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}
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index d156352..6ac157f 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -565,19 +565,19 @@
return kInvalidPointer;
}
- int64_t receive_time_ms = clock_->TimeInMilliseconds();
+ Timestamp receive_time = clock_->CurrentTime();
stats_->ReceivedPacket();
PacketList packet_list;
// Insert packet in a packet list.
- packet_list.push_back([&rtp_header, &payload, &receive_time_ms] {
+ packet_list.push_back([&rtp_header, &payload, &receive_time] {
// Convert to Packet.
Packet packet;
packet.payload_type = rtp_header.payloadType;
packet.sequence_number = rtp_header.sequenceNumber;
packet.timestamp = rtp_header.timestamp;
packet.payload.SetData(payload.data(), payload.size());
- packet.packet_info = RtpPacketInfo(rtp_header, receive_time_ms);
+ packet.packet_info = RtpPacketInfo(rtp_header, receive_time);
// Waiting time will be set upon inserting the packet in the buffer.
RTC_DCHECK(!packet.waiting_time);
return packet;