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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/video_coding/packet.h | |
parent | Initial commit. (diff) | |
download | firefox-upstream.tar.xz firefox-upstream.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/video_coding/packet.h')
-rw-r--r-- | third_party/libwebrtc/modules/video_coding/packet.h | 80 |
1 files changed, 80 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/video_coding/packet.h b/third_party/libwebrtc/modules/video_coding/packet.h new file mode 100644 index 0000000000..9aa2d5ce08 --- /dev/null +++ b/third_party/libwebrtc/modules/video_coding/packet.h @@ -0,0 +1,80 @@ +/* + * 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_PACKET_H_ +#define MODULES_VIDEO_CODING_PACKET_H_ + +#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 { + 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<RtpGenericFrameDescriptor> generic_descriptor; + + RtpPacketInfo packet_info; +}; + +} // namespace webrtc +#endif // MODULES_VIDEO_CODING_PACKET_H_ |