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 */
10#include <algorithm>
11#include <sstream>
12#include <string>
13
14#include "testing/gtest/include/gtest/gtest.h"
15
16#include "webrtc/base/checks.h"
17#include "webrtc/base/scoped_ptr.h"
18#include "webrtc/base/thread_annotations.h"
19#include "webrtc/call.h"
20#include "webrtc/call/transport_adapter.h"
21#include "webrtc/common.h"
22#include "webrtc/config.h"
23#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
24#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
25#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
26#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
27#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
28#include "webrtc/test/call_test.h"
29#include "webrtc/test/direct_transport.h"
30#include "webrtc/test/encoder_settings.h"
31#include "webrtc/test/fake_audio_device.h"
32#include "webrtc/test/fake_decoder.h"
33#include "webrtc/test/fake_encoder.h"
34#include "webrtc/test/frame_generator.h"
35#include "webrtc/test/frame_generator_capturer.h"
36#include "webrtc/test/rtp_rtcp_observer.h"
37#include "webrtc/test/testsupport/fileutils.h"
38#include "webrtc/test/testsupport/perf_test.h"
39#include "webrtc/voice_engine/include/voe_base.h"
40#include "webrtc/voice_engine/include/voe_codec.h"
41#include "webrtc/voice_engine/include/voe_network.h"
42#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
43#include "webrtc/voice_engine/include/voe_video_sync.h"
44
45namespace webrtc {
46
47class CallPerfTest : public test::CallTest {
48 protected:
49  void TestAudioVideoSync(bool fec, bool create_audio_first);
50
51  void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
52
53  void TestMinTransmitBitrate(bool pad_to_min_bitrate);
54
55  void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
56                          int threshold_ms,
57                          int start_time_ms,
58                          int run_time_ms);
59};
60
61class SyncRtcpObserver : public test::RtpRtcpObserver {
62 public:
63  SyncRtcpObserver() : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs) {}
64
65  Action OnSendRtcp(const uint8_t* packet, size_t length) override {
66    RTCPUtility::RTCPParserV2 parser(packet, length, true);
67    EXPECT_TRUE(parser.IsValid());
68
69    for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
70         packet_type != RTCPUtility::RTCPPacketTypes::kInvalid;
71         packet_type = parser.Iterate()) {
72      if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
73        const RTCPUtility::RTCPPacket& packet = parser.Packet();
74        RtcpMeasurement ntp_rtp_pair(
75            packet.SR.NTPMostSignificant,
76            packet.SR.NTPLeastSignificant,
77            packet.SR.RTPTimestamp);
78        StoreNtpRtpPair(ntp_rtp_pair);
79      }
80    }
81    return SEND_PACKET;
82  }
83
84  int64_t RtpTimestampToNtp(uint32_t timestamp) const {
85    rtc::CritScope lock(&crit_);
86    int64_t timestamp_in_ms = -1;
87    if (ntp_rtp_pairs_.size() == 2) {
88      // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the
89      // RTCP sender where it sends RTCP SR before any RTP packets, which leads
90      // to a bogus NTP/RTP mapping.
91      RtpToNtpMs(timestamp, ntp_rtp_pairs_, &timestamp_in_ms);
92      return timestamp_in_ms;
93    }
94    return -1;
95  }
96
97 private:
98  void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) {
99    rtc::CritScope lock(&crit_);
100    for (RtcpList::iterator it = ntp_rtp_pairs_.begin();
101         it != ntp_rtp_pairs_.end();
102         ++it) {
103      if (ntp_rtp_pair.ntp_secs == it->ntp_secs &&
104          ntp_rtp_pair.ntp_frac == it->ntp_frac) {
105        // This RTCP has already been added to the list.
106        return;
107      }
108    }
109    // We need two RTCP SR reports to map between RTP and NTP. More than two
110    // will not improve the mapping.
