stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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 { |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 16 | |
| 17 | enum { kMtu = 1200 }; |
| 18 | |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 19 | namespace testing { |
| 20 | |
pbos@webrtc.org | 988a5b3 | 2013-07-31 15:16:52 +0000 | [diff] [blame] | 21 | void TestBitrateObserver::OnReceiveBitrateChanged( |
| 22 | const std::vector<unsigned int>& ssrcs, |
| 23 | unsigned int bitrate) { |
| 24 | latest_bitrate_ = bitrate; |
| 25 | updated_ = true; |
| 26 | } |
| 27 | |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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. |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 52 | int64_t RtpStream::GenerateFrame(int64_t time_now_us, PacketList* packets) { |
| 53 | if (time_now_us < next_rtp_time_) { |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 54 | return next_rtp_time_; |
| 55 | } |
| 56 | assert(packets != NULL); |
| 57 | int bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_; |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 58 | int n_packets = std::max((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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; |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 63 | packet->send_time = time_now_us + kSendSideOffsetUs; |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 64 | packet->size = packet_size; |
| 65 | packet->rtp_timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>( |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 66 | ((frequency_ / 1000) * packet->send_time + 500) / 1000); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 67 | packet->ssrc = ssrc_; |
| 68 | packets->push_back(packet); |
| 69 | } |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 70 | next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_; |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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. |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 80 | RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) { |
| 81 | if (time_now_us < next_rtcp_time_) { |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 82 | return NULL; |
| 83 | } |
| 84 | RtcpPacket* rtcp = new RtcpPacket; |
solenberg@webrtc.org | de0b5fa | 2013-05-22 20:53:42 +0000 | [diff] [blame] | 85 | int64_t send_time_us = time_now_us + kSendSideOffsetUs; |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 86 | rtcp->timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>( |
solenberg@webrtc.org | de0b5fa | 2013-05-22 20:53:42 +0000 | [diff] [blame] | 87 | ((frequency_ / 1000) * send_time_us + 500) / 1000); |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 88 | rtcp->ntp_secs = send_time_us / 1000000; |
solenberg@webrtc.org | de0b5fa | 2013-05-22 20:53:42 +0000 | [diff] [blame] | 89 | rtcp->ntp_frac = static_cast<int64_t>((send_time_us % 1000000) * |
| 90 | kNtpFracPerMs); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 91 | rtcp->ssrc = ssrc_; |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 92 | next_rtcp_time_ = time_now_us + kRtcpIntervalUs; |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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), |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 116 | prev_arrival_time_us_(time_now) {} |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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 |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 138 | // is decided by the current allocation ratios. |
| 139 | void StreamGenerator::SetBitrateBps(int bitrate_bps) { |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 140 | ASSERT_GE(streams_.size(), 0u); |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 141 | int total_bitrate_before = 0; |
| 142 | for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) { |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 143 | total_bitrate_before += it->second->bitrate_bps(); |
