blob: b08c40588222c8003347c657a3038c3f0d3222f0 [file] [log] [blame]
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +00001/*
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 */
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +000010#include <algorithm> // max
11
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +000012#include "testing/gtest/include/gtest/gtest.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000013
14#include "webrtc/call.h"
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +000015#include "webrtc/common_video/interface/i420_video_frame.h"
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +000016#include "webrtc/common_video/interface/native_handle.h"
17#include "webrtc/common_video/interface/texture_video_frame.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000018#include "webrtc/frame_callback.h"
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +000019#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
pbos@webrtc.org709e2972014-03-19 10:59:52 +000020#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +000021#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
pbos@webrtc.org013d9942013-08-22 09:42:17 +000022#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +000023#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +000024#include "webrtc/system_wrappers/interface/event_wrapper.h"
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +000025#include "webrtc/system_wrappers/interface/ref_count.h"
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +000026#include "webrtc/system_wrappers/interface/scoped_ptr.h"
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +000027#include "webrtc/system_wrappers/interface/scoped_vector.h"
pbos@webrtc.org29023282013-09-11 10:14:56 +000028#include "webrtc/system_wrappers/interface/sleep.h"
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +000029#include "webrtc/system_wrappers/interface/thread_wrapper.h"
stefan@webrtc.org168f23f2014-07-11 13:44:02 +000030#include "webrtc/system_wrappers/interface/logging.h"
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000031#include "webrtc/test/call_test.h"
pbos@webrtc.org709e2972014-03-19 10:59:52 +000032#include "webrtc/test/configurable_frame_size_encoder.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000033#include "webrtc/test/null_transport.h"
pbos@webrtc.org709e2972014-03-19 10:59:52 +000034#include "webrtc/test/testsupport/perf_test.h"
pbos@webrtc.org16e03b72013-10-28 16:32:01 +000035#include "webrtc/video/transport_adapter.h"
36#include "webrtc/video_send_stream.h"
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +000037
38namespace webrtc {
39
sprang@webrtc.org346094c2014-02-18 08:40:33 +000040enum VideoFormat { kGeneric, kVP8, };
41
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +000042void ExpectEqualFrames(const I420VideoFrame& frame1,
43 const I420VideoFrame& frame2);
44void ExpectEqualTextureFrames(const I420VideoFrame& frame1,
45 const I420VideoFrame& frame2);
46void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
47 const I420VideoFrame& frame2);
48void ExpectEqualFramesVector(const std::vector<I420VideoFrame*>& frames1,
49 const std::vector<I420VideoFrame*>& frames2);
50I420VideoFrame* CreateI420VideoFrame(int width, int height, uint8_t data);
51
52class FakeNativeHandle : public NativeHandle {
53 public:
54 FakeNativeHandle() {}
55 virtual ~FakeNativeHandle() {}
56 virtual void* GetHandle() { return NULL; }
57};
58
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000059class VideoSendStreamTest : public test::CallTest {
pbos@webrtc.org013d9942013-08-22 09:42:17 +000060 protected:
stefan@webrtc.org69969e22013-11-15 12:32:15 +000061 void TestNackRetransmission(uint32_t retransmit_ssrc,
stefan@webrtc.orgcb254aa2014-06-12 15:12:25 +000062 uint8_t retransmit_payload_type);
sprang@webrtc.org346094c2014-02-18 08:40:33 +000063 void TestPacketFragmentationSize(VideoFormat format, bool with_fec);
pbos@webrtc.org013d9942013-08-22 09:42:17 +000064};
65
pbos@webrtc.orgf777cf22014-01-10 18:47:32 +000066TEST_F(VideoSendStreamTest, CanStartStartedStream) {
67 test::NullTransport transport;
68 Call::Config call_config(&transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000069 CreateSenderCall(call_config);
pbos@webrtc.orgf777cf22014-01-10 18:47:32 +000070
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000071 CreateSendConfig(1);
72 CreateStreams();
73 send_stream_->Start();
74 send_stream_->Start();
75 DestroyStreams();
pbos@webrtc.orgf777cf22014-01-10 18:47:32 +000076}
77
78TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
79 test::NullTransport transport;
80 Call::Config call_config(&transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000081 CreateSenderCall(call_config);
pbos@webrtc.orgf777cf22014-01-10 18:47:32 +000082
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000083 CreateSendConfig(1);
84 CreateStreams();
85 send_stream_->Stop();
86 send_stream_->Stop();
87 DestroyStreams();
pbos@webrtc.orgf777cf22014-01-10 18:47:32 +000088}
89
pbos@webrtc.org013d9942013-08-22 09:42:17 +000090TEST_F(VideoSendStreamTest, SupportsCName) {
91 static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000092 class CNameObserver : public test::SendTest {
pbos@webrtc.org013d9942013-08-22 09:42:17 +000093 public:
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000094 CNameObserver() : SendTest(kDefaultTimeoutMs) {}
pbos@webrtc.org013d9942013-08-22 09:42:17 +000095
pbos@webrtc.org994d0b72014-06-27 08:47:52 +000096 private:
stefan@webrtc.org69969e22013-11-15 12:32:15 +000097 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.org013d9942013-08-22 09:42:17 +000098 RTCPUtility::RTCPParserV2 parser(packet, length, true);
99 EXPECT_TRUE(parser.IsValid());
100
101 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
102 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
103 if (packet_type == RTCPUtility::kRtcpSdesChunkCode) {
104 EXPECT_EQ(parser.Packet().CName.CName, kCName);
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000105 observation_complete_->Set();
pbos@webrtc.org013d9942013-08-22 09:42:17 +0000106 }
107
108 packet_type = parser.Iterate();
109 }
110
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000111 return SEND_PACKET;
pbos@webrtc.org013d9942013-08-22 09:42:17 +0000112 }
pbos@webrtc.org013d9942013-08-22 09:42:17 +0000113
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000114 virtual void ModifyConfigs(
115 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000116 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000117 std::vector<VideoStream>* video_streams) OVERRIDE {
118 send_config->rtp.c_name = kCName;
119 }
pbos@webrtc.org013d9942013-08-22 09:42:17 +0000120
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000121 virtual void PerformTest() OVERRIDE {
122 EXPECT_EQ(kEventSignaled, Wait())
123 << "Timed out while waiting for RTCP with CNAME.";
124 }
125 } test;
pbos@webrtc.org013d9942013-08-22 09:42:17 +0000126
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000127 RunBaseTest(&test);
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +0000128}
129
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000130TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
131 static const uint8_t kAbsSendTimeExtensionId = 13;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000132 class AbsoluteSendTimeObserver : public test::SendTest {
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000133 public:
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000134 AbsoluteSendTimeObserver() : SendTest(kDefaultTimeoutMs) {
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000135 EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000136 kRtpExtensionAbsoluteSendTime, kAbsSendTimeExtensionId));
137 }
138
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000139 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000140 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000141 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000142
pbos@webrtc.org5ab75672013-12-16 12:24:44 +0000143 EXPECT_FALSE(header.extension.hasTransmissionTimeOffset);
144 EXPECT_TRUE(header.extension.hasAbsoluteSendTime);
145 EXPECT_EQ(header.extension.transmissionTimeOffset, 0);
146 EXPECT_GT(header.extension.absoluteSendTime, 0u);
147 observation_complete_->Set();
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000148
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000149 return SEND_PACKET;
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000150 }
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000151
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000152 virtual void ModifyConfigs(
