Send and Receive VideoFrameTrackingid RTP header extension.
Bug: webrtc:12594
Change-Id: I2372a361e55d0fdadf9847081644b6a3359a2928
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212283
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/master@{#33570}
diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc
index 92f99e9..8a18f89 100644
--- a/api/rtp_parameters.cc
+++ b/api/rtp_parameters.cc
@@ -130,6 +130,7 @@
constexpr char RtpExtension::kMidUri[];
constexpr char RtpExtension::kRidUri[];
constexpr char RtpExtension::kRepairedRidUri[];
+constexpr char RtpExtension::kVideoFrameTrackingIdUri[];
constexpr int RtpExtension::kMinId;
constexpr int RtpExtension::kMaxId;
@@ -164,7 +165,8 @@
uri == webrtc::RtpExtension::kColorSpaceUri ||
uri == webrtc::RtpExtension::kRidUri ||
uri == webrtc::RtpExtension::kRepairedRidUri ||
- uri == webrtc::RtpExtension::kVideoLayersAllocationUri;
+ uri == webrtc::RtpExtension::kVideoLayersAllocationUri ||
+ uri == webrtc::RtpExtension::kVideoFrameTrackingIdUri;
}
bool RtpExtension::IsEncryptionSupported(absl::string_view uri) {
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index df0e7a9..7fe9f2b 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -353,6 +353,10 @@
static constexpr char kRepairedRidUri[] =
"urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
+ // Header extension to propagate webrtc::VideoFrame id field
+ static constexpr char kVideoFrameTrackingIdUri[] =
+ "http://www.webrtc.org/experiments/rtp-hdrext/video-frame-tracking-id";
+
// Inclusive min and max IDs for two-byte header extensions and one-byte
// header extensions, per RFC8285 Section 4.2-4.3.
static constexpr int kMinId = 1;
diff --git a/api/video/encoded_image.h b/api/video/encoded_image.h
index 3063366..dae4e3a 100644
--- a/api/video/encoded_image.h
+++ b/api/video/encoded_image.h
@@ -109,6 +109,15 @@
color_space_ = color_space;
}
+ // These methods along with the private member video_frame_tracking_id_ are
+ // meant for media quality testing purpose only.
+ absl::optional<uint16_t> VideoFrameTrackingId() const {
+ return video_frame_tracking_id_;
+ }
+ void SetVideoFrameTrackingId(absl::optional<uint16_t> tracking_id) {
+ video_frame_tracking_id_ = tracking_id;
+ }
+
const RtpPacketInfos& PacketInfos() const { return packet_infos_; }
void SetPacketInfos(RtpPacketInfos packet_infos) {
packet_infos_ = std::move(packet_infos);
@@ -182,6 +191,9 @@
absl::optional<int> spatial_index_;
std::map<int, size_t> spatial_layer_frame_size_bytes_;
absl::optional<webrtc::ColorSpace> color_space_;
+ // This field is meant for media quality testing purpose only. When enabled it
+ // carries the webrtc::VideoFrame id field from the sender to the receiver.
