1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ 12 #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ 13 14 #include <stdint.h> 15 16 #include <memory> 17 #include <vector> 18 19 #include "absl/types/optional.h" 20 #include "api/array_view.h" 21 #include "modules/rtp_rtcp/source/rtp_video_header.h" 22 23 namespace webrtc { 24 25 class RtpPacketToSend; 26 27 class RtpPacketizer { 28 public: 29 struct PayloadSizeLimits { 30 int max_payload_len = 1200; 31 int first_packet_reduction_len = 0; 32 int last_packet_reduction_len = 0; 33 // Reduction len for packet that is first & last at the same time. 34 int single_packet_reduction_len = 0; 35 }; 36 37 // If type is not set, returns a raw packetizer. 38 static std::unique_ptr<RtpPacketizer> Create( 39 absl::optional<VideoCodecType> type, 40 rtc::ArrayView<const uint8_t> payload, 41 PayloadSizeLimits limits, 42 // Codec-specific details. 43 const RTPVideoHeader& rtp_video_header); 44 45 virtual ~RtpPacketizer() = default; 46 47 // Returns number of remaining packets to produce by the packetizer. 48 virtual size_t NumPackets() const = 0; 49 50 // Get the next payload with payload header. 51 // Write payload and set marker bit of the `packet`. 52 // Returns true on success, false otherwise. 53 virtual bool NextPacket(RtpPacketToSend* packet) = 0; 54 55 // Split payload_len into sum of integers with respect to `limits`. 56 // Returns empty vector on failure. 57 static std::vector<int> SplitAboutEqually(int payload_len, 58 const PayloadSizeLimits& limits); 59 }; 60 } // namespace webrtc 61 #endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_ 62