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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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70 | 70 |
71 DeliveryStatus DeliverPacket(MediaType media_type, | 71 DeliveryStatus DeliverPacket(MediaType media_type, |
72 const uint8_t* packet, | 72 const uint8_t* packet, |
73 size_t length, | 73 size_t length, |
74 const PacketTime& packet_time) override; | 74 const PacketTime& packet_time) override; |
75 | 75 |
76 void SetBitrateConfig( | 76 void SetBitrateConfig( |
77 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; | 77 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; |
78 void SignalNetworkState(NetworkState state) override; | 78 void SignalNetworkState(NetworkState state) override; |
79 | 79 |
| 80 void OnSentPacket(const rtc::SentPacket& sent_packet) override; |
| 81 |
80 private: | 82 private: |
81 DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet, | 83 DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet, |
82 size_t length); | 84 size_t length); |
83 DeliveryStatus DeliverRtp(MediaType media_type, | 85 DeliveryStatus DeliverRtp(MediaType media_type, |
84 const uint8_t* packet, | 86 const uint8_t* packet, |
85 size_t length, | 87 size_t length, |
86 const PacketTime& packet_time); | 88 const PacketTime& packet_time); |
87 | 89 |
88 void ConfigureSync(const std::string& sync_group) | 90 void ConfigureSync(const std::string& sync_group) |
89 EXCLUSIVE_LOCKS_REQUIRED(receive_crit_); | 91 EXCLUSIVE_LOCKS_REQUIRED(receive_crit_); |
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404 } | 406 } |
405 } | 407 } |
406 { | 408 { |
407 ReadLockScoped write_lock(*receive_crit_); | 409 ReadLockScoped write_lock(*receive_crit_); |
408 for (auto& kv : video_receive_ssrcs_) { | 410 for (auto& kv : video_receive_ssrcs_) { |
409 kv.second->SignalNetworkState(state); | 411 kv.second->SignalNetworkState(state); |
410 } | 412 } |
411 } | 413 } |
412 } | 414 } |
413 | 415 |
| 416 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) { |
| 417 channel_group_->OnSentPacket(sent_packet); |
| 418 } |
| 419 |
414 void Call::ConfigureSync(const std::string& sync_group) { | 420 void Call::ConfigureSync(const std::string& sync_group) { |
415 // Set sync only if there was no previous one. | 421 // Set sync only if there was no previous one. |
416 if (config_.voice_engine == nullptr || sync_group.empty()) | 422 if (config_.voice_engine == nullptr || sync_group.empty()) |
417 return; | 423 return; |
418 | 424 |
419 AudioReceiveStream* sync_audio_stream = nullptr; | 425 AudioReceiveStream* sync_audio_stream = nullptr; |
420 // Find existing audio stream. | 426 // Find existing audio stream. |
421 const auto it = sync_stream_mapping_.find(sync_group); | 427 const auto it = sync_stream_mapping_.find(sync_group); |
422 if (it != sync_stream_mapping_.end()) { | 428 if (it != sync_stream_mapping_.end()) { |
423 sync_audio_stream = it->second; | 429 sync_audio_stream = it->second; |
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532 size_t length, | 538 size_t length, |
533 const PacketTime& packet_time) { | 539 const PacketTime& packet_time) { |
534 if (RtpHeaderParser::IsRtcp(packet, length)) | 540 if (RtpHeaderParser::IsRtcp(packet, length)) |
535 return DeliverRtcp(media_type, packet, length); | 541 return DeliverRtcp(media_type, packet, length); |
536 | 542 |
537 return DeliverRtp(media_type, packet, length, packet_time); | 543 return DeliverRtp(media_type, packet, length, packet_time); |
538 } | 544 } |
539 | 545 |
540 } // namespace internal | 546 } // namespace internal |
541 } // namespace webrtc | 547 } // namespace webrtc |
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