111    if (ntp_rtp_pairs_.size() == 2) {
112      ntp_rtp_pairs_.pop_back();
113    }
114    ntp_rtp_pairs_.push_front(ntp_rtp_pair);
115  }
116
117  mutable rtc::CriticalSection crit_;
118  RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
119};
120
121class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
122  static const int kInSyncThresholdMs = 50;
123  static const int kStartupTimeMs = 2000;
124  static const int kMinRunTimeMs = 30000;
125
126 public:
127  VideoRtcpAndSyncObserver(Clock* clock,
128                           int voe_channel,
129                           VoEVideoSync* voe_sync,
130                           SyncRtcpObserver* audio_observer)
131      : clock_(clock),
132        voe_channel_(voe_channel),
133        voe_sync_(voe_sync),
134        audio_observer_(audio_observer),
135        creation_time_ms_(clock_->TimeInMilliseconds()),
136        first_time_in_sync_(-1) {}
137
138  void RenderFrame(const VideoFrame& video_frame,
139                   int time_to_render_ms) override {
140    int64_t now_ms = clock_->TimeInMilliseconds();
141    uint32_t playout_timestamp = 0;
142    if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0)
143      return;
144    int64_t latest_audio_ntp =
145        audio_observer_->RtpTimestampToNtp(playout_timestamp);
146    int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp());
147    if (latest_audio_ntp < 0 || latest_video_ntp < 0)
148      return;
149    int time_until_render_ms =
150        std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms));
151    latest_video_ntp += time_until_render_ms;
152    int64_t stream_offset = latest_audio_ntp - latest_video_ntp;
153    std::stringstream ss;
154    ss << stream_offset;
155    webrtc::test::PrintResult("stream_offset",
156                              "",
157                              "synchronization",
158                              ss.str(),
159                              "ms",
160                              false);
161    int64_t time_since_creation = now_ms - creation_time_ms_;
162    // During the first couple of seconds audio and video can falsely be
163    // estimated as being synchronized. We don't want to trigger on those.
164    if (time_since_creation < kStartupTimeMs)
165      return;
166    if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
167      if (first_time_in_sync_ == -1) {
168        first_time_in_sync_ = now_ms;
169        webrtc::test::PrintResult("sync_convergence_time",
170                                  "",
171                                  "synchronization",
172                                  time_since_creation,
173                                  "ms",
174                                  false);
175      }
176      if (time_since_creation > kMinRunTimeMs)
177        observation_complete_.Set();
178    }
179  }
180
181  bool IsTextureSupported() const override { return false; }
182
183 private:
184  Clock* const clock_;
185  const int voe_channel_;
186  VoEVideoSync* const voe_sync_;
187  SyncRtcpObserver* const audio_observer_;
188  const int64_t creation_time_ms_;
189  int64_t first_time_in_sync_;
190};
191
192void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
193  const char* kSyncGroup = "av_sync";
194  const uint32_t kAudioSendSsrc = 1234;
195  const uint32_t kAudioRecvSsrc = 5678;
196  class AudioPacketReceiver : public PacketReceiver {
197   public:
198    AudioPacketReceiver(int channel, VoENetwork* voe_network)
199        : channel_(channel),
200          voe_network_(voe_network),
201          parser_(RtpHeaderParser::Create()) {}
202    DeliveryStatus DeliverPacket(MediaType media_type,
203                                 const uint8_t* packet,
204                                 size_t length,
205                                 const PacketTime& packet_time) override {
206      EXPECT_TRUE(media_type == MediaType::ANY ||
207                  media_type == MediaType::AUDIO);
208      int ret;
209      if (parser_->IsRtcp(packet, length)) {
210        ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length);
211      } else {
212        ret = voe_network_->ReceivedRTPPacket(channel_, packet, length,
213                                              PacketTime());
214      }
215      return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
216    }
217
218   private:
219    int channel_;
220    VoENetwork* voe_network_;
221    rtc::scoped_ptr<RtpHeaderParser> parser_;
222  };
223
224  VoiceEngine* voice_engine = VoiceEngine::Create();
