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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 #include <vector> |
17 | 17 |
18 #include "webrtc/modules/include/module_common_types.h" | 18 #include "webrtc/modules/include/module_common_types.h" |
19 #include "webrtc/system_wrappers/include/clock.h" | 19 #include "webrtc/system_wrappers/include/clock.h" |
20 #include "webrtc/typedefs.h" | 20 #include "webrtc/typedefs.h" |
21 | 21 |
22 #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 |
23 #define IP_PACKET_SIZE 1500 // we assume ethernet | 23 #define IP_PACKET_SIZE 1500 // we assume ethernet |
24 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 | 24 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 |
25 #define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds | |
26 | 25 |
27 namespace webrtc { | 26 namespace webrtc { |
28 namespace rtcp { | 27 namespace rtcp { |
29 class TransportFeedback; | 28 class TransportFeedback; |
30 } | 29 } |
31 | 30 |
32 const int kVideoPayloadTypeFrequency = 90000; | 31 const int kVideoPayloadTypeFrequency = 90000; |
| 32 // TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy |
| 33 // and should be fixed. |
| 34 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458 |
| 35 const int kBogusRtpRateForAudioRtcp = 8000; |
33 | 36 |
34 // Minimum RTP header size in bytes. | 37 // Minimum RTP header size in bytes. |
35 const uint8_t kRtpHeaderSize = 12; | 38 const uint8_t kRtpHeaderSize = 12; |
36 | 39 |
37 struct AudioPayload { | 40 struct AudioPayload { |
38 uint32_t frequency; | 41 uint32_t frequency; |
39 size_t channels; | 42 size_t channels; |
40 uint32_t rate; | 43 uint32_t rate; |
41 }; | 44 }; |
42 | 45 |
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389 class TransportSequenceNumberAllocator { | 392 class TransportSequenceNumberAllocator { |
390 public: | 393 public: |
391 TransportSequenceNumberAllocator() {} | 394 TransportSequenceNumberAllocator() {} |
392 virtual ~TransportSequenceNumberAllocator() {} | 395 virtual ~TransportSequenceNumberAllocator() {} |
393 | 396 |
394 virtual uint16_t AllocateSequenceNumber() = 0; | 397 virtual uint16_t AllocateSequenceNumber() = 0; |
395 }; | 398 }; |
396 | 399 |
397 } // namespace webrtc | 400 } // namespace webrtc |
398 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ | 401 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ |
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