Do not propagate generic descriptor on receiving frame
It was used only for the frame decryptor. Decryptor needs only raw representation that it can recreate in a way compatible with the new version of the descriptor. This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f. with adjustments. Change-Id: I935977179bef31d8e1023964b967658e9a7db92d Bug: webrtc:10342 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30532}
This commit is contained in:

committed by
Commit Bot

parent
135d9a386e
commit
e209fe6c68
@ -165,6 +165,8 @@ rtc_library("rtp_rtcp") {
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"source/rtcp_receiver.h",
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"source/rtcp_sender.cc",
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"source/rtcp_sender.h",
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"source/rtp_descriptor_authentication.cc",
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"source/rtp_descriptor_authentication.h",
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"source/rtp_format.cc",
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"source/rtp_format.h",
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"source/rtp_format_h264.cc",
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58
modules/rtp_rtcp/source/rtp_descriptor_authentication.cc
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58
modules/rtp_rtcp/source/rtp_descriptor_authentication.cc
Normal file
@ -0,0 +1,58 @@
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/*
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* Copyright (c) 2020 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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#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
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#include <cstdint>
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#include <vector>
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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namespace webrtc {
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std::vector<uint8_t> RtpDescriptorAuthentication(
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const RTPVideoHeader& rtp_video_header) {
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if (!rtp_video_header.generic) {
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return {};
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}
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const RTPVideoHeader::GenericDescriptorInfo& descriptor =
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*rtp_video_header.generic;
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// Default way of creating additional data for an encrypted frame.
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if (descriptor.spatial_index < 0 || descriptor.temporal_index < 0 ||
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descriptor.spatial_index >=
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RtpGenericFrameDescriptor::kMaxSpatialLayers ||
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descriptor.temporal_index >=
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RtpGenericFrameDescriptor::kMaxTemporalLayers ||
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descriptor.dependencies.size() >
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RtpGenericFrameDescriptor::kMaxNumFrameDependencies) {
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return {};
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}
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RtpGenericFrameDescriptor frame_descriptor;
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frame_descriptor.SetFirstPacketInSubFrame(true);
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frame_descriptor.SetLastPacketInSubFrame(false);
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frame_descriptor.SetTemporalLayer(descriptor.temporal_index);
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frame_descriptor.SetSpatialLayersBitmask(1 << descriptor.spatial_index);
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frame_descriptor.SetFrameId(descriptor.frame_id & 0xFFFF);
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for (int64_t dependency : descriptor.dependencies) {
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frame_descriptor.AddFrameDependencyDiff(descriptor.frame_id - dependency);
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}
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if (descriptor.dependencies.empty()) {
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frame_descriptor.SetResolution(rtp_video_header.width,
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rtp_video_header.height);
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}
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std::vector<uint8_t> result(
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RtpGenericFrameDescriptorExtension00::ValueSize(frame_descriptor));
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RtpGenericFrameDescriptorExtension00::Write(result, frame_descriptor);
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return result;
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}
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} // namespace webrtc
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27
modules/rtp_rtcp/source/rtp_descriptor_authentication.h
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27
modules/rtp_rtcp/source/rtp_descriptor_authentication.h
Normal file
@ -0,0 +1,27 @@
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/*
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* Copyright (c) 2020 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_RTP_RTCP_SOURCE_RTP_DESCRIPTOR_AUTHENTICATION_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_DESCRIPTOR_AUTHENTICATION_H_
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#include <cstdint>
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#include <vector>
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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namespace webrtc {
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// Converts frame dependencies into array of bytes for authentication.
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std::vector<uint8_t> RtpDescriptorAuthentication(
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const RTPVideoHeader& rtp_video_header);
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_DESCRIPTOR_AUTHENTICATION_H_
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@ -97,22 +97,4 @@ bool RtpGenericFrameDescriptor::AddFrameDependencyDiff(uint16_t fdiff) {
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return true;
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}
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void RtpGenericFrameDescriptor::SetByteRepresentation(
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rtc::ArrayView<const uint8_t> byte_representation) {
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RTC_CHECK(!byte_representation.empty());
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byte_representation_.assign(byte_representation.begin(),
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byte_representation.end());
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// Clear end_of_subframe bit.
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// Because ByteRepresentation is used for frame authentication, bit describing
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// position of the packet in the frame shouldn't be part of it.
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// This match RtpVideoSender where descriptor is passed for authentication
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// before end_of_subframe bit is decided and set, i.e. it is always 0.
