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Side by Side Diff: webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc

Issue 1392513002: Disable pacer disabling. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Remove RTP FIR + test refactoring Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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202 // happens if the prober wants to probe, but there's no packet to send. 202 // happens if the prober wants to probe, but there's no packet to send.
203 if (time_until_next_event_ms == 0 && last_run_time_ms == 0) 203 if (time_until_next_event_ms == 0 && last_run_time_ms == 0)
204 time_until_next_event_ms = 1; 204 time_until_next_event_ms = 1;
205 last_run_time_ms = time_until_next_event_ms; 205 last_run_time_ms = time_until_next_event_ms;
206 206
207 Packets generated_packets; 207 Packets generated_packets;
208 source_->RunFor(time_until_next_event_ms, &generated_packets); 208 source_->RunFor(time_until_next_event_ms, &generated_packets);
209 if (!generated_packets.empty()) { 209 if (!generated_packets.empty()) {
210 for (Packet* packet : generated_packets) { 210 for (Packet* packet : generated_packets) {
211 MediaPacket* media_packet = static_cast<MediaPacket*>(packet); 211 MediaPacket* media_packet = static_cast<MediaPacket*>(packet);
212 pacer_.SendPacket( 212 pacer_.InsertPacket(
213 PacedSender::kNormalPriority, media_packet->header().ssrc, 213 PacedSender::kNormalPriority, media_packet->header().ssrc,
214 media_packet->header().sequenceNumber, media_packet->send_time_ms(), 214 media_packet->header().sequenceNumber, media_packet->send_time_ms(),
215 media_packet->payload_size(), false); 215 media_packet->payload_size(), false);
216 pacer_queue_.push_back(packet); 216 pacer_queue_.push_back(packet);
217 assert(pacer_queue_.size() < 10000); 217 assert(pacer_queue_.size() < 10000);
218 } 218 }
219 } 219 }
220 220
221 clock_.AdvanceTimeMilliseconds(time_until_next_event_ms); 221 clock_.AdvanceTimeMilliseconds(time_until_next_event_ms);
222 222
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485 RecordBitrate(); 485 RecordBitrate();
486 } 486 }
487 487
488 uint32_t TcpSender::TargetBitrateKbps() { 488 uint32_t TcpSender::TargetBitrateKbps() {
489 return bitrate_kbps_; 489 return bitrate_kbps_;
490 } 490 }
491 491
492 } // namespace bwe 492 } // namespace bwe
493 } // namespace testing 493 } // namespace testing
494 } // namespace webrtc 494 } // namespace webrtc
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