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/video_coding/packet_buffer.cc b/modules/video_coding/packet_buffer.cc
index 1291f78..99f2891 100644
--- a/modules/video_coding/packet_buffer.cc
+++ b/modules/video_coding/packet_buffer.cