Timeout tests for TMMBR
Added a test that injects 3 packets and then times out. Rewrote the packet construction in test to use a builder format. This makes tests a lot more readable. Odd behaviour of timeout found; documented as test. BUG= TEST= Review URL: https://webrtc-codereview.appspot.com/553004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2161 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -25,6 +25,128 @@ 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(WebRtc_UWord8 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(WebRtc_UWord16 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(WebRtc_UWord32 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(WebRtc_UWord32 upper_half, WebRtc_UWord32 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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WebRtc_UWord32 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(WebRtc_UWord32 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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const WebRtc_UWord8* 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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WebRtc_UWord8 buffer_[kMaxPacketSize];
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};
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// Fake system clock, controllable to the millisecond.
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// The Epoch for this clock is Jan 1, 1970, as evidenced
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// by the NTP calculation.
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class FakeSystemClock : public RtpRtcpClock {
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public:
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FakeSystemClock()
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: time_in_ms_(1335900000) {} // A nonzero, but fake, value.
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virtual WebRtc_UWord32 GetTimeInMS() {
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return time_in_ms_;
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}
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virtual void CurrentNTP(WebRtc_UWord32& secs,
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WebRtc_UWord32& frac) {
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secs = (time_in_ms_ / 1000) + ModuleRTPUtility::NTP_JAN_1970;
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// NTP_FRAC is 2^32 - number of ticks per second in the NTP fraction.
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frac = (WebRtc_UWord32)((time_in_ms_ % 1000)
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* ModuleRTPUtility::NTP_FRAC / 1000);
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}
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void AdvanceClock(int ms_to_advance) {
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time_in_ms_ += ms_to_advance;
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}
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private:
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WebRtc_UWord32 time_in_ms_;
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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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@ -56,7 +178,8 @@ class TestTransport : public Transport,
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class RtcpReceiverTest : public ::testing::Test {
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protected:
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RtcpReceiverTest() {
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system_clock_ = ModuleRTPUtility::GetSystemClock();
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// system_clock_ = ModuleRTPUtility::GetSystemClock();
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system_clock_ = new FakeSystemClock();
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rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(0, false, system_clock_);
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rtcp_receiver_ = new RTCPReceiver(0, system_clock_, rtp_rtcp_impl_);
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test_transport_ = new TestTransport(rtcp_receiver_);
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@ -84,7 +207,7 @@ class RtcpReceiverTest : public ::testing::Test {
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return result;
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}
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RtpRtcpClock* system_clock_;
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FakeSystemClock* 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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@ -99,19 +222,13 @@ TEST_F(RtcpReceiverTest, BrokenPacketIsIgnored) {
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}
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TEST_F(RtcpReceiverTest, InjectSrPacket) {
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const WebRtc_UWord8 sr_packet[] = {
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0x81, 200, // Type 200, report count = 0
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0, 6, // length
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0, 1, 2, 3, // SSRC of sender
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0, 1, 2, 3, 4, 5, 6, 7, // NTP timestamp
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0, 1, 2, 3, // RTP timestamp
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0, 0, 0, 0, // Sender's packet count
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0, 0, 0, 0 // Sender's octet count
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};
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EXPECT_EQ(0, InjectRtcpPacket(sr_packet, sizeof(sr_packet)));
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const WebRtc_UWord32 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(0x010203U, rtcp_packet_info_.remoteSSRC);
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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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@ -122,76 +239,107 @@ TEST_F(RtcpReceiverTest, TmmbrReceivedWithNoIncomingPacket) {
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}
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TEST_F(RtcpReceiverTest, TmmbrPacketAccepted) {
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const WebRtc_UWord8 tmmbr_packet[] = {
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0x81, 200, // Type 200 SR, report count = 0
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0, 6, // length
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0, 1, 2, 3, // SSRC of sender
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0, 1, 2, 3, 4, 5, 6, 7, // NTP timestamp
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0, 1, 2, 3, // RTP timestamp
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0, 0, 0, 0, // Sender's packet count
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0, 0, 0, 0, // Sender's octet count
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// TMMBR
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0x83, 205, // Type 205 RTPFB, FMT 3 TMMBR
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0, 4, // length
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0, 1, 2, 3, // SSRC of sender
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0, 0, 1, 1, // SSRC of media source
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2, 4, 6, 8, // SSRC we ask to rate-control. Must match "our" SSRC.
