mirror of
https://github.com/mollyim/webrtc.git
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426a80ce08
Also changes default value of frame ID in RTPVideoHeader to kNoPictureId. Special care should be take so that picture ID will not be set in RTPVideoHeader unless the client on the end supports deserializing extended generic header. Bug: webrtc:9582 Change-Id: Ib096373ed187f31e51d481193a2bda56de68f167 Reviewed-on: https://webrtc-review.googlesource.com/92084 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24250}
288 lines
10 KiB
C++
288 lines
10 KiB
C++
/*
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* Copyright (c) 2013 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 <algorithm>
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#include <iterator>
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#include <list>
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#include <memory>
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#include <set>
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#include "api/call/transport.h"
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#include "call/rtp_stream_receiver_controller.h"
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#include "call/rtx_receive_stream.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "rtc_base/rate_limiter.h"
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#include "test/gtest.h"
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namespace webrtc {
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const int kVideoNackListSize = 30;
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const uint32_t kTestSsrc = 3456;
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const uint32_t kTestRtxSsrc = kTestSsrc + 1;
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const uint16_t kTestSequenceNumber = 2345;
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const uint32_t kTestNumberOfPackets = 1350;
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const int kTestNumberOfRtxPackets = 149;
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const int kNumFrames = 30;
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const int kPayloadType = 123;
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const int kRtxPayloadType = 98;
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const int64_t kMaxRttMs = 1000;
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class VerifyingMediaStream : public RtpPacketSinkInterface {
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public:
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VerifyingMediaStream() {}
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void OnRtpPacket(const RtpPacketReceived& packet) override {
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if (!sequence_numbers_.empty())
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EXPECT_EQ(kTestSsrc, packet.Ssrc());
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sequence_numbers_.push_back(packet.SequenceNumber());
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}
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std::list<uint16_t> sequence_numbers_;
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};
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class RtxLoopBackTransport : public webrtc::Transport {
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public:
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explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
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: count_(0),
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packet_loss_(0),
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consecutive_drop_start_(0),
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consecutive_drop_end_(0),
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rtx_ssrc_(rtx_ssrc),
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count_rtx_ssrc_(0),
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module_(NULL) {}
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void SetSendModule(RtpRtcp* rtpRtcpModule) { module_ = rtpRtcpModule; }
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void DropEveryNthPacket(int n) { packet_loss_ = n; }
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void DropConsecutivePackets(int start, int total) {
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consecutive_drop_start_ = start;
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consecutive_drop_end_ = start + total;
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packet_loss_ = 0;
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}
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bool SendRtp(const uint8_t* data,
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size_t len,
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const PacketOptions& options) override {
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count_++;
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RtpPacketReceived packet;
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if (!packet.Parse(data, len))
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return false;
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if (packet.Ssrc() == rtx_ssrc_) {
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count_rtx_ssrc_++;
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} else {
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// For non-RTX packets only.
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expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
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packet.SequenceNumber());
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}
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if (packet_loss_ > 0) {
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if ((count_ % packet_loss_) == 0) {
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return true;
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}
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} else if (count_ >= consecutive_drop_start_ &&
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count_ < consecutive_drop_end_) {
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return true;
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}
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EXPECT_TRUE(stream_receiver_controller_.OnRtpPacket(packet));
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return true;
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}
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bool SendRtcp(const uint8_t* data, size_t len) override {
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module_->IncomingRtcpPacket((const uint8_t*)data, len);
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return true;
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}
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int count_;
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int packet_loss_;
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int consecutive_drop_start_;
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int consecutive_drop_end_;
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uint32_t rtx_ssrc_;
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int count_rtx_ssrc_;
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RtpRtcp* module_;
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RtpStreamReceiverController stream_receiver_controller_;
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std::set<uint16_t> expected_sequence_numbers_;
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};
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class RtpRtcpRtxNackTest : public ::testing::Test {
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protected:
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RtpRtcpRtxNackTest()
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: rtp_rtcp_module_(nullptr),
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transport_(kTestRtxSsrc),
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rtx_stream_(&media_stream_, rtx_associated_payload_types_, kTestSsrc),
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payload_data_length(sizeof(payload_data)),
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fake_clock(123456),
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retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
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~RtpRtcpRtxNackTest() override {}
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void SetUp() override {
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RtpRtcp::Configuration configuration;
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configuration.audio = false;
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configuration.clock = &fake_clock;
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receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
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configuration.receive_statistics = receive_statistics_.get();
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configuration.outgoing_transport = &transport_;
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configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
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rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
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rtp_rtcp_module_->SetSSRC(kTestSsrc);
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rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
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rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
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EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
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rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
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rtp_rtcp_module_->SetStartTimestamp(111111);
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// Used for NACK processing.
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// TODO(nisse): Unclear on which side? It's confusing to use a
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// single rtp_rtcp module for both send and receive side.