153 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000154 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000155 std::vector<VideoStream>* video_streams) OVERRIDE {
156 send_config->rtp.extensions.push_back(
157 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
158 }
pbos@webrtc.orgdde16f12014-08-05 23:35:43 +0000159
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000160 virtual void PerformTest() OVERRIDE {
161 EXPECT_EQ(kEventSignaled, Wait())
162 << "Timed out while waiting for single RTP packet.";
163 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000164 } test;
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000165
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000166 RunBaseTest(&test);
pbos@webrtc.org5c678ea2013-09-11 19:00:39 +0000167}
168
pbos@webrtc.org29023282013-09-11 10:14:56 +0000169TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
170 static const uint8_t kTOffsetExtensionId = 13;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000171 class TransmissionTimeOffsetObserver : public test::SendTest {
pbos@webrtc.org29023282013-09-11 10:14:56 +0000172 public:
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000173 TransmissionTimeOffsetObserver()
174 : SendTest(kDefaultTimeoutMs), encoder_(Clock::GetRealTimeClock()) {
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000175 EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
pbos@webrtc.org29023282013-09-11 10:14:56 +0000176 kRtpExtensionTransmissionTimeOffset, kTOffsetExtensionId));
177 }
178
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000179 private:
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000180 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.org29023282013-09-11 10:14:56 +0000181 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000182 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org29023282013-09-11 10:14:56 +0000183
pbos@webrtc.org5ab75672013-12-16 12:24:44 +0000184 EXPECT_TRUE(header.extension.hasTransmissionTimeOffset);
185 EXPECT_FALSE(header.extension.hasAbsoluteSendTime);
pbos@webrtc.org29023282013-09-11 10:14:56 +0000186 EXPECT_GT(header.extension.transmissionTimeOffset, 0);
pbos@webrtc.org5ab75672013-12-16 12:24:44 +0000187 EXPECT_EQ(header.extension.absoluteSendTime, 0u);
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000188 observation_complete_->Set();
pbos@webrtc.org29023282013-09-11 10:14:56 +0000189
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000190 return SEND_PACKET;
pbos@webrtc.org29023282013-09-11 10:14:56 +0000191 }
pbos@webrtc.org29023282013-09-11 10:14:56 +0000192
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000193 virtual void ModifyConfigs(
194 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000195 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000196 std::vector<VideoStream>* video_streams) OVERRIDE {
197 send_config->encoder_settings.encoder = &encoder_;
198 send_config->rtp.extensions.push_back(
199 RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
200 }
pbos@webrtc.org29023282013-09-11 10:14:56 +0000201
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000202 virtual void PerformTest() OVERRIDE {
203 EXPECT_EQ(kEventSignaled, Wait())
204 << "Timed out while waiting single RTP packet.";
205 }
pbos@webrtc.org29023282013-09-11 10:14:56 +0000206
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000207 class DelayedEncoder : public test::FakeEncoder {
208 public:
209 explicit DelayedEncoder(Clock* clock) : test::FakeEncoder(clock) {}
210 virtual int32_t Encode(
211 const I420VideoFrame& input_image,
212 const CodecSpecificInfo* codec_specific_info,
213 const std::vector<VideoFrameType>* frame_types) OVERRIDE {
214 // A delay needs to be introduced to assure that we get a timestamp
215 // offset.
216 SleepMs(5);
217 return FakeEncoder::Encode(
218 input_image, codec_specific_info, frame_types);
219 }
220 };
221
222 DelayedEncoder encoder_;
223 } test;
224
225 RunBaseTest(&test);
pbos@webrtc.org29023282013-09-11 10:14:56 +0000226}
227
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000228class FakeReceiveStatistics : public NullReceiveStatistics {
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000229 public:
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000230 FakeReceiveStatistics(uint32_t send_ssrc,
231 uint32_t last_sequence_number,
232 uint32_t cumulative_lost,
233 uint8_t fraction_lost)
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000234 : lossy_stats_(new LossyStatistician(last_sequence_number,
235 cumulative_lost,
236 fraction_lost)) {
237 stats_map_[send_ssrc] = lossy_stats_.get();
238 }
239
240 virtual StatisticianMap GetActiveStatisticians() const OVERRIDE {
241 return stats_map_;
242 }
243
244 virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE {
245 return lossy_stats_.get();
246 }
247
248 private:
249 class LossyStatistician : public StreamStatistician {
250 public:
251 LossyStatistician(uint32_t extended_max_sequence_number,
252 uint32_t cumulative_lost,
253 uint8_t fraction_lost) {
254 stats_.fraction_lost = fraction_lost;
255 stats_.cumulative_lost = cumulative_lost;
256 stats_.extended_max_sequence_number = extended_max_sequence_number;
257 }
sprang@webrtc.org54ae4ff2013-12-19 13:26:02 +0000258 virtual bool GetStatistics(RtcpStatistics* statistics,
259 bool reset) OVERRIDE {
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000260 *statistics = stats_;
261 return true;
262 }
263 virtual void GetDataCounters(uint32_t* bytes_received,
264 uint32_t* packets_received) const OVERRIDE {
265 *bytes_received = 0;
266 *packets_received = 0;
267 }
268 virtual uint32_t BitrateReceived() const OVERRIDE { return 0; }
269 virtual void ResetStatistics() OVERRIDE {}
270 virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
271 int min_rtt) const OVERRIDE {
272 return false;
273 }
274
275 virtual bool IsPacketInOrder(uint16_t sequence_number) const OVERRIDE {
276 return true;
277 }
sprang@webrtc.org54ae4ff2013-12-19 13:26:02 +0000278
279 RtcpStatistics stats_;
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000280 };
281
282 scoped_ptr<LossyStatistician> lossy_stats_;
283 StatisticianMap stats_map_;
284};
285
pbos@webrtc.org724947b2013-12-11 16:26:16 +0000286TEST_F(VideoSendStreamTest, SwapsI420VideoFrames) {
287 static const size_t kWidth = 320;
288 static const size_t kHeight = 240;
289
290 test::NullTransport transport;
291 Call::Config call_config(&transport);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000292 CreateSenderCall(call_config);
pbos@webrtc.org724947b2013-12-11 16:26:16 +0000293
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000294 CreateSendConfig(1);
295 CreateStreams();
296 send_stream_->Start();
pbos@webrtc.org724947b2013-12-11 16:26:16 +0000297
298 I420VideoFrame frame;
299 frame.CreateEmptyFrame(
300 kWidth, kHeight, kWidth, (kWidth + 1) / 2, (kWidth + 1) / 2);
301 uint8_t* old_y_buffer = frame.buffer(kYPlane);
302
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000303 send_stream_->Input()->SwapFrame(&frame);
pbos@webrtc.org724947b2013-12-11 16:26:16 +0000304
305 EXPECT_NE(frame.buffer(kYPlane), old_y_buffer);
306
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000307 DestroyStreams();
pbos@webrtc.org724947b2013-12-11 16:26:16 +0000308}
309
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000310TEST_F(VideoSendStreamTest, SupportsFec) {
311 static const int kRedPayloadType = 118;
312 static const int kUlpfecPayloadType = 119;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000313 class FecObserver : public test::SendTest {
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000314 public:
315 FecObserver()
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000316 : SendTest(kDefaultTimeoutMs),
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000317 transport_adapter_(SendTransport()),
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000318 send_count_(0),
319 received_media_(false),
sprang@webrtc.orgd9b95602014-01-27 13:03:02 +0000320 received_fec_(false) {
321 transport_adapter_.Enable();
322 }
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000323
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000324 private:
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000325 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000326 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000327 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000328
329 // Send lossy receive reports to trigger FEC enabling.
330 if (send_count_++ % 2 != 0) {
331 // Receive statistics reporting having lost 50% of the packets.
