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}
This commit is contained in:
Chen Xing
2019-06-20 10:05:55 +02:00
committed by Commit Bot
parent 7953ad5dab
commit f00bf42d1c
30 changed files with 187 additions and 29 deletions

View File

@ -122,5 +122,31 @@ TEST_F(GenericDecoderTest, PassesColorSpaceForDelayedDecoders) {
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