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rtp_rtcp_module_->SetRemoteSSRC(kTestSsrc);
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rtp_rtcp_module_->RegisterVideoSendPayload(kPayloadType, "video");
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rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
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transport_.SetSendModule(rtp_rtcp_module_);
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media_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
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kTestSsrc, &media_stream_);
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for (size_t n = 0; n < payload_data_length; n++) {
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payload_data[n] = n % 10;
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}
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}
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int BuildNackList(uint16_t* nack_list) {
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media_stream_.sequence_numbers_.sort();
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std::list<uint16_t> missing_sequence_numbers;
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std::list<uint16_t>::iterator it = media_stream_.sequence_numbers_.begin();
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while (it != media_stream_.sequence_numbers_.end()) {
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uint16_t sequence_number_1 = *it;
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++it;
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if (it != media_stream_.sequence_numbers_.end()) {
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uint16_t sequence_number_2 = *it;
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// Add all missing sequence numbers to list
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for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
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missing_sequence_numbers.push_back(i);
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}
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}
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}
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int n = 0;
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for (it = missing_sequence_numbers.begin();
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it != missing_sequence_numbers.end(); ++it) {
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nack_list[n++] = (*it);
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}
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return n;
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}
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bool ExpectedPacketsReceived() {
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std::list<uint16_t> received_sorted;
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std::copy(media_stream_.sequence_numbers_.begin(),
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media_stream_.sequence_numbers_.end(),
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std::back_inserter(received_sorted));
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received_sorted.sort();
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return received_sorted.size() ==
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transport_.expected_sequence_numbers_.size() &&
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std::equal(received_sorted.begin(), received_sorted.end(),
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transport_.expected_sequence_numbers_.begin());
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}
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void RunRtxTest(RtxMode rtx_method, int loss) {
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rtx_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
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kTestRtxSsrc, &rtx_stream_);
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rtp_rtcp_module_->SetRtxSendStatus(rtx_method);
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rtp_rtcp_module_->SetRtxSsrc(kTestRtxSsrc);
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transport_.DropEveryNthPacket(loss);
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uint32_t timestamp = 3000;
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uint16_t nack_list[kVideoNackListSize];
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for (int frame = 0; frame < kNumFrames; ++frame) {
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RTPVideoHeader video_header;
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EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
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webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
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payload_data, payload_data_length, nullptr, &video_header, nullptr));
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// Min required delay until retransmit = 5 + RTT ms (RTT = 0).
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fake_clock.AdvanceTimeMilliseconds(5);
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int length = BuildNackList(nack_list);
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if (length > 0)
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rtp_rtcp_module_->SendNACK(nack_list, length);
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fake_clock.AdvanceTimeMilliseconds(28); // 33ms - 5ms delay.
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rtp_rtcp_module_->Process();
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// Prepare next frame.
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timestamp += 3000;
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}
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media_stream_.sequence_numbers_.sort();
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}
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void TearDown() override { delete rtp_rtcp_module_; }
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std::unique_ptr<ReceiveStatistics> receive_statistics_;
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RtpRtcp* rtp_rtcp_module_;
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RtxLoopBackTransport transport_;
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const std::map<int, int> rtx_associated_payload_types_ = {
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{kRtxPayloadType, kPayloadType}};
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VerifyingMediaStream media_stream_;
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RtxReceiveStream rtx_stream_;
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uint8_t payload_data[65000];
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size_t payload_data_length;
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SimulatedClock fake_clock;
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RateLimiter retransmission_rate_limiter_;
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std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
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std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
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};
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TEST_F(RtpRtcpRtxNackTest, LongNackList) {
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const int kNumPacketsToDrop = 900;
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const int kNumRequiredRtcp = 4;
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uint32_t timestamp = 3000;
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uint16_t nack_list[kNumPacketsToDrop];
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// Disable StorePackets to be able to set a larger packet history.
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rtp_rtcp_module_->SetStorePacketsStatus(false, 0);
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// Enable StorePackets with a packet history of 2000 packets.
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rtp_rtcp_module_->SetStorePacketsStatus(true, 2000);
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// Drop 900 packets from the second one so that we get a NACK list which is
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// big enough to require 4 RTCP packets to be fully transmitted to the sender.
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transport_.DropConsecutivePackets(2, kNumPacketsToDrop);
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// Send 30 frames which at the default size is roughly what we need to get
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// enough packets.
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for (int frame = 0; frame < kNumFrames; ++frame) {
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RTPVideoHeader video_header;
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EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
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webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
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payload_data, payload_data_length, nullptr, &video_header, nullptr));
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// Prepare next frame.
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timestamp += 3000;
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fake_clock.AdvanceTimeMilliseconds(33);
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rtp_rtcp_module_->Process();
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}
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EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
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EXPECT_FALSE(media_stream_.sequence_numbers_.empty());
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size_t last_receive_count = media_stream_.sequence_numbers_.size();
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int length = BuildNackList(nack_list);
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for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
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rtp_rtcp_module_->SendNACK(nack_list, length);
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EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
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last_receive_count = media_stream_.sequence_numbers_.size();
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EXPECT_FALSE(ExpectedPacketsReceived());
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}
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rtp_rtcp_module_->SendNACK(nack_list, length);
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EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
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EXPECT_TRUE(ExpectedPacketsReceived());
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}
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TEST_F(RtpRtcpRtxNackTest, RtxNack) {
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RunRtxTest(kRtxRetransmitted, 10);
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EXPECT_EQ(kTestSequenceNumber, *(media_stream_.sequence_numbers_.begin()));
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EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
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*(media_stream_.sequence_numbers_.rbegin()));
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EXPECT_EQ(kTestNumberOfPackets, media_stream_.sequence_numbers_.size());
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EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
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EXPECT_TRUE(ExpectedPacketsReceived());
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}
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} // namespace webrtc
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