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0, 55, 0, 0 // MxTBR
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};
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rtcp_receiver_->SetSSRC(0x2040608); // Matches "media source" above.
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EXPECT_EQ(0, InjectRtcpPacket(tmmbr_packet, sizeof(tmmbr_packet)));
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const WebRtc_UWord32 kMediaFlowSsrc = 0x2040608;
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const WebRtc_UWord32 kSenderSsrc = 0x10203;
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const WebRtc_UWord32 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.ptrTmmbrSet[0]);
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EXPECT_EQ(0x101U, candidate_set.ptrSsrcSet[0]);
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EXPECT_EQ(kMediaRecipientSsrc, candidate_set.ptrSsrcSet[0]);
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}
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TEST_F(RtcpReceiverTest, TmmbrPacketNotForUsIgnored) {
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const WebRtc_UWord8 tmmbr_packet[] = {
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0x81, 200, // Type 200 SR, report count = 0
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0, 6, // length
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0, 1, 2, 3, // SSRC of sender
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0, 1, 2, 3, 4, 5, 6, 7, // NTP timestamp
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0, 1, 2, 3, // RTP timestamp
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0, 0, 0, 0, // Sender's packet count
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0, 0, 0, 0, // Sender's octet count
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// TMMBR
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0x83, 205, // Type 205 RTPFB, FMT 3 TMMBR
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0, 4, // length
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0, 1, 2, 3, // SSRC of sender
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0, 0, 1, 1, // SSRC of media source
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99, 99, 99, 99, // SSRC we ask to rate-control. Different from 0x2040608.
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0, 55, 0, 0 // MxTBR
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};
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rtcp_receiver_->SetSSRC(0x2040608); // Matches "media source" above.
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EXPECT_EQ(0, InjectRtcpPacket(tmmbr_packet, sizeof(tmmbr_packet)));
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const WebRtc_UWord32 kMediaFlowSsrc = 0x2040608;
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const WebRtc_UWord32 kSenderSsrc = 0x10203;
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const WebRtc_UWord32 kMediaRecipientSsrc = 0x101;
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const WebRtc_UWord32 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 WebRtc_UWord8 tmmbr_packet[] = {
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0x81, 200, // Type 200 SR, report count = 0
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0, 6, // length
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0, 1, 2, 3, // SSRC of sender
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0, 1, 2, 3, 4, 5, 6, 7, // NTP timestamp
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0, 1, 2, 3, // RTP timestamp
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0, 0, 0, 0, // Sender's packet count
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0, 0, 0, 0, // Sender's octet count
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// TMMBR
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0x83, 205, // Type 205 RTPFB, FMT 3 TMMBR
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0, 4, // length
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0, 1, 2, 3, // SSRC of sender
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0, 0, 1, 1, // SSRC of media source
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2, 4, 6, 8, // SSRC we ask to rate-control. Must match "our" SSRC.
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0, 0, 0, 0 // MxTBR == zero
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};
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rtcp_receiver_->SetSSRC(0x2040608); // Matches "media source" above.
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EXPECT_EQ(0, InjectRtcpPacket(tmmbr_packet, sizeof(tmmbr_packet)));
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const WebRtc_UWord32 kMediaFlowSsrc = 0x2040608;
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const WebRtc_UWord32 kSenderSsrc = 0x10203;
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const WebRtc_UWord32 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 WebRtc_UWord32 kMediaFlowSsrc = 0x2040608;
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const WebRtc_UWord32 kSenderSsrc = 0x10203;
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const WebRtc_UWord32 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 (WebRtc_UWord32 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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system_clock_->AdvanceClock(5000); // 5 seconds between each packet.
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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.ptrTmmbrSet[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_->AdvanceClock(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.ptrSsrcSet[0]);
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}
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} // Anonymous namespace
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} // namespace webrtc
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