225  VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
226  VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
227  VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine);
228  VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine);
229  const std::string audio_filename =
230      test::ResourcePath("voice_engine/audio_long16", "pcm");
231  ASSERT_STRNE("", audio_filename.c_str());
232  test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
233                                          audio_filename);
234  EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
235  Config voe_config;
236  voe_config.Set<VoicePacing>(new VoicePacing(true));
237  int send_channel_id = voe_base->CreateChannel(voe_config);
238  int recv_channel_id = voe_base->CreateChannel();
239
240  SyncRtcpObserver audio_observer;
241
242  AudioState::Config send_audio_state_config;
243  send_audio_state_config.voice_engine = voice_engine;
244  Call::Config sender_config;
245  sender_config.audio_state = AudioState::Create(send_audio_state_config);
246  Call::Config receiver_config;
247  receiver_config.audio_state = sender_config.audio_state;
248  CreateCalls(sender_config, receiver_config);
249
250  AudioPacketReceiver voe_send_packet_receiver(send_channel_id, voe_network);
251  AudioPacketReceiver voe_recv_packet_receiver(recv_channel_id, voe_network);
252
253  FakeNetworkPipe::Config net_config;
254  net_config.queue_delay_ms = 500;
255  net_config.loss_percent = 5;
256  test::PacketTransport audio_send_transport(
257      nullptr, &audio_observer, test::PacketTransport::kSender, net_config);
258  audio_send_transport.SetReceiver(&voe_recv_packet_receiver);
259  test::PacketTransport audio_receive_transport(
260      nullptr, &audio_observer, test::PacketTransport::kReceiver, net_config);
261  audio_receive_transport.SetReceiver(&voe_send_packet_receiver);
262
263  internal::TransportAdapter send_transport_adapter(&audio_send_transport);
264  send_transport_adapter.Enable();
265  EXPECT_EQ(0, voe_network->RegisterExternalTransport(send_channel_id,
266                                                      send_transport_adapter));
267
268  internal::TransportAdapter recv_transport_adapter(&audio_receive_transport);
269  recv_transport_adapter.Enable();
270  EXPECT_EQ(0, voe_network->RegisterExternalTransport(recv_channel_id,
271                                                      recv_transport_adapter));
272
273  VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), recv_channel_id,
274                                    voe_sync, &audio_observer);
275
276  test::PacketTransport sync_send_transport(sender_call_.get(), &observer,
277                                            test::PacketTransport::kSender,
278                                            FakeNetworkPipe::Config());
279  sync_send_transport.SetReceiver(receiver_call_->Receiver());
280  test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer,
281                                               test::PacketTransport::kReceiver,
282                                               FakeNetworkPipe::Config());
283  sync_receive_transport.SetReceiver(sender_call_->Receiver());
284
285  test::FakeDecoder fake_decoder;
286
287  CreateSendConfig(1, 0, &sync_send_transport);
288  CreateMatchingReceiveConfigs(&sync_receive_transport);
289
290  AudioSendStream::Config audio_send_config(&audio_send_transport);
291  audio_send_config.voe_channel_id = send_channel_id;
292  audio_send_config.rtp.ssrc = kAudioSendSsrc;
293  AudioSendStream* audio_send_stream =
294      sender_call_->CreateAudioSendStream(audio_send_config);
295
296  CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
297  EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
298
299  video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
300  if (fec) {
301    video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
302    video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
303    video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
304    video_receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
305  }
306  video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
307  video_receive_configs_[0].renderer = &observer;
308  video_receive_configs_[0].sync_group = kSyncGroup;
309
310  AudioReceiveStream::Config audio_recv_config;
311  audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
312  audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
313  audio_recv_config.voe_channel_id = recv_channel_id;
