Upstream version 9.37.195.0
[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / remote_bitrate_estimator / remote_bitrate_estimator_unittest_helper.cc
1 /*
2  *  Copyright (c) 2012 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 "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
11
12 #include <algorithm>
13 #include <utility>
14
15 namespace webrtc {
16
17 enum { kMtu = 1200 };
18
19 namespace testing {
20
21 void TestBitrateObserver::OnReceiveBitrateChanged(
22     const std::vector<unsigned int>& ssrcs,
23     unsigned int bitrate) {
24   latest_bitrate_ = bitrate;
25   updated_ = true;
26 }
27
28 RtpStream::RtpStream(int fps,
29                      int bitrate_bps,
30                      unsigned int ssrc,
31                      unsigned int frequency,
32                      uint32_t timestamp_offset,
33                      int64_t rtcp_receive_time)
34     : fps_(fps),
35       bitrate_bps_(bitrate_bps),
36       ssrc_(ssrc),
37       frequency_(frequency),
38       next_rtp_time_(0),
39       next_rtcp_time_(rtcp_receive_time),
40       rtp_timestamp_offset_(timestamp_offset),
41       kNtpFracPerMs(4.294967296E6) {
42   assert(fps_ > 0);
43 }
44
45 void RtpStream::set_rtp_timestamp_offset(uint32_t offset) {
46   rtp_timestamp_offset_ = offset;
47 }
48
49 // Generates a new frame for this stream. If called too soon after the
50 // previous frame, no frame will be generated. The frame is split into
51 // packets.
52 int64_t RtpStream::GenerateFrame(int64_t time_now_us, PacketList* packets) {
53   if (time_now_us < next_rtp_time_) {
54     return next_rtp_time_;
55   }
56   assert(packets != NULL);
57   int bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
58   int n_packets = std::max((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1);
59   int packet_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
60   assert(n_packets >= 0);
61   for (int i = 0; i < n_packets; ++i) {
62     RtpPacket* packet = new RtpPacket;
63     packet->send_time = time_now_us + kSendSideOffsetUs;
64     packet->size = packet_size;
65     packet->rtp_timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
66         ((frequency_ / 1000) * packet->send_time + 500) / 1000);
67     packet->ssrc = ssrc_;
68     packets->push_back(packet);
69   }
70   next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
71   return next_rtp_time_;
72 }
73
74 // The send-side time when the next frame can be generated.
75 double RtpStream::next_rtp_time() const {
76   return next_rtp_time_;
77 }
78
79 // Generates an RTCP packet.
80 RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) {
81   if (time_now_us < next_rtcp_time_) {
82     return NULL;
83   }
84   RtcpPacket* rtcp = new RtcpPacket;
85   int64_t send_time_us = time_now_us + kSendSideOffsetUs;
86   rtcp->timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
87       ((frequency_ / 1000) * send_time_us + 500) / 1000);
88   rtcp->ntp_secs = send_time_us / 1000000;
89   rtcp->ntp_frac = static_cast<int64_t>((send_time_us % 1000000) *
90       kNtpFracPerMs);
91   rtcp->ssrc = ssrc_;
92   next_rtcp_time_ = time_now_us + kRtcpIntervalUs;
93   return rtcp;
94 }
95
96 void RtpStream::set_bitrate_bps(int bitrate_bps) {
97   ASSERT_GE(bitrate_bps, 0);
98   bitrate_bps_ = bitrate_bps;
99 }
100
101 int RtpStream::bitrate_bps() const {
102   return bitrate_bps_;
103 }
104
105 unsigned int RtpStream::ssrc() const {
106   return ssrc_;
107 }
108
109 bool RtpStream::Compare(const std::pair<unsigned int, RtpStream*>& left,
110                         const std::pair<unsigned int, RtpStream*>& right) {
111   return left.second->next_rtp_time_ < right.second->next_rtp_time_;
112 }
113
114 StreamGenerator::StreamGenerator(int capacity, double time_now)
115     : capacity_(capacity),
116       prev_arrival_time_us_(time_now) {}
117
118 StreamGenerator::~StreamGenerator() {
119   for (StreamMap::iterator it = streams_.begin(); it != streams_.end();
120       ++it) {
121     delete it->second;
122   }
123   streams_.clear();
124 }
125
126 // Add a new stream.