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000332 FakeReceiveStatistics lossy_receive_stats(
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000333 kSendSsrcs[0], header.sequenceNumber, send_count_ / 2, 127);
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000334 RTCPSender rtcp_sender(
335 0, false, Clock::GetRealTimeClock(), &lossy_receive_stats);
336 EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
337
338 rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000339 rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000340
341 RTCPSender::FeedbackState feedback_state;
342
343 EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
344 }
345
346 EXPECT_EQ(kRedPayloadType, header.payloadType);
347
348 uint8_t encapsulated_payload_type = packet[header.headerLength];
349
350 if (encapsulated_payload_type == kUlpfecPayloadType) {
351 received_fec_ = true;
352 } else {
353 received_media_ = true;
354 }
355
356 if (received_media_ && received_fec_)
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000357 observation_complete_->Set();
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000358
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000359 return SEND_PACKET;
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000360 }
361
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000362 virtual void ModifyConfigs(
363 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000364 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000365 std::vector<VideoStream>* video_streams) OVERRIDE {
366 send_config->rtp.fec.red_payload_type = kRedPayloadType;
367 send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
368 }
369
370 virtual void PerformTest() OVERRIDE {
371 EXPECT_TRUE(Wait()) << "Timed out waiting for FEC and media packets.";
372 }
373
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000374 internal::TransportAdapter transport_adapter_;
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000375 int send_count_;
376 bool received_media_;
377 bool received_fec_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000378 } test;
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000379
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000380 RunBaseTest(&test);
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000381}
382
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000383void VideoSendStreamTest::TestNackRetransmission(
384 uint32_t retransmit_ssrc,
stefan@webrtc.orgcb254aa2014-06-12 15:12:25 +0000385 uint8_t retransmit_payload_type) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000386 class NackObserver : public test::SendTest {
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000387 public:
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000388 explicit NackObserver(uint32_t retransmit_ssrc,
389 uint8_t retransmit_payload_type)
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000390 : SendTest(kDefaultTimeoutMs),
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000391 transport_adapter_(SendTransport()),
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000392 send_count_(0),
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000393 retransmit_ssrc_(retransmit_ssrc),
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000394 retransmit_payload_type_(retransmit_payload_type),
sprang@webrtc.orgd9b95602014-01-27 13:03:02 +0000395 nacked_sequence_number_(-1) {
396 transport_adapter_.Enable();
397 }
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000398
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000399 private:
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000400 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000401 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000402 EXPECT_TRUE(parser_->Parse(packet, length, &header));
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000403
404 // Nack second packet after receiving the third one.
405 if (++send_count_ == 3) {
pbos@webrtc.orge7223e72014-01-23 16:14:34 +0000406 uint16_t nack_sequence_number = header.sequenceNumber - 1;
407 nacked_sequence_number_ = nack_sequence_number;
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000408 NullReceiveStatistics null_stats;
409 RTCPSender rtcp_sender(
410 0, false, Clock::GetRealTimeClock(), &null_stats);
411 EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
412
413 rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000414 rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000415
416 RTCPSender::FeedbackState feedback_state;
417
418 EXPECT_EQ(0,
419 rtcp_sender.SendRTCP(
pbos@webrtc.orge7223e72014-01-23 16:14:34 +0000420 feedback_state, kRtcpNack, 1, &nack_sequence_number));
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000421 }
422
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000423 uint16_t sequence_number = header.sequenceNumber;
424
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000425 if (header.ssrc == retransmit_ssrc_ &&
426 retransmit_ssrc_ != kSendSsrcs[0]) {
427 // Not kSendSsrcs[0], assume correct RTX packet. Extract sequence
428 // number.
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000429 const uint8_t* rtx_header = packet + header.headerLength;
430 sequence_number = (rtx_header[0] << 8) + rtx_header[1];
431 }
432
433 if (sequence_number == nacked_sequence_number_) {
434 EXPECT_EQ(retransmit_ssrc_, header.ssrc);
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000435 EXPECT_EQ(retransmit_payload_type_, header.payloadType);
436 observation_complete_->Set();
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000437 }
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000438
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000439 return SEND_PACKET;
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000440 }
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000441
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000442 virtual void ModifyConfigs(
443 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000444 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000445 std::vector<VideoStream>* video_streams) OVERRIDE {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000446 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000447 send_config->rtp.rtx.payload_type = retransmit_payload_type_;
448 if (retransmit_ssrc_ != kSendSsrcs[0])
449 send_config->rtp.rtx.ssrcs.push_back(retransmit_ssrc_);
450 }
451
452 virtual void PerformTest() OVERRIDE {
453 EXPECT_EQ(kEventSignaled, Wait())
454 << "Timed out while waiting for NACK retransmission.";
455 }
456
pbos@webrtc.org0e63e762013-09-20 11:56:26 +0000457 internal::TransportAdapter transport_adapter_;
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000458 int send_count_;
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000459 uint32_t retransmit_ssrc_;
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000460 uint8_t retransmit_payload_type_;
pbos@webrtc.orge7223e72014-01-23 16:14:34 +0000461 int nacked_sequence_number_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000462 } test(retransmit_ssrc, retransmit_payload_type);
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000463
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000464 RunBaseTest(&test);
pbos@webrtc.orgdf531a22013-09-10 14:56:33 +0000465}
466
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000467TEST_F(VideoSendStreamTest, RetransmitsNack) {
468 // Normal NACKs should use the send SSRC.
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000469 TestNackRetransmission(kSendSsrcs[0], kFakeSendPayloadType);
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000470}
471
472TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
473 // NACKs over RTX should use a separate SSRC.
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +0000474 TestNackRetransmission(kSendRtxSsrcs[0], kSendRtxPayloadType);
pbos@webrtc.org5860de02013-09-16 13:01:47 +0000475}
476
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000477void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
478 bool with_fec) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000479 // Use a fake encoder to output a frame of every size in the range [90, 290],
480 // for each size making sure that the exact number of payload bytes received
481 // is correct and that packets are fragmented to respect max packet size.
482 static const uint32_t kMaxPacketSize = 128;
483 static const uint32_t start = 90;
484 static const uint32_t stop = 290;
485
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000486 static const int kRedPayloadType = 118;
487 static const int kUlpfecPayloadType = 119;
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000488 // Observer that verifies that the expected number of packets and bytes
489 // arrive for each frame size, from start_size to stop_size.
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000490 class FrameFragmentationTest : public test::SendTest,
491 public EncodedFrameObserver {
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000492 public:
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000493 FrameFragmentationTest(uint32_t max_packet_size,
494 uint32_t start_size,
495 uint32_t stop_size,
496 bool test_generic_packetization,
497 bool use_fec)
498 : SendTest(kLongTimeoutMs),
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000499 transport_adapter_(SendTransport()),
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000500 encoder_(stop),
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000501 max_packet_size_(max_packet_size),
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000502 stop_size_(stop_size),
503 test_generic_packetization_(test_generic_packetization),
504 use_fec_(use_fec),
505 packet_count_(0),
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000506 accumulated_size_(0),
507 accumulated_payload_(0),
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000508 fec_packet_received_(false),
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000509 current_size_rtp_(start_size),
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000510 current_size_frame_(start_size) {
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000511 // Fragmentation required, this test doesn't make sense without it.
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000512 encoder_.SetFrameSize(start);
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000513 assert(stop_size > max_packet_size);
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000514 transport_adapter_.Enable();
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000515 }
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000516
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000517 private:
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000518 virtual Action OnSendRtp(const uint8_t* packet, size_t size) OVERRIDE {
519 uint32_t length = static_cast<int>(size);
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000520 RTPHeader header;
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000521 EXPECT_TRUE(parser_->Parse(packet, length, &header));
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000522
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000523 EXPECT_LE(length, max_packet_size_);
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000524
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000525 if (use_fec_) {
526 uint8_t payload_type = packet[header.headerLength];
527 bool is_fec = header.payloadType == kRedPayloadType &&
528 payload_type == kUlpfecPayloadType;
529 if (is_fec) {
530 fec_packet_received_ = true;
531 return SEND_PACKET;
532 }
533 }
534
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000535 accumulated_size_ += length;
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000536
537 if (use_fec_)
538 TriggerLossReport(header);
539
540 if (test_generic_packetization_) {
541 uint32_t overhead = header.headerLength + header.paddingLength +
542 (1 /* Generic header */);
543 if (use_fec_)
544 overhead += 1; // RED for FEC header.
545 accumulated_payload_ += length - overhead;
546 }
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000547
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000548 // Marker bit set indicates last packet of a frame.
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000549 if (header.markerBit) {
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000550 if (use_fec_ && accumulated_payload_ == current_size_rtp_ - 1) {
551 // With FEC enabled, frame size is incremented asynchronously, so
552 // "old" frames one byte too small may arrive. Accept, but don't
553 // increase expected frame size.
