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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(FrameType frame_type, size_t max_payload_len); |
29 | 30 |
30 virtual ~RtpPacketizerH264(); | 31 virtual ~RtpPacketizerH264(); |
31 | 32 |
32 void SetPayloadData(const uint8_t* payload_data, | 33 void SetPayloadData(const uint8_t* payload_data, |
33 size_t payload_size, | 34 size_t payload_size, |
34 const RTPFragmentationHeader* fragmentation) override; | 35 const RTPFragmentationHeader* fragmentation) override; |
35 | 36 |
36 // Get the next payload with H264 payload header. | 37 // Get the next payload with H264 payload header. |
37 // buffer is a pointer to where the output will be written. | 38 // Write payload and set marker bit of the |packet|. |
38 // bytes_to_send is an output variable that will contain number of bytes | 39 // The parameter |last_packet| is true for the last packet of the frame, false |
39 // written to buffer. The parameter last_packet is true for the last packet of | 40 // otherwise (i.e., call the function again to get the next packet). |
40 // the frame, false otherwise (i.e., call the function again to get the | 41 // Returns true on success, false otherwise. |
41 // next packet). | 42 bool NextPacket(RtpPacketToSend* rtp_packet, bool* last_packet) override; |
42 // Returns true on success or false if there was no payload to packetize. | |
43 bool NextPacket(uint8_t* buffer, | |
44 size_t* bytes_to_send, | |
45 bool* last_packet) override; | |
46 | 43 |
47 ProtectionType GetProtectionType() override; | 44 ProtectionType GetProtectionType() override; |
48 | 45 |
49 StorageType GetStorageType(uint32_t retransmission_settings) override; | 46 StorageType GetStorageType(uint32_t retransmission_settings) override; |
50 | 47 |
51 std::string ToString() override; | 48 std::string ToString() override; |
52 | 49 |
53 private: | 50 private: |
54 // Input fragments (NAL units), with an optionally owned temporary buffer, | 51 // Input fragments (NAL units), with an optionally owned temporary buffer, |
55 // used in case the fragment gets modified. | 52 // used in case the fragment gets modified. |
(...skipping 26 matching lines...) Expand all Loading... |
82 const Fragment source_fragment; | 79 const Fragment source_fragment; |
83 bool first_fragment; | 80 bool first_fragment; |
84 bool last_fragment; | 81 bool last_fragment; |
85 bool aggregated; | 82 bool aggregated; |
86 uint8_t header; | 83 uint8_t header; |
87 }; | 84 }; |
88 | 85 |
89 void GeneratePackets(); | 86 void GeneratePackets(); |
90 void PacketizeFuA(size_t fragment_index); | 87 void PacketizeFuA(size_t fragment_index); |
91 size_t PacketizeStapA(size_t fragment_index); | 88 size_t PacketizeStapA(size_t fragment_index); |
92 void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send); | 89 void NextAggregatePacket(RtpPacketToSend* rtp_packet); |
93 void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send); | 90 void NextFragmentPacket(RtpPacketToSend* rtp_packet); |
94 | 91 |
95 const size_t max_payload_len_; | 92 const size_t max_payload_len_; |
96 std::deque<Fragment> input_fragments_; | 93 std::deque<Fragment> input_fragments_; |
97 std::queue<PacketUnit> packets_; | 94 std::queue<PacketUnit> packets_; |
98 | 95 |
99 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); | 96 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264); |
100 }; | 97 }; |
101 | 98 |
102 // Depacketizer for H264. | 99 // Depacketizer for H264. |
103 class RtpDepacketizerH264 : public RtpDepacketizer { | 100 class RtpDepacketizerH264 : public RtpDepacketizer { |
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114 const uint8_t* payload_data); | 111 const uint8_t* payload_data); |
115 bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, | 112 bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload, |
116 const uint8_t* payload_data); | 113 const uint8_t* payload_data); |
117 | 114 |
118 size_t offset_; | 115 size_t offset_; |
119 size_t length_; | 116 size_t length_; |
120 std::unique_ptr<rtc::Buffer> modified_buffer_; | 117 std::unique_ptr<rtc::Buffer> modified_buffer_; |
121 }; | 118 }; |
122 } // namespace webrtc | 119 } // namespace webrtc |
123 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ | 120 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_ |
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