127 void StreamGenerator::AddStream(RtpStream* stream) {
128   streams_[stream->ssrc()] = stream;
129 }
130
131 // Set the link capacity.
132 void StreamGenerator::set_capacity_bps(int capacity_bps) {
133   ASSERT_GT(capacity_bps, 0);
134   capacity_ = capacity_bps;
135 }
136
137 // Divides |bitrate_bps| among all streams. The allocated bitrate per stream
138 // is decided by the current allocation ratios.
139 void StreamGenerator::SetBitrateBps(int bitrate_bps) {
140   ASSERT_GE(streams_.size(), 0u);
141   int total_bitrate_before = 0;
142   for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) {
143     total_bitrate_before += it->second->bitrate_bps();
144   }
145   int64_t bitrate_before = 0;
146   int total_bitrate_after = 0;
147   for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) {
148     bitrate_before += it->second->bitrate_bps();
149     int64_t bitrate_after = (bitrate_before * bitrate_bps +
150         total_bitrate_before / 2) / total_bitrate_before;
151     it->second->set_bitrate_bps(bitrate_after - total_bitrate_after);
152     total_bitrate_after += it->second->bitrate_bps();
153   }
154   ASSERT_EQ(bitrate_before, total_bitrate_before);
155   EXPECT_EQ(total_bitrate_after, bitrate_bps);
156 }
157
158 // Set the RTP timestamp offset for the stream identified by |ssrc|.
159 void StreamGenerator::set_rtp_timestamp_offset(unsigned int ssrc,
160                                                uint32_t offset) {
161   streams_[ssrc]->set_rtp_timestamp_offset(offset);
162 }
163
164 // TODO(holmer): Break out the channel simulation part from this class to make
165 // it possible to simulate different types of channels.
166 int64_t StreamGenerator::GenerateFrame(RtpStream::PacketList* packets,
167                                        int64_t time_now_us) {
168   assert(packets != NULL);
169   assert(packets->empty());
170   assert(capacity_ > 0);
171   StreamMap::iterator it = std::min_element(streams_.begin(), streams_.end(),
172                                             RtpStream::Compare);
173   (*it).second->GenerateFrame(time_now_us, packets);
174   int i = 0;
175   for (RtpStream::PacketList::iterator packet_it = packets->begin();
176       packet_it != packets->end(); ++packet_it) {
177     int capacity_bpus = capacity_ / 1000;
178     int64_t required_network_time_us =
179         (8 * 1000 * (*packet_it)->size + capacity_bpus / 2) / capacity_bpus;
180     prev_arrival_time_us_ = std::max(time_now_us + required_network_time_us,
181         prev_arrival_time_us_ + required_network_time_us);
182     (*packet_it)->arrival_time = prev_arrival_time_us_;
183     ++i;
184   }
185   it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
186   return (*it).second->next_rtp_time();
187 }
188 }  // namespace testing
189
190 RemoteBitrateEstimatorTest::RemoteBitrateEstimatorTest()
191     : clock_(0),
192       bitrate_observer_(new testing::TestBitrateObserver),
193       stream_generator_(new testing::StreamGenerator(
194           1e6,  // Capacity.
195           clock_.TimeInMicroseconds())) {}
196
197 RemoteBitrateEstimatorTest::~RemoteBitrateEstimatorTest() {}
198
199 void RemoteBitrateEstimatorTest::AddDefaultStream() {
200   stream_generator_->AddStream(new testing::RtpStream(
201     30,          // Frames per second.
202     3e5,         // Bitrate.
203     1,           // SSRC.
204     90000,       // RTP frequency.
205     0xFFFFF000,  // Timestamp offset.
206     0));         // RTCP receive time.
207 }
208
209 uint32_t RemoteBitrateEstimatorTest::AbsSendTime(int64_t t, int64_t denom) {
210   return (((t << 18) + (denom >> 1)) / denom) & 0x00fffffful;
211 }
212
213 uint32_t RemoteBitrateEstimatorTest::AddAbsSendTime(uint32_t t1, uint32_t t2) {
214   return (t1 + t2) & 0x00fffffful;
215 }
216
217 const unsigned int RemoteBitrateEstimatorTest::kDefaultSsrc = 1;
218
219 void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
220                                                 uint32_t payload_size,
221                                                 int64_t arrival_time,
222                                                 uint32_t rtp_timestamp,
223                                                 uint32_t absolute_send_time) {
224   RTPHeader header;
225   memset(&header, 0, sizeof(header));
226   header.ssrc = ssrc;
227   header.timestamp = rtp_timestamp;
228   header.extension.absoluteSendTime = absolute_send_time;
229   bitrate_estimator_->IncomingPacket(arrival_time + kArrivalTimeClockOffsetMs,
230       payload_size, header);
231 }
232
233 // Generates a frame of packets belonging to a stream at a given bitrate and
234 // with a given ssrc. The stream is pushed through a very simple simulated
235 // network, and is then given to the receive-side bandwidth estimator.