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byte_representation_[0] &= ~0x40;
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}
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rtc::ArrayView<const uint8_t>
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RtpGenericFrameDescriptor::GetByteRepresentation() {
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return byte_representation_;
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}
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} // namespace webrtc
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@ -70,9 +70,6 @@ class RtpGenericFrameDescriptor {
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// Returns false on failure, i.e. number of dependencies is too large.
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bool AddFrameDependencyDiff(uint16_t fdiff);
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void SetByteRepresentation(rtc::ArrayView<const uint8_t> representation);
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rtc::ArrayView<const uint8_t> GetByteRepresentation();
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private:
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bool beginning_of_subframe_ = false;
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bool end_of_subframe_ = false;
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@ -86,8 +83,6 @@ class RtpGenericFrameDescriptor {
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uint16_t frame_deps_id_diffs_[kMaxNumFrameDependencies];
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int width_ = 0;
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int height_ = 0;
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std::vector<uint8_t> byte_representation_;
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};
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} // namespace webrtc
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@ -84,6 +84,7 @@ rtc_library("video_coding") {
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"../../api/video:encoded_image",
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"../../api/video:video_bitrate_allocation",
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"../../api/video:video_bitrate_allocator_factory",
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"../../rtc_base:deprecation",
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"../../system_wrappers:field_trial",
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"../../system_wrappers:metrics",
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"../rtp_rtcp:rtp_video_header",
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@ -37,7 +37,6 @@ RtpFrameObject::RtpFrameObject(
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VideoContentType content_type,
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const RTPVideoHeader& video_header,
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const absl::optional<webrtc::ColorSpace>& color_space,
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const absl::optional<RtpGenericFrameDescriptor>& generic_descriptor,
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RtpPacketInfos packet_infos,
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rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
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: first_seq_num_(first_seq_num),
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@ -45,7 +44,6 @@ RtpFrameObject::RtpFrameObject(
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last_packet_received_time_(last_packet_received_time),
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times_nacked_(times_nacked) {
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rtp_video_header_ = video_header;
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rtp_generic_frame_descriptor_ = generic_descriptor;
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// EncodedFrame members
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codec_type_ = codec;
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@ -92,6 +90,43 @@ RtpFrameObject::RtpFrameObject(
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is_last_spatial_layer = markerBit;
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}
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RtpFrameObject::RtpFrameObject(
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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bool markerBit,
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int times_nacked,
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int64_t first_packet_received_time,
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int64_t last_packet_received_time,
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uint32_t rtp_timestamp,
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int64_t ntp_time_ms,
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const VideoSendTiming& timing,
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uint8_t payload_type,
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VideoCodecType codec,
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VideoRotation rotation,
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VideoContentType content_type,
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const RTPVideoHeader& video_header,
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const absl::optional<webrtc::ColorSpace>& color_space,
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const absl::optional<RtpGenericFrameDescriptor>& /*generic_descriptor*/,
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RtpPacketInfos packet_infos,
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rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
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: RtpFrameObject(first_seq_num,
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last_seq_num,
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markerBit,
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times_nacked,
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first_packet_received_time,
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last_packet_received_time,
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rtp_timestamp,
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ntp_time_ms,
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timing,
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payload_type,
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codec,
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rotation,
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content_type,
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video_header,
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color_space,
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std::move(packet_infos),
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std::move(image_buffer)) {}
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RtpFrameObject::~RtpFrameObject() {
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}
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@ -131,11 +166,6 @@ const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const {
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return rtp_video_header_;
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}
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const absl::optional<RtpGenericFrameDescriptor>&
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RtpFrameObject::GetGenericFrameDescriptor() const {
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return rtp_generic_frame_descriptor_;
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}
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const FrameMarking& RtpFrameObject::GetFrameMarking() const {
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return rtp_video_header_.frame_marking;
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}
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@ -14,12 +14,32 @@
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#include "absl/types/optional.h"
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#include "api/video/encoded_frame.h"
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
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#include "rtc_base/deprecation.h"
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namespace webrtc {
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namespace video_coding {
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class RtpFrameObject : public EncodedFrame {
