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Side by Side Diff: webrtc/voice_engine/channel.h

Issue 1267683002: Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Processed more comments from Henrik. Created 5 years, 4 months ago
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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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