236 // Returns true if an over-use was seen, false otherwise.
237 // The StreamGenerator::updated() should be used to check for any changes in
238 // target bitrate after the call to this function.
239 bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(unsigned int ssrc,
240     unsigned int bitrate_bps) {
241   stream_generator_->SetBitrateBps(bitrate_bps);
242   testing::RtpStream::PacketList packets;
243   int64_t next_time_us = stream_generator_->GenerateFrame(
244       &packets, clock_.TimeInMicroseconds());
245   bool overuse = false;
246   while (!packets.empty()) {
247     testing::RtpStream::RtpPacket* packet = packets.front();
248     bitrate_observer_->Reset();
249     // The simulated clock should match the time of packet->arrival_time
250     // since both are used in IncomingPacket().
251     clock_.AdvanceTimeMicroseconds(packet->arrival_time -
252                                    clock_.TimeInMicroseconds());
253     IncomingPacket(packet->ssrc,
254                    packet->size,
255                    (packet->arrival_time + 500) / 1000,
256                    packet->rtp_timestamp,
257                    AbsSendTime(packet->send_time, 1000000));
258     if (bitrate_observer_->updated()) {
259       // Verify that new estimates only are triggered by an overuse and a
260       // rate decrease.
261       overuse = true;
262       EXPECT_LE(bitrate_observer_->latest_bitrate(), bitrate_bps);
263     }
264     delete packet;
265     packets.pop_front();
266   }
267   bitrate_estimator_->Process();
268   clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
269   return overuse;
270 }
271
272 // Run the bandwidth estimator with a stream of |number_of_frames| frames, or
273 // until it reaches |target_bitrate|.
274 // Can for instance be used to run the estimator for some time to get it
275 // into a steady state.
276 unsigned int RemoteBitrateEstimatorTest::SteadyStateRun(
277     unsigned int ssrc,
278     int max_number_of_frames,
279     unsigned int start_bitrate,
280     unsigned int min_bitrate,
281     unsigned int max_bitrate,
282     unsigned int target_bitrate) {
283   unsigned int bitrate_bps = start_bitrate;
284   bool bitrate_update_seen = false;
285   // Produce |number_of_frames| frames and give them to the estimator.
286   for (int i = 0; i < max_number_of_frames; ++i) {
287     bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
288     if (overuse) {
289       EXPECT_LT(bitrate_observer_->latest_bitrate(), max_bitrate);
290       EXPECT_GT(bitrate_observer_->latest_bitrate(), min_bitrate);
291       bitrate_bps = bitrate_observer_->latest_bitrate();
292       bitrate_update_seen = true;
293     } else if (bitrate_observer_->updated()) {
294       bitrate_bps = bitrate_observer_->latest_bitrate();
295       bitrate_observer_->Reset();
296     }
297     if (bitrate_update_seen && bitrate_bps > target_bitrate) {
298       break;
299     }
300   }
301   EXPECT_TRUE(bitrate_update_seen);
302   return bitrate_bps;
303 }
304
305 void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper(
306     unsigned int expected_converge_bitrate) {
307   const int kFramerate = 50;  // 50 fps to avoid rounding errors.
308   const int kFrameIntervalMs = 1000 / kFramerate;
309   const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate);
310   unsigned int bitrate_bps = 0;
311   uint32_t timestamp = 0;
312   uint32_t absolute_send_time = 0;
313   std::vector<unsigned int> ssrcs;
314   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
315   EXPECT_EQ(0u, ssrcs.size());
316   clock_.AdvanceTimeMilliseconds(1000);
317   bitrate_estimator_->Process();
318   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
319   EXPECT_FALSE(bitrate_observer_->updated());
320   bitrate_observer_->Reset();
321   clock_.AdvanceTimeMilliseconds(1000);
322   // Inserting a packet. Still no valid estimate. We need to wait 1 second.
