pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 10 | #include <assert.h> |
| 11 | |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 12 | #include <algorithm> |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 13 | #include <map> |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 14 | #include <sstream> |
| 15 | #include <string> |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 16 | |
| 17 | #include "testing/gtest/include/gtest/gtest.h" |
| 18 | |
pbos@webrtc.org | 24e2089 | 2013-10-28 16:32:01 +0000 | [diff] [blame] | 19 | #include "webrtc/call.h" |
pbos@webrtc.org | c5b5ad1 | 2013-10-21 09:02:30 +0000 | [diff] [blame] | 20 | #include "webrtc/common_video/test/frame_generator.h" |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 21 | #include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h" |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 22 | #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| 23 | #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| 24 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 25 | #include "webrtc/system_wrappers/interface/event_wrapper.h" |
pbos@webrtc.org | fa996f2 | 2013-09-10 09:26:25 +0000 | [diff] [blame] | 26 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 27 | #include "webrtc/video/transport_adapter.h" |
| 28 | #include "webrtc/voice_engine/include/voe_base.h" |
| 29 | #include "webrtc/voice_engine/include/voe_codec.h" |
| 30 | #include "webrtc/voice_engine/include/voe_network.h" |
| 31 | #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| 32 | #include "webrtc/voice_engine/include/voe_video_sync.h" |
| 33 | #include "webrtc/voice_engine/test/auto_test/resource_manager.h" |
pbos@webrtc.org | 24e2089 | 2013-10-28 16:32:01 +0000 | [diff] [blame] | 34 | #include "webrtc/test/direct_transport.h" |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 35 | #include "webrtc/test/fake_audio_device.h" |
pbos@webrtc.org | 24e2089 | 2013-10-28 16:32:01 +0000 | [diff] [blame] | 36 | #include "webrtc/test/fake_decoder.h" |
| 37 | #include "webrtc/test/fake_encoder.h" |
| 38 | #include "webrtc/test/frame_generator_capturer.h" |
| 39 | #include "webrtc/test/generate_ssrcs.h" |
| 40 | #include "webrtc/test/rtp_rtcp_observer.h" |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 41 | #include "webrtc/test/testsupport/perf_test.h" |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 42 | |
| 43 | namespace webrtc { |
| 44 | |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 45 | static unsigned int kDefaultTimeoutMs = 30 * 1000; |
| 46 | static unsigned int kLongTimeoutMs = 120 * 1000; |
stefan@webrtc.org | 4985c7b | 2013-11-15 12:32:15 +0000 | [diff] [blame] | 47 | static const uint8_t kSendPayloadType = 125; |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 48 | |
pbos@webrtc.org | 362e3e5 | 2013-09-27 10:54:10 +0000 | [diff] [blame] | 49 | class CallTest : public ::testing::Test { |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 50 | public: |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 51 | CallTest() |
pbos@webrtc.org | 618a0ec | 2013-09-09 08:26:30 +0000 | [diff] [blame] | 52 | : send_stream_(NULL), |
| 53 | receive_stream_(NULL), |
| 54 | fake_encoder_(Clock::GetRealTimeClock()) {} |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 55 | |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 56 | ~CallTest() { |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 57 | EXPECT_EQ(NULL, send_stream_); |
| 58 | EXPECT_EQ(NULL, receive_stream_); |
| 59 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 60 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 61 | protected: |
pbos@webrtc.org | b5d2d16 | 2013-10-02 13:36:09 +0000 | [diff] [blame] | 62 | void CreateCalls(const Call::Config& sender_config, |
| 63 | const Call::Config& receiver_config) { |
pbos@webrtc.org | bf6d572 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 64 | sender_call_.reset(Call::Create(sender_config)); |
| 65 | receiver_call_.reset(Call::Create(receiver_config)); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 66 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 67 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 68 | void CreateTestConfigs() { |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 69 | send_config_ = sender_call_->GetDefaultSendConfig(); |
| 70 | receive_config_ = receiver_call_->GetDefaultReceiveConfig(); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 71 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 72 | test::GenerateRandomSsrcs(&send_config_, &reserved_ssrcs_); |
pbos@webrtc.org | 618a0ec | 2013-09-09 08:26:30 +0000 | [diff] [blame] | 73 | send_config_.encoder = &fake_encoder_; |
| 74 | send_config_.internal_source = false; |
| 75 | test::FakeEncoder::SetCodecSettings(&send_config_.codec, 1); |
stefan@webrtc.org | 4985c7b | 2013-11-15 12:32:15 +0000 | [diff] [blame] | 76 | send_config_.codec.plType = kSendPayloadType; |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 77 | |
pbos@webrtc.org | 618a0ec | 2013-09-09 08:26:30 +0000 | [diff] [blame] | 78 | receive_config_.codecs.clear(); |
| 79 | receive_config_.codecs.push_back(send_config_.codec); |
| 80 | ExternalVideoDecoder decoder; |
| 81 | decoder.decoder = &fake_decoder_; |
| 82 | decoder.payload_type = send_config_.codec.plType; |
| 83 | receive_config_.external_decoders.push_back(decoder); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 84 | receive_config_.rtp.ssrc = send_config_.rtp.ssrcs[0]; |
| 85 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 86 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 87 | void CreateStreams() { |
| 88 | assert(send_stream_ == NULL); |
| 89 | assert(receive_stream_ == NULL); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 90 | |
pbos@webrtc.org | 964d78e | 2013-11-20 10:40:25 +0000 | [diff] [blame^] | 91 | send_stream_ = sender_call_->CreateVideoSendStream(send_config_); |
| 92 | receive_stream_ = receiver_call_->CreateVideoReceiveStream(receive_config_); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 93 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 94 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 95 | void CreateFrameGenerator() { |