314  audio_recv_config.sync_group = kSyncGroup;
315
316  AudioReceiveStream* audio_receive_stream;
317
318  if (create_audio_first) {
319    audio_receive_stream =
320        receiver_call_->CreateAudioReceiveStream(audio_recv_config);
321    CreateVideoStreams();
322  } else {
323    CreateVideoStreams();
324    audio_receive_stream =
325        receiver_call_->CreateAudioReceiveStream(audio_recv_config);
326  }
327
328  CreateFrameGeneratorCapturer();
329
330  Start();
331
332  fake_audio_device.Start();
333  EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
334  EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
335  EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
336
337  EXPECT_TRUE(observer.Wait())
338      << "Timed out while waiting for audio and video to be synchronized.";
339
340  EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
341  EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
342  EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
343  fake_audio_device.Stop();
344
345  Stop();
346  sync_send_transport.StopSending();
347  sync_receive_transport.StopSending();
348  audio_send_transport.StopSending();
349  audio_receive_transport.StopSending();
350
351  DestroyStreams();
352
353  sender_call_->DestroyAudioSendStream(audio_send_stream);
354  receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
355
356  voe_base->DeleteChannel(send_channel_id);
357  voe_base->DeleteChannel(recv_channel_id);
358  voe_base->Release();
359  voe_codec->Release();
360  voe_network->Release();
361  voe_sync->Release();
362
363  DestroyCalls();
364
365  VoiceEngine::Delete(voice_engine);
366}
367
368TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
369  TestAudioVideoSync(false, true);
370}
371
372TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
373  TestAudioVideoSync(false, false);
374}
375
376TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
377  TestAudioVideoSync(true, false);
378}
379
380void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
381                                      int threshold_ms,
382                                      int start_time_ms,
383                                      int run_time_ms) {
384  class CaptureNtpTimeObserver : public test::EndToEndTest,
385                                 public VideoRenderer {
386   public:
387    CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
388                           int threshold_ms,
389                           int start_time_ms,
390                           int run_time_ms)
391        : EndToEndTest(kLongTimeoutMs),
392          net_config_(net_config),
393          clock_(Clock::GetRealTimeClock()),
394          threshold_ms_(threshold_ms),
395          start_time_ms_(start_time_ms),
396          run_time_ms_(run_time_ms),
397          creation_time_ms_(clock_->TimeInMilliseconds()),
398          capturer_(nullptr),
399          rtp_start_timestamp_set_(false),
400          rtp_start_timestamp_(0) {}
401
402   private:
403    test::PacketTransport* CreateSendTransport(Call* sender_call) override {
404      return new test::PacketTransport(
405          sender_call, this, test::PacketTransport::kSender, net_config_);
406    }
407
408    test::PacketTransport* CreateReceiveTransport() override {
409      return new test::PacketTransport(
410          nullptr, this, test::PacketTransport::kReceiver, net_config_);
411    }
412
413    void RenderFrame(const VideoFrame& video_frame,
414                     int time_to_render_ms) override {
415      rtc::CritScope lock(&crit_);
416      if (video_frame.ntp_time_ms() <= 0) {
417        // Haven't got enough RTCP SR in order to calculate the capture ntp
418        // time.
419        return;
420      }
421
422      int64_t now_ms = clock_->TimeInMilliseconds();
423      int64_t time_since_creation = now_ms - creation_time_ms_;
424      if (time_since_creation < start_time_ms_) {
425        // Wait for |start_time_ms_| before start measuring.
426        return;
427      }
428
429      if (time_since_creation > run_time_ms_) {
430        observation_complete_.Set();
431      }
432
433      FrameCaptureTimeList::iterator iter =
434          capture_time_list_.find(video_frame.timestamp());
435      EXPECT_TRUE(iter != capture_time_list_.end());
436
437      // The real capture time has been wrapped to uint32_t before converted
438      // to rtp timestamp in the sender side. So here we convert the estimated
439      // capture time to a uint32_t 90k timestamp also for comparing.
