Delete struct webrtc::PacketTime.
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index 51635f6..94271a3 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -236,11 +236,11 @@
const int kTransportSequenceNumberValue = 1234;
std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
- PacketTime packet_time(5678000, 0);
+ constexpr int64_t packet_time_us = 5678000;
RtpPacketReceived parsed_packet;
ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
- parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000);
+ parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
EXPECT_CALL(*helper.channel_proxy(),
OnRtpPacket(testing::Ref(parsed_packet)));