+ absl::optional<uint16_t> video_frame_tracking_id_;
// Information about packets used to assemble this video frame. This is needed
// by |SourceTracker| when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
diff --git a/call/rtp_payload_params.cc b/call/rtp_payload_params.cc
index bb086de..18b1138 100644
--- a/call/rtp_payload_params.cc
+++ b/call/rtp_payload_params.cc
@@ -165,6 +165,7 @@
rtp_video_header.color_space = image.ColorSpace()
? absl::make_optional(*image.ColorSpace())
: absl::nullopt;
+ rtp_video_header.video_frame_tracking_id = image.VideoFrameTrackingId();
SetVideoTiming(image, &rtp_video_header.video_timing);
const bool is_keyframe = image._frameType == VideoFrameType::kVideoFrameKey;
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 8fbd5ec..c260de4 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -686,6 +686,12 @@
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
+ result.emplace_back(
+ webrtc::RtpExtension::kVideoFrameTrackingIdUri, id++,
+ IsEnabled(trials_, "WebRTC-VideoFrameTrackingIdAdvertised")
+ ? webrtc::RtpTransceiverDirection::kSendRecv
+ : webrtc::RtpTransceiverDirection::kStopped);
+
return result;
}
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index cf03490..a4176ba 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -397,6 +397,17 @@
ExpectRtpCapabilitySupport(RtpExtension::kVideoLayersAllocationUri, true);
}
+class WebRtcVideoFrameTrackingId : public WebRtcVideoEngineTest {
+ public:
+ WebRtcVideoFrameTrackingId()
+ : WebRtcVideoEngineTest(
+ "WebRTC-VideoFrameTrackingIdAdvertised/Enabled/") {}
+};
+
+TEST_F(WebRtcVideoFrameTrackingId, AdvertiseVideoFrameTrackingId) {
+ ExpectRtpCapabilitySupport(RtpExtension::kVideoFrameTrackingIdUri, true);
+}
+
TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) {
// Allocate the source first to prevent early destruction before channel's
// dtor is called.
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 602cf9d..3b992dc 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -447,6 +447,11 @@
send_allocation_ == SendVideoLayersAllocation::kSendWithResolution;
packet->SetExtension<RtpVideoLayersAllocationExtension>(allocation);
}
+
+ if (first_packet && video_header.video_frame_tracking_id) {
+ packet->SetExtension<VideoFrameTrackingIdExtension>(
+ *video_header.video_frame_tracking_id);
+ }
}
bool RTPSenderVideo::SendVideo(
diff --git a/modules/rtp_rtcp/source/rtp_video_header.h b/modules/rtp_rtcp/source/rtp_video_header.h
index 8a2fcba..c1be76f 100644
--- a/modules/rtp_rtcp/source/rtp_video_header.h
+++ b/modules/rtp_rtcp/source/rtp_video_header.h
@@ -77,6 +77,9 @@
VideoPlayoutDelay playout_delay;
VideoSendTiming video_timing;
absl::optional<ColorSpace> color_space;
+ // This field is meant for media quality testing purpose only. When enabled it
+ // carries the webrtc::VideoFrame id field from the sender to the receiver.
+ absl::optional<uint16_t> video_frame_tracking_id;
RTPVideoTypeHeader video_type_header;
};
diff --git a/modules/video_coding/frame_object.cc b/modules/video_coding/frame_object.cc
index 8d011b9..d226dcd 100644
--- a/modules/video_coding/frame_object.cc
+++ b/modules/video_coding/frame_object.cc
@@ -68,6 +68,7 @@
rotation_ = rotation;
SetColorSpace(color_space);
+ SetVideoFrameTrackingId(rtp_video_header_.video_frame_tracking_id);
content_type_ = content_type;
if (timing.flags != VideoSendTiming::kInvalid) {
// ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
diff --git a/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc b/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
index 0cfc99d..4593df7 100644
--- a/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
+++ b/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
@@ -51,6 +51,7 @@
buffer->data()[insertion_pos + 2] = info.sub_id;
EncodedImage out = source;
+ out.SetVideoFrameTrackingId(id);
out.SetEncodedData(buffer);
return out;
}
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 12ac57a..be208ce 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -561,6 +561,8 @@
video_header.color_space = last_color_space_;
}
}
+ video_header.video_frame_tracking_id =
+ rtp_packet.GetExtension<VideoFrameTrackingIdExtension>();
if (loss_notification_controller_) {
if (rtp_packet.recovered()) {
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index f02dccd..8df76d5 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -528,6 +528,8 @@
video_header.color_space = last_color_space_;
}
}
+ video_header.video_frame_tracking_id =
+ rtp_packet.GetExtension<VideoFrameTrackingIdExtension>();
if (loss_notification_controller_) {
if (rtp_packet.recovered()) {
