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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/common_types.h" | |
19 #include "webrtc/modules/include/module_common_types.h" | 18 #include "webrtc/modules/include/module_common_types.h" |
20 #include "webrtc/system_wrappers/include/clock.h" | 19 #include "webrtc/system_wrappers/include/clock.h" |
21 #include "webrtc/typedefs.h" | 20 #include "webrtc/typedefs.h" |
22 | 21 |
23 #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 |
24 #define IP_PACKET_SIZE 1500 // we assume ethernet | 23 #define IP_PACKET_SIZE 1500 // we assume ethernet |
25 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 | 24 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 |
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; |
33 // TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy | 32 // TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy |
34 // and should be fixed. | 33 // and should be fixed. |
35 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458 | 34 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458 |
36 const int kBogusRtpRateForAudioRtcp = 8000; | 35 const int kBogusRtpRateForAudioRtcp = 8000; |
37 | 36 |
38 // Minimum RTP header size in bytes. | 37 // Minimum RTP header size in bytes. |
39 const uint8_t kRtpHeaderSize = 12; | 38 const uint8_t kRtpHeaderSize = 12; |
40 | 39 |
41 struct AudioPayload { | 40 struct AudioPayload { |
42 uint32_t frequency; | 41 uint32_t frequency; |
43 size_t channels; | 42 size_t channels; |
44 uint32_t rate; | 43 uint32_t rate; |
45 }; | 44 }; |
46 | 45 |
47 struct VideoPayload { | 46 struct VideoPayload { |
48 RtpVideoCodecTypes videoCodecType; | 47 RtpVideoCodecTypes videoCodecType; |
49 // The H264 profile only matters if videoCodecType == kRtpVideoH264. | |
50 H264::Profile h264_profile; | |
51 }; | 48 }; |
52 | 49 |
53 union PayloadUnion { | 50 union PayloadUnion { |
54 AudioPayload Audio; | 51 AudioPayload Audio; |
55 VideoPayload Video; | 52 VideoPayload Video; |
56 }; | 53 }; |
57 | 54 |
58 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 }; | 55 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 }; |
59 | 56 |
60 enum ProtectionType { | 57 enum ProtectionType { |
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396 class TransportSequenceNumberAllocator { | 393 class TransportSequenceNumberAllocator { |
397 public: | 394 public: |
398 TransportSequenceNumberAllocator() {} | 395 TransportSequenceNumberAllocator() {} |
399 virtual ~TransportSequenceNumberAllocator() {} | 396 virtual ~TransportSequenceNumberAllocator() {} |
400 | 397 |
401 virtual uint16_t AllocateSequenceNumber() = 0; | 398 virtual uint16_t AllocateSequenceNumber() = 0; |
402 }; | 399 }; |
403 | 400 |
404 } // namespace webrtc | 401 } // namespace webrtc |
405 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ | 402 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ |
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