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

Issue 1702043002: Remove video-codec max bitrate from TMMBN. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 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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34 const uint8_t kRtpHeaderSize = 12; 34 const uint8_t kRtpHeaderSize = 12;
35 35
36 struct AudioPayload { 36 struct AudioPayload {
37 uint32_t frequency; 37 uint32_t frequency;
38 size_t channels; 38 size_t channels;
39 uint32_t rate; 39 uint32_t rate;
40 }; 40 };
41 41
42 struct VideoPayload { 42 struct VideoPayload {
43 RtpVideoCodecTypes videoCodecType; 43 RtpVideoCodecTypes videoCodecType;
44 uint32_t maxRate;
45 }; 44 };
46 45
47 union PayloadUnion { 46 union PayloadUnion {
48 AudioPayload Audio; 47 AudioPayload Audio;
49 VideoPayload Video; 48 VideoPayload Video;
50 }; 49 };
51 50
52 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 }; 51 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
53 52
54 enum ProtectionType { 53 enum ProtectionType {
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407 class TransportSequenceNumberAllocator { 406 class TransportSequenceNumberAllocator {
408 public: 407 public:
409 TransportSequenceNumberAllocator() {} 408 TransportSequenceNumberAllocator() {}
410 virtual ~TransportSequenceNumberAllocator() {} 409 virtual ~TransportSequenceNumberAllocator() {}
411 410
412 virtual uint16_t AllocateSequenceNumber() = 0; 411 virtual uint16_t AllocateSequenceNumber() = 0;
413 }; 412 };
414 413
415 } // namespace webrtc 414 } // namespace webrtc
416 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ 415 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
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