andresp@webrtc.org | 28631e7 | 2013-09-19 12:14:03 +0000 | [diff] [blame] | 96 | frame_generator_capturer_.reset( |
| 97 | test::FrameGeneratorCapturer::Create(send_stream_->Input(), |
| 98 | send_config_.codec.width, |
| 99 | send_config_.codec.height, |
| 100 | 30, |
| 101 | Clock::GetRealTimeClock())); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 102 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 103 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 104 | void StartSending() { |
| 105 | receive_stream_->StartReceive(); |
| 106 | send_stream_->StartSend(); |
pbos@webrtc.org | c5b5ad1 | 2013-10-21 09:02:30 +0000 | [diff] [blame] | 107 | if (frame_generator_capturer_.get() != NULL) |
| 108 | frame_generator_capturer_->Start(); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 109 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 110 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 111 | void StopSending() { |
pbos@webrtc.org | c5b5ad1 | 2013-10-21 09:02:30 +0000 | [diff] [blame] | 112 | if (frame_generator_capturer_.get() != NULL) |
| 113 | frame_generator_capturer_->Stop(); |
pbos@webrtc.org | 0020858 | 2013-09-05 12:38:54 +0000 | [diff] [blame] | 114 | if (send_stream_ != NULL) |
| 115 | send_stream_->StopSend(); |
| 116 | if (receive_stream_ != NULL) |
| 117 | receive_stream_->StopReceive(); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 118 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 119 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 120 | void DestroyStreams() { |
pbos@webrtc.org | 0020858 | 2013-09-05 12:38:54 +0000 | [diff] [blame] | 121 | if (send_stream_ != NULL) |
| 122 | sender_call_->DestroySendStream(send_stream_); |
| 123 | if (receive_stream_ != NULL) |
| 124 | receiver_call_->DestroyReceiveStream(receive_stream_); |
pbos@webrtc.org | bf6d572 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 125 | send_stream_ = NULL; |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 126 | receive_stream_ = NULL; |
| 127 | } |
| 128 | |
| 129 | void ReceivesPliAndRecovers(int rtp_history_ms); |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 130 | void RespectsRtcpMode(newapi::RtcpMode rtcp_mode); |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 131 | void PlaysOutAudioAndVideoInSync(); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 132 | |
pbos@webrtc.org | bf6d572 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 133 | scoped_ptr<Call> sender_call_; |
| 134 | scoped_ptr<Call> receiver_call_; |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 135 | |
pbos@webrtc.org | c179706 | 2013-08-23 09:19:30 +0000 | [diff] [blame] | 136 | VideoSendStream::Config send_config_; |
| 137 | VideoReceiveStream::Config receive_config_; |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 138 | |
pbos@webrtc.org | c179706 | 2013-08-23 09:19:30 +0000 | [diff] [blame] | 139 | VideoSendStream* send_stream_; |
| 140 | VideoReceiveStream* receive_stream_; |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 141 | |
| 142 | scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_; |
| 143 | |
pbos@webrtc.org | 618a0ec | 2013-09-09 08:26:30 +0000 | [diff] [blame] | 144 | test::FakeEncoder fake_encoder_; |
| 145 | test::FakeDecoder fake_decoder_; |
| 146 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 147 | std::map<uint32_t, bool> reserved_ssrcs_; |
| 148 | }; |
| 149 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 150 | class NackObserver : public test::RtpRtcpObserver { |
| 151 | static const int kNumberOfNacksToObserve = 4; |
| 152 | static const int kInverseProbabilityToStartLossBurst = 20; |
| 153 | static const int kMaxLossBurst = 10; |
pbos@webrtc.org | bf6d572 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 154 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 155 | public: |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 156 | NackObserver() |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 157 | : test::RtpRtcpObserver(kLongTimeoutMs), |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 158 | rtp_parser_(RtpHeaderParser::Create()), |
| 159 | drop_burst_count_(0), |
| 160 | sent_rtp_packets_(0), |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 161 | nacks_left_(kNumberOfNacksToObserve) {} |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 162 | |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 163 | private: |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 164 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 165 | EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))); |
| 166 | |
| 167 | RTPHeader header; |
| 168 | EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); |
| 169 | |
| 170 | // Never drop retransmitted packets. |
| 171 | if (dropped_packets_.find(header.sequenceNumber) != |
| 172 | dropped_packets_.end()) { |
| 173 | retransmitted_packets_.insert(header.sequenceNumber); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 174 | return SEND_PACKET; |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 175 | } |
| 176 | |
| 177 | // Enough NACKs received, stop dropping packets. |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 178 | if (nacks_left_ == 0) { |
| 179 | ++sent_rtp_packets_; |
| 180 | return SEND_PACKET; |
| 181 | } |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 182 | |
| 183 | // Still dropping packets. |
| 184 | if (drop_burst_count_ > 0) { |
| 185 | --drop_burst_count_; |
| 186 | dropped_packets_.insert(header.sequenceNumber); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 187 | return DROP_PACKET; |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 188 | } |
| 189 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 190 | // Should we start dropping packets? |
| 191 | if (sent_rtp_packets_ > 0 && |
| 192 | rand() % kInverseProbabilityToStartLossBurst == 0) { |
| 193 | drop_burst_count_ = rand() % kMaxLossBurst; |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 194 | dropped_packets_.insert(header.sequenceNumber); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 195 | return DROP_PACKET; |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 196 | } |