440      uint32_t estimated_capture_timestamp =
441          90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
442      uint32_t real_capture_timestamp = iter->second;
443      int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
444      time_offset_ms = time_offset_ms / 90;
445      std::stringstream ss;
446      ss << time_offset_ms;
447
448      webrtc::test::PrintResult(
449          "capture_ntp_time", "", "real - estimated", ss.str(), "ms", true);
450      EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
451    }
452
453    bool IsTextureSupported() const override { return false; }
454
455    virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
456      rtc::CritScope lock(&crit_);
457      RTPHeader header;
458      EXPECT_TRUE(parser_->Parse(packet, length, &header));
459
460      if (!rtp_start_timestamp_set_) {
461        // Calculate the rtp timestamp offset in order to calculate the real
462        // capture time.
463        uint32_t first_capture_timestamp =
464            90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
465        rtp_start_timestamp_ = header.timestamp - first_capture_timestamp;
466        rtp_start_timestamp_set_ = true;
467      }
468
469      uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_;
470      capture_time_list_.insert(
471          capture_time_list_.end(),
472          std::make_pair(header.timestamp, capture_timestamp));
473      return SEND_PACKET;
474    }
475
476    void OnFrameGeneratorCapturerCreated(
477        test::FrameGeneratorCapturer* frame_generator_capturer) override {
478      capturer_ = frame_generator_capturer;
479    }
480
481    void ModifyVideoConfigs(
482        VideoSendStream::Config* send_config,
483        std::vector<VideoReceiveStream::Config>* receive_configs,
484        VideoEncoderConfig* encoder_config) override {
485      (*receive_configs)[0].renderer = this;
486      // Enable the receiver side rtt calculation.
487      (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
488    }
489
490    void PerformTest() override {
491      EXPECT_TRUE(Wait()) << "Timed out while waiting for "
492                             "estimated capture NTP time to be "
493                             "within bounds.";
494    }
495
496    rtc::CriticalSection crit_;
497    const FakeNetworkPipe::Config net_config_;
498    Clock* const clock_;
499    int threshold_ms_;
500    int start_time_ms_;
501    int run_time_ms_;
502    int64_t creation_time_ms_;
503    test::FrameGeneratorCapturer* capturer_;
504    bool rtp_start_timestamp_set_;
505    uint32_t rtp_start_timestamp_;
506    typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
507    FrameCaptureTimeList capture_time_list_ GUARDED_BY(&crit_);
508  } test(net_config, threshold_ms, start_time_ms, run_time_ms);
509
510  RunBaseTest(&test);
511}
512
513TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
514  FakeNetworkPipe::Config net_config;
515  net_config.queue_delay_ms = 100;
516  // TODO(wu): lower the threshold as the calculation/estimatation becomes more
517  // accurate.
518  const int kThresholdMs = 100;
519  const int kStartTimeMs = 10000;
520  const int kRunTimeMs = 20000;
521  TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
522}
523
524TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
525  FakeNetworkPipe::Config net_config;
526  net_config.queue_delay_ms = 100;
527  net_config.delay_standard_deviation_ms = 10;
528  // TODO(wu): lower the threshold as the calculation/estimatation becomes more
529  // accurate.
