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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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