commit f7de74c58c05f83f921e4c72693830357df52f44
author Johannes Kron <kron@webrtc.org> Fri Apr 30 13:10:56 2021 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Tue May 04 13:16:54 2021 +0000
tree 1afea447d92632606c0d98c55efe4e223cae5782
parent c27c047e3edb9e0a28b8f265d06e627419c6fc34

Use Timestamp to represent packet receive timestamps

Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.

Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}

modules/audio_coding/neteq/neteq_impl_unittest.cc

diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index c2a2bee..b3c25ca 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -497,7 +497,7 @@
 
   // Insert one packet.
   clock_.AdvanceTimeMilliseconds(123456);
-  int64_t expected_receive_time_ms = clock_.TimeInMilliseconds();
+  Timestamp expected_receive_time = clock_.CurrentTime();
   EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
 
   // Pull audio once.
@@ -518,7 +518,7 @@
     EXPECT_THAT(packet_info.csrcs(), ElementsAre(43, 65, 17));
     EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
     EXPECT_FALSE(packet_info.audio_level().has_value());
-    EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms);
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
   }
 
   // Start with a simple check that the fake decoder is behaving as expected.
@@ -590,7 +590,7 @@
 
   // Insert one packet.
   clock_.AdvanceTimeMilliseconds(123456);
-  int64_t expected_receive_time_ms = clock_.TimeInMilliseconds();
+  Timestamp expected_receive_time = clock_.CurrentTime();
   EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
 
   // Pull audio once.
@@ -610,7 +610,7 @@
     EXPECT_THAT(packet_info.csrcs(), IsEmpty());
     EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
     EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
-    EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms);
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
   }
 
   // Insert two more packets. The first one is out of order, and is already too
@@ -626,7 +626,7 @@
   rtp_header.extension.audioLevel = 2;
   payload[0] = 2;
   clock_.AdvanceTimeMilliseconds(2000);
-  expected_receive_time_ms = clock_.TimeInMilliseconds();
+  expected_receive_time = clock_.CurrentTime();
   EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
 
   // Expect only the second packet to be decoded (the one with "2" as the first
@@ -656,7 +656,7 @@
     EXPECT_THAT(packet_info.csrcs(), IsEmpty());
     EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
     EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
-    EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms);
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
   }
 
   EXPECT_CALL(mock_decoder, Die());