530  const int kThresholdMs = 100;
531  const int kStartTimeMs = 10000;
532  const int kRunTimeMs = 20000;
533  TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
534}
535
536void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,
537                                  int encode_delay_ms) {
538  class LoadObserver : public test::SendTest, public webrtc::LoadObserver {
539   public:
540    LoadObserver(LoadObserver::Load tested_load, int encode_delay_ms)
541        : SendTest(kLongTimeoutMs),
542          tested_load_(tested_load),
543          encoder_(Clock::GetRealTimeClock(), encode_delay_ms) {}
544
545    void OnLoadUpdate(Load load) override {
546      if (load == tested_load_)
547        observation_complete_.Set();
548    }
549
550    void ModifyVideoConfigs(
551        VideoSendStream::Config* send_config,
552        std::vector<VideoReceiveStream::Config>* receive_configs,
553        VideoEncoderConfig* encoder_config) override {
554      send_config->overuse_callback = this;
555      send_config->encoder_settings.encoder = &encoder_;
556    }
557
558    void PerformTest() override {
559      EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
560    }
561
562    LoadObserver::Load tested_load_;
563    test::DelayedEncoder encoder_;
564  } test(tested_load, encode_delay_ms);
565
566  RunBaseTest(&test);
567}
568
569TEST_F(CallPerfTest, ReceivesCpuUnderuse) {
570  const int kEncodeDelayMs = 2;
571  TestCpuOveruse(LoadObserver::kUnderuse, kEncodeDelayMs);
572}
573
574TEST_F(CallPerfTest, ReceivesCpuOveruse) {
575  const int kEncodeDelayMs = 35;
576  TestCpuOveruse(LoadObserver::kOveruse, kEncodeDelayMs);
577}
578
579void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
580  static const int kMaxEncodeBitrateKbps = 30;
581  static const int kMinTransmitBitrateBps = 150000;
582  static const int kMinAcceptableTransmitBitrate = 130;
583  static const int kMaxAcceptableTransmitBitrate = 170;
584  static const int kNumBitrateObservationsInRange = 100;
585  static const int kAcceptableBitrateErrorMargin = 15;  // +- 7
586  class BitrateObserver : public test::EndToEndTest {
587   public:
588    explicit BitrateObserver(bool using_min_transmit_bitrate)
589        : EndToEndTest(kLongTimeoutMs),
590          send_stream_(nullptr),
591          pad_to_min_bitrate_(using_min_transmit_bitrate),
592          num_bitrate_observations_in_range_(0) {}
593
594   private:
595    // TODO(holmer): Run this with a timer instead of once per packet.
596    Action OnSendRtp(const uint8_t* packet, size_t length) override {
597      VideoSendStream::Stats stats = send_stream_->GetStats();
598      if (stats.substreams.size() > 0) {
599        RTC_DCHECK_EQ(1u, stats.substreams.size());
600        int bitrate_kbps =
601            stats.substreams.begin()->second.total_bitrate_bps / 1000;
602        if (bitrate_kbps > 0) {
603          test::PrintResult(
604              "bitrate_stats_",
605              (pad_to_min_bitrate_ ? "min_transmit_bitrate"
606                                   : "without_min_transmit_bitrate"),
607              "bitrate_kbps",
608              static_cast<size_t>(bitrate_kbps),
609              "kbps",
610              false);
611          if (pad_to_min_bitrate_) {
612            if (bitrate_kbps > kMinAcceptableTransmitBitrate &&
613                bitrate_kbps < kMaxAcceptableTransmitBitrate) {
614              ++num_bitrate_observations_in_range_;
615            }
616          } else {
617            // Expect bitrate stats to roughly match the max encode bitrate.
