| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ |
| |
| #include <string> |
| |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| namespace RtpFormatVideoGeneric { |
| static const uint8_t kKeyFrameBit = 0x01; |
| static const uint8_t kFirstPacketBit = 0x02; |
| } // namespace RtpFormatVideoGeneric |
| |
| class RtpPacketizerGeneric : public RtpPacketizer { |
| public: |
| // Initialize with payload from encoder. |
| // The payload_data must be exactly one encoded generic frame. |
| RtpPacketizerGeneric(FrameType frametype, |
| size_t max_payload_len, |
| size_t last_packet_reduction_len); |
| |
| virtual ~RtpPacketizerGeneric(); |
| |
| // Returns total number of packets to be generated. |
| size_t SetPayloadData(const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| // Get the next payload with generic payload header. |
| // Write payload and set marker bit of the |packet|. |
| // Returns true on success, false otherwise. |
| bool NextPacket(RtpPacketToSend* packet) override; |
| |
| std::string ToString() override; |
| |
| private: |
| const uint8_t* payload_data_; |
| size_t payload_size_; |
| const size_t max_payload_len_; |
| const size_t last_packet_reduction_len_; |
| FrameType frame_type_; |
| size_t payload_len_per_packet_; |
| uint8_t generic_header_; |
| // Number of packets yet to be retrieved by NextPacket() call. |
| size_t num_packets_left_; |
| // Number of packets, which will be 1 byte more than the rest. |
| size_t num_larger_packets_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric); |
| }; |
| |
| // Depacketizer for generic codec. |
| class RtpDepacketizerGeneric : public RtpDepacketizer { |
| public: |
| virtual ~RtpDepacketizerGeneric() {} |
| |
| bool Parse(ParsedPayload* parsed_payload, |
| const uint8_t* payload_data, |
| size_t payload_data_length) override; |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_ |