| 144 | } |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 145 | int64_t bitrate_before = 0; |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 146 | int total_bitrate_after = 0; |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 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); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 152 | total_bitrate_after += it->second->bitrate_bps(); |
| 153 | } |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 154 | ASSERT_EQ(bitrate_before, total_bitrate_before); |
| 155 | EXPECT_EQ(total_bitrate_after, bitrate_bps); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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. |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 166 | int64_t StreamGenerator::GenerateFrame(RtpStream::PacketList* packets, |
| 167 | int64_t time_now_us) { |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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); |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 173 | (*it).second->GenerateFrame(time_now_us, packets); |
| 174 | int i = 0; |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 175 | for (RtpStream::PacketList::iterator packet_it = packets->begin(); |
| 176 | packet_it != packets->end(); ++packet_it) { |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 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; |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 184 | } |
| 185 | it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare); |
| 186 | return (*it).second->next_rtp_time(); |
| 187 | } |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 188 | } // namespace testing |
| 189 | |
| 190 | RemoteBitrateEstimatorTest::RemoteBitrateEstimatorTest() |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 191 | : clock_(0), |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 192 | bitrate_observer_(new testing::TestBitrateObserver), |
| 193 | stream_generator_(new testing::StreamGenerator( |
| 194 | 1e6, // Capacity. |
| 195 | clock_.TimeInMicroseconds())) {} |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 196 | |
pbos@webrtc.org | 988a5b3 | 2013-07-31 15:16:52 +0000 | [diff] [blame] | 197 | RemoteBitrateEstimatorTest::~RemoteBitrateEstimatorTest() {} |
| 198 | |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 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 | |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 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 | |
pbos@webrtc.org | 98b2011 | 2013-05-07 12:36:21 +0000 | [diff] [blame] | 217 | const unsigned int RemoteBitrateEstimatorTest::kDefaultSsrc = 1; |
| 218 | |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 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) { |
stefan@webrtc.org | d8ecee5 | 2013-06-04 12:15:40 +0000 | [diff] [blame] | 224 | RTPHeader header; |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 225 | memset(&header, 0, sizeof(header)); |
stefan@webrtc.org | d8ecee5 | 2013-06-04 12:15:40 +0000 | [diff] [blame] | 226 | header.ssrc = ssrc; |
| 227 | header.timestamp = rtp_timestamp; |
| 228 | header.extension.absoluteSendTime = absolute_send_time; |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 229 | bitrate_estimator_->IncomingPacket(arrival_time, payload_size, header); |
| 230 | } |
| 231 | |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 232 | // Generates a frame of packets belonging to a stream at a given bitrate and |
| 233 | // with a given ssrc. The stream is pushed through a very simple simulated |
| 234 | // network, and is then given to the receive-side bandwidth estimator. |
| 235 | // Returns true if an over-use was seen, false otherwise. |
| 236 | // The StreamGenerator::updated() should be used to check for any changes in |
| 237 | // target bitrate after the call to this function. |
| 238 | bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(unsigned int ssrc, |
| 239 | unsigned int bitrate_bps) { |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 240 | stream_generator_->SetBitrateBps(bitrate_bps); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 241 | testing::RtpStream::PacketList packets; |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 242 | int64_t next_time_us = stream_generator_->GenerateFrame( |
