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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 |
170 size_t PayloadRouter::MaxPayloadLength() const { | 189 size_t PayloadRouter::MaxPayloadLength() const { |
171 size_t min_payload_length = DefaultMaxPayloadLength(); | 190 size_t min_payload_length = DefaultMaxPayloadLength(); |
172 rtc::CritScope lock(&crit_); | 191 rtc::CritScope lock(&crit_); |
173 for (size_t i = 0; i < num_sending_modules_; ++i) { | 192 for (size_t i = 0; i < num_sending_modules_; ++i) { |
174 size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength(); | 193 size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength(); |
175 if (module_payload_length < min_payload_length) | 194 if (module_payload_length < min_payload_length) |
176 min_payload_length = module_payload_length; | 195 min_payload_length = module_payload_length; |
177 } | 196 } |
178 return min_payload_length; | 197 return min_payload_length; |
179 } | 198 } |
180 | 199 |
181 } // namespace webrtc | 200 } // namespace webrtc |
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