Parse extension and store it in RTPVideoHeader
Bug: webrtc:358039777
Change-Id: Ib70046662877efa5f8d0cbe559b44d138f4733e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364481
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43146}
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 0eb6d6f..fdbe2fa 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -117,6 +117,9 @@
"../call:rtp_sender",
"../call:video_stream_api",
"../common_video",
+ "../common_video:corruption_detection_converters",
+ "../common_video:corruption_detection_message",
+ "../common_video:frame_instrumentation_data",
"../media:media_constants",
"../media:rtc_sdp_video_format_utils",
"../modules:module_api",
@@ -170,6 +173,7 @@
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:variant",
]
if (!build_with_mozilla) {
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 818353c..9c8f28c 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -19,12 +19,17 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
+#include "absl/types/variant.h"
#include "api/video/video_codec_type.h"
+#include "common_video/corruption_detection_converters.h"
+#include "common_video/corruption_detection_message.h"
+#include "common_video/frame_instrumentation_data.h"
#include "media/base/media_constants.h"
#include "modules/pacing/packet_router.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
+#include "modules/rtp_rtcp/source/corruption_detection_extension.h"
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/frame_object.h"
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
@@ -497,6 +502,21 @@
return kHasGenericDescriptor;
}
+void RtpVideoStreamReceiver2::SetLastCorruptionDetectionIndex(
+ const absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>&
+ frame_instrumentation_data) {
+ if (const auto* sync_data = absl::get_if<FrameInstrumentationSyncData>(
+ &frame_instrumentation_data)) {
+ last_corruption_detection_index_ = sync_data->sequence_index;
+ } else if (const auto* data = absl::get_if<FrameInstrumentationData>(
+ &frame_instrumentation_data)) {
+ last_corruption_detection_index_ =
+ data->sequence_index + data->sample_values.size();
+ } else {
+ RTC_DCHECK_NOTREACHED();
+ }
+}
+
bool RtpVideoStreamReceiver2::OnReceivedPayloadData(
rtc::CopyOnWriteBuffer codec_payload,
const RtpPacketReceived& rtp_packet,
@@ -582,9 +602,7 @@
return false;
}
- // Color space should only be transmitted in the last packet of a frame,
- // therefore, neglect it otherwise so that last_color_space_ is not reset by
- // mistake.
+ // Extensions that should only be transmitted in the last packet of a frame.
if (video_header.is_last_packet_in_frame) {
video_header.color_space = rtp_packet.GetExtension<ColorSpaceExtension>();
if (video_header.color_space ||
@@ -596,6 +614,20 @@
} else if (last_color_space_) {
video_header.color_space = last_color_space_;
}
+ CorruptionDetectionMessage message;
+ rtp_packet.GetExtension<CorruptionDetectionExtension>(&message);
+ if (message.sample_values().empty()) {
+ // TODO: bugs.webrtc.org/358039777 - Convert message to sync data and
+ // use that for assignment instead of `std::nullptr`.
+ video_header.frame_instrumentation_data = std::nullopt;
+ } else {
+ video_header.frame_instrumentation_data =
+ ConvertCorruptionDetectionMessageToFrameInstrumentationData(
+ message, last_corruption_detection_index_);
+ }
+ if (video_header.frame_instrumentation_data.has_value()) {
+ SetLastCorruptionDetectionIndex(*video_header.frame_instrumentation_data);
+ }
}
video_header.video_frame_tracking_id =
rtp_packet.GetExtension<VideoFrameTrackingIdExtension>();
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 141f15c..e996bb2 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -17,6 +17,7 @@
#include <string>
#include <vector>
+#include "absl/types/variant.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/environment/environment.h"
#include "api/sequence_checker.h"
@@ -26,6 +27,7 @@
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "call/video_receive_stream.h"
+#include "common_video/frame_instrumentation_data.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/recovered_packet_receiver.h"
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
@@ -313,6 +315,10 @@
void UpdatePacketReceiveTimestamps(const RtpPacketReceived& packet,
bool is_keyframe)
RTC_RUN_ON(packet_sequence_checker_);
+ void SetLastCorruptionDetectionIndex(
+ const absl::variant<FrameInstrumentationSyncData,
+ FrameInstrumentationData>&
+ frame_instrumentation_data);
const Environment env_;
TaskQueueBase* const worker_queue_;
@@ -445,6 +451,7 @@
Timestamp next_keyframe_request_for_missing_video_structure_ =
Timestamp::MinusInfinity();
bool sps_pps_idr_is_h264_keyframe_ = false;
+ int last_corruption_detection_index_ = 0;
};
} // namespace webrtc
