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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" |
18 #include "webrtc/modules/include/module_common_types.h" | 19 #include "webrtc/modules/include/module_common_types.h" |
19 #include "webrtc/system_wrappers/include/clock.h" | 20 #include "webrtc/system_wrappers/include/clock.h" |
20 #include "webrtc/typedefs.h" | 21 #include "webrtc/typedefs.h" |
21 | 22 |
22 #define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination | 23 #define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination |
23 #define IP_PACKET_SIZE 1500 // we assume ethernet | 24 #define IP_PACKET_SIZE 1500 // we assume ethernet |
24 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 | 25 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 |
25 | 26 |
26 namespace webrtc { | 27 namespace webrtc { |
27 namespace rtcp { | 28 namespace rtcp { |
28 class TransportFeedback; | 29 class TransportFeedback; |
29 } | 30 } |
30 | 31 |
31 const int kVideoPayloadTypeFrequency = 90000; | 32 const int kVideoPayloadTypeFrequency = 90000; |
32 // TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy | 33 // TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy |
33 // and should be fixed. | 34 // and should be fixed. |
34 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458 | 35 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458 |
35 const int kBogusRtpRateForAudioRtcp = 8000; | 36 const int kBogusRtpRateForAudioRtcp = 8000; |
36 | 37 |
37 // Minimum RTP header size in bytes. | 38 // Minimum RTP header size in bytes. |
38 const uint8_t kRtpHeaderSize = 12; | 39 const uint8_t kRtpHeaderSize = 12; |
39 | 40 |
40 struct AudioPayload { | 41 struct AudioPayload { |
41 uint32_t frequency; | 42 uint32_t frequency; |
42 size_t channels; | 43 size_t channels; |
43 uint32_t rate; | 44 uint32_t rate; |
44 }; | 45 }; |
45 | 46 |
46 struct VideoPayload { | 47 struct VideoPayload { |
47 RtpVideoCodecTypes videoCodecType; | 48 RtpVideoCodecTypes videoCodecType; |
| 49 // The H264 profile only matters if videoCodecType == kRtpVideoH264. |
| 50 H264::Profile h264_profile; |
48 }; | 51 }; |
49 | 52 |
50 union PayloadUnion { | 53 union PayloadUnion { |
51 AudioPayload Audio; | 54 AudioPayload Audio; |
52 VideoPayload Video; | 55 VideoPayload Video; |
53 }; | 56 }; |
54 | 57 |
55 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 }; | 58 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 }; |
56 | 59 |
57 enum ProtectionType { | 60 enum ProtectionType { |
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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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