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public:
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RtpFrameObject(uint16_t first_seq_num,
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uint16_t last_seq_num,
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bool markerBit,
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int times_nacked,
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int64_t first_packet_received_time,
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int64_t last_packet_received_time,
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uint32_t rtp_timestamp,
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int64_t ntp_time_ms,
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const VideoSendTiming& timing,
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uint8_t payload_type,
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VideoCodecType codec,
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VideoRotation rotation,
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VideoContentType content_type,
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const RTPVideoHeader& video_header,
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const absl::optional<webrtc::ColorSpace>& color_space,
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RtpPacketInfos packet_infos,
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rtc::scoped_refptr<EncodedImageBuffer> image_buffer);
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RTC_DEPRECATED
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RtpFrameObject(
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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@ -50,13 +70,10 @@ class RtpFrameObject : public EncodedFrame {
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int64_t RenderTime() const override;
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bool delayed_by_retransmission() const override;
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const RTPVideoHeader& GetRtpVideoHeader() const;
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const absl::optional<RtpGenericFrameDescriptor>& GetGenericFrameDescriptor()
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const;
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const FrameMarking& GetFrameMarking() const;
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private:
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RTPVideoHeader rtp_video_header_;
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absl::optional<RtpGenericFrameDescriptor> rtp_generic_frame_descriptor_;
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VideoCodecType codec_type_;
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uint16_t first_seq_num_;
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uint16_t last_seq_num_;
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@ -473,7 +473,6 @@ std::unique_ptr<RtpFrameObject> PacketBuffer::AssembleFrame(
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last_packet.video_header.content_type, //
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first_packet.video_header, //
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last_packet.video_header.color_space, //
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first_packet.generic_descriptor, //
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RtpPacketInfos(std::move(packet_infos)), //
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std::move(bitstream));
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}
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@ -19,7 +19,6 @@
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#include "absl/base/attributes.h"
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#include "api/rtp_packet_info.h"
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#include "api/video/encoded_image.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_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.h"
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#include "modules/video_coding/frame_object.h"
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@ -67,7 +66,6 @@ class PacketBuffer {
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rtc::CopyOnWriteBuffer video_payload;
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RTPVideoHeader video_header;
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absl::optional<RtpGenericFrameDescriptor> generic_descriptor;
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RtpPacketInfo packet_info;
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};
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@ -57,7 +57,6 @@ std::unique_ptr<RtpFrameObject> CreateFrame(
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VideoContentType::UNSPECIFIED,
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video_header,
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/*color_space=*/absl::nullopt,
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/*generic_descriptor=*/absl::nullopt,
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RtpPacketInfos(),
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EncodedImageBuffer::Create(/*size=*/0));
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// clang-format on
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@ -63,25 +63,31 @@ class NullCallback : public video_coding::OnCompleteFrameCallback {
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std::unique_ptr<video_coding::EncodedFrame> frame) override {}
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};
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RtpGenericFrameDescriptor GenerateRtpGenericFrameDescriptor(
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DataReader* reader) {
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RtpGenericFrameDescriptor res;
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res.SetFirstPacketInSubFrame(true);
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res.SetFrameId(reader->GetNum<uint16_t>());
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int spatial_layer =
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reader->GetNum<uint8_t>() % RtpGenericFrameDescriptor::kMaxSpatialLayers;
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res.SetSpatialLayersBitmask(1 << spatial_layer);
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res.SetTemporalLayer(reader->GetNum<uint8_t>() %
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RtpGenericFrameDescriptor::kMaxTemporalLayers);
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int num_diffs = (reader->GetNum<uint8_t>() %
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RtpGenericFrameDescriptor::kMaxNumFrameDependencies);
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for (int i = 0; i < num_diffs; ++i) {
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res.AddFrameDependencyDiff(reader->GetNum<uint16_t>() % (1 << 14));
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absl::optional<RTPVideoHeader::GenericDescriptorInfo>
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GenerateGenericFrameDependencies(DataReader* reader) {
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absl::optional<RTPVideoHeader::GenericDescriptorInfo> result;
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uint8_t flags = reader->GetNum<uint8_t>();
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if (flags & 0b1000'0000) {
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// i.e. with 50% chance there are no generic dependencies.
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// in such case codec-specfic code path of the RtpFrameReferenceFinder will
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// be validated.
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return result;
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}
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return res;
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result.emplace();
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result->frame_id = reader->GetNum<int64_t>();
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result->spatial_index = (flags & 0b0111'0000) >> 4;
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result->temporal_index = (flags & 0b0000'1110) >> 1;
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result->discardable = (flags & 0b0000'0001);
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// Larger than supported by the RtpFrameReferenceFinder.
