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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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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359 rtc::TaskQueue worker_queue_; | 359 rtc::TaskQueue worker_queue_; |
360 | 360 |
361 // The config mask set by SetBitrateConfigMask. | 361 // The config mask set by SetBitrateConfigMask. |
362 // 0 <= min <= start <= max | 362 // 0 <= min <= start <= max |
363 Config::BitrateConfigMask bitrate_config_mask_; | 363 Config::BitrateConfigMask bitrate_config_mask_; |
364 | 364 |
365 // The config set by SetBitrateConfig. | 365 // The config set by SetBitrateConfig. |
366 // min >= 0, start != 0, max == -1 || max > 0 | 366 // min >= 0, start != 0, max == -1 || max > 0 |
367 Config::BitrateConfig base_bitrate_config_; | 367 Config::BitrateConfig base_bitrate_config_; |
368 | 368 |
| 369 void PrintDebugStuff() override; |
| 370 |
369 RTC_DISALLOW_COPY_AND_ASSIGN(Call); | 371 RTC_DISALLOW_COPY_AND_ASSIGN(Call); |
370 }; | 372 }; |
371 } // namespace internal | 373 } // namespace internal |
372 | 374 |
373 std::string Call::Stats::ToString(int64_t time_ms) const { | 375 std::string Call::Stats::ToString(int64_t time_ms) const { |
374 std::stringstream ss; | 376 std::stringstream ss; |
375 ss << "Call stats: " << time_ms << ", {"; | 377 ss << "Call stats: " << time_ms << ", {"; |
376 ss << "send_bw_bps: " << send_bandwidth_bps << ", "; | 378 ss << "send_bw_bps: " << send_bandwidth_bps << ", "; |
377 ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; | 379 ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; |
378 ss << "max_pad_bps: " << max_padding_bitrate_bps << ", "; | 380 ss << "max_pad_bps: " << max_padding_bitrate_bps << ", "; |
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1433 } | 1435 } |
1434 // For audio, we only support send side BWE. | 1436 // For audio, we only support send side BWE. |
1435 if (media_type == MediaType::VIDEO || | 1437 if (media_type == MediaType::VIDEO || |
1436 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { | 1438 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { |
1437 receive_side_cc_.OnReceivedPacket( | 1439 receive_side_cc_.OnReceivedPacket( |
1438 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), | 1440 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), |
1439 header); | 1441 header); |
1440 } | 1442 } |
1441 } | 1443 } |
1442 | 1444 |
| 1445 void Call::PrintDebugStuff() { |
| 1446 transport_send_->send_side_cc()->PrintDebugStuff(); |
| 1447 print_ = true; |
| 1448 } |
| 1449 |
1443 } // namespace internal | 1450 } // namespace internal |
1444 | 1451 |
1445 } // namespace webrtc | 1452 } // namespace webrtc |
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