECT_EQ(3, rtcp_receiver_->TMMBRReceived(0, 0, NULL));
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// Odd behaviour: There's only one left after timeout, not 2.
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EXPECT_EQ(1, rtcp_receiver_->TMMBRReceived(3, 0, &candidate_set));
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EXPECT_EQ(kMediaRecipientSsrc + 2, candidate_set.Ssrc(0));
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}
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} // Anonymous namespace
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} // namespace webrtc
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