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int num_diffs = (reader->GetNum<uint8_t>() % 16);
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for (int i = 0; i < num_diffs; ++i) {
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result->dependencies.push_back(result->frame_id -
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(reader->GetNum<uint16_t>() % (1 << 14)));
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}
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return result;
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}
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} // namespace
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@ -90,7 +96,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
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NullCallback cb;
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video_coding::RtpFrameReferenceFinder reference_finder(&cb);
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auto codec = static_cast<VideoCodecType>(reader.GetNum<uint8_t>() % 4);
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auto codec = static_cast<VideoCodecType>(reader.GetNum<uint8_t>() % 5);
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while (reader.MoreToRead()) {
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uint16_t first_seq_num = reader.GetNum<uint16_t>();
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@ -128,6 +134,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
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}
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reader.CopyTo(&video_header.frame_marking);
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video_header.generic = GenerateGenericFrameDependencies(&reader);
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// clang-format off
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auto frame = std::make_unique<video_coding::RtpFrameObject>(
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@ -146,7 +153,6 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
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VideoContentType::UNSPECIFIED,
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video_header,
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/*color_space=*/absl::nullopt,
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GenerateRtpGenericFrameDescriptor(&reader),
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RtpPacketInfos(),
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EncodedImageBuffer::Create(/*size=*/0));
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// clang-format on
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@ -11,7 +11,10 @@
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#include "video/buffered_frame_decryptor.h"
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#include <utility>
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#include <vector>
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#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
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#include "modules/video_coding/frame_object.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/field_trial.h"
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@ -60,9 +63,7 @@ BufferedFrameDecryptor::FrameDecision BufferedFrameDecryptor::DecryptFrame(
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return FrameDecision::kStash;
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}
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// When using encryption we expect the frame to have the generic descriptor.
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absl::optional<RtpGenericFrameDescriptor> descriptor =
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frame->GetGenericFrameDescriptor();
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if (!descriptor) {
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if (frame->GetRtpVideoHeader().generic == absl::nullopt) {
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RTC_LOG(LS_ERROR) << "No generic frame descriptor found dropping frame.";
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return FrameDecision::kDrop;
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}
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@ -76,9 +77,9 @@ BufferedFrameDecryptor::FrameDecision BufferedFrameDecryptor::DecryptFrame(
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max_plaintext_byte_size);
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// Only enable authenticating the header if the field trial is enabled.
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rtc::ArrayView<const uint8_t> additional_data;
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std::vector<uint8_t> additional_data;
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if (generic_descriptor_auth_experiment_) {
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additional_data = descriptor->GetByteRepresentation();
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additional_data = RtpDescriptorAuthentication(frame->GetRtpVideoHeader());
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}
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// Attempt to decrypt the video frame.
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|
@ -57,6 +57,8 @@ class BufferedFrameDecryptorTest : public ::testing::Test,