| 197 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 198 | ++sent_rtp_packets_; |
| 199 | return SEND_PACKET; |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 200 | } |
| 201 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 202 | virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 203 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 204 | EXPECT_TRUE(parser.IsValid()); |
| 205 | |
| 206 | bool received_nack = false; |
| 207 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 208 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 209 | if (packet_type == RTCPUtility::kRtcpRtpfbNackCode) |
| 210 | received_nack = true; |
| 211 | |
| 212 | packet_type = parser.Iterate(); |
| 213 | } |
| 214 | |
| 215 | if (received_nack) { |
| 216 | ReceivedNack(); |
| 217 | } else { |
| 218 | RtcpWithoutNack(); |
| 219 | } |
| 220 | return SEND_PACKET; |
| 221 | } |
| 222 | |
| 223 | private: |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 224 | void ReceivedNack() { |
| 225 | if (nacks_left_ > 0) |
| 226 | --nacks_left_; |
| 227 | rtcp_without_nack_count_ = 0; |
| 228 | } |
| 229 | |
| 230 | void RtcpWithoutNack() { |
| 231 | if (nacks_left_ > 0) |
| 232 | return; |
| 233 | ++rtcp_without_nack_count_; |
| 234 | |
| 235 | // All packets retransmitted and no recent NACKs. |
| 236 | if (dropped_packets_.size() == retransmitted_packets_.size() && |
| 237 | rtcp_without_nack_count_ >= kRequiredRtcpsWithoutNack) { |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 238 | observation_complete_->Set(); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 239 | } |
| 240 | } |
| 241 | |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 242 | scoped_ptr<RtpHeaderParser> rtp_parser_; |
| 243 | std::set<uint16_t> dropped_packets_; |
| 244 | std::set<uint16_t> retransmitted_packets_; |
| 245 | int drop_burst_count_; |
| 246 | uint64_t sent_rtp_packets_; |
| 247 | int nacks_left_; |
| 248 | int rtcp_without_nack_count_; |
| 249 | static const int kRequiredRtcpsWithoutNack = 2; |
| 250 | }; |
| 251 | |
pbos@webrtc.org | b5d2d16 | 2013-10-02 13:36:09 +0000 | [diff] [blame] | 252 | TEST_F(CallTest, UsesTraceCallback) { |
| 253 | const unsigned int kSenderTraceFilter = kTraceDebug; |
| 254 | const unsigned int kReceiverTraceFilter = kTraceDefault & (~kTraceDebug); |
pbos@webrtc.org | 63301bd | 2013-10-21 10:34:43 +0000 | [diff] [blame] | 255 | class TraceObserver : public TraceCallback { |
pbos@webrtc.org | b5d2d16 | 2013-10-02 13:36:09 +0000 | [diff] [blame] | 256 | public: |
| 257 | TraceObserver(unsigned int filter) |
| 258 | : filter_(filter), messages_left_(50), done_(EventWrapper::Create()) {} |
| 259 | |
| 260 | virtual void Print(TraceLevel level, |
| 261 | const char* message, |
| 262 | int length) OVERRIDE { |
| 263 | EXPECT_EQ(0u, level & (~filter_)); |
| 264 | if (--messages_left_ == 0) |
| 265 | done_->Set(); |
| 266 | } |
| 267 | |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 268 | EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); } |
pbos@webrtc.org | b5d2d16 | 2013-10-02 13:36:09 +0000 | [diff] [blame] | 269 | |
| 270 | private: |
| 271 | unsigned int filter_; |
| 272 | unsigned int messages_left_; |
| 273 | scoped_ptr<EventWrapper> done_; |
| 274 | } sender_trace(kSenderTraceFilter), receiver_trace(kReceiverTraceFilter); |
| 275 | |
| 276 | test::DirectTransport send_transport, receive_transport; |
| 277 | Call::Config sender_call_config(&send_transport); |
| 278 | sender_call_config.trace_callback = &sender_trace; |
| 279 | sender_call_config.trace_filter = kSenderTraceFilter; |
| 280 | Call::Config receiver_call_config(&receive_transport); |
| 281 | receiver_call_config.trace_callback = &receiver_trace; |
| 282 | receiver_call_config.trace_filter = kReceiverTraceFilter; |
| 283 | CreateCalls(sender_call_config, receiver_call_config); |
| 284 | send_transport.SetReceiver(receiver_call_->Receiver()); |
| 285 | receive_transport.SetReceiver(sender_call_->Receiver()); |
| 286 | |
| 287 | CreateTestConfigs(); |
| 288 | |
| 289 | CreateStreams(); |
| 290 | CreateFrameGenerator(); |
| 291 | StartSending(); |
| 292 | |
| 293 | // Wait() waits for a couple of trace callbacks to occur. |
| 294 | EXPECT_EQ(kEventSignaled, sender_trace.Wait()); |
| 295 | EXPECT_EQ(kEventSignaled, receiver_trace.Wait()); |
| 296 | |
| 297 | StopSending(); |
| 298 | send_transport.StopSending(); |
| 299 | receive_transport.StopSending(); |
| 300 | DestroyStreams(); |
| 301 | |
| 302 | // The TraceCallback instance MUST outlive Calls, destroy Calls explicitly. |
| 303 | sender_call_.reset(); |
| 304 | receiver_call_.reset(); |
| 305 | } |
| 306 | |
pbos@webrtc.org | c5b5ad1 | 2013-10-21 09:02:30 +0000 | [diff] [blame] | 307 | TEST_F(CallTest, TransmitsFirstFrame) { |
| 308 | class Renderer : public VideoRenderer { |
| 309 | public: |
| 310 | Renderer() : event_(EventWrapper::Create()) {} |
| 311 | |
| 312 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 313 | int /*time_to_render_ms*/) OVERRIDE { |
| 314 | event_->Set(); |
| 315 | } |
| 316 | |
| 317 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 318 | |
| 319 | scoped_ptr<EventWrapper> event_; |
| 320 | } renderer; |
| 321 | |
| 322 | test::DirectTransport sender_transport, receiver_transport; |
| 323 | |
| 324 | CreateCalls(Call::Config(&sender_transport), |
| 325 | Call::Config(&receiver_transport)); |
| 326 | |
| 327 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 328 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 329 | |
| 330 | CreateTestConfigs(); |
| 331 | receive_config_.renderer = &renderer; |
| 332 | |
| 333 | CreateStreams(); |
| 334 | StartSending(); |
| 335 | |
| 336 | scoped_ptr<test::FrameGenerator> frame_generator(test::FrameGenerator::Create( |
| 337 | send_config_.codec.width, send_config_.codec.height)); |
| 338 | send_stream_->Input()->PutFrame(frame_generator->NextFrame(), 0); |
| 339 | |
| 340 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 341 | << "Timed out while waiting for the frame to render."; |
| 342 | |
| 343 | StopSending(); |