554 accumulated_size_ = 0;
555 accumulated_payload_ = 0;
556 return SEND_PACKET;
557 }
558
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000559 EXPECT_GE(accumulated_size_, current_size_rtp_);
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000560 if (test_generic_packetization_) {
561 EXPECT_EQ(current_size_rtp_, accumulated_payload_);
562 }
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000563
564 // Last packet of frame; reset counters.
565 accumulated_size_ = 0;
566 accumulated_payload_ = 0;
567 if (current_size_rtp_ == stop_size_) {
568 // Done! (Don't increase size again, might arrive more @ stop_size).
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000569 observation_complete_->Set();
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000570 } else {
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000571 // Increase next expected frame size. If testing with FEC, make sure
572 // a FEC packet has been received for this frame size before
573 // proceeding, to make sure that redundancy packets don't exceed
574 // size limit.
575 if (!use_fec_) {
576 ++current_size_rtp_;
577 } else if (fec_packet_received_) {
578 fec_packet_received_ = false;
579 ++current_size_rtp_;
580 ++current_size_frame_;
581 }
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000582 }
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000583 }
584
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000585 return SEND_PACKET;
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000586 }
587
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000588 void TriggerLossReport(const RTPHeader& header) {
589 // Send lossy receive reports to trigger FEC enabling.
590 if (packet_count_++ % 2 != 0) {
591 // Receive statistics reporting having lost 50% of the packets.
592 FakeReceiveStatistics lossy_receive_stats(
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000593 kSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000594 RTCPSender rtcp_sender(
595 0, false, Clock::GetRealTimeClock(), &lossy_receive_stats);
596 EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
597
598 rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000599 rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000600
601 RTCPSender::FeedbackState feedback_state;
602
603 EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
604 }
605 }
606
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000607 virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
608 // Increase frame size for next encoded frame, in the context of the
609 // encoder thread.
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000610 if (!use_fec_ &&
611 current_size_frame_.Value() < static_cast<int32_t>(stop_size_)) {
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000612 ++current_size_frame_;
613 }
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000614 encoder_.SetFrameSize(current_size_frame_.Value());
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000615 }
616
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000617 virtual void ModifyConfigs(
618 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000619 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000620 std::vector<VideoStream>* video_streams) OVERRIDE {
621 if (use_fec_) {
622 send_config->rtp.fec.red_payload_type = kRedPayloadType;
623 send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
624 }
625
626 if (!test_generic_packetization_)
627 send_config->encoder_settings.payload_name = "VP8";
628
629 send_config->encoder_settings.encoder = &encoder_;
630 send_config->rtp.max_packet_size = kMaxPacketSize;
631 send_config->post_encode_callback = this;
632
633 // Add an extension header, to make the RTP header larger than the base
634 // length of 12 bytes.
635 static const uint8_t kAbsSendTimeExtensionId = 13;
636 send_config->rtp.extensions.push_back(
637 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
638 }
639
640 virtual void PerformTest() OVERRIDE {
641 EXPECT_EQ(kEventSignaled, Wait())
642 << "Timed out while observing incoming RTP packets.";
643 }
644
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000645 internal::TransportAdapter transport_adapter_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000646 test::ConfigurableFrameSizeEncoder encoder_;
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000647
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000648 const uint32_t max_packet_size_;
649 const uint32_t stop_size_;
650 const bool test_generic_packetization_;
651 const bool use_fec_;
652
653 uint32_t packet_count_;
654 uint32_t accumulated_size_;
655 uint32_t accumulated_payload_;
656 bool fec_packet_received_;
657
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000658 uint32_t current_size_rtp_;
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000659 Atomic32 current_size_frame_;
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000660 };
661
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000662 // Don't auto increment if FEC is used; continue sending frame size until
663 // a FEC packet has been received.
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000664 FrameFragmentationTest test(
665 kMaxPacketSize, start, stop, format == kGeneric, with_fec);
sprang@webrtc.org8b881922013-12-10 10:05:17 +0000666
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000667 RunBaseTest(&test);
sprang@webrtc.org5d957e22013-10-16 11:37:54 +0000668}
669
sprang@webrtc.org346094c2014-02-18 08:40:33 +0000670// TODO(sprang): Is there any way of speeding up these tests?
671TEST_F(VideoSendStreamTest, FragmentsGenericAccordingToMaxPacketSize) {
672 TestPacketFragmentationSize(kGeneric, false);
673}
674
675TEST_F(VideoSendStreamTest, FragmentsGenericAccordingToMaxPacketSizeWithFec) {
676 TestPacketFragmentationSize(kGeneric, true);
677}
678
679TEST_F(VideoSendStreamTest, FragmentsVp8AccordingToMaxPacketSize) {
680 TestPacketFragmentationSize(kVP8, false);
681}
682
683TEST_F(VideoSendStreamTest, FragmentsVp8AccordingToMaxPacketSizeWithFec) {
684 TestPacketFragmentationSize(kVP8, true);
685}
686
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000687// The test will go through a number of phases.
688// 1. Start sending packets.
689// 2. As soon as the RTP stream has been detected, signal a low REMB value to
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000690// suspend the stream.
691// 3. Wait until |kSuspendTimeFrames| have been captured without seeing any RTP
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000692// packets.
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000693// 4. Signal a high REMB and then wait for the RTP stream to start again.
henrik.lundin@webrtc.orged8b2812014-03-18 08:43:29 +0000694// When the stream is detected again, and the stats show that the stream
695// is no longer suspended, the test ends.
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000696TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
697 static const int kSuspendTimeFrames = 60; // Suspend for 2 seconds @ 30 fps.
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000698
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000699 class RembObserver : public test::SendTest, public I420FrameCallback {
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000700 public:
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000701 RembObserver()
702 : SendTest(kDefaultTimeoutMs),
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000703 transport_adapter_(&transport_),
704 clock_(Clock::GetRealTimeClock()),
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000705 crit_(CriticalSectionWrapper::CreateCriticalSection()),
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000706 test_state_(kBeforeSuspend),
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000707 rtp_count_(0),
708 last_sequence_number_(0),
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000709 suspended_frame_count_(0),
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +0000710 low_remb_bps_(0),
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000711 high_remb_bps_(0) {
sprang@webrtc.orgd9b95602014-01-27 13:03:02 +0000712 transport_adapter_.Enable();
713 }
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000714
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000715 private:
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000716 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000717 // Receive statistics reporting having lost 0% of the packets.
718 // This is needed for the send-side bitrate controller to work properly.
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000719 CriticalSectionScoped lock(crit_.get());
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000720 SendRtcpFeedback(0); // REMB is only sent if value is > 0.
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000721 return SEND_PACKET;
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000722 }
723
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000724 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000725 CriticalSectionScoped lock(crit_.get());
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000726 ++rtp_count_;
727 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +0000728 EXPECT_TRUE(parser_->Parse(packet, length, &header));
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000729 last_sequence_number_ = header.sequenceNumber;
730
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000731 if (test_state_ == kBeforeSuspend) {
732 // The stream has started. Try to suspend it.
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +0000733 SendRtcpFeedback(low_remb_bps_);
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000734 test_state_ = kDuringSuspend;
735 } else if (test_state_ == kDuringSuspend) {
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000736 if (header.paddingLength == 0) {
737 // Received non-padding packet during suspension period. Reset the
738 // counter.
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000739 suspended_frame_count_ = 0;
740 }
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000741 } else if (test_state_ == kWaitingForPacket) {
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000742 if (header.paddingLength == 0) {
henrik.lundin@webrtc.orged8b2812014-03-18 08:43:29 +0000743 // Non-padding packet observed. Test is almost complete. Will just
744 // have to wait for the stats to change.
745 test_state_ = kWaitingForStats;
746 }
747 } else if (test_state_ == kWaitingForStats) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000748 VideoSendStream::Stats stats = stream_->GetStats();
henrik.lundin@webrtc.orged8b2812014-03-18 08:43:29 +0000749 if (stats.suspended == false) {
750 // Stats flipped to false. Test is complete.
henrik.lundin@webrtc.org331d4402013-11-21 14:05:40 +0000751 observation_complete_->Set();
752 }
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000753 }
754
stefan@webrtc.org69969e22013-11-15 12:32:15 +0000755 return SEND_PACKET;
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000756 }
757
758 // This method implements the I420FrameCallback.