618            if (bitrate_kbps > (kMaxEncodeBitrateKbps -
619                                kAcceptableBitrateErrorMargin / 2) &&
620                bitrate_kbps < (kMaxEncodeBitrateKbps +
621                                kAcceptableBitrateErrorMargin / 2)) {
622              ++num_bitrate_observations_in_range_;
623            }
624          }
625          if (num_bitrate_observations_in_range_ ==
626              kNumBitrateObservationsInRange)
627            observation_complete_.Set();
628        }
629      }
630      return SEND_PACKET;
631    }
632
633    void OnVideoStreamsCreated(
634        VideoSendStream* send_stream,
635        const std::vector<VideoReceiveStream*>& receive_streams) override {
636      send_stream_ = send_stream;
637    }
638
639    void ModifyVideoConfigs(
640        VideoSendStream::Config* send_config,
641        std::vector<VideoReceiveStream::Config>* receive_configs,
642        VideoEncoderConfig* encoder_config) override {
643      if (pad_to_min_bitrate_) {
644        encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
645      } else {
646        RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
647      }
648    }
649
650    void PerformTest() override {
651      EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
652    }
653
654    VideoSendStream* send_stream_;
655    const bool pad_to_min_bitrate_;
656    int num_bitrate_observations_in_range_;
657  } test(pad_to_min_bitrate);
658
659  fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
660  RunBaseTest(&test);
661}
662
663TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }
664
665TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) {
666  TestMinTransmitBitrate(false);
667}
668
669TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
670  static const uint32_t kInitialBitrateKbps = 400;
671  static const uint32_t kReconfigureThresholdKbps = 600;
672  static const uint32_t kPermittedReconfiguredBitrateDiffKbps = 100;
673
674  class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder {
675   public:
676    BitrateObserver()
677        : EndToEndTest(kDefaultTimeoutMs),
678          FakeEncoder(Clock::GetRealTimeClock()),
679          time_to_reconfigure_(false, false),
680          encoder_inits_(0),
681          last_set_bitrate_(0),
682          send_stream_(nullptr) {}
683
684    int32_t InitEncode(const VideoCodec* config,
685                       int32_t number_of_cores,
686                       size_t max_payload_size) override {
687      if (encoder_inits_ == 0) {
688        EXPECT_EQ(kInitialBitrateKbps, config->startBitrate)
689            << "Encoder not initialized at expected bitrate.";
690      }
691      ++encoder_inits_;
692      if (encoder_inits_ == 2) {
693        EXPECT_GE(last_set_bitrate_, kReconfigureThresholdKbps);
694        EXPECT_NEAR(config->startBitrate,
695                    last_set_bitrate_,
696                    kPermittedReconfiguredBitrateDiffKbps)
697            << "Encoder reconfigured with bitrate too far away from last set.";
698        observation_complete_.Set();
699      }
700      return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
701    }
702
703    int32_t SetRates(uint32_t new_target_bitrate_kbps,
704                     uint32_t framerate) override {
705      last_set_bitrate_ = new_target_bitrate_kbps;
706      if (encoder_inits_ == 1 &&
707          new_target_bitrate_kbps > kReconfigureThresholdKbps) {
708        time_to_reconfigure_.Set();
709      }
710      return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
711    }
712
713    Call::Config GetSenderCallConfig() override {
714      Call::Config config = EndToEndTest::GetSenderCallConfig();
715      config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
716      return config;
717    }
718
719    void ModifyVideoConfigs(
720        VideoSendStream::Config* send_config,
721        std::vector<VideoReceiveStream::Config>* receive_configs,
722        VideoEncoderConfig* encoder_config) override {
723      send_config->encoder_settings.encoder = this;
724      encoder_config->streams[0].min_bitrate_bps = 50000;
725      encoder_config->streams[0].target_bitrate_bps =
726          encoder_config->streams[0].max_bitrate_bps = 2000000;
727
728      encoder_config_ = *encoder_config;
729    }
730
731    void OnVideoStreamsCreated(
732        VideoSendStream* send_stream,
733        const std::vector<VideoReceiveStream*>& receive_streams) override {
734      send_stream_ = send_stream;
735    }
736
737    void PerformTest() override {
738      ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
739          << "Timed out before receiving an initial high bitrate.";
740      encoder_config_.streams[0].width *= 2;
741      encoder_config_.streams[0].height *= 2;
742      EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_));
743      EXPECT_TRUE(Wait())
744          << "Timed out while waiting for a couple of high bitrate estimates "
745             "after reconfiguring the send stream.";
746    }
747
748   private:
749    rtc::Event time_to_reconfigure_;
750    int encoder_inits_;
751    uint32_t last_set_bitrate_;
752    VideoSendStream* send_stream_;
753    VideoEncoderConfig encoder_config_;
754  } test;
755
756  RunBaseTest(&test);
757}
758
759}  // namespace webrtc
760