blob: 9aa2d5ce08c4ee88bb2ea5db81888f91099e3dd5 [file] [log] [blame]
* 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.
#include <stddef.h>
#include <stdint.h>
#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 {
VCMPacket(const uint8_t* ptr,
size_t size,
const RTPHeader& rtp_header,
const RTPVideoHeader& video_header,
int64_t ntp_time_ms,
Timestamp receive_time);
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<RtpGenericFrameDescriptor> generic_descriptor;
RtpPacketInfo packet_info;
} // namespace webrtc