759 void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000760 CriticalSectionScoped lock(crit_.get());
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000761 if (test_state_ == kDuringSuspend &&
762 ++suspended_frame_count_ > kSuspendTimeFrames) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000763 VideoSendStream::Stats stats = stream_->GetStats();
henrik.lundin@webrtc.orgb10363f2014-03-13 13:31:21 +0000764 EXPECT_TRUE(stats.suspended);
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +0000765 SendRtcpFeedback(high_remb_bps_);
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000766 test_state_ = kWaitingForPacket;
767 }
768 }
769
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000770 void set_low_remb_bps(int value) {
771 CriticalSectionScoped lock(crit_.get());
772 low_remb_bps_ = value;
773 }
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +0000774
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000775 void set_high_remb_bps(int value) {
776 CriticalSectionScoped lock(crit_.get());
777 high_remb_bps_ = value;
778 }
henrik.lundin@webrtc.org1a3a6e52013-10-28 10:16:14 +0000779
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000780 virtual void SetReceivers(
781 PacketReceiver* send_transport_receiver,
782 PacketReceiver* receive_transport_receiver) OVERRIDE {
783 transport_.SetReceiver(send_transport_receiver);
784 }
pbos@webrtc.orgdef22b42013-10-29 10:12:10 +0000785
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000786 virtual void OnStreamsCreated(
787 VideoSendStream* send_stream,
788 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000789 stream_ = send_stream;
790 }
791
792 virtual void ModifyConfigs(
793 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000794 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000795 std::vector<VideoStream>* video_streams) OVERRIDE {
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000796 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000797 send_config->pre_encode_callback = this;
798 send_config->suspend_below_min_bitrate = true;
799 int min_bitrate_bps = (*video_streams)[0].min_bitrate_bps;
800 set_low_remb_bps(min_bitrate_bps - 10000);
801 int threshold_window = std::max(min_bitrate_bps / 10, 10000);
802 ASSERT_GT((*video_streams)[0].max_bitrate_bps,
803 min_bitrate_bps + threshold_window + 5000);
804 set_high_remb_bps(min_bitrate_bps + threshold_window + 5000);
805 }
806
807 virtual void PerformTest() OVERRIDE {
808 EXPECT_EQ(kEventSignaled, Wait())
809 << "Timed out during suspend-below-min-bitrate test.";
810 transport_.StopSending();
811 }
812
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000813 enum TestState {
henrik.lundin@webrtc.orgce8e0932013-11-18 12:18:43 +0000814 kBeforeSuspend,
815 kDuringSuspend,
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000816 kWaitingForPacket,
henrik.lundin@webrtc.orged8b2812014-03-18 08:43:29 +0000817 kWaitingForStats
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000818 };
819
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000820 virtual void SendRtcpFeedback(int remb_value)
821 EXCLUSIVE_LOCKS_REQUIRED(crit_) {
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000822 FakeReceiveStatistics receive_stats(
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000823 kSendSsrcs[0], last_sequence_number_, rtp_count_, 0);
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000824 RTCPSender rtcp_sender(0, false, clock_, &receive_stats);
825 EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
826
827 rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000828 rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000829 if (remb_value > 0) {
830 rtcp_sender.SetREMBStatus(true);
831 rtcp_sender.SetREMBData(remb_value, 0, NULL);
832 }
833 RTCPSender::FeedbackState feedback_state;
834 EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
835 }
836
837 internal::TransportAdapter transport_adapter_;
838 test::DirectTransport transport_;
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000839 Clock* const clock_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000840 VideoSendStream* stream_;
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000841
842 const scoped_ptr<CriticalSectionWrapper> crit_;
843 TestState test_state_ GUARDED_BY(crit_);
844 int rtp_count_ GUARDED_BY(crit_);
845 int last_sequence_number_ GUARDED_BY(crit_);
846 int suspended_frame_count_ GUARDED_BY(crit_);
847 int low_remb_bps_ GUARDED_BY(crit_);
848 int high_remb_bps_ GUARDED_BY(crit_);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000849 } test;
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000850
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000851 RunBaseTest(&test);
henrik.lundin@webrtc.orgba975e22013-10-23 11:04:57 +0000852}
853
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000854TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000855 class NoPaddingWhenVideoIsMuted : public test::SendTest {
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000856 public:
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000857 NoPaddingWhenVideoIsMuted()
858 : SendTest(kDefaultTimeoutMs),
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000859 clock_(Clock::GetRealTimeClock()),
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000860 transport_adapter_(ReceiveTransport()),
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000861 crit_(CriticalSectionWrapper::CreateCriticalSection()),
862 last_packet_time_ms_(-1),
863 capturer_(NULL) {
sprang@webrtc.orgd9b95602014-01-27 13:03:02 +0000864 transport_adapter_.Enable();
865 }
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000866
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000867 private:
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000868 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000869 CriticalSectionScoped lock(crit_.get());
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000870 last_packet_time_ms_ = clock_->TimeInMilliseconds();
871 capturer_->Stop();
872 return SEND_PACKET;
873 }
874
875 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000876 CriticalSectionScoped lock(crit_.get());
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000877 const int kVideoMutedThresholdMs = 10000;
pbos@webrtc.orgc279a5d2014-01-24 09:30:53 +0000878 if (last_packet_time_ms_ > 0 &&
879 clock_->TimeInMilliseconds() - last_packet_time_ms_ >
880 kVideoMutedThresholdMs)
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000881 observation_complete_->Set();
882 // Receive statistics reporting having lost 50% of the packets.
883 FakeReceiveStatistics receive_stats(kSendSsrcs[0], 1, 1, 0);
884 RTCPSender rtcp_sender(
885 0, false, Clock::GetRealTimeClock(), &receive_stats);
886 EXPECT_EQ(0, rtcp_sender.RegisterSendTransport(&transport_adapter_));
887
888 rtcp_sender.SetRTCPStatus(kRtcpNonCompound);
889 rtcp_sender.SetRemoteSSRC(kSendSsrcs[0]);
890
891 RTCPSender::FeedbackState feedback_state;
892
893 EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
894 return SEND_PACKET;
895 }
896
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000897 virtual void SetReceivers(
898 PacketReceiver* send_transport_receiver,
899 PacketReceiver* receive_transport_receiver) OVERRIDE {
900 RtpRtcpObserver::SetReceivers(send_transport_receiver,
901 send_transport_receiver);
902 }
903
904 virtual size_t GetNumStreams() const OVERRIDE { return 3; }
905
906 virtual void OnFrameGeneratorCapturerCreated(
907 test::FrameGeneratorCapturer* frame_generator_capturer) {
908 CriticalSectionScoped lock(crit_.get());
909 capturer_ = frame_generator_capturer;
910 }
911
912 virtual void PerformTest() OVERRIDE {
913 EXPECT_EQ(kEventSignaled, Wait())
914 << "Timed out while waiting for RTP packets to stop being sent.";
915 }
916
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000917 Clock* const clock_;
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000918 internal::TransportAdapter transport_adapter_;
pbos@webrtc.orgde1429e2014-04-28 13:00:21 +0000919 const scoped_ptr<CriticalSectionWrapper> crit_;
920 int64_t last_packet_time_ms_ GUARDED_BY(crit_);
921 test::FrameGeneratorCapturer* capturer_ GUARDED_BY(crit_);
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000922 } test;
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000923
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000924 RunBaseTest(&test);
stefan@webrtc.org4ab4fc02013-11-25 11:54:24 +0000925}
926
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000927TEST_F(VideoSendStreamTest, ProducesStats) {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000928 class ProducesStats : public test::SendTest {
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000929 public:
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000930 ProducesStats()
931 : SendTest(kDefaultTimeoutMs),
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +0000932 stream_(NULL),
933 event_(EventWrapper::Create()) {}
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000934
935 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +0000936 event_->Set();
937
938 return SEND_PACKET;
939 }
940
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000941 private:
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +0000942 bool WaitForFilledStats() {
943 Clock* clock = Clock::GetRealTimeClock();
944 int64_t now = clock->TimeInMilliseconds();
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000945 int64_t stop_time = now + kDefaultTimeoutMs;
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +0000946 while (now < stop_time) {
947 int64_t time_left = stop_time - now;
948 if (time_left > 0 && event_->Wait(time_left) == kEventSignaled &&
949 CheckStats()) {
950 return true;
951 }
952 now = clock->TimeInMilliseconds();
953 }
954 return false;
955 }
956
957 bool CheckStats() {
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000958 VideoSendStream::Stats stats = stream_->GetStats();
959 // Check that all applicable data sources have been used.