| 344 | |
| 345 | sender_transport.StopSending(); |
| 346 | receiver_transport.StopSending(); |
| 347 | |
| 348 | DestroyStreams(); |
| 349 | } |
| 350 | |
pbos@webrtc.org | 362e3e5 | 2013-09-27 10:54:10 +0000 | [diff] [blame] | 351 | TEST_F(CallTest, ReceivesAndRetransmitsNack) { |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 352 | NackObserver observer; |
| 353 | |
pbos@webrtc.org | b5d2d16 | 2013-10-02 13:36:09 +0000 | [diff] [blame] | 354 | CreateCalls(Call::Config(observer.SendTransport()), |
| 355 | Call::Config(observer.ReceiveTransport())); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 356 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 357 | observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver()); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 358 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 359 | CreateTestConfigs(); |
| 360 | int rtp_history_ms = 1000; |
| 361 | send_config_.rtp.nack.rtp_history_ms = rtp_history_ms; |
| 362 | receive_config_.rtp.nack.rtp_history_ms = rtp_history_ms; |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 363 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 364 | CreateStreams(); |
| 365 | CreateFrameGenerator(); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 366 | StartSending(); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 367 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 368 | // Wait() waits for an event triggered when NACKs have been received, NACKed |
| 369 | // packets retransmitted and frames rendered again. |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 370 | EXPECT_EQ(kEventSignaled, observer.Wait()); |
| 371 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 372 | StopSending(); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 373 | |
pbos@webrtc.org | fe881f6 | 2013-08-12 12:59:04 +0000 | [diff] [blame] | 374 | observer.StopSending(); |
pbos@webrtc.org | 618a0ec | 2013-09-09 08:26:30 +0000 | [diff] [blame] | 375 | |
| 376 | DestroyStreams(); |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 377 | } |
| 378 | |
pbos@webrtc.org | 63301bd | 2013-10-21 10:34:43 +0000 | [diff] [blame] | 379 | TEST_F(CallTest, UsesFrameCallbacks) { |
| 380 | static const int kWidth = 320; |
| 381 | static const int kHeight = 240; |
| 382 | |
| 383 | class Renderer : public VideoRenderer { |
| 384 | public: |
| 385 | Renderer() : event_(EventWrapper::Create()) {} |
| 386 | |
| 387 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 388 | int /*time_to_render_ms*/) OVERRIDE { |
| 389 | EXPECT_EQ(0, *video_frame.buffer(kYPlane)) |
| 390 | << "Rendered frame should have zero luma which is applied by the " |
| 391 | "pre-render callback."; |
| 392 | event_->Set(); |
| 393 | } |
| 394 | |
| 395 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 396 | scoped_ptr<EventWrapper> event_; |
| 397 | } renderer; |
| 398 | |
| 399 | class TestFrameCallback : public I420FrameCallback { |
| 400 | public: |
| 401 | TestFrameCallback(int expected_luma_byte, int next_luma_byte) |
| 402 | : event_(EventWrapper::Create()), |
| 403 | expected_luma_byte_(expected_luma_byte), |
| 404 | next_luma_byte_(next_luma_byte) {} |
| 405 | |
| 406 | EventTypeWrapper Wait() { return event_->Wait(kDefaultTimeoutMs); } |
| 407 | |
| 408 | private: |
| 409 | virtual void FrameCallback(I420VideoFrame* frame) { |
| 410 | EXPECT_EQ(kWidth, frame->width()) |
| 411 | << "Width not as expected, callback done before resize?"; |
| 412 | EXPECT_EQ(kHeight, frame->height()) |
| 413 | << "Height not as expected, callback done before resize?"; |
| 414 | |
| 415 | // Previous luma specified, observed luma should be fairly close. |
| 416 | if (expected_luma_byte_ != -1) { |
| 417 | EXPECT_NEAR(expected_luma_byte_, *frame->buffer(kYPlane), 10); |
| 418 | } |
| 419 | |
| 420 | memset(frame->buffer(kYPlane), |
| 421 | next_luma_byte_, |
| 422 | frame->allocated_size(kYPlane)); |
| 423 | |
| 424 | event_->Set(); |
| 425 | } |
| 426 | |
| 427 | scoped_ptr<EventWrapper> event_; |
| 428 | int expected_luma_byte_; |
| 429 | int next_luma_byte_; |
| 430 | }; |
| 431 | |
| 432 | TestFrameCallback pre_encode_callback(-1, 255); // Changes luma to 255. |
| 433 | TestFrameCallback pre_render_callback(255, 0); // Changes luma from 255 to 0. |
| 434 | |
| 435 | test::DirectTransport sender_transport, receiver_transport; |
| 436 | |
| 437 | CreateCalls(Call::Config(&sender_transport), |
| 438 | Call::Config(&receiver_transport)); |
| 439 | |
| 440 | sender_transport.SetReceiver(receiver_call_->Receiver()); |
| 441 | receiver_transport.SetReceiver(sender_call_->Receiver()); |
| 442 | |
| 443 | CreateTestConfigs(); |
| 444 | send_config_.encoder = NULL; |
| 445 | send_config_.codec = sender_call_->GetVideoCodecs()[0]; |
| 446 | send_config_.codec.width = kWidth; |
| 447 | send_config_.codec.height = kHeight; |
| 448 | send_config_.pre_encode_callback = &pre_encode_callback; |
| 449 | receive_config_.pre_render_callback = &pre_render_callback; |
| 450 | receive_config_.renderer = &renderer; |
| 451 | |
| 452 | CreateStreams(); |
| 453 | StartSending(); |
| 454 | |
| 455 | // Create frames that are smaller than the send width/height, this is done to |
| 456 | // check that the callbacks are done after processing video. |
| 457 | scoped_ptr<test::FrameGenerator> frame_generator( |
| 458 | test::FrameGenerator::Create(kWidth / 2, kHeight / 2)); |
| 459 | send_stream_->Input()->PutFrame(frame_generator->NextFrame(), 0); |
| 460 | |
| 461 | EXPECT_EQ(kEventSignaled, pre_encode_callback.Wait()) |
| 462 | << "Timed out while waiting for pre-encode callback."; |
| 463 | EXPECT_EQ(kEventSignaled, pre_render_callback.Wait()) |
| 464 | << "Timed out while waiting for pre-render callback."; |
| 465 | EXPECT_EQ(kEventSignaled, renderer.Wait()) |
| 466 | << "Timed out while waiting for the frame to render."; |
| 467 | |
| 468 | StopSending(); |
| 469 | |
| 470 | sender_transport.StopSending(); |
| 471 | receiver_transport.StopSending(); |
| 472 | |
| 473 | DestroyStreams(); |
| 474 | } |
