2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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.
10 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
21 void TestBitrateObserver::OnReceiveBitrateChanged(
22 const std::vector<unsigned int>& ssrcs,
23 unsigned int bitrate) {
24 latest_bitrate_ = bitrate;
28 RtpStream::RtpStream(int fps,
31 unsigned int frequency,
32 uint32_t timestamp_offset,
33 int64_t rtcp_receive_time)
35 bitrate_bps_(bitrate_bps),
37 frequency_(frequency),
39 next_rtcp_time_(rtcp_receive_time),
40 rtp_timestamp_offset_(timestamp_offset),
41 kNtpFracPerMs(4.294967296E6) {
45 void RtpStream::set_rtp_timestamp_offset(uint32_t offset) {
46 rtp_timestamp_offset_ = offset;
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
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_;
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);
68 packets->push_back(packet);
70 next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
71 return next_rtp_time_;
74 // The send-side time when the next frame can be generated.
75 double RtpStream::next_rtp_time() const {
76 return next_rtp_time_;
79 // Generates an RTCP packet.
80 RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) {
81 if (time_now_us < next_rtcp_time_) {
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) *
92 next_rtcp_time_ = time_now_us + kRtcpIntervalUs;
96 void RtpStream::set_bitrate_bps(int bitrate_bps) {
97 ASSERT_GE(bitrate_bps, 0);
98 bitrate_bps_ = bitrate_bps;
101 int RtpStream::bitrate_bps() const {
105 unsigned int RtpStream::ssrc() const {
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_;
114 StreamGenerator::StreamGenerator(int capacity, double time_now)
115 : capacity_(capacity),
116 prev_arrival_time_us_(time_now) {}
118 StreamGenerator::~StreamGenerator() {
119 for (StreamMap::iterator it = streams_.begin(); it != streams_.end();
127 void StreamGenerator::AddStream(RtpStream* stream) {
128 streams_[stream->ssrc()] = stream;
131 // Set the link capacity.
132 void StreamGenerator::set_capacity_bps(int capacity_bps) {
133 ASSERT_GT(capacity_bps, 0);
134 capacity_ = capacity_bps;
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();
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();
154 ASSERT_EQ(bitrate_before, total_bitrate_before);
155 EXPECT_EQ(total_bitrate_after, bitrate_bps);
158 // Set the RTP timestamp offset for the stream identified by |ssrc|.
159 void StreamGenerator::set_rtp_timestamp_offset(unsigned int ssrc,
161 streams_[ssrc]->set_rtp_timestamp_offset(offset);
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(),
173 (*it).second->GenerateFrame(time_now_us, packets);
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_;
185 it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
186 return (*it).second->next_rtp_time();
188 } // namespace testing
190 RemoteBitrateEstimatorTest::RemoteBitrateEstimatorTest()
192 bitrate_observer_(new testing::TestBitrateObserver),
193 stream_generator_(new testing::StreamGenerator(
195 clock_.TimeInMicroseconds())) {}
197 RemoteBitrateEstimatorTest::~RemoteBitrateEstimatorTest() {}
199 void RemoteBitrateEstimatorTest::AddDefaultStream() {
200 stream_generator_->AddStream(new testing::RtpStream(
201 30, // Frames per second.
204 90000, // RTP frequency.
205 0xFFFFF000, // Timestamp offset.
206 0)); // RTCP receive time.
209 uint32_t RemoteBitrateEstimatorTest::AbsSendTime(int64_t t, int64_t denom) {
210 return (((t << 18) + (denom >> 1)) / denom) & 0x00fffffful;
213 uint32_t RemoteBitrateEstimatorTest::AddAbsSendTime(uint32_t t1, uint32_t t2) {
214 return (t1 + t2) & 0x00fffffful;
217 const unsigned int RemoteBitrateEstimatorTest::kDefaultSsrc = 1;
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) {
225 memset(&header, 0, sizeof(header));
227 header.timestamp = rtp_timestamp;
228 header.extension.absoluteSendTime = absolute_send_time;
229 bitrate_estimator_->IncomingPacket(arrival_time + kArrivalTimeClockOffsetMs,
230 payload_size, header);
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,
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
262 EXPECT_LE(bitrate_observer_->latest_bitrate(), bitrate_bps);
267 bitrate_estimator_->Process();
268 clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
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(
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);
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();
297 if (bitrate_update_seen && bitrate_bps > target_bitrate) {
301 EXPECT_TRUE(bitrate_update_seen);
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,
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,
334 clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
335 timestamp += 90 * kFrameIntervalMs;
336 absolute_send_time = AddAbsSendTime(absolute_send_time,
337 kFrameIntervalAbsSendTime);
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);
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,
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,
368 clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
369 timestamp += 90 * kFrameIntervalMs;
370 absolute_send_time = AddAbsSendTime(absolute_send_time,
371 kFrameIntervalAbsSendTime);
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,
383 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(),
384 timestamp - 90 * kFrameIntervalMs,
385 AddAbsSendTime(absolute_send_time,
386 -int(kFrameIntervalAbsSendTime)));
388 bitrate_estimator_->Process();
389 EXPECT_TRUE(bitrate_observer_->updated());
390 EXPECT_EQ(expected_bitrate_bps, bitrate_observer_->latest_bitrate());
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;
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);
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();
415 ASSERT_LE(iterations, kExpectedIterations);
417 ASSERT_EQ(kExpectedIterations, iterations);
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;
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;
439 steady_state_time = 8 * number_of_streams;
440 expected_overuse_start_time = 8000;
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;
448 bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
450 stream_generator_->AddStream(new testing::RtpStream(
451 kFramerate, // Frames per second.
453 kDefaultSsrc + i, // SSRC.
454 90000, // RTP frequency.
455 0xFFFFF000 ^ (~0 << (32 - i)), // Timestamp offset.
456 0)); // RTCP receive time.
457 bitrate_sum += bitrate;
459 ASSERT_EQ(bitrate_sum, kStartBitrate);
461 if (wrap_time_stamp) {
462 stream_generator_->set_rtp_timestamp_offset(kDefaultSsrc,
463 std::numeric_limits<uint32_t>::max() - steady_state_time * 90000);
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,
473 kInitialCapacityBps);
474 EXPECT_EQ(expected_converge_bitrate, bitrate_bps);
475 bitrate_observer_->Reset();
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();
490 bitrate_bps = bitrate_observer_->latest_bitrate();
491 bitrate_observer_->Reset();
495 EXPECT_EQ(expected_bitrate_drop_delta,
496 bitrate_drop_time - overuse_start_time);
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]);
508 bitrate_estimator_->RemoveStream(kDefaultSsrc + i);
510 EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &latest_bps));
511 EXPECT_EQ(0u, ssrcs.size());
512 EXPECT_EQ(0u, latest_bps);
514 } // namespace webrtc