| /* |
| * Copyright (c) 2015 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. |
| */ |
| |
| // |
| // This file contains the declaration of the VP9 packetizer class. |
| // A packetizer object is created for each encoded video frame. The |
| // constructor is called with the payload data and size. |
| // |
| // After creating the packetizer, the method NextPacket is called |
| // repeatedly to get all packets for the frame. The method returns |
| // false as long as there are more packets left to fetch. |
| // |
| |
| #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_ |
| |
| #include <queue> |
| #include <string> |
| |
| #include "modules/include/module_common_types.h" |
| #include "modules/rtp_rtcp/source/rtp_format.h" |
| #include "rtc_base/constructormagic.h" |
| |
| namespace webrtc { |
| |
| class RtpPacketizerVp9 : public RtpPacketizer { |
| public: |
| RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr, |
| size_t max_payload_length, |
| size_t last_packet_reduction_len); |
| |
| ~RtpPacketizerVp9() override; |
| |
| // The payload data must be one encoded VP9 layer frame. |
| size_t SetPayloadData(const uint8_t* payload, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation); |
| |
| size_t NumPackets() const override; |
| |
| // Gets the next payload with VP9 payload header. |
| // Write payload and set marker bit of the |packet|. |
| // Returns true on success, false otherwise. |
| bool NextPacket(RtpPacketToSend* packet) override; |
| |
| typedef struct { |
| size_t payload_start_pos; |
| size_t size; |
| bool layer_begin; |
| bool layer_end; |
| } PacketInfo; |
| typedef std::queue<PacketInfo> PacketInfoQueue; |
| |
| private: |
| // Calculates all packet sizes and loads info to packet queue. |
| void GeneratePackets(); |
| |
| // Writes the payload descriptor header and copies payload to the |buffer|. |
| // |packet_info| determines which part of the payload to write. |
| // |last| indicates if the packet is the last packet in the frame. |
| // Returns true on success, false otherwise. |
| bool WriteHeaderAndPayload(const PacketInfo& packet_info, |
| RtpPacketToSend* packet, |
| bool last) const; |
| |
| // Writes payload descriptor header to |buffer|. |
| // Returns true on success, false otherwise. |
| bool WriteHeader(const PacketInfo& packet_info, |
| uint8_t* buffer, |
| size_t* header_length) const; |
| |
| const RTPVideoHeaderVP9 hdr_; |
| const size_t max_payload_length_; // The max length in bytes of one packet. |
| const uint8_t* payload_; // The payload data to be packetized. |
| size_t payload_size_; // The size in bytes of the payload data. |
| const size_t last_packet_reduction_len_; |
| PacketInfoQueue packets_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9); |
| }; |
| |
| class RtpDepacketizerVp9 : public RtpDepacketizer { |
| public: |
| ~RtpDepacketizerVp9() override = default; |
| |
| bool Parse(ParsedPayload* parsed_payload, |
| const uint8_t* payload, |
| size_t payload_length) override; |
| }; |
| |
| } // namespace webrtc |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_ |