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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ | 10 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ |
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29 // The payload_data must be exactly one encoded generic frame. | 29 // The payload_data must be exactly one encoded generic frame. |
30 RtpPacketizerGeneric(FrameType frametype, size_t max_payload_len); | 30 RtpPacketizerGeneric(FrameType frametype, size_t max_payload_len); |
31 | 31 |
32 virtual ~RtpPacketizerGeneric(); | 32 virtual ~RtpPacketizerGeneric(); |
33 | 33 |
34 void SetPayloadData(const uint8_t* payload_data, | 34 void SetPayloadData(const uint8_t* payload_data, |
35 size_t payload_size, | 35 size_t payload_size, |
36 const RTPFragmentationHeader* fragmentation) override; | 36 const RTPFragmentationHeader* fragmentation) override; |
37 | 37 |
38 // Get the next payload with generic payload header. | 38 // Get the next payload with generic payload header. |
39 // buffer is a pointer to where the output will be written. | 39 // Write payload and set marker bit of the |packet|. |
40 // bytes_to_send is an output variable that will contain number of bytes | 40 // The parameter |last_packet| is true for the last packet of the frame, false |
41 // written to buffer. The parameter last_packet is true for the last packet of | 41 // otherwise (i.e., call the function again to get the next packet). |
42 // the frame, false otherwise (i.e., call the function again to get the | 42 // Returns true on success, false otherwise. |
43 // next packet). | 43 bool NextPacket(RtpPacketToSend* packet, bool* last_packet) override; |
44 // Returns true on success or false if there was no payload to packetize. | |
45 bool NextPacket(uint8_t* buffer, | |
46 size_t* bytes_to_send, | |
47 bool* last_packet) override; | |
48 | 44 |
49 ProtectionType GetProtectionType() override; | 45 ProtectionType GetProtectionType() override; |
50 | 46 |
51 StorageType GetStorageType(uint32_t retransmission_settings) override; | 47 StorageType GetStorageType(uint32_t retransmission_settings) override; |
52 | 48 |
53 std::string ToString() override; | 49 std::string ToString() override; |
54 | 50 |
55 private: | 51 private: |
56 const uint8_t* payload_data_; | 52 const uint8_t* payload_data_; |
57 size_t payload_size_; | 53 size_t payload_size_; |
58 const size_t max_payload_len_; | 54 const size_t max_payload_len_; |
59 FrameType frame_type_; | 55 FrameType frame_type_; |
60 size_t payload_length_; | 56 size_t payload_length_; |
61 uint8_t generic_header_; | 57 uint8_t generic_header_; |
62 | 58 |
63 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric); | 59 RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric); |
64 }; | 60 }; |
65 | 61 |
66 // Depacketizer for generic codec. | 62 // Depacketizer for generic codec. |
67 class RtpDepacketizerGeneric : public RtpDepacketizer { | 63 class RtpDepacketizerGeneric : public RtpDepacketizer { |
68 public: | 64 public: |
69 virtual ~RtpDepacketizerGeneric() {} | 65 virtual ~RtpDepacketizerGeneric() {} |
70 | 66 |
71 bool Parse(ParsedPayload* parsed_payload, | 67 bool Parse(ParsedPayload* parsed_payload, |
72 const uint8_t* payload_data, | 68 const uint8_t* payload_data, |
73 size_t payload_data_length) override; | 69 size_t payload_data_length) override; |
74 }; | 70 }; |
75 } // namespace webrtc | 71 } // namespace webrtc |
76 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ | 72 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ |
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