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Side by Side Diff: webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

Issue 1350163005: Avoid circular dependency rtp_rtcp <-> paced_sender (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years, 3 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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388 // The number of packets lost in events where no adjacent packets were also 388 // The number of packets lost in events where no adjacent packets were also
389 // lost. 389 // lost.
390 uint64_t single_packet_loss_count; 390 uint64_t single_packet_loss_count;
391 // The number of events in which more than one adjacent packet was lost. 391 // The number of events in which more than one adjacent packet was lost.
392 uint64_t multiple_packet_loss_event_count; 392 uint64_t multiple_packet_loss_event_count;
393 // The number of packets lost in events where more than one adjacent packet 393 // The number of packets lost in events where more than one adjacent packet
394 // was lost. 394 // was lost.
395 uint64_t multiple_packet_loss_packet_count; 395 uint64_t multiple_packet_loss_packet_count;
396 }; 396 };
397 397
398 class RtpPacketSender {
399 public:
400 RtpPacketSender() {}
401 virtual ~RtpPacketSender() {}
402
403 enum Priority {
404 kHighPriority = 0, // Pass through; will be sent immediately.
405 kNormalPriority = 2, // Put in back of the line.
406 kLowPriority = 3, // Put in back of the low priority line.
407 };
408 // Low priority packets are mixed with the normal priority packets
409 // while we are paused.
410
411 // Returns true if we send the packet now, else it will add the packet
412 // information to the queue and call TimeToSendPacket when it's time to send.
413 virtual bool SendPacket(Priority priority,
414 uint32_t ssrc,
415 uint16_t sequence_number,
416 int64_t capture_time_ms,
417 size_t bytes,
418 bool retransmission) = 0;
419 };
420
421 class TransportSequenceNumberAllocator {
422 public:
423 TransportSequenceNumberAllocator() {}
424 virtual ~TransportSequenceNumberAllocator() {}
425
426 virtual uint16_t AllocateSequenceNumber() = 0;
427 };
428
398 } // namespace webrtc 429 } // namespace webrtc
399 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ 430 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
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