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000960 if (stats.input_frame_rate > 0 && stats.encode_frame_rate > 0
961 && !stats.substreams.empty()) {
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000962 uint32_t ssrc = stats.substreams.begin()->first;
963 EXPECT_NE(
964 config_.rtp.ssrcs.end(),
965 std::find(
966 config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc));
967 // Check for data populated by various sources. RTCP excluded as this
968 // data is received from remote side. Tested in call tests instead.
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +0000969 const StreamStats& entry = stats.substreams[ssrc];
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000970 if (entry.key_frames > 0u && entry.bitrate_bps > 0 &&
stefan@webrtc.org168f23f2014-07-11 13:44:02 +0000971 entry.rtp_stats.packets > 0u && entry.avg_delay_ms > 0 &&
972 entry.max_delay_ms > 0) {
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +0000973 return true;
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000974 }
975 }
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +0000976 return false;
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000977 }
978
979 void SetConfig(const VideoSendStream::Config& config) { config_ = config; }
980
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000981 virtual void ModifyConfigs(
982 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000983 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000984 std::vector<VideoStream>* video_streams) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000985 SetConfig(*send_config);
986 }
987
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +0000988 virtual void OnStreamsCreated(
989 VideoSendStream* send_stream,
990 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +0000991 stream_ = send_stream;
992 }
993
994 virtual void PerformTest() OVERRIDE {
995 EXPECT_TRUE(WaitForFilledStats())
996 << "Timed out waiting for filled statistics.";
997 }
sprang@webrtc.orgccd42842014-01-07 09:54:34 +0000998
999 VideoSendStream* stream_;
1000 VideoSendStream::Config config_;
sprang@webrtc.org60ad5fd2014-03-06 10:03:36 +00001001 scoped_ptr<EventWrapper> event_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001002 } test;
sprang@webrtc.orgccd42842014-01-07 09:54:34 +00001003
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001004 RunBaseTest(&test);
sprang@webrtc.orgccd42842014-01-07 09:54:34 +00001005}
1006
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001007// This test first observes "high" bitrate use at which point it sends a REMB to
1008// indicate that it should be lowered significantly. The test then observes that
1009// the bitrate observed is sinking well below the min-transmit-bitrate threshold
1010// to verify that the min-transmit bitrate respects incoming REMB.
andresp@webrtc.org44caf012014-03-26 21:00:21 +00001011//
1012// Note that the test starts at "high" bitrate and does not ramp up to "higher"
1013// bitrate since no receiver block or remb is sent in the initial phase.
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001014TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
1015 static const int kMinTransmitBitrateBps = 400000;
andresp@webrtc.org44caf012014-03-26 21:00:21 +00001016 static const int kHighBitrateBps = 150000;
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001017 static const int kRembBitrateBps = 80000;
1018 static const int kRembRespectedBitrateBps = 100000;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001019 class BitrateObserver : public test::SendTest, public PacketReceiver {
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001020 public:
1021 BitrateObserver()
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001022 : SendTest(kDefaultTimeoutMs),
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001023 feedback_transport_(ReceiveTransport()),
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001024 bitrate_capped_(false) {
1025 RtpRtcp::Configuration config;
1026 feedback_transport_.Enable();
1027 config.outgoing_transport = &feedback_transport_;
1028 rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
1029 rtp_rtcp_->SetREMBStatus(true);
1030 rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
1031 }
1032
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001033 virtual void OnStreamsCreated(
1034 VideoSendStream* send_stream,
1035 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001036 stream_ = send_stream;
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001037 }
1038
1039 private:
pbos@webrtc.orgcaba2d22014-05-14 13:57:12 +00001040 virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
1041 size_t length) OVERRIDE {
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001042 if (RtpHeaderParser::IsRtcp(packet, length))
pbos@webrtc.orgcaba2d22014-05-14 13:57:12 +00001043 return DELIVERY_OK;
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001044
1045 RTPHeader header;
pbos@webrtc.org62bafae2014-07-08 12:10:51 +00001046 if (!parser_->Parse(packet, length, &header))
pbos@webrtc.orgcaba2d22014-05-14 13:57:12 +00001047 return DELIVERY_PACKET_ERROR;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001048 assert(stream_ != NULL);
1049 VideoSendStream::Stats stats = stream_->GetStats();
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001050 if (!stats.substreams.empty()) {
1051 EXPECT_EQ(1u, stats.substreams.size());
1052 int bitrate_bps = stats.substreams.begin()->second.bitrate_bps;
1053 test::PrintResult(
1054 "bitrate_stats_",
1055 "min_transmit_bitrate_low_remb",
1056 "bitrate_bps",
1057 static_cast<size_t>(bitrate_bps),
1058 "bps",
1059 false);
1060 if (bitrate_bps > kHighBitrateBps) {
1061 rtp_rtcp_->SetREMBData(kRembBitrateBps, 1, &header.ssrc);
1062 rtp_rtcp_->Process();
1063 bitrate_capped_ = true;
1064 } else if (bitrate_capped_ &&
1065 bitrate_bps < kRembRespectedBitrateBps) {
1066 observation_complete_->Set();
1067 }
1068 }
pbos@webrtc.orgcaba2d22014-05-14 13:57:12 +00001069 return DELIVERY_OK;
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001070 }
1071
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001072 virtual void SetReceivers(
1073 PacketReceiver* send_transport_receiver,
1074 PacketReceiver* receive_transport_receiver) OVERRIDE {
1075 RtpRtcpObserver::SetReceivers(this, send_transport_receiver);
1076 }
1077
1078 virtual void ModifyConfigs(
1079 VideoSendStream::Config* send_config,
pbos@webrtc.orgbe9d2a42014-06-30 13:19:09 +00001080 std::vector<VideoReceiveStream::Config>* receive_configs,
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001081 std::vector<VideoStream>* video_streams) OVERRIDE {
1082 send_config->rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps;
1083 }
1084
1085 virtual void PerformTest() OVERRIDE {
1086 EXPECT_EQ(kEventSignaled, Wait())
1087 << "Timeout while waiting for low bitrate stats after REMB.";
1088 }
1089
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001090 scoped_ptr<RtpRtcp> rtp_rtcp_;
1091 internal::TransportAdapter feedback_transport_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001092 VideoSendStream* stream_;
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001093 bool bitrate_capped_;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001094 } test;
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001095
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001096 RunBaseTest(&test);
pbos@webrtc.org709e2972014-03-19 10:59:52 +00001097}
1098
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +00001099TEST_F(VideoSendStreamTest, CapturesTextureAndI420VideoFrames) {
1100 class FrameObserver : public I420FrameCallback {
1101 public:
1102 FrameObserver() : output_frame_event_(EventWrapper::Create()) {}
1103
1104 void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
1105 // Clone the frame because the caller owns it.
1106 output_frames_.push_back(video_frame->CloneFrame());
1107 output_frame_event_->Set();
1108 }
1109
1110 void WaitOutputFrame() {
1111 const unsigned long kWaitFrameTimeoutMs = 3000;
1112 EXPECT_EQ(kEventSignaled, output_frame_event_->Wait(kWaitFrameTimeoutMs))
1113 << "Timeout while waiting for output frames.";
1114 }
1115
1116 const std::vector<I420VideoFrame*>& output_frames() const {
1117 return output_frames_.get();
1118 }
1119
1120 private:
1121 // Delivered output frames.
