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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

Issue 1342543004: Consolidate constructormagic macros with Chromium version and remove Chromium override. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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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77 size_t fragment_offset, 77 size_t fragment_offset,
78 size_t fragment_length); 78 size_t fragment_length);
79 void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send); 79 void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send);
80 void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send); 80 void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send);
81 81
82 const uint8_t* payload_data_; 82 const uint8_t* payload_data_;
83 size_t payload_size_; 83 size_t payload_size_;
84 const size_t max_payload_len_; 84 const size_t max_payload_len_;
85 RTPFragmentationHeader fragmentation_; 85 RTPFragmentationHeader fragmentation_;
86 PacketQueue packets_; 86 PacketQueue packets_;
87 FrameType frame_type_;
88 87
89 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); 88 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
90 }; 89 };
91 90
92 // Depacketizer for H264. 91 // Depacketizer for H264.
93 class RtpDepacketizerH264 : public RtpDepacketizer { 92 class RtpDepacketizerH264 : public RtpDepacketizer {
94 public: 93 public:
95 virtual ~RtpDepacketizerH264() {} 94 virtual ~RtpDepacketizerH264() {}
96 95
97 bool Parse(ParsedPayload* parsed_payload, 96 bool Parse(ParsedPayload* parsed_payload,
98 const uint8_t* payload_data, 97 const uint8_t* payload_data,
99 size_t payload_data_length) override; 98 size_t payload_data_length) override;
100 }; 99 };
101 } // namespace webrtc 100 } // namespace webrtc
102 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ 101 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
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