323   IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
324                  absolute_send_time);
325   bitrate_estimator_->Process();
326   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
327   EXPECT_EQ(0u, ssrcs.size());
328   EXPECT_FALSE(bitrate_observer_->updated());
329   bitrate_observer_->Reset();
330   // Inserting packets for one second to get a valid estimate.
331   for (int i = 0; i < kFramerate; ++i) {
332     IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
333                    absolute_send_time);
334     clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
335     timestamp += 90 * kFrameIntervalMs;
336     absolute_send_time = AddAbsSendTime(absolute_send_time,
337                                         kFrameIntervalAbsSendTime);
338   }
339   bitrate_estimator_->Process();
340   EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
341   ASSERT_EQ(1u, ssrcs.size());
342   EXPECT_EQ(kDefaultSsrc, ssrcs.front());
343   EXPECT_EQ(expected_converge_bitrate, bitrate_bps);
344   EXPECT_TRUE(bitrate_observer_->updated());
345   bitrate_observer_->Reset();
346   EXPECT_EQ(bitrate_observer_->latest_bitrate(), bitrate_bps);
347   bitrate_estimator_->RemoveStream(kDefaultSsrc);
348   EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
349   ASSERT_EQ(0u, ssrcs.size());
350   EXPECT_EQ(0u, bitrate_bps);
351 }
352
353 void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
354     uint32_t expected_bitrate_bps) {
355   const int kFramerate = 50;  // 50 fps to avoid rounding errors.
356   const int kFrameIntervalMs = 1000 / kFramerate;
357   const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate);
358   uint32_t timestamp = 0;
359   uint32_t absolute_send_time = 0;
360   IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
361                  absolute_send_time);
362   bitrate_estimator_->Process();
363   EXPECT_FALSE(bitrate_observer_->updated());  // No valid estimate.
364   // Inserting packets for one second to get a valid estimate.
365   for (int i = 0; i < kFramerate; ++i) {
366     IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
367                    absolute_send_time);
368     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
369     timestamp += 90 * kFrameIntervalMs;
370     absolute_send_time = AddAbsSendTime(absolute_send_time,
371                                         kFrameIntervalAbsSendTime);
372   }
373   bitrate_estimator_->Process();
374   EXPECT_TRUE(bitrate_observer_->updated());
375   EXPECT_EQ(expected_bitrate_bps, bitrate_observer_->latest_bitrate());
376   for (int i = 0; i < 10; ++i) {
377     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
378     timestamp += 2 * 90 * kFrameIntervalMs;
379     absolute_send_time = AddAbsSendTime(absolute_send_time,
380                                         2 * kFrameIntervalAbsSendTime);
381     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
382                    absolute_send_time);
383     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
384                    timestamp - 90 * kFrameIntervalMs,
385                    AddAbsSendTime(absolute_send_time,
386                                   -int(kFrameIntervalAbsSendTime)));
387   }
388   bitrate_estimator_->Process();
389   EXPECT_TRUE(bitrate_observer_->updated());
390   EXPECT_EQ(expected_bitrate_bps, bitrate_observer_->latest_bitrate());
391 }
392
393 // Make sure we initially increase the bitrate as expected.
394 void RemoteBitrateEstimatorTest::RateIncreaseRtpTimestampsTestHelper() {
395   // This threshold corresponds approximately to increasing linearly with
396   // bitrate(i) = 1.04 * bitrate(i-1) + 1000
397   // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
398   const int kExpectedIterations = 1621;
399   unsigned int bitrate_bps = 30000;
400   int iterations = 0;
401   AddDefaultStream();
402   // Feed the estimator with a stream of packets and verify that it reaches
403   // 500 kbps at the expected time.