1122 ScopedVector<I420VideoFrame> output_frames_;
1123
1124 // Indicate an output frame has arrived.
1125 scoped_ptr<EventWrapper> output_frame_event_;
1126 };
1127
1128 // Initialize send stream.
1129 test::NullTransport transport;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001130 CreateSenderCall(Call::Config(&transport));
1131
1132 CreateSendConfig(1);
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +00001133 FrameObserver observer;
1134 send_config_.pre_encode_callback = &observer;
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001135 CreateStreams();
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +00001136
1137 // Prepare five input frames. Send I420VideoFrame and TextureVideoFrame
1138 // alternatively.
1139 ScopedVector<I420VideoFrame> input_frames;
1140 int width = static_cast<int>(video_streams_[0].width);
1141 int height = static_cast<int>(video_streams_[0].height);
1142 webrtc::RefCountImpl<FakeNativeHandle>* handle1 =
1143 new webrtc::RefCountImpl<FakeNativeHandle>();
1144 webrtc::RefCountImpl<FakeNativeHandle>* handle2 =
1145 new webrtc::RefCountImpl<FakeNativeHandle>();
1146 webrtc::RefCountImpl<FakeNativeHandle>* handle3 =
1147 new webrtc::RefCountImpl<FakeNativeHandle>();
1148 input_frames.push_back(new TextureVideoFrame(handle1, width, height, 1, 1));
1149 input_frames.push_back(new TextureVideoFrame(handle2, width, height, 2, 2));
1150 input_frames.push_back(CreateI420VideoFrame(width, height, 1));
1151 input_frames.push_back(CreateI420VideoFrame(width, height, 2));
1152 input_frames.push_back(new TextureVideoFrame(handle3, width, height, 3, 3));
1153
1154 send_stream_->Start();
1155 for (size_t i = 0; i < input_frames.size(); i++) {
1156 // Make a copy of the input frame because the buffer will be swapped.
1157 scoped_ptr<I420VideoFrame> frame(input_frames[i]->CloneFrame());
1158 send_stream_->Input()->SwapFrame(frame.get());
1159 // Do not send the next frame too fast, so the frame dropper won't drop it.
1160 if (i < input_frames.size() - 1)
1161 SleepMs(1000 / video_streams_[0].max_framerate);
1162 // Wait until the output frame is received before sending the next input
1163 // frame. Or the previous input frame may be replaced without delivering.
1164 observer.WaitOutputFrame();
1165 }
1166 send_stream_->Stop();
1167
1168 // Test if the input and output frames are the same. render_time_ms and
1169 // timestamp are not compared because capturer sets those values.
1170 ExpectEqualFramesVector(input_frames.get(), observer.output_frames());
1171
pbos@webrtc.org994d0b72014-06-27 08:47:52 +00001172 DestroyStreams();
wuchengli@chromium.orgf425b552014-06-20 12:04:05 +00001173}
1174
1175void ExpectEqualFrames(const I420VideoFrame& frame1,
1176 const I420VideoFrame& frame2) {
1177 if (frame1.native_handle() != NULL || frame2.native_handle() != NULL)
1178 ExpectEqualTextureFrames(frame1, frame2);
1179 else
1180 ExpectEqualBufferFrames(frame1, frame2);
1181}
1182
1183void ExpectEqualTextureFrames(const I420VideoFrame& frame1,
1184 const I420VideoFrame& frame2) {
1185 EXPECT_EQ(frame1.native_handle(), frame2.native_handle());
1186 EXPECT_EQ(frame1.width(), frame2.width());
1187 EXPECT_EQ(frame1.height(), frame2.height());
1188}
1189
1190void ExpectEqualBufferFrames(const I420VideoFrame& frame1,
1191 const I420VideoFrame& frame2) {
1192 EXPECT_EQ(frame1.width(), frame2.width());
1193 EXPECT_EQ(frame1.height(), frame2.height());
1194 EXPECT_EQ(frame1.stride(kYPlane), frame2.stride(kYPlane));
1195 EXPECT_EQ(frame1.stride(kUPlane), frame2.stride(kUPlane));
1196 EXPECT_EQ(frame1.stride(kVPlane), frame2.stride(kVPlane));
1197 EXPECT_EQ(frame1.ntp_time_ms(), frame2.ntp_time_ms());
1198 ASSERT_EQ(frame1.allocated_size(kYPlane), frame2.allocated_size(kYPlane));
1199 EXPECT_EQ(0,
1200 memcmp(frame1.buffer(kYPlane),
1201 frame2.buffer(kYPlane),
1202 frame1.allocated_size(kYPlane)));
1203 ASSERT_EQ(frame1.allocated_size(kUPlane), frame2.allocated_size(kUPlane));
1204 EXPECT_EQ(0,
1205 memcmp(frame1.buffer(kUPlane),
1206 frame2.buffer(kUPlane),
1207 frame1.allocated_size(kUPlane)));
1208 ASSERT_EQ(frame1.allocated_size(kVPlane), frame2.allocated_size(kVPlane));
1209 EXPECT_EQ(0,
1210 memcmp(frame1.buffer(kVPlane),
1211 frame2.buffer(kVPlane),
1212 frame1.allocated_size(kVPlane)));
1213}
1214
1215void ExpectEqualFramesVector(const std::vector<I420VideoFrame*>& frames1,
1216 const std::vector<I420VideoFrame*>& frames2) {
1217 EXPECT_EQ(frames1.size(), frames2.size());
1218 for (size_t i = 0; i < std::min(frames1.size(), frames2.size()); ++i)
1219 ExpectEqualFrames(*frames1[i], *frames2[i]);
1220}
1221
1222I420VideoFrame* CreateI420VideoFrame(int width, int height, uint8_t data) {
1223 I420VideoFrame* frame = new I420VideoFrame();
1224 const int kSizeY = width * height * 2;
1225 const int kSizeUV = width * height;
1226 scoped_ptr<uint8_t[]> buffer(new uint8_t[kSizeY]);
1227 memset(buffer.get(), data, kSizeY);
1228 frame->CreateFrame(kSizeY,
1229 buffer.get(),
1230 kSizeUV,
1231 buffer.get(),
1232 kSizeUV,
1233 buffer.get(),
1234 width,
1235 height,
1236 width,
1237 width / 2,
1238 width / 2);
1239 frame->set_timestamp(data);
1240 frame->set_ntp_time_ms(data);
1241 frame->set_render_time_ms(data);
1242 return frame;
1243}
1244
pbos@webrtc.org161f8082014-07-07 14:22:35 +00001245TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
1246 class EncoderStateObserver : public test::SendTest, public VideoEncoder {
1247 public:
1248 EncoderStateObserver()
1249 : SendTest(kDefaultTimeoutMs),
1250 crit_(CriticalSectionWrapper::CreateCriticalSection()),
1251 initialized_(false),
1252 callback_registered_(false),
1253 num_releases_(0),
1254 released_(false) {}
1255
1256 bool IsReleased() {
1257 CriticalSectionScoped lock(crit_.get());
1258 return released_;
1259 }
1260
1261 bool IsReadyForEncode() {
1262 CriticalSectionScoped lock(crit_.get());
1263 return initialized_ && callback_registered_;
1264 }
1265
1266 size_t num_releases() {
1267 CriticalSectionScoped lock(crit_.get());
1268 return num_releases_;
1269 }
1270
1271 private:
1272 virtual int32_t InitEncode(const VideoCodec* codecSettings,
1273 int32_t numberOfCores,
1274 uint32_t maxPayloadSize) OVERRIDE {
1275 CriticalSectionScoped lock(crit_.get());
1276 EXPECT_FALSE(initialized_);
1277 initialized_ = true;
1278 released_ = false;
1279 return 0;
1280 }
1281
1282 virtual int32_t Encode(
1283 const I420VideoFrame& inputImage,
1284 const CodecSpecificInfo* codecSpecificInfo,
1285 const std::vector<VideoFrameType>* frame_types) OVERRIDE {
1286 EXPECT_TRUE(IsReadyForEncode());
1287
1288 observation_complete_->Set();
1289 return 0;
1290 }
1291
1292 virtual int32_t RegisterEncodeCompleteCallback(
1293 EncodedImageCallback* callback) OVERRIDE {
1294 CriticalSectionScoped lock(crit_.get());
1295 EXPECT_TRUE(initialized_);
1296 callback_registered_ = true;
1297 return 0;
1298 }
1299
1300 virtual int32_t Release() OVERRIDE {
1301 CriticalSectionScoped lock(crit_.get());
1302 EXPECT_TRUE(IsReadyForEncode());
1303 EXPECT_FALSE(released_);
1304 initialized_ = false;
1305 callback_registered_ = false;
1306 released_ = true;
1307 ++num_releases_;
1308 return 0;
1309 }
1310
1311 virtual int32_t SetChannelParameters(uint32_t packetLoss,
1312 int rtt) OVERRIDE {
1313 EXPECT_TRUE(IsReadyForEncode());
1314 return 0;
1315 }
1316
1317 virtual int32_t SetRates(uint32_t newBitRate, uint32_t frameRate) OVERRIDE {
1318 EXPECT_TRUE(IsReadyForEncode());
1319 return 0;
1320 }
1321
1322 virtual void OnStreamsCreated(
1323 VideoSendStream* send_stream,
1324 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
1325 // Encoder initialization should be done in stream construction before
1326 // starting.
