/* * Copyright (c) 2011 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_VIDEO_CODING_DEPRECATED_PACKET_H_ #define MODULES_VIDEO_CODING_DEPRECATED_PACKET_H_ #include #include #include "absl/types/optional.h" #include "api/rtp_headers.h" #include "api/rtp_packet_info.h" #include "api/units/timestamp.h" #include "api/video/video_frame_type.h" #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h" #include "modules/rtp_rtcp/source/rtp_video_header.h" namespace webrtc { // Used to indicate if a received packet contain a complete NALU (or equivalent) enum VCMNaluCompleteness { kNaluUnset = 0, // Packet has not been filled. kNaluComplete = 1, // Packet can be decoded as is. kNaluStart, // Packet contain beginning of NALU kNaluIncomplete, // Packet is not beginning or end of NALU kNaluEnd, // Packet is the end of a NALU }; class VCMPacket { public: VCMPacket(); VCMPacket(const uint8_t* ptr, size_t size, const RTPHeader& rtp_header, const RTPVideoHeader& video_header, int64_t ntp_time_ms, Timestamp receive_time); ~VCMPacket(); VideoCodecType codec() const { return video_header.codec; } int width() const { return video_header.width; } int height() const { return video_header.height; } bool is_first_packet_in_frame() const { return video_header.is_first_packet_in_frame; } bool is_last_packet_in_frame() const { return video_header.is_last_packet_in_frame; } uint8_t payloadType; uint32_t timestamp; // NTP time of the capture time in local timebase in milliseconds. int64_t ntp_time_ms_; uint16_t seqNum; const uint8_t* dataPtr; size_t sizeBytes; bool markerBit; int timesNacked; VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete. bool insertStartCode; // True if a start code should be inserted before this // packet. RTPVideoHeader video_header; absl::optional generic_descriptor; RtpPacketInfo packet_info; }; } // namespace webrtc #endif // MODULES_VIDEO_CODING_DEPRECATED_PACKET_H_