| 243 | &packets, clock_.TimeInMicroseconds()); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 244 | bool overuse = false; |
| 245 | while (!packets.empty()) { |
| 246 | testing::RtpStream::RtpPacket* packet = packets.front(); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 247 | bitrate_observer_->Reset(); |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 248 | IncomingPacket(packet->ssrc, |
| 249 | packet->size, |
| 250 | (packet->arrival_time + 500) / 1000, |
| 251 | packet->rtp_timestamp, |
| 252 | AbsSendTime(packet->send_time, 1000000)); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 253 | if (bitrate_observer_->updated()) { |
| 254 | // Verify that new estimates only are triggered by an overuse and a |
| 255 | // rate decrease. |
| 256 | overuse = true; |
| 257 | EXPECT_LE(bitrate_observer_->latest_bitrate(), bitrate_bps); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 258 | } |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 259 | clock_.AdvanceTimeMicroseconds(packet->arrival_time - |
| 260 | clock_.TimeInMicroseconds()); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 261 | delete packet; |
| 262 | packets.pop_front(); |
| 263 | } |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 264 | bitrate_estimator_->Process(); |
| 265 | clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds()); |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 266 | return overuse; |
| 267 | } |
| 268 | |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 269 | // Run the bandwidth estimator with a stream of |number_of_frames| frames, or |
| 270 | // until it reaches |target_bitrate|. |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 271 | // Can for instance be used to run the estimator for some time to get it |
| 272 | // into a steady state. |
| 273 | unsigned int RemoteBitrateEstimatorTest::SteadyStateRun( |
| 274 | unsigned int ssrc, |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 275 | int max_number_of_frames, |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 276 | unsigned int start_bitrate, |
| 277 | unsigned int min_bitrate, |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 278 | unsigned int max_bitrate, |
| 279 | unsigned int target_bitrate) { |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 280 | unsigned int bitrate_bps = start_bitrate; |
| 281 | bool bitrate_update_seen = false; |
| 282 | // Produce |number_of_frames| frames and give them to the estimator. |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 283 | for (int i = 0; i < max_number_of_frames; ++i) { |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 284 | bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps); |
| 285 | if (overuse) { |
| 286 | EXPECT_LT(bitrate_observer_->latest_bitrate(), max_bitrate); |
| 287 | EXPECT_GT(bitrate_observer_->latest_bitrate(), min_bitrate); |
| 288 | bitrate_bps = bitrate_observer_->latest_bitrate(); |
| 289 | bitrate_update_seen = true; |
| 290 | } else if (bitrate_observer_->updated()) { |
| 291 | bitrate_bps = bitrate_observer_->latest_bitrate(); |
| 292 | bitrate_observer_->Reset(); |
| 293 | } |
stefan@webrtc.org | 2a5dbce | 2013-02-01 14:33:42 +0000 | [diff] [blame] | 294 | if (bitrate_update_seen && bitrate_bps > target_bitrate) { |
| 295 | break; |
| 296 | } |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 297 | } |
| 298 | EXPECT_TRUE(bitrate_update_seen); |
| 299 | return bitrate_bps; |
| 300 | } |
solenberg@webrtc.org | b7716d8 | 2013-05-22 19:04:19 +0000 | [diff] [blame] | 301 | |
| 302 | void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper( |
| 303 | unsigned int expected_converge_bitrate) { |
| 304 | const int kFramerate = 50; // 50 fps to avoid rounding errors. |
| 305 | const int kFrameIntervalMs = 1000 / kFramerate; |
| 306 | const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate); |
| 307 | unsigned int bitrate_bps = 0; |
| 308 | uint32_t timestamp = 0; |
| 309 | uint32_t absolute_send_time = 0; |
| 310 | std::vector<unsigned int> ssrcs; |
| 311 | EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); |
| 312 | EXPECT_EQ(0u, ssrcs.size()); |
