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0cb858c7f2
Bug: None Change-Id: I84898d3c1ce71d907bb91e560223a31384534a39 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123231 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26996}
73 lines
2.1 KiB
C++
73 lines
2.1 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_VIDEO_CODING_PACKET_H_
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#define MODULES_VIDEO_CODING_PACKET_H_
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#include <stddef.h>
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#include <stdint.h>
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#include "absl/types/optional.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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namespace webrtc {
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class VCMPacket {
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public:
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VCMPacket();
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VCMPacket(const uint8_t* ptr, size_t size, const WebRtcRTPHeader& rtpHeader);
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VCMPacket(const uint8_t* ptr,
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size_t size,
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const RTPHeader& rtp_header,
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const RTPVideoHeader& video_header,
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FrameType frame_type,
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int64_t ntp_time_ms);
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~VCMPacket();
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VideoCodecType codec() const { return video_header.codec; }
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int width() const { return video_header.width; }
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int height() const { return video_header.height; }
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bool is_first_packet_in_frame() const {
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return video_header.is_first_packet_in_frame;
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}
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bool is_last_packet_in_frame() const {
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return video_header.is_last_packet_in_frame;
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}
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uint8_t payloadType;
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uint32_t timestamp;
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// NTP time of the capture time in local timebase in milliseconds.
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int64_t ntp_time_ms_;
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uint16_t seqNum;
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const uint8_t* dataPtr;
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size_t sizeBytes;
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bool markerBit;
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int timesNacked;
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FrameType frameType;
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VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
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bool insertStartCode; // True if a start code should be inserted before this
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// packet.
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RTPVideoHeader video_header;
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absl::optional<RtpGenericFrameDescriptor> generic_descriptor;
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int64_t receive_time_ms;
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};
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_PACKET_H_
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