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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 <queue> | 16 #include <queue> |
16 #include <string> | 17 #include <string> |
17 | 18 |
18 #include "webrtc/base/buffer.h" | 19 #include "webrtc/base/buffer.h" |
19 #include "webrtc/base/constructormagic.h" | 20 #include "webrtc/base/constructormagic.h" |
20 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" |
21 | 22 |
22 namespace webrtc { | 23 namespace webrtc { |
23 | 24 |
24 class RtpPacketizerH264 : public RtpPacketizer { | 25 class RtpPacketizerH264 : public RtpPacketizer { |
25 public: | 26 public: |
26 // Initialize with payload from encoder. | 27 // Initialize with payload from encoder. |
27 // The payload_data must be exactly one encoded H264 frame. | 28 // The payload_data must be exactly one encoded H264 frame. |
28 RtpPacketizerH264(FrameType frame_type, size_t max_payload_len); | 29 RtpPacketizerH264(const RTPVideoHeaderH264& parameters, |
| 30 size_t max_payload_len); |
29 | 31 |
30 virtual ~RtpPacketizerH264(); | 32 virtual ~RtpPacketizerH264(); |
31 | 33 |
32 void SetPayloadData(const uint8_t* payload_data, | 34 void SetPayloadData(const uint8_t* payload_data, |
33 size_t payload_size, | 35 size_t payload_size, |
34 const RTPFragmentationHeader* fragmentation) override; | 36 const RTPFragmentationHeader* fragmentation) override; |
35 | 37 |
36 // Get the next payload with H264 payload header. | 38 // Get the next payload with H264 payload header. |
37 // buffer is a pointer to where the output will be written. | 39 // buffer is a pointer to where the output will be written. |
38 // bytes_to_send is an output variable that will contain number of bytes | 40 // bytes_to_send is an output variable that will contain number of bytes |
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85 bool aggregated; | 87 bool aggregated; |
86 uint8_t header; | 88 uint8_t header; |
87 }; | 89 }; |
88 | 90 |
89 void GeneratePackets(); | 91 void GeneratePackets(); |
90 void PacketizeFuA(size_t fragment_index); | 92 void PacketizeFuA(size_t fragment_index); |
91 size_t PacketizeStapA(size_t fragment_index); | 93 size_t PacketizeStapA(size_t fragment_index); |
92 void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send); | 94 void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send); |
93 void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send); | 95 void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send); |
94 | 96 |
| 97 H264PacketizationMode packetization_mode_; |
95 const size_t max_payload_len_; | 98 const size_t max_payload_len_; |
96 std::deque<Fragment> input_fragments_; | 99 std::deque<Fragment> input_fragments_; |
97 std::queue<PacketUnit> packets_; | 100 std::queue<PacketUnit> packets_; |
98 | 101 |
99 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); | 102 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); |
100 }; | 103 }; |
101 | 104 |
102 // Depacketizer for H264. | 105 // Depacketizer for H264. |
103 class RtpDepacketizerH264 : public RtpDepacketizer { | 106 class RtpDepacketizerH264 : public RtpDepacketizer { |
104 public: | 107 public: |
105 RtpDepacketizerH264(); | 108 RtpDepacketizerH264(); |
106 virtual ~RtpDepacketizerH264(); | 109 virtual ~RtpDepacketizerH264(); |
107 | 110 |
108 bool Parse(ParsedPayload* parsed_payload, | 111 bool Parse(ParsedPayload* parsed_payload, |
109 const uint8_t* payload_data, | 112 const uint8_t* payload_data, |
110 size_t payload_data_length) override; | 113 size_t payload_data_length) override; |
111 | 114 |
112 private: | 115 private: |
113 bool ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload, | 116 bool ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload, |
114 const uint8_t* payload_data); | 117 const uint8_t* payload_data); |
115 bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, | 118 bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, |
116 const uint8_t* payload_data); | 119 const uint8_t* payload_data); |
117 | 120 |
118 size_t offset_; | 121 size_t offset_; |
119 size_t length_; | 122 size_t length_; |
120 std::unique_ptr<rtc::Buffer> modified_buffer_; | 123 std::unique_ptr<rtc::Buffer> modified_buffer_; |
121 }; | 124 }; |
122 } // namespace webrtc | 125 } // namespace webrtc |
123 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 126 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |
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