404   while (bitrate_bps < 5e5) {
405     bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
406     if (overuse) {
407       EXPECT_GT(bitrate_observer_->latest_bitrate(), bitrate_bps);
408       bitrate_bps = bitrate_observer_->latest_bitrate();
409       bitrate_observer_->Reset();
410     } else if (bitrate_observer_->updated()) {
411       bitrate_bps = bitrate_observer_->latest_bitrate();
412       bitrate_observer_->Reset();
413     }
414     ++iterations;
415     ASSERT_LE(iterations, kExpectedIterations);
416   }
417   ASSERT_EQ(kExpectedIterations, iterations);
418 }
419
420 void RemoteBitrateEstimatorTest::CapacityDropTestHelper(
421     int number_of_streams,
422     bool wrap_time_stamp,
423     unsigned int expected_converge_bitrate,
424     unsigned int expected_bitrate_drop_delta) {
425   const int kFramerate = 30;
426   const int kStartBitrate = 900e3;
427   const int kMinExpectedBitrate = 800e3;
428   const int kMaxExpectedBitrate = 1100e3;
429   const unsigned int kInitialCapacityBps = 1000e3;
430   const unsigned int kReducedCapacityBps = 500e3;
431
432   int steady_state_time = 0;
433   int expected_overuse_start_time = 0;
434   if (number_of_streams <= 1) {
435     steady_state_time = 10;
436     expected_overuse_start_time = 10000;
437     AddDefaultStream();
438   } else {
439     steady_state_time = 8 * number_of_streams;
440     expected_overuse_start_time = 8000;
441     int bitrate_sum = 0;
442     int kBitrateDenom = number_of_streams * (number_of_streams - 1);
443     for (int i = 0; i < number_of_streams; i++) {
444       // First stream gets half available bitrate, while the rest share the
445       // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
446       int bitrate = kStartBitrate / 2;
447       if (i > 0) {
448         bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
449       }
450       stream_generator_->AddStream(new testing::RtpStream(
451           kFramerate,                     // Frames per second.
452           bitrate,                        // Bitrate.
453           kDefaultSsrc + i,               // SSRC.
454           90000,                          // RTP frequency.
455           0xFFFFF000 ^ (~0 << (32 - i)),  // Timestamp offset.
456           0));                            // RTCP receive time.
457       bitrate_sum += bitrate;
458     }
459     ASSERT_EQ(bitrate_sum, kStartBitrate);
460   }
461   if (wrap_time_stamp) {
462     stream_generator_->set_rtp_timestamp_offset(kDefaultSsrc,
463         std::numeric_limits<uint32_t>::max() - steady_state_time * 90000);
464   }
465
466   // Run in steady state to make the estimator converge.
467   stream_generator_->set_capacity_bps(kInitialCapacityBps);
468   unsigned int bitrate_bps = SteadyStateRun(kDefaultSsrc,
469                                             steady_state_time * kFramerate,
470                                             kStartBitrate,
471                                             kMinExpectedBitrate,
472                                             kMaxExpectedBitrate,
473                                             kInitialCapacityBps);
474   EXPECT_EQ(expected_converge_bitrate, bitrate_bps);
475   bitrate_observer_->Reset();
476
477   // Reduce the capacity and verify the decrease time.
478   stream_generator_->set_capacity_bps(kReducedCapacityBps);
479   int64_t overuse_start_time = clock_.TimeInMilliseconds();
480   EXPECT_EQ(expected_overuse_start_time, overuse_start_time);
481   int64_t bitrate_drop_time = -1;
482   for (int i = 0; i < 100 * number_of_streams; ++i) {
483     GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
484     // Check for either increase or decrease.
485     if (bitrate_observer_->updated()) {
486       if (bitrate_drop_time == -1 &&
487           bitrate_observer_->latest_bitrate() <= kReducedCapacityBps) {
488         bitrate_drop_time = clock_.TimeInMilliseconds();
489       }
490       bitrate_bps = bitrate_observer_->latest_bitrate();
491       bitrate_observer_->Reset();
492     }
493   }
494
495   EXPECT_EQ(expected_bitrate_drop_delta,
496             bitrate_drop_time - overuse_start_time);
497
498   // Remove stream one by one.
499   unsigned int latest_bps = 0;
500   std::vector<unsigned int> ssrcs;
501   for (int i = 0; i < number_of_streams; i++) {
502     EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &latest_bps));
503     EXPECT_EQ(number_of_streams - i, static_cast<int>(ssrcs.size()));
504     EXPECT_EQ(bitrate_bps, latest_bps);
505     for (int j = i; j < number_of_streams; j++) {
506       EXPECT_EQ(kDefaultSsrc + j, ssrcs[j - i]);
507     }
508     bitrate_estimator_->RemoveStream(kDefaultSsrc + i);
509   }
510   EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &latest_bps));
511   EXPECT_EQ(0u, ssrcs.size());
512   EXPECT_EQ(0u, latest_bps);
513 }
514 }  // namespace webrtc