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

Issue 2310943002: Add time line for acked bitrate. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 4 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
11 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
13 13
14 #include <stddef.h> 14 #include <stddef.h>
15 #include <list> 15 #include <list>
16 #include <vector>
16 17
17 #include "webrtc/modules/include/module_common_types.h" 18 #include "webrtc/modules/include/module_common_types.h"
18 #include "webrtc/system_wrappers/include/clock.h" 19 #include "webrtc/system_wrappers/include/clock.h"
19 #include "webrtc/typedefs.h" 20 #include "webrtc/typedefs.h"
20 21
21 #define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination 22 #define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination
22 #define IP_PACKET_SIZE 1500 // we assume ethernet 23 #define IP_PACKET_SIZE 1500 // we assume ethernet
23 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 24 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
24 #define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds 25 #define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds
25 26
(...skipping 277 matching lines...) Expand 10 before | Expand all | Expand 10 after
303 TransportFeedbackObserver() {} 304 TransportFeedbackObserver() {}
304 virtual ~TransportFeedbackObserver() {} 305 virtual ~TransportFeedbackObserver() {}
305 306
306 // Note: Transport-wide sequence number as sequence number. Arrival time 307 // Note: Transport-wide sequence number as sequence number. Arrival time
307 // must be set to 0. 308 // must be set to 0.
308 virtual void AddPacket(uint16_t sequence_number, 309 virtual void AddPacket(uint16_t sequence_number,
309 size_t length, 310 size_t length,
310 int probe_cluster_id) = 0; 311 int probe_cluster_id) = 0;
311 312
312 virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0; 313 virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
314
315 virtual std::vector<PacketInfo> GetTransportFeedbackVector() const = 0;
313 }; 316 };
314 317
315 class RtcpRttStats { 318 class RtcpRttStats {
316 public: 319 public:
317 virtual void OnRttUpdate(int64_t rtt) = 0; 320 virtual void OnRttUpdate(int64_t rtt) = 0;
318 321
319 virtual int64_t LastProcessedRtt() const = 0; 322 virtual int64_t LastProcessedRtt() const = 0;
320 323
321 virtual ~RtcpRttStats() {} 324 virtual ~RtcpRttStats() {}
322 }; 325 };
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393 class TransportSequenceNumberAllocator { 396 class TransportSequenceNumberAllocator {
394 public: 397 public:
395 TransportSequenceNumberAllocator() {} 398 TransportSequenceNumberAllocator() {}
396 virtual ~TransportSequenceNumberAllocator() {} 399 virtual ~TransportSequenceNumberAllocator() {}
397 400
398 virtual uint16_t AllocateSequenceNumber() = 0; 401 virtual uint16_t AllocateSequenceNumber() = 0;
399 }; 402 };
400 403
401 } // namespace webrtc 404 } // namespace webrtc
402 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ 405 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
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