| 475 | |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 476 | class PliObserver : public test::RtpRtcpObserver, public VideoRenderer { |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 477 | static const int kInverseDropProbability = 16; |
pbos@webrtc.org | bf6d572 | 2013-09-09 15:04:25 +0000 | [diff] [blame] | 478 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 479 | public: |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 480 | explicit PliObserver(bool nack_enabled) |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 481 | : test::RtpRtcpObserver(kLongTimeoutMs), |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 482 | rtp_header_parser_(RtpHeaderParser::Create()), |
| 483 | nack_enabled_(nack_enabled), |
| 484 | first_retransmitted_timestamp_(0), |
| 485 | last_send_timestamp_(0), |
| 486 | rendered_frame_(false), |
| 487 | received_pli_(false) {} |
| 488 | |
| 489 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 490 | RTPHeader header; |
pbos@webrtc.org | 96ff6ab | 2013-08-19 16:35:36 +0000 | [diff] [blame] | 491 | EXPECT_TRUE( |
| 492 | rtp_header_parser_->Parse(packet, static_cast<int>(length), &header)); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 493 | |
| 494 | // Drop all NACK retransmissions. This is to force transmission of a PLI. |
| 495 | if (header.timestamp < last_send_timestamp_) |
| 496 | return DROP_PACKET; |
| 497 | |
| 498 | if (received_pli_) { |
| 499 | if (first_retransmitted_timestamp_ == 0) { |
| 500 | first_retransmitted_timestamp_ = header.timestamp; |
| 501 | } |
| 502 | } else if (rendered_frame_ && rand() % kInverseDropProbability == 0) { |
| 503 | return DROP_PACKET; |
| 504 | } |
| 505 | |
| 506 | last_send_timestamp_ = header.timestamp; |
| 507 | return SEND_PACKET; |
| 508 | } |
| 509 | |
| 510 | virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 511 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 512 | EXPECT_TRUE(parser.IsValid()); |
| 513 | |
| 514 | for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 515 | packet_type != RTCPUtility::kRtcpNotValidCode; |
| 516 | packet_type = parser.Iterate()) { |
| 517 | if (!nack_enabled_) |
| 518 | EXPECT_NE(packet_type, RTCPUtility::kRtcpRtpfbNackCode); |
| 519 | |
| 520 | if (packet_type == RTCPUtility::kRtcpPsfbPliCode) { |
| 521 | received_pli_ = true; |
| 522 | break; |
| 523 | } |
| 524 | } |
| 525 | return SEND_PACKET; |
| 526 | } |
| 527 | |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 528 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 529 | int time_to_render_ms) OVERRIDE { |
| 530 | CriticalSectionScoped crit_(lock_.get()); |
| 531 | if (first_retransmitted_timestamp_ != 0 && |
| 532 | video_frame.timestamp() > first_retransmitted_timestamp_) { |
| 533 | EXPECT_TRUE(received_pli_); |
| 534 | observation_complete_->Set(); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 535 | } |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 536 | rendered_frame_ = true; |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 537 | } |
| 538 | |
| 539 | private: |
| 540 | scoped_ptr<RtpHeaderParser> rtp_header_parser_; |
| 541 | bool nack_enabled_; |
| 542 | |
| 543 | uint32_t first_retransmitted_timestamp_; |
| 544 | uint32_t last_send_timestamp_; |
| 545 | |
| 546 | bool rendered_frame_; |
| 547 | bool received_pli_; |
| 548 | }; |
| 549 | |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 550 | void CallTest::ReceivesPliAndRecovers(int rtp_history_ms) { |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 551 | PliObserver observer(rtp_history_ms > 0); |
| 552 | |
pbos@webrtc.org | b5d2d16 | 2013-10-02 13:36:09 +0000 | [diff] [blame] | 553 | CreateCalls(Call::Config(observer.SendTransport()), |
| 554 | Call::Config(observer.ReceiveTransport())); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 555 | |
| 556 | observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver()); |
| 557 | |
| 558 | CreateTestConfigs(); |
| 559 | send_config_.rtp.nack.rtp_history_ms = rtp_history_ms; |
| 560 | receive_config_.rtp.nack.rtp_history_ms = rtp_history_ms; |
pbos@webrtc.org | 28a1166 | 2013-09-19 14:22:12 +0000 | [diff] [blame] | 561 | receive_config_.renderer = &observer; |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 562 | |
| 563 | CreateStreams(); |
| 564 | CreateFrameGenerator(); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 565 | StartSending(); |
| 566 | |
| 567 | // Wait() waits for an event triggered when Pli has been received and frames |
| 568 | // have been rendered afterwards. |
| 569 | EXPECT_EQ(kEventSignaled, observer.Wait()); |
| 570 | |
| 571 | StopSending(); |
| 572 | |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 573 | observer.StopSending(); |
pbos@webrtc.org | 618a0ec | 2013-09-09 08:26:30 +0000 | [diff] [blame] | 574 | |
| 575 | DestroyStreams(); |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 576 | } |
| 577 | |
pbos@webrtc.org | 362e3e5 | 2013-09-27 10:54:10 +0000 | [diff] [blame] | 578 | TEST_F(CallTest, ReceivesPliAndRecoversWithNack) { |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 579 | ReceivesPliAndRecovers(1000); |
| 580 | } |
| 581 | |
| 582 | // TODO(pbos): Enable this when 2250 is resolved. |
pbos@webrtc.org | 362e3e5 | 2013-09-27 10:54:10 +0000 | [diff] [blame] | 583 | TEST_F(CallTest, DISABLED_ReceivesPliAndRecoversWithoutNack) { |
pbos@webrtc.org | 8ce445e | 2013-08-19 16:09:34 +0000 | [diff] [blame] | 584 | ReceivesPliAndRecovers(0); |
| 585 | } |
| 586 | |
pbos@webrtc.org | 362e3e5 | 2013-09-27 10:54:10 +0000 | [diff] [blame] | 587 | TEST_F(CallTest, SurvivesIncomingRtpPacketsToDestroyedReceiveStream) { |
pbos@webrtc.org | 0020858 | 2013-09-05 12:38:54 +0000 | [diff] [blame] | 588 | class PacketInputObserver : public PacketReceiver { |
| 589 | public: |
| 590 | explicit PacketInputObserver(PacketReceiver* receiver) |
| 591 | : receiver_(receiver), delivered_packet_(EventWrapper::Create()) {} |
| 592 | |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 593 | EventTypeWrapper Wait() { |
| 594 | return delivered_packet_->Wait(kDefaultTimeoutMs); |
| 595 | } |
pbos@webrtc.org | 0020858 | 2013-09-05 12:38:54 +0000 | [diff] [blame] | 596 | |
| 597 | private: |
| 598 | virtual bool DeliverPacket(const uint8_t* packet, size_t length) { |
