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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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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118 | 118 |
119 // Helpers to insert a single packet into the delay-based BWE. | 119 // Helpers to insert a single packet into the delay-based BWE. |
120 void IncomingFeedback(int64_t arrival_time_ms, | 120 void IncomingFeedback(int64_t arrival_time_ms, |
121 int64_t send_time_ms, | 121 int64_t send_time_ms, |
122 uint16_t sequence_number, | 122 uint16_t sequence_number, |
123 size_t payload_size); | 123 size_t payload_size); |
124 void IncomingFeedback(int64_t arrival_time_ms, | 124 void IncomingFeedback(int64_t arrival_time_ms, |
125 int64_t send_time_ms, | 125 int64_t send_time_ms, |
126 uint16_t sequence_number, | 126 uint16_t sequence_number, |
127 size_t payload_size, | 127 size_t payload_size, |
128 int probe_cluster_id); | 128 const PacedPacketInfo& pacing_info); |
129 | 129 |
130 // Generates a frame of packets belonging to a stream at a given bitrate and | 130 // Generates a frame of packets belonging to a stream at a given bitrate and |
131 // with a given ssrc. The stream is pushed through a very simple simulated | 131 // with a given ssrc. The stream is pushed through a very simple simulated |
132 // network, and is then given to the receive-side bandwidth estimator. | 132 // network, and is then given to the receive-side bandwidth estimator. |
133 // Returns true if an over-use was seen, false otherwise. | 133 // Returns true if an over-use was seen, false otherwise. |
134 // The StreamGenerator::updated() should be used to check for any changes in | 134 // The StreamGenerator::updated() should be used to check for any changes in |
135 // target bitrate after the call to this function. | 135 // target bitrate after the call to this function. |
136 bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps); | 136 bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps); |
137 | 137 |
138 // Run the bandwidth estimator with a stream of |number_of_frames| frames, or | 138 // Run the bandwidth estimator with a stream of |number_of_frames| frames, or |
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165 std::unique_ptr<DelayBasedBwe> bitrate_estimator_; | 165 std::unique_ptr<DelayBasedBwe> bitrate_estimator_; |
166 std::unique_ptr<test::StreamGenerator> stream_generator_; | 166 std::unique_ptr<test::StreamGenerator> stream_generator_; |
167 int64_t arrival_time_offset_ms_; | 167 int64_t arrival_time_offset_ms_; |
168 bool first_update_; | 168 bool first_update_; |
169 | 169 |
170 RTC_DISALLOW_COPY_AND_ASSIGN(DelayBasedBweTest); | 170 RTC_DISALLOW_COPY_AND_ASSIGN(DelayBasedBweTest); |
171 }; | 171 }; |
172 } // namespace webrtc | 172 } // namespace webrtc |
173 | 173 |
174 #endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_
H_ | 174 #endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_
H_ |
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