Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )

Reason for revert:
Reland with appropriate changes to API to not break depending projects.

Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/64e739aeae5629cbbebf2a19e1d3e6b452bb6d0b
>
> [email protected],[email protected],[email protected],[email protected],[email protected]
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: https://chromium.googlesource.com/external/webrtc/+/57218668083fce36a55efb8c8f1d31785253b8c0

[email protected],[email protected],[email protected],[email protected],[email protected]
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420

Review-Url: https://codereview.webrtc.org/2812913002
Cr-Commit-Position: refs/heads/master@{#17651}

Author: ilnik

webrtc/api/video/video_content_type.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2017 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 WEBRTC_API_VIDEO_VIDEO_CONTENT_TYPE_H_
+#define WEBRTC_API_VIDEO_VIDEO_CONTENT_TYPE_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+enum class VideoContentType : uint8_t {
+  UNSPECIFIED = 0,
+  SCREENSHARE = 1,
+  TOTAL_CONTENT_TYPES  // Must be the last value in the enum.
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_VIDEO_VIDEO_CONTENT_TYPE_H_

webrtc/common_types.cc

@@ -31,7 +31,9 @@ RTPHeaderExtension::RTPHeaderExtension()
       voiceActivity(false),
       audioLevel(0),
       hasVideoRotation(false),
-      videoRotation(kVideoRotation_0) {}
+      videoRotation(kVideoRotation_0),
+      hasVideoContentType(false),
+      videoContentType(VideoContentType::UNSPECIFIED) {}
 
 RTPHeader::RTPHeader()
     : markerBit(false),

webrtc/common_types.h

@@ -18,6 +18,7 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/api/video/video_content_type.h"
 #include "webrtc/api/video/video_rotation.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/optional.h"
@@ -716,6 +717,11 @@ struct RTPHeaderExtension {
   bool hasVideoRotation;
   VideoRotation videoRotation;
 
+  // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
+  // a corresponding bool flag.
+  bool hasVideoContentType;
+  VideoContentType videoContentType;
+
   PlayoutDelay playout_delay = {-1, -1};
 };
 

webrtc/config.cc

@@ -64,6 +64,10 @@ const char* RtpExtension::kTransportSequenceNumberUri =
     "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
 const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
 
+const char* RtpExtension::kVideoContentTypeUri =
+    "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type";
+const int RtpExtension::kVideoContentTypeDefaultId = 6;
+
 // This extension allows applications to adaptively limit the playout delay
 // on frames as per the current needs. For example, a gaming application
 // has very different needs on end-to-end delay compared to a video-conference
@@ -85,7 +89,8 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
          uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kVideoRotationUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
-         uri == webrtc::RtpExtension::kPlayoutDelayUri;
+         uri == webrtc::RtpExtension::kPlayoutDelayUri ||
+         uri == webrtc::RtpExtension::kVideoContentTypeUri;
 }
 
 VideoStream::VideoStream()

webrtc/config.h

@@ -88,6 +88,10 @@ struct RtpExtension {
   static const char* kVideoRotationUri;
   static const int kVideoRotationDefaultId;
 
+  // Header extension for video content type. E.g. default or screenshare.
+  static const char* kVideoContentTypeUri;
+  static const int kVideoContentTypeDefaultId;
+
   // Header extension for transport sequence number, see url for details:
   // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
   static const char* kTransportSequenceNumberUri;

webrtc/media/engine/webrtcvideoengine2_unittest.cc

@@ -3852,7 +3852,7 @@ TEST_F(WebRtcVideoChannel2Test, ReceiveDifferentUnsignaledSsrc) {
   EXPECT_EQ(rtpHeader.ssrc, recv_stream->GetConfig().rtp.remote_ssrc);
   // Verify that the receive stream sinks to a renderer.
   webrtc::VideoFrame video_frame2(CreateBlackFrameBuffer(4, 4), 200, 0,
-                                 webrtc::kVideoRotation_0);
+                                  webrtc::kVideoRotation_0);
   recv_stream->InjectFrame(video_frame2);
   EXPECT_EQ(2, renderer.num_rendered_frames());
 
@@ -3869,7 +3869,7 @@ TEST_F(WebRtcVideoChannel2Test, ReceiveDifferentUnsignaledSsrc) {
   EXPECT_EQ(rtpHeader.ssrc, recv_stream->GetConfig().rtp.remote_ssrc);
   // Verify that the receive stream sinks to a renderer.
   webrtc::VideoFrame video_frame3(CreateBlackFrameBuffer(4, 4), 300, 0,
-                                 webrtc::kVideoRotation_0);
+                                  webrtc::kVideoRotation_0);
   recv_stream->InjectFrame(video_frame3);
   EXPECT_EQ(3, renderer.num_rendered_frames());
 #endif

webrtc/modules/include/module_common_types.h

@@ -58,6 +58,8 @@ struct RTPVideoHeader {
 
   PlayoutDelay playout_delay;
 
+  VideoContentType content_type;
+
   union {
     bool is_first_packet_in_frame;
     RTC_DEPRECATED bool isFirstPacket;  // first packet in frame
@@ -87,7 +89,7 @@ class RTPFragmentationHeader {
         fragmentationOffset(NULL),
         fragmentationLength(NULL),
         fragmentationTimeDiff(NULL),
-        fragmentationPlType(NULL) {};
+        fragmentationPlType(NULL) {}
 
   ~RTPFragmentationHeader() {
     delete[] fragmentationOffset;

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h

@@ -76,7 +76,8 @@ enum RTPExtensionType {
   kRtpExtensionVideoRotation,
   kRtpExtensionTransportSequenceNumber,
   kRtpExtensionPlayoutDelay,
-  kRtpExtensionNumberOfExtensions,
+  kRtpExtensionVideoContentType,
+  kRtpExtensionNumberOfExtensions  // Must be the last entity in the enum.
 };
 
 enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };

webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc

@@ -39,6 +39,7 @@ constexpr ExtensionInfo kExtensions[] = {
     CreateExtensionInfo<VideoOrientation>(),
     CreateExtensionInfo<TransportSequenceNumber>(),
     CreateExtensionInfo<PlayoutDelayLimits>(),
+    CreateExtensionInfo<VideoContentTypeExtension>(),
 };
 
 // Because of kRtpExtensionNone, NumberOfExtension is 1 bigger than the actual

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc

@@ -215,4 +215,33 @@ bool PlayoutDelayLimits::Write(uint8_t* data,
   return true;
 }
 
+// Video Content Type.
+//
+// E.g. default video or screenshare.
+//
+//    0                   1
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |  ID   | len=0 | Content type  |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+constexpr RTPExtensionType VideoContentTypeExtension::kId;
+constexpr uint8_t VideoContentTypeExtension::kValueSizeBytes;
+constexpr const char* VideoContentTypeExtension::kUri;
+
+bool VideoContentTypeExtension::Parse(rtc::ArrayView<const uint8_t> data,
+                                      VideoContentType* content_type) {
+  if (data.size() == 1 &&
+      data[0] < static_cast<uint8_t>(VideoContentType::TOTAL_CONTENT_TYPES)) {
+    *content_type = static_cast<VideoContentType>(data[0]);
+    return true;
+  }
+  return false;
+}
+
+bool VideoContentTypeExtension::Write(uint8_t* data,
+                                      VideoContentType content_type) {
+  data[0] = static_cast<uint8_t>(content_type);
+  return true;
+}
+
 }  // namespace webrtc