| 599 | if (RtpHeaderParser::IsRtcp(packet, static_cast<int>(length))) { |
| 600 | return receiver_->DeliverPacket(packet, length); |
| 601 | } else { |
| 602 | EXPECT_FALSE(receiver_->DeliverPacket(packet, length)); |
| 603 | delivered_packet_->Set(); |
| 604 | return false; |
| 605 | } |
| 606 | } |
| 607 | |
| 608 | PacketReceiver* receiver_; |
| 609 | scoped_ptr<EventWrapper> delivered_packet_; |
| 610 | }; |
| 611 | |
| 612 | test::DirectTransport send_transport, receive_transport; |
| 613 | |
pbos@webrtc.org | b5d2d16 | 2013-10-02 13:36:09 +0000 | [diff] [blame] | 614 | CreateCalls(Call::Config(&send_transport), Call::Config(&receive_transport)); |
pbos@webrtc.org | 0020858 | 2013-09-05 12:38:54 +0000 | [diff] [blame] | 615 | PacketInputObserver input_observer(receiver_call_->Receiver()); |
| 616 | |
| 617 | send_transport.SetReceiver(&input_observer); |
| 618 | receive_transport.SetReceiver(sender_call_->Receiver()); |
| 619 | |
| 620 | CreateTestConfigs(); |
| 621 | |
| 622 | CreateStreams(); |
| 623 | CreateFrameGenerator(); |
pbos@webrtc.org | 0020858 | 2013-09-05 12:38:54 +0000 | [diff] [blame] | 624 | StartSending(); |
| 625 | |
| 626 | receiver_call_->DestroyReceiveStream(receive_stream_); |
| 627 | receive_stream_ = NULL; |
| 628 | |
| 629 | // Wait() waits for a received packet. |
| 630 | EXPECT_EQ(kEventSignaled, input_observer.Wait()); |
| 631 | |
| 632 | StopSending(); |
| 633 | |
| 634 | DestroyStreams(); |
| 635 | |
| 636 | send_transport.StopSending(); |
| 637 | receive_transport.StopSending(); |
| 638 | } |
pbos@webrtc.org | c5080a9 | 2013-10-01 11:33:24 +0000 | [diff] [blame] | 639 | |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 640 | void CallTest::RespectsRtcpMode(newapi::RtcpMode rtcp_mode) { |
| 641 | static const int kRtpHistoryMs = 1000; |
| 642 | static const int kNumCompoundRtcpPacketsToObserve = 10; |
| 643 | class RtcpModeObserver : public test::RtpRtcpObserver { |
| 644 | public: |
| 645 | RtcpModeObserver(newapi::RtcpMode rtcp_mode) |
| 646 | : test::RtpRtcpObserver(kDefaultTimeoutMs), |
| 647 | rtcp_mode_(rtcp_mode), |
| 648 | sent_rtp_(0), |
| 649 | sent_rtcp_(0) {} |
| 650 | |
| 651 | private: |
| 652 | virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| 653 | if (++sent_rtp_ % 3 == 0) |
| 654 | return DROP_PACKET; |
| 655 | |
| 656 | return SEND_PACKET; |
| 657 | } |
| 658 | |
| 659 | virtual Action OnReceiveRtcp(const uint8_t* packet, |
| 660 | size_t length) OVERRIDE { |
| 661 | ++sent_rtcp_; |
| 662 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 663 | EXPECT_TRUE(parser.IsValid()); |
| 664 | |
| 665 | RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 666 | bool has_report_block = false; |
| 667 | while (packet_type != RTCPUtility::kRtcpNotValidCode) { |
| 668 | EXPECT_NE(RTCPUtility::kRtcpSrCode, packet_type); |
| 669 | if (packet_type == RTCPUtility::kRtcpRrCode) { |
| 670 | has_report_block = true; |
| 671 | break; |
| 672 | } |
| 673 | packet_type = parser.Iterate(); |
| 674 | } |
| 675 | |
| 676 | switch (rtcp_mode_) { |
| 677 | case newapi::kRtcpCompound: |
| 678 | if (!has_report_block) { |
| 679 | ADD_FAILURE() << "Received RTCP packet without receiver report for " |
| 680 | "kRtcpCompound."; |
| 681 | observation_complete_->Set(); |
| 682 | } |
| 683 | |
| 684 | if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve) |
| 685 | observation_complete_->Set(); |
| 686 | |
| 687 | break; |
| 688 | case newapi::kRtcpReducedSize: |
| 689 | if (!has_report_block) |
| 690 | observation_complete_->Set(); |
| 691 | break; |
| 692 | } |
| 693 | |
| 694 | return SEND_PACKET; |
| 695 | } |
| 696 | |
| 697 | newapi::RtcpMode rtcp_mode_; |
| 698 | int sent_rtp_; |
| 699 | int sent_rtcp_; |
| 700 | } observer(rtcp_mode); |
| 701 | |
| 702 | CreateCalls(Call::Config(observer.SendTransport()), |
| 703 | Call::Config(observer.ReceiveTransport())); |
| 704 | |
| 705 | observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver()); |
| 706 | |
| 707 | CreateTestConfigs(); |
| 708 | send_config_.rtp.nack.rtp_history_ms = kRtpHistoryMs; |
| 709 | receive_config_.rtp.nack.rtp_history_ms = kRtpHistoryMs; |
| 710 | receive_config_.rtp.rtcp_mode = rtcp_mode; |
| 711 | |
| 712 | CreateStreams(); |
| 713 | CreateFrameGenerator(); |
| 714 | StartSending(); |
| 715 | |
| 716 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 717 | << (rtcp_mode == newapi::kRtcpCompound |
| 718 | ? "Timed out before observing enough compound packets." |
| 719 | : "Timed out before receiving a non-compound RTCP packet."); |
| 720 | |
| 721 | StopSending(); |
| 722 | observer.StopSending(); |
| 723 | DestroyStreams(); |
| 724 | } |
| 725 | |
| 726 | TEST_F(CallTest, UsesRtcpCompoundMode) { |
| 727 | RespectsRtcpMode(newapi::kRtcpCompound); |
| 728 | } |
| 729 | |
| 730 | TEST_F(CallTest, UsesRtcpReducedSizeMode) { |
| 731 | RespectsRtcpMode(newapi::kRtcpReducedSize); |
| 732 | } |
| 733 | |
pbos@webrtc.org | c5080a9 | 2013-10-01 11:33:24 +0000 | [diff] [blame] | 734 | // Test sets up a Call multiple senders with different resolutions and SSRCs. |
| 735 | // Another is set up to receive all three of these with different renderers. |
| 736 | // Each renderer verifies that it receives the expected resolution, and as soon |
| 737 | // as every renderer has received a frame, the test finishes. |
| 738 | TEST_F(CallTest, SendsAndReceivesMultipleStreams) { |
| 739 | static const size_t kNumStreams = 3; |
| 740 | |
| 741 | class VideoOutputObserver : public VideoRenderer { |
| 742 | public: |
| 743 | VideoOutputObserver(int width, int height) |
| 744 | : width_(width), height_(height), done_(EventWrapper::Create()) {} |
| 745 | |
| 746 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 747 | int time_to_render_ms) OVERRIDE { |
| 748 | EXPECT_EQ(width_, video_frame.width()); |
| 749 | EXPECT_EQ(height_, video_frame.height()); |
| 750 | done_->Set(); |
| 751 | } |
| 752 | |
pbos@webrtc.org | 51e0101 | 2013-10-17 14:14:42 +0000 | [diff] [blame] | 753 | void Wait() { done_->Wait(kDefaultTimeoutMs); } |
pbos@webrtc.org | c5080a9 | 2013-10-01 11:33:24 +0000 | [diff] [blame] | 754 | |
| 755 | private: |
