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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 |
11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |
13 | 13 |
14 #include <deque> | 14 #include <deque> |
15 #include <memory> | 15 #include <memory> |
16 #include <queue> | 16 #include <queue> |
17 #include <string> | 17 #include <string> |
18 | 18 |
19 #include "webrtc/base/buffer.h" | |
20 #include "webrtc/base/constructormagic.h" | |
21 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" | 19 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" |
| 20 #include "webrtc/rtc_base/buffer.h" |
| 21 #include "webrtc/rtc_base/constructormagic.h" |
22 | 22 |
23 namespace webrtc { | 23 namespace webrtc { |
24 | 24 |
25 class RtpPacketizerH264 : public RtpPacketizer { | 25 class RtpPacketizerH264 : public RtpPacketizer { |
26 public: | 26 public: |
27 // Initialize with payload from encoder. | 27 // Initialize with payload from encoder. |
28 // The payload_data must be exactly one encoded H264 frame. | 28 // The payload_data must be exactly one encoded H264 frame. |
29 RtpPacketizerH264(size_t max_payload_len, | 29 RtpPacketizerH264(size_t max_payload_len, |
30 size_t last_packet_reduction_len, | 30 size_t last_packet_reduction_len, |
31 H264PacketizationMode packetization_mode); | 31 H264PacketizationMode packetization_mode); |
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115 const uint8_t* payload_data); | 115 const uint8_t* payload_data); |
116 bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, | 116 bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, |
117 const uint8_t* payload_data); | 117 const uint8_t* payload_data); |
118 | 118 |
119 size_t offset_; | 119 size_t offset_; |
120 size_t length_; | 120 size_t length_; |
121 std::unique_ptr<rtc::Buffer> modified_buffer_; | 121 std::unique_ptr<rtc::Buffer> modified_buffer_; |
122 }; | 122 }; |
123 } // namespace webrtc | 123 } // namespace webrtc |
124 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 124 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |
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