| 313 | clock_.AdvanceTimeMilliseconds(1000); |
| 314 | bitrate_estimator_->Process(); |
| 315 | EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); |
| 316 | EXPECT_FALSE(bitrate_observer_->updated()); |
| 317 | bitrate_observer_->Reset(); |
| 318 | clock_.AdvanceTimeMilliseconds(1000); |
| 319 | // Inserting a packet. Still no valid estimate. We need to wait 1 second. |
| 320 | IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, |
| 321 | absolute_send_time); |
| 322 | bitrate_estimator_->Process(); |
| 323 | EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); |
| 324 | EXPECT_EQ(0u, ssrcs.size()); |
| 325 | EXPECT_FALSE(bitrate_observer_->updated()); |
| 326 | bitrate_observer_->Reset(); |
| 327 | // Inserting packets for one second to get a valid estimate. |
| 328 | for (int i = 0; i < kFramerate; ++i) { |
| 329 | IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, |
| 330 | absolute_send_time); |
| 331 | clock_.AdvanceTimeMilliseconds(1000 / kFramerate); |
| 332 | timestamp += 90 * kFrameIntervalMs; |
| 333 | absolute_send_time = AddAbsSendTime(absolute_send_time, |
| 334 | kFrameIntervalAbsSendTime); |
| 335 | } |
| 336 | bitrate_estimator_->Process(); |
| 337 | EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps)); |
| 338 | ASSERT_EQ(1u, ssrcs.size()); |
| 339 | EXPECT_EQ(kDefaultSsrc, ssrcs.front()); |
| 340 | EXPECT_EQ(expected_converge_bitrate, bitrate_bps); |
| 341 | EXPECT_TRUE(bitrate_observer_->updated()); |
| 342 | bitrate_observer_->Reset(); |
| 343 | EXPECT_EQ(bitrate_observer_->latest_bitrate(), bitrate_bps); |
| 344 | } |
| 345 | |
| 346 | void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper() { |
| 347 | const int kFramerate = 50; // 50 fps to avoid rounding errors. |
| 348 | const int kFrameIntervalMs = 1000 / kFramerate; |
| 349 | const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate); |
| 350 | uint32_t timestamp = 0; |
| 351 | uint32_t absolute_send_time = 0; |
| 352 | IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp, |
| 353 | absolute_send_time); |
| 354 | bitrate_estimator_->Process(); |
| 355 | EXPECT_FALSE(bitrate_observer_->updated()); // No valid estimate. |
| 356 | // Inserting packets for one second to get a valid estimate. |
| 357 | for (int i = 0; i < kFramerate; ++i) { |
| 358 | IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, |
| 359 | absolute_send_time); |
| 360 | clock_.AdvanceTimeMilliseconds(kFrameIntervalMs); |
| 361 | timestamp += 90 * kFrameIntervalMs; |
| 362 | absolute_send_time = AddAbsSendTime(absolute_send_time, |
| 363 | kFrameIntervalAbsSendTime); |
| 364 | } |
| 365 | bitrate_estimator_->Process(); |
| 366 | EXPECT_TRUE(bitrate_observer_->updated()); |
| 367 | EXPECT_EQ(498136u, bitrate_observer_->latest_bitrate()); |
| 368 | for (int i = 0; i < 10; ++i) { |
| 369 | clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs); |
| 370 | timestamp += 2 * 90 * kFrameIntervalMs; |
| 371 | absolute_send_time = AddAbsSendTime(absolute_send_time, |
| 372 | 2 * kFrameIntervalAbsSendTime); |
| 373 | IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp, |
| 374 | absolute_send_time); |
| 375 | IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), |
| 376 | timestamp - 90 * kFrameIntervalMs, |
| 377 | AddAbsSendTime(absolute_send_time, |
| 378 | -int(kFrameIntervalAbsSendTime))); |
| 379 | } |
| 380 | bitrate_estimator_->Process(); |
| 381 | EXPECT_TRUE(bitrate_observer_->updated()); |
| 382 | EXPECT_EQ(498136u, bitrate_observer_->latest_bitrate()); |
| 383 | } |
| 384 | |
| 385 | // Make sure we initially increase the bitrate as expected. |
| 386 | void RemoteBitrateEstimatorTest::RateIncreaseRtpTimestampsTestHelper() { |
| 387 | // This threshold corresponds approximately to increasing linearly with |
| 388 | // bitrate(i) = 1.04 * bitrate(i-1) + 1000 |
| 389 | // until bitrate(i) > 500000, with bitrate(1) ~= 30000. |
| 390 | const int kExpectedIterations = 1621; |
| 391 | unsigned int bitrate_bps = 30000; |
| 392 | int iterations = 0; |
| 393 | AddDefaultStream(); |
| 394 | // Feed the estimator with a stream of packets and verify that it reaches |
| 395 | // 500 kbps at the expected time. |