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std::unique_ptr<video_coding::RtpFrameObject> CreateRtpFrameObject(
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bool key_frame) {
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seq_num_++;
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RTPVideoHeader rtp_video_header;
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rtp_video_header.generic.emplace();
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// clang-format off
|
||||
return std::make_unique<video_coding::RtpFrameObject>(
|
||||
@ -73,9 +75,8 @@ class BufferedFrameDecryptorTest : public ::testing::Test,
|
||||
kVideoCodecGeneric,
|
||||
kVideoRotation_0,
|
||||
VideoContentType::UNSPECIFIED,
|
||||
RTPVideoHeader(),
|
||||
rtp_video_header,
|
||||
/*color_space=*/absl::nullopt,
|
||||
RtpGenericFrameDescriptor(),
|
||||
RtpPacketInfos(),
|
||||
EncodedImageBuffer::Create(/*size=*/0));
|
||||
// clang-format on
|
||||
|
@ -28,6 +28,7 @@
|
||||
#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
|
||||
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_format.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
@ -367,51 +368,43 @@ void RtpVideoStreamReceiver::OnReceivedPayloadData(
|
||||
rtp_packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay);
|
||||
rtp_packet.GetExtension<FrameMarkingExtension>(&video_header.frame_marking);
|
||||
|
||||
RtpGenericFrameDescriptor& generic_descriptor =
|
||||
packet->generic_descriptor.emplace();
|
||||
if (rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension01>(
|
||||
&generic_descriptor)) {
|
||||
if (rtp_packet.HasExtension<RtpGenericFrameDescriptorExtension00>()) {
|
||||
RTC_LOG(LS_WARNING) << "RTP packet had two different GFD versions.";
|
||||
return;
|
||||
}
|
||||
generic_descriptor.SetByteRepresentation(
|
||||
rtp_packet.GetRawExtension<RtpGenericFrameDescriptorExtension01>());
|
||||
} else if ((rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>(
|
||||
&generic_descriptor))) {
|
||||
generic_descriptor.SetByteRepresentation(
|
||||
rtp_packet.GetRawExtension<RtpGenericFrameDescriptorExtension00>());
|
||||
} else {
|
||||
packet->generic_descriptor = absl::nullopt;
|
||||
if (rtp_packet.HasExtension<RtpGenericFrameDescriptorExtension00>() &&
|
||||
rtp_packet.HasExtension<RtpGenericFrameDescriptorExtension01>()) {
|
||||
RTC_LOG(LS_WARNING) << "RTP packet had two different GFD versions.";
|
||||
return;
|
||||
}
|
||||
if (packet->generic_descriptor != absl::nullopt) {
|
||||
video_header.is_first_packet_in_frame =
|
||||
packet->generic_descriptor->FirstPacketInSubFrame();
|
||||
video_header.is_last_packet_in_frame =
|
||||
packet->generic_descriptor->LastPacketInSubFrame();
|
||||
|
||||
if (packet->generic_descriptor->FirstPacketInSubFrame()) {
|
||||
RtpGenericFrameDescriptor generic_descriptor;
|
||||
bool has_generic_descriptor =
|
||||
rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension01>(
|
||||
&generic_descriptor) ||
|
||||
rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>(
|
||||
&generic_descriptor);
|
||||
if (has_generic_descriptor) {
|
||||
video_header.is_first_packet_in_frame =
|
||||
generic_descriptor.FirstPacketInSubFrame();
|
||||
video_header.is_last_packet_in_frame =
|
||||
generic_descriptor.LastPacketInSubFrame();
|
||||
|
||||
if (generic_descriptor.FirstPacketInSubFrame()) {
|
||||
video_header.frame_type =
|
||||
packet->generic_descriptor->FrameDependenciesDiffs().empty()
|
||||
generic_descriptor.FrameDependenciesDiffs().empty()
|
||||
? VideoFrameType::kVideoFrameKey
|
||||
: VideoFrameType::kVideoFrameDelta;
|
||||
|
||||
auto& descriptor = video_header.generic.emplace();
|
||||
int64_t frame_id =
|
||||
frame_id_unwrapper_.Unwrap(packet->generic_descriptor->FrameId());
|
||||
frame_id_unwrapper_.Unwrap(generic_descriptor.FrameId());
|
||||
descriptor.frame_id = frame_id;
|
||||
descriptor.spatial_index = packet->generic_descriptor->SpatialLayer();
|
||||
descriptor.temporal_index = packet->generic_descriptor->TemporalLayer();
|
||||
descriptor.discardable =
|
||||
packet->generic_descriptor->Discardable().value_or(false);
|
||||
for (uint16_t fdiff :
|
||||
packet->generic_descriptor->FrameDependenciesDiffs()) {
|
||||
descriptor.spatial_index = generic_descriptor.SpatialLayer();
|
||||
descriptor.temporal_index = generic_descriptor.TemporalLayer();
|
||||
descriptor.discardable = generic_descriptor.Discardable().value_or(false);
|
||||
for (uint16_t fdiff : generic_descriptor.FrameDependenciesDiffs()) {
|
||||
descriptor.dependencies.push_back(frame_id - fdiff);
|
||||
}
|
||||
}
|
||||
|
||||
video_header.width = packet->generic_descriptor->Width();
|
||||
video_header.height = packet->generic_descriptor->Height();
|
||||
video_header.width = generic_descriptor.Width();
|
||||
video_header.height = generic_descriptor.Height();
|
||||
}
|
||||
|
||||
// Color space should only be transmitted in the last packet of a frame,
|
||||
@ -435,7 +428,7 @@ void RtpVideoStreamReceiver::OnReceivedPayloadData(
|
||||
// TODO(bugs.webrtc.org/10336): Implement support for reordering.
|
||||
RTC_LOG(LS_INFO)
|
||||
<< "LossNotificationController does not support reordering.";
|
||||
} else if (!packet->generic_descriptor) {
|
||||
} else if (!has_generic_descriptor) {
|
||||
RTC_LOG(LS_WARNING) << "LossNotificationController requires generic "
|
||||
"frame descriptor, but it is missing.";
|
||||
} else {
|
||||
|
Reference in New Issue
Block a user