| 756 | int width_; |
| 757 | int height_; |
| 758 | scoped_ptr<EventWrapper> done_; |
| 759 | }; |
| 760 | |
| 761 | struct { |
| 762 | uint32_t ssrc; |
| 763 | int width; |
| 764 | int height; |
| 765 | } codec_settings[kNumStreams] = {{1, 640, 480}, {2, 320, 240}, {3, 240, 160}}; |
| 766 | |
| 767 | test::DirectTransport sender_transport, receiver_transport; |
| 768 | scoped_ptr<Call> sender_call(Call::Create(Call::Config(&sender_transport))); |
| 769 | scoped_ptr<Call> receiver_call( |
| 770 | Call::Create(Call::Config(&receiver_transport))); |
| 771 | sender_transport.SetReceiver(receiver_call->Receiver()); |
| 772 | receiver_transport.SetReceiver(sender_call->Receiver()); |
| 773 | |
| 774 | VideoSendStream* send_streams[kNumStreams]; |
| 775 | VideoReceiveStream* receive_streams[kNumStreams]; |
| 776 | |
| 777 | VideoOutputObserver* observers[kNumStreams]; |
| 778 | test::FrameGeneratorCapturer* frame_generators[kNumStreams]; |
| 779 | |
| 780 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 781 | uint32_t ssrc = codec_settings[i].ssrc; |
| 782 | int width = codec_settings[i].width; |
| 783 | int height = codec_settings[i].height; |
| 784 | observers[i] = new VideoOutputObserver(width, height); |
| 785 | |
| 786 | VideoReceiveStream::Config receive_config = |
| 787 | receiver_call->GetDefaultReceiveConfig(); |
| 788 | receive_config.renderer = observers[i]; |
| 789 | receive_config.rtp.ssrc = ssrc; |
pbos@webrtc.org | 964d78e | 2013-11-20 10:40:25 +0000 | [diff] [blame^] | 790 | receive_streams[i] = |
| 791 | receiver_call->CreateVideoReceiveStream(receive_config); |
pbos@webrtc.org | c5080a9 | 2013-10-01 11:33:24 +0000 | [diff] [blame] | 792 | receive_streams[i]->StartReceive(); |
| 793 | |
| 794 | VideoSendStream::Config send_config = sender_call->GetDefaultSendConfig(); |
| 795 | send_config.rtp.ssrcs.push_back(ssrc); |
| 796 | send_config.codec.width = width; |
| 797 | send_config.codec.height = height; |
pbos@webrtc.org | 964d78e | 2013-11-20 10:40:25 +0000 | [diff] [blame^] | 798 | send_streams[i] = sender_call->CreateVideoSendStream(send_config); |
pbos@webrtc.org | c5080a9 | 2013-10-01 11:33:24 +0000 | [diff] [blame] | 799 | send_streams[i]->StartSend(); |
| 800 | |
| 801 | frame_generators[i] = test::FrameGeneratorCapturer::Create( |
| 802 | send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock()); |
| 803 | frame_generators[i]->Start(); |
| 804 | } |
| 805 | |
| 806 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 807 | observers[i]->Wait(); |
| 808 | } |
| 809 | |
| 810 | for (size_t i = 0; i < kNumStreams; ++i) { |
| 811 | frame_generators[i]->Stop(); |
| 812 | delete frame_generators[i]; |
| 813 | sender_call->DestroySendStream(send_streams[i]); |
| 814 | receiver_call->DestroyReceiveStream(receive_streams[i]); |
| 815 | delete observers[i]; |
| 816 | } |
| 817 | |
| 818 | sender_transport.StopSending(); |
| 819 | receiver_transport.StopSending(); |
| 820 | } |
stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 821 | |
| 822 | class SyncRtcpObserver : public test::RtpRtcpObserver { |
| 823 | public: |
| 824 | SyncRtcpObserver(int delay_ms) |
| 825 | : test::RtpRtcpObserver(kLongTimeoutMs, delay_ms), |
| 826 | critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {} |
| 827 | |
| 828 | virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| 829 | RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| 830 | EXPECT_TRUE(parser.IsValid()); |
| 831 | |
| 832 | for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| 833 | packet_type != RTCPUtility::kRtcpNotValidCode; |
| 834 | packet_type = parser.Iterate()) { |
| 835 | if (packet_type == RTCPUtility::kRtcpSrCode) { |
| 836 | const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| 837 | synchronization::RtcpMeasurement ntp_rtp_pair( |
| 838 | packet.SR.NTPMostSignificant, |
| 839 | packet.SR.NTPLeastSignificant, |
| 840 | packet.SR.RTPTimestamp); |
| 841 | StoreNtpRtpPair(ntp_rtp_pair); |
| 842 | } |
| 843 | } |
| 844 | return SEND_PACKET; |
| 845 | } |
| 846 | |
| 847 | int64_t RtpTimestampToNtp(uint32_t timestamp) const { |
| 848 | CriticalSectionScoped cs(critical_section_.get()); |
| 849 | int64_t timestamp_in_ms = -1; |
| 850 | if (ntp_rtp_pairs_.size() == 2) { |
| 851 | // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the |
| 852 | // RTCP sender where it sends RTCP SR before any RTP packets, which leads |
| 853 | // to a bogus NTP/RTP mapping. |
| 854 | synchronization::RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms); |
| 855 | return timestamp_in_ms; |
| 856 | } |
| 857 | return -1; |
| 858 | } |
| 859 | |
| 860 | private: |
| 861 | void StoreNtpRtpPair(synchronization::RtcpMeasurement ntp_rtp_pair) { |
| 862 | CriticalSectionScoped cs(critical_section_.get()); |
| 863 | for (synchronization::RtcpList::iterator it = ntp_rtp_pairs_.begin(); |
| 864 | it != ntp_rtp_pairs_.end(); |
| 865 | ++it) { |
| 866 | if (ntp_rtp_pair.ntp_secs == it->ntp_secs && |
| 867 | ntp_rtp_pair.ntp_frac == it->ntp_frac) { |
| 868 | // This RTCP has already been added to the list. |
| 869 | return; |
| 870 | } |
| 871 | } |
| 872 | // We need two RTCP SR reports to map between RTP and NTP. More than two |
| 873 | // will not improve the mapping. |
| 874 | if (ntp_rtp_pairs_.size() == 2) { |
| 875 | ntp_rtp_pairs_.pop_back(); |
| 876 | } |
| 877 | ntp_rtp_pairs_.push_front(ntp_rtp_pair); |
| 878 | } |
| 879 | |
| 880 | scoped_ptr<CriticalSectionWrapper> critical_section_; |
| 881 | synchronization::RtcpList ntp_rtp_pairs_; |
| 882 | }; |
| 883 | |
| 884 | class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer { |
| 885 | static const int kInSyncThresholdMs = 50; |
| 886 | static const int kStartupTimeMs = 2000; |
| 887 | static const int kMinRunTimeMs = 30000; |
| 888 | |
| 889 | public: |
| 890 | VideoRtcpAndSyncObserver(Clock* clock, |
| 891 | int voe_channel, |
| 892 | VoEVideoSync* voe_sync, |
| 893 | SyncRtcpObserver* audio_observer) |
| 894 | : SyncRtcpObserver(0), |
| 895 | clock_(clock), |
| 896 | voe_channel_(voe_channel), |
| 897 | voe_sync_(voe_sync), |
| 898 | audio_observer_(audio_observer), |
| 899 | creation_time_ms_(clock_->TimeInMilliseconds()), |