| 396 | while (bitrate_bps < 5e5) { |
| 397 | bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps); |
| 398 | if (overuse) { |
| 399 | EXPECT_GT(bitrate_observer_->latest_bitrate(), bitrate_bps); |
| 400 | bitrate_bps = bitrate_observer_->latest_bitrate(); |
| 401 | bitrate_observer_->Reset(); |
| 402 | } else if (bitrate_observer_->updated()) { |
| 403 | bitrate_bps = bitrate_observer_->latest_bitrate(); |
| 404 | bitrate_observer_->Reset(); |
| 405 | } |
| 406 | ++iterations; |
| 407 | ASSERT_LE(iterations, kExpectedIterations); |
| 408 | } |
| 409 | ASSERT_EQ(kExpectedIterations, iterations); |
| 410 | } |
| 411 | |
| 412 | void RemoteBitrateEstimatorTest::CapacityDropTestHelper( |
| 413 | int number_of_streams, |
| 414 | bool wrap_time_stamp, |
| 415 | unsigned int expected_converge_bitrate, |
| 416 | unsigned int expected_bitrate_drop_delta) { |
| 417 | const int kFramerate = 30; |
| 418 | const int kStartBitrate = 900e3; |
| 419 | const int kMinExpectedBitrate = 800e3; |
| 420 | const int kMaxExpectedBitrate = 1100e3; |
| 421 | const unsigned int kInitialCapacityBps = 1000e3; |
| 422 | const unsigned int kReducedCapacityBps = 500e3; |
| 423 | |
| 424 | int steady_state_time = 0; |
| 425 | int expected_overuse_start_time = 0; |
| 426 | if (number_of_streams <= 1) { |
| 427 | steady_state_time = 10; |
| 428 | expected_overuse_start_time = 10000; |
| 429 | AddDefaultStream(); |
| 430 | } else { |
| 431 | steady_state_time = 8 * number_of_streams; |
| 432 | expected_overuse_start_time = 8000; |
| 433 | int bitrate_sum = 0; |
| 434 | int kBitrateDenom = number_of_streams * (number_of_streams - 1); |
| 435 | for (int i = 0; i < number_of_streams; i++) { |
| 436 | // First stream gets half available bitrate, while the rest share the |
| 437 | // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up) |
| 438 | int bitrate = kStartBitrate / 2; |
| 439 | if (i > 0) { |
| 440 | bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom; |
| 441 | } |
| 442 | stream_generator_->AddStream(new testing::RtpStream( |
| 443 | kFramerate, // Frames per second. |
| 444 | bitrate, // Bitrate. |
| 445 | kDefaultSsrc + i, // SSRC. |
| 446 | 90000, // RTP frequency. |
| 447 | 0xFFFFF000 ^ (~0 << (32 - i)), // Timestamp offset. |
| 448 | 0)); // RTCP receive time. |
| 449 | bitrate_sum += bitrate; |
| 450 | } |
| 451 | ASSERT_EQ(bitrate_sum, kStartBitrate); |
| 452 | } |
| 453 | if (wrap_time_stamp) { |
| 454 | stream_generator_->set_rtp_timestamp_offset(kDefaultSsrc, |
| 455 | std::numeric_limits<uint32_t>::max() - steady_state_time * 90000); |
| 456 | } |
| 457 | |
| 458 | // Run in steady state to make the estimator converge. |
| 459 | stream_generator_->set_capacity_bps(kInitialCapacityBps); |
| 460 | unsigned int bitrate_bps = SteadyStateRun(kDefaultSsrc, |
| 461 | steady_state_time * kFramerate, |
| 462 | kStartBitrate, |
| 463 | kMinExpectedBitrate, |
| 464 | kMaxExpectedBitrate, |
| 465 | kInitialCapacityBps); |
| 466 | EXPECT_EQ(expected_converge_bitrate, bitrate_bps); |
| 467 | bitrate_observer_->Reset(); |
| 468 | |
| 469 | // Reduce the capacity and verify the decrease time. |
| 470 | stream_generator_->set_capacity_bps(kReducedCapacityBps); |
| 471 | int64_t overuse_start_time = clock_.TimeInMilliseconds(); |
| 472 | EXPECT_EQ(expected_overuse_start_time, overuse_start_time); |
| 473 | int64_t bitrate_drop_time = -1; |
| 474 | for (int i = 0; i < 100 * number_of_streams; ++i) { |
| 475 | GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps); |
| 476 | // Check for either increase or decrease. |
| 477 | if (bitrate_observer_->updated()) { |
| 478 | if (bitrate_drop_time == -1 && |
| 479 | bitrate_observer_->latest_bitrate() <= kReducedCapacityBps) { |
| 480 | bitrate_drop_time = clock_.TimeInMilliseconds(); |
| 481 | } |
| 482 | bitrate_bps = bitrate_observer_->latest_bitrate(); |
| 483 | bitrate_observer_->Reset(); |
| 484 | } |
| 485 | } |
| 486 | |
| 487 | EXPECT_EQ(expected_bitrate_drop_delta, |
| 488 | bitrate_drop_time - overuse_start_time); |
| 489 | } |
stefan@webrtc.org | 686a447 | 2012-11-07 18:35:30 +0000 | [diff] [blame] | 490 | } // namespace webrtc |