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| 1 /* | 1 /* | 
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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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| 44 | 44 | 
| 45 namespace webrtc { | 45 namespace webrtc { | 
| 46 | 46 | 
| 47 class AudioDeviceModule; | 47 class AudioDeviceModule; | 
| 48 class Config; | 48 class Config; | 
| 49 class CriticalSectionWrapper; | 49 class CriticalSectionWrapper; | 
| 50 class FileWrapper; | 50 class FileWrapper; | 
| 51 class ProcessThread; | 51 class ProcessThread; | 
| 52 class ReceiveStatistics; | 52 class ReceiveStatistics; | 
| 53 class RemoteNtpTimeEstimator; | 53 class RemoteNtpTimeEstimator; | 
| 54 class RtcEventLog; | |
| 54 class RTPPayloadRegistry; | 55 class RTPPayloadRegistry; | 
| 55 class RtpReceiver; | 56 class RtpReceiver; | 
| 56 class RTPReceiverAudio; | 57 class RTPReceiverAudio; | 
| 57 class RtpRtcp; | 58 class RtpRtcp; | 
| 58 class TelephoneEventHandler; | 59 class TelephoneEventHandler; | 
| 59 class VoEMediaProcess; | 60 class VoEMediaProcess; | 
| 60 class VoERTPObserver; | 61 class VoERTPObserver; | 
| 61 class VoiceEngineObserver; | 62 class VoiceEngineObserver; | 
| 62 | 63 | 
| 63 struct CallStatistics; | 64 struct CallStatistics; | 
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| 441 // Used for obtaining RTT for a receive-only channel. | 442 // Used for obtaining RTT for a receive-only channel. | 
| 442 void set_associate_send_channel(const ChannelOwner& channel) { | 443 void set_associate_send_channel(const ChannelOwner& channel) { | 
| 443 assert(_channelId != channel.channel()->ChannelId()); | 444 assert(_channelId != channel.channel()->ChannelId()); | 
| 444 CriticalSectionScoped lock(assoc_send_channel_lock_.get()); | 445 CriticalSectionScoped lock(assoc_send_channel_lock_.get()); | 
| 445 associate_send_channel_ = channel; | 446 associate_send_channel_ = channel; | 
| 446 } | 447 } | 
| 447 | 448 | 
| 448 // Disassociate a send channel if it was associated. | 449 // Disassociate a send channel if it was associated. | 
| 449 void DisassociateSendChannel(int channel_id); | 450 void DisassociateSendChannel(int channel_id); | 
| 450 | 451 | 
| 452 // Sets an RtcEventLog object to enable logging of debug events in the Audio | |
| 453 // Coding Module. Calling this function with a nullptr is allowed and will | |
| 454 // stop all logging activity. | |
| 455 void SetEventLog(RtcEventLog* event_log); | |
| 
 
Henrik Grunell WebRTC
2015/08/05 09:44:35
Use a separate function for disabling the logging.
 
Henrik Grunell WebRTC
2015/08/07 11:30:03
Please address this.
 
terelius
2015/08/11 09:04:34
The main way to stop logging is to call StopLoggin
 
Henrik Grunell WebRTC
2015/08/12 12:38:50
OK, well, the comment says differently. :) We need
 
ivoc
2015/08/12 13:16:41
The SetEventLog function is removed in the latest
 
Henrik Grunell WebRTC
2015/08/12 15:29:53
Acknowledged.
 
 | |
| 456 | |
| 451 protected: | 457 protected: | 
| 452 void OnIncomingFractionLoss(int fraction_lost); | 458 void OnIncomingFractionLoss(int fraction_lost); | 
| 453 | 459 | 
| 454 private: | 460 private: | 
| 455 bool ReceivePacket(const uint8_t* packet, size_t packet_length, | 461 bool ReceivePacket(const uint8_t* packet, size_t packet_length, | 
| 456 const RTPHeader& header, bool in_order); | 462 const RTPHeader& header, bool in_order); | 
| 457 bool HandleRtxPacket(const uint8_t* packet, | 463 bool HandleRtxPacket(const uint8_t* packet, | 
| 458 size_t packet_length, | 464 size_t packet_length, | 
| 459 const RTPHeader& header); | 465 const RTPHeader& header); | 
| 460 bool IsPacketInOrder(const RTPHeader& header) const; | 466 bool IsPacketInOrder(const RTPHeader& header) const; | 
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| 578 rtc::scoped_ptr<NetworkPredictor> network_predictor_; | 584 rtc::scoped_ptr<NetworkPredictor> network_predictor_; | 
| 579 // An associated send channel. | 585 // An associated send channel. | 
| 580 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; | 586 rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_; | 
| 581 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); | 587 ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_); | 
| 582 }; | 588 }; | 
| 583 | 589 | 
| 584 } // namespace voe | 590 } // namespace voe | 
| 585 } // namespace webrtc | 591 } // namespace webrtc | 
| 586 | 592 | 
| 587 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 593 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 
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