commit f00bf42d1ceb7e72008f51968b13af8763e3f2a7
author Chen Xing <chxg@google.com> Thu Jun 20 10:05:55 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Thu Jun 20 10:24:29 2019 +0000
tree ae70c4800ccb0cde69ad9e88e0f76120741854d2
parent 7953ad5daba30b5b004ee07549d7ec2c6dfd3742

Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.

This change adds the plumbing of RtpPacketInfo from RtpVideoStreamReceiver::OnRtpPacket() to VideoReceiveStream::OnFrame() for video. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.

Bug: webrtc:10668
Change-Id: Ib97d430530c5a8487d3b129936c7c51e118889bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139891
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28332}

modules/video_coding/generic_decoder_unittest.cc

diff --git a/modules/video_coding/generic_decoder_unittest.cc b/modules/video_coding/generic_decoder_unittest.cc
index adc945c..691561d 100644
--- a/modules/video_coding/generic_decoder_unittest.cc
+++ b/modules/video_coding/generic_decoder_unittest.cc
@@ -122,5 +122,31 @@
   EXPECT_EQ(*decoded_color_space, color_space);
 }
 
+TEST_F(GenericDecoderTest, PassesPacketInfos) {
+  RtpPacketInfos packet_infos = CreatePacketInfos(3);
+  VCMEncodedFrame encoded_frame;
+  encoded_frame.SetPacketInfos(packet_infos);
+  generic_decoder_.Decode(encoded_frame, clock_.TimeInMilliseconds());
+  absl::optional<VideoFrame> decoded_frame = user_callback_.WaitForFrame(10);
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_EQ(decoded_frame->packet_infos().size(), 3U);
+}
+
+TEST_F(GenericDecoderTest, PassesPacketInfosForDelayedDecoders) {
+  RtpPacketInfos packet_infos = CreatePacketInfos(3);
+  decoder_.SetDelayedDecoding(100);
+
+  {
+    // Ensure the original frame is destroyed before the decoding is completed.
+    VCMEncodedFrame encoded_frame;
+    encoded_frame.SetPacketInfos(packet_infos);
+    generic_decoder_.Decode(encoded_frame, clock_.TimeInMilliseconds());
+  }
+
+  absl::optional<VideoFrame> decoded_frame = user_callback_.WaitForFrame(200);
+  ASSERT_TRUE(decoded_frame.has_value());
+  EXPECT_EQ(decoded_frame->packet_infos().size(), 3U);
+}
+
 }  // namespace video_coding
 }  // namespace webrtc