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/video_coding/packet_buffer.cc

@@ -36,7 +36,7 @@ namespace video_coding {
 
 PacketBuffer::Packet::Packet(const RtpPacketReceived& rtp_packet,
                              const RTPVideoHeader& video_header,
-                             int64_t receive_time_ms)
+                             Timestamp receive_time)
     : marker_bit(rtp_packet.Marker()),
       payload_type(rtp_packet.PayloadType()),
       seq_num(rtp_packet.SequenceNumber()),
@@ -48,7 +48,7 @@ PacketBuffer::Packet::Packet(const RtpPacketReceived& rtp_packet,
                   rtp_packet.Timestamp(),
                   /*audio_level=*/absl::nullopt,
                   rtp_packet.GetExtension<AbsoluteCaptureTimeExtension>(),
-                  receive_time_ms) {}
+                  receive_time) {}
 
 PacketBuffer::PacketBuffer(size_t start_buffer_size, size_t max_buffer_size)
     : max_size_(max_buffer_size),