1327 EXPECT_TRUE(IsReadyForEncode());
1328 stream_ = send_stream;
1329 }
1330
1331 virtual void ModifyConfigs(
1332 VideoSendStream::Config* send_config,
1333 std::vector<VideoReceiveStream::Config>* receive_configs,
1334 std::vector<VideoStream>* video_streams) OVERRIDE {
1335 send_config->encoder_settings.encoder = this;
1336 video_streams_ = *video_streams;
1337 }
1338
1339 virtual void PerformTest() OVERRIDE {
1340 EXPECT_EQ(kEventSignaled, Wait())
1341 << "Timed out while waiting for Encode.";
1342 EXPECT_EQ(0u, num_releases());
1343 stream_->ReconfigureVideoEncoder(video_streams_, NULL);
1344 EXPECT_EQ(0u, num_releases());
1345 stream_->Stop();
1346 // Encoder should not be released before destroying the VideoSendStream.
1347 EXPECT_FALSE(IsReleased());
1348 EXPECT_TRUE(IsReadyForEncode());
1349 stream_->Start();
1350 // Sanity check, make sure we still encode frames with this encoder.
1351 EXPECT_EQ(kEventSignaled, Wait())
1352 << "Timed out while waiting for Encode.";
1353 }
1354
1355 scoped_ptr<CriticalSectionWrapper> crit_;
1356 VideoSendStream* stream_;
1357 bool initialized_ GUARDED_BY(crit_);
1358 bool callback_registered_ GUARDED_BY(crit_);
1359 size_t num_releases_ GUARDED_BY(crit_);
1360 bool released_ GUARDED_BY(crit_);
1361 std::vector<VideoStream> video_streams_;
1362 } test_encoder;
1363
1364 RunBaseTest(&test_encoder);
1365
1366 EXPECT_TRUE(test_encoder.IsReleased());
1367 EXPECT_EQ(1u, test_encoder.num_releases());
1368}
1369
pbos@webrtc.org91f17522014-07-10 10:13:37 +00001370TEST_F(VideoSendStreamTest, EncoderSetupPropagatesVp8Config) {
1371 class VideoCodecConfigObserver : public test::SendTest,
1372 public test::FakeEncoder {
1373 public:
1374 VideoCodecConfigObserver()
1375 : SendTest(kDefaultTimeoutMs),
1376 FakeEncoder(Clock::GetRealTimeClock()),
1377 num_initializations_(0) {
1378 memset(&vp8_settings_, 0, sizeof(vp8_settings_));
1379 }
1380
1381 private:
1382 virtual void ModifyConfigs(
1383 VideoSendStream::Config* send_config,
1384 std::vector<VideoReceiveStream::Config>* receive_configs,
1385 std::vector<VideoStream>* video_streams) OVERRIDE {
1386 send_config->encoder_settings.encoder = this;
1387 send_config->encoder_settings.payload_name = "VP8";
1388
1389 video_streams_ = *video_streams;
1390 }
1391
1392 virtual void OnStreamsCreated(
1393 VideoSendStream* send_stream,
1394 const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
1395 stream_ = send_stream;
1396 }
1397
1398 virtual int32_t InitEncode(const VideoCodec* config,
1399 int32_t number_of_cores,
1400 uint32_t max_payload_size) OVERRIDE {
1401 EXPECT_EQ(kVideoCodecVP8, config->codecType);
1402 EXPECT_EQ(0,
1403 memcmp(&config->codecSpecific.VP8,
1404 &vp8_settings_,
1405 sizeof(vp8_settings_)));
1406 ++num_initializations_;
1407 return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
1408 }
1409
1410 virtual void PerformTest() OVERRIDE {
1411 EXPECT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
1412
1413 vp8_settings_.denoisingOn = true;
1414 stream_->ReconfigureVideoEncoder(video_streams_, &vp8_settings_);
1415 EXPECT_EQ(2u, num_initializations_)
1416 << "ReconfigureVideoEncoder did not reinitialize the encoder with "
1417 "new encoder settings.";
1418 }
1419
pbos@webrtc.orgbd9c0922014-07-10 13:21:40 +00001420 int32_t Encode(
1421 const I420VideoFrame& input_image,
1422 const CodecSpecificInfo* codec_specific_info,
1423 const std::vector<VideoFrameType>* frame_types) {
1424 // Silently skip the encode, FakeEncoder::Encode doesn't produce VP8.
1425 return 0;
1426 }
1427
pbos@webrtc.org91f17522014-07-10 10:13:37 +00001428 virtual const void* GetEncoderSettings() OVERRIDE { return &vp8_settings_; }
1429
1430 VideoCodecVP8 vp8_settings_;
1431 size_t num_initializations_;
1432 VideoSendStream* stream_;
1433 std::vector<VideoStream> video_streams_;
1434 } test;
1435
1436 RunBaseTest(&test);
1437}
1438
pbos@webrtc.org2f4b14e2014-07-15 15:25:39 +00001439TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
1440 class RtcpByeTest : public test::SendTest {
1441 public:
1442 RtcpByeTest() : SendTest(kDefaultTimeoutMs), media_bytes_sent_(0) {}
1443
1444 private:
1445 virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
1446 RTPHeader header;
1447 EXPECT_TRUE(parser_->Parse(packet, length, &header));
1448 media_bytes_sent_ += length - header.headerLength - header.paddingLength;
1449 return SEND_PACKET;
1450 }
1451
1452 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
1453 RTCPUtility::RTCPParserV2 parser(packet, length, true);
1454 EXPECT_TRUE(parser.IsValid());
1455
1456 RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
1457 uint32_t sender_octet_count = 0;
1458 while (packet_type != RTCPUtility::kRtcpNotValidCode) {
1459 if (packet_type == RTCPUtility::kRtcpSrCode) {
1460 sender_octet_count = parser.Packet().SR.SenderOctetCount;
1461 EXPECT_EQ(sender_octet_count, media_bytes_sent_);
1462 if (sender_octet_count > 0)
1463 observation_complete_->Set();
1464 }
1465
1466 packet_type = parser.Iterate();
1467 }
1468
1469 return SEND_PACKET;
1470 }
1471
1472 virtual void PerformTest() OVERRIDE {
1473 EXPECT_EQ(kEventSignaled, Wait())
1474 << "Timed out while waiting for RTCP sender report.";
1475 }
1476
1477 size_t media_bytes_sent_;
1478 } test;
1479
1480 RunBaseTest(&test);
1481}
1482
pbos@webrtc.org119a1cc2013-08-20 13:14:07 +00001483} // namespace webrtc