| 900 | first_time_in_sync_(-1) {} |
| 901 | |
| 902 | virtual void RenderFrame(const I420VideoFrame& video_frame, |
| 903 | int time_to_render_ms) OVERRIDE { |
| 904 | int64_t now_ms = clock_->TimeInMilliseconds(); |
| 905 | uint32_t playout_timestamp = 0; |
| 906 | if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0) |
| 907 | return; |
| 908 | int64_t latest_audio_ntp = |
| 909 | audio_observer_->RtpTimestampToNtp(playout_timestamp); |
| 910 | int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp()); |
| 911 | if (latest_audio_ntp < 0 || latest_video_ntp < 0) |
| 912 | return; |
| 913 | int time_until_render_ms = |
| 914 | std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms)); |
| 915 | latest_video_ntp += time_until_render_ms; |
| 916 | int64_t stream_offset = latest_audio_ntp - latest_video_ntp; |
| 917 | std::stringstream ss; |
| 918 | ss << stream_offset; |
| 919 | webrtc::test::PrintResult( |
| 920 | "stream_offset", "", "synchronization", ss.str(), "ms", false); |
| 921 | int64_t time_since_creation = now_ms - creation_time_ms_; |
| 922 | // During the first couple of seconds audio and video can falsely be |
| 923 | // estimated as being synchronized. We don't want to trigger on those. |
| 924 | if (time_since_creation < kStartupTimeMs) |
| 925 | return; |
| 926 | if (abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) { |
| 927 | if (first_time_in_sync_ == -1) { |
| 928 | first_time_in_sync_ = now_ms; |
| 929 | webrtc::test::PrintResult("sync_convergence_time", |
| 930 | "", |
| 931 | "synchronization", |
| 932 | time_since_creation, |
| 933 | "ms", |
| 934 | false); |
| 935 | } |
| 936 | if (time_since_creation > kMinRunTimeMs) |
| 937 | observation_complete_->Set(); |
| 938 | } |
| 939 | } |
| 940 | |
| 941 | private: |
| 942 | Clock* clock_; |
| 943 | int voe_channel_; |
| 944 | VoEVideoSync* voe_sync_; |
| 945 | SyncRtcpObserver* audio_observer_; |
| 946 | int64_t creation_time_ms_; |
| 947 | int64_t first_time_in_sync_; |
| 948 | }; |
| 949 | |
| 950 | TEST_F(CallTest, PlaysOutAudioAndVideoInSync) { |
| 951 | VoiceEngine* voice_engine = VoiceEngine::Create(); |
| 952 | VoEBase* voe_base = VoEBase::GetInterface(voice_engine); |
| 953 | VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); |
| 954 | VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine); |
| 955 | VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine); |
| 956 | ResourceManager resource_manager; |
| 957 | const std::string audio_filename = resource_manager.long_audio_file_path(); |
| 958 | ASSERT_STRNE("", audio_filename.c_str()); |
| 959 | test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), |
| 960 | audio_filename); |
| 961 | EXPECT_EQ(0, voe_base->Init(&fake_audio_device, NULL)); |
| 962 | int channel = voe_base->CreateChannel(); |
| 963 | |
| 964 | const int kVoiceDelayMs = 500; |
| 965 | SyncRtcpObserver audio_observer(kVoiceDelayMs); |
| 966 | VideoRtcpAndSyncObserver observer( |
| 967 | Clock::GetRealTimeClock(), channel, voe_sync, &audio_observer); |
| 968 | |
| 969 | Call::Config receiver_config(observer.ReceiveTransport()); |
| 970 | receiver_config.voice_engine = voice_engine; |
| 971 | CreateCalls(Call::Config(observer.SendTransport()), receiver_config); |
| 972 | CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; |
| 973 | EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac)); |
| 974 | |
| 975 | class VoicePacketReceiver : public PacketReceiver { |
| 976 | public: |
| 977 | VoicePacketReceiver(int channel, VoENetwork* voe_network) |
| 978 | : channel_(channel), |
| 979 | voe_network_(voe_network), |
| 980 | parser_(RtpHeaderParser::Create()) {} |
| 981 | virtual bool DeliverPacket(const uint8_t* packet, size_t length) { |
| 982 | int ret; |
| 983 | if (parser_->IsRtcp(packet, static_cast<int>(length))) { |
| 984 | ret = voe_network_->ReceivedRTCPPacket( |
| 985 | channel_, packet, static_cast<unsigned int>(length)); |
| 986 | } else { |
| 987 | ret = voe_network_->ReceivedRTPPacket( |
| 988 | channel_, packet, static_cast<unsigned int>(length)); |
| 989 | } |
| 990 | return ret == 0; |
| 991 | } |
| 992 | |
| 993 | private: |
| 994 | int channel_; |
| 995 | VoENetwork* voe_network_; |
| 996 | scoped_ptr<RtpHeaderParser> parser_; |
| 997 | } voe_packet_receiver(channel, voe_network); |
| 998 | |
| 999 | audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver); |
| 1000 | |
| 1001 | internal::TransportAdapter transport_adapter(audio_observer.SendTransport()); |
| 1002 | EXPECT_EQ(0, |
| 1003 | voe_network->RegisterExternalTransport(channel, transport_adapter)); |
| 1004 | |
| 1005 | observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver()); |
| 1006 | |
| 1007 | CreateTestConfigs(); |
| 1008 | send_config_.rtp.nack.rtp_history_ms = 1000; |
| 1009 | receive_config_.rtp.nack.rtp_history_ms = 1000; |
| 1010 | receive_config_.renderer = &observer; |
| 1011 | receive_config_.audio_channel_id = channel; |
| 1012 | |
| 1013 | CreateStreams(); |
| 1014 | CreateFrameGenerator(); |
| 1015 | StartSending(); |
| 1016 | |
| 1017 | fake_audio_device.Start(); |
| 1018 | EXPECT_EQ(0, voe_base->StartPlayout(channel)); |
| 1019 | EXPECT_EQ(0, voe_base->StartReceive(channel)); |
| 1020 | EXPECT_EQ(0, voe_base->StartSend(channel)); |
| 1021 | |
| 1022 | EXPECT_EQ(kEventSignaled, observer.Wait()) |
| 1023 | << "Timed out while waiting for audio and video to be synchronized."; |
| 1024 | |
| 1025 | EXPECT_EQ(0, voe_base->StopSend(channel)); |
| 1026 | EXPECT_EQ(0, voe_base->StopReceive(channel)); |
| 1027 | EXPECT_EQ(0, voe_base->StopPlayout(channel)); |
| 1028 | fake_audio_device.Stop(); |
| 1029 | |
| 1030 | StopSending(); |
| 1031 | observer.StopSending(); |
| 1032 | audio_observer.StopSending(); |
| 1033 | |
| 1034 | voe_base->DeleteChannel(channel); |
| 1035 | voe_base->Release(); |
| 1036 | voe_codec->Release(); |
| 1037 | voe_network->Release(); |
| 1038 | voe_sync->Release(); |
| 1039 | DestroyStreams(); |
| 1040 | VoiceEngine::Delete(voice_engine); |
| 1041 | } |
| 1042 | |
pbos@webrtc.org | ce85109 | 2013-08-05 12:01:36 +0000 | [diff] [blame] | 1043 | } // namespace webrtc |