| /* | |
| * Copyright (c) 2014 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 MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | |
| #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ | |
| #include <stdint.h> | |
| #include <cstddef> | |
| #include <memory> | |
| #include <span> | |
| #include <vector> | |
| #include "modules/rtp_rtcp/source/rtp_video_header.h" | |
| namespace webrtc { | |
| class RtpPacketToSend; | |
| class RtpPacketizer { | |
| public: | |
| struct PayloadSizeLimits { | |
| int max_payload_len = 1200; | |
| int first_packet_reduction_len = 0; | |
| int last_packet_reduction_len = 0; | |
| // Reduction len for packet that is first & last at the same time. | |
| int single_packet_reduction_len = 0; | |
| }; | |
| enum class PacketizationFormat { | |
| kGeneric, | |
| kRaw, | |
| kH264, | |
| kH265, | |
| kVP8, | |
| kVP9, | |
| kAV1, | |
| }; | |
| static std::unique_ptr<RtpPacketizer> Create( | |
| PacketizationFormat format, | |
| std::span<const uint8_t> payload, | |
| PayloadSizeLimits limits, | |
| // Codec-specific details. | |
| const RTPVideoHeader& rtp_video_header); | |
| virtual ~RtpPacketizer() = default; | |
| // Returns number of remaining packets to produce by the packetizer. | |
| virtual size_t NumPackets() const = 0; | |
| // Get the next payload with payload header. | |
| // Write payload and set marker bit of the `packet`. | |
| // Returns true on success, false otherwise. | |
| virtual bool NextPacket(RtpPacketToSend* packet) = 0; | |
| // Split payload_len into sum of integers with respect to `limits`. | |
| // Returns empty vector on failure. | |
| static std::vector<int> SplitAboutEqually(int payload_len, | |
| const PayloadSizeLimits& limits); | |
| }; | |
| } // namespace webrtc | |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ |