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Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed nit Created 4 years, 5 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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160 if (rtp_video_header.simulcastIdx >= num_sending_modules_) 160 if (rtp_video_header.simulcastIdx >= num_sending_modules_)
161 return -1; 161 return -1;
162 stream_idx = rtp_video_header.simulcastIdx; 162 stream_idx = rtp_video_header.simulcastIdx;
163 163
164 return rtp_modules_[stream_idx]->SendOutgoingData( 164 return rtp_modules_[stream_idx]->SendOutgoingData(
165 encoded_image._frameType, payload_type_, encoded_image._timeStamp, 165 encoded_image._frameType, payload_type_, encoded_image._timeStamp,
166 encoded_image.capture_time_ms_, encoded_image._buffer, 166 encoded_image.capture_time_ms_, encoded_image._buffer,
167 encoded_image._length, fragmentation, &rtp_video_header); 167 encoded_image._length, fragmentation, &rtp_video_header);
168 } 168 }
169 169
170 void PayloadRouter::SetTargetSendBitrate(uint32_t bitrate_bps) {
171 rtc::CritScope lock(&crit_);
172 RTC_DCHECK_LE(streams_.size(), rtp_modules_.size());
173
174 // TODO(sprang): Rebase https://codereview.webrtc.org/1913073002/ on top of
175 // this.
176 int bitrate_remainder = bitrate_bps;
177 for (size_t i = 0; i < streams_.size() && bitrate_remainder > 0; ++i) {
178 int stream_bitrate = 0;
179 if (streams_[i].max_bitrate_bps > bitrate_remainder) {
180 stream_bitrate = bitrate_remainder;
181 } else {
182 stream_bitrate = streams_[i].max_bitrate_bps;
183 }
184 bitrate_remainder -= stream_bitrate;
185 rtp_modules_[i]->SetTargetSendBitrate(stream_bitrate);
186 }
187 }
188
189 size_t PayloadRouter::MaxPayloadLength() const { 170 size_t PayloadRouter::MaxPayloadLength() const {
190 size_t min_payload_length = DefaultMaxPayloadLength(); 171 size_t min_payload_length = DefaultMaxPayloadLength();
191 rtc::CritScope lock(&crit_); 172 rtc::CritScope lock(&crit_);
192 for (size_t i = 0; i < num_sending_modules_; ++i) { 173 for (size_t i = 0; i < num_sending_modules_; ++i) {
193 size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength(); 174 size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength();
194 if (module_payload_length < min_payload_length) 175 if (module_payload_length < min_payload_length)
195 min_payload_length = module_payload_length; 176 min_payload_length = module_payload_length;
196 } 177 }
197 return min_payload_length; 178 return min_payload_length;
198 } 179 }
199 180
200 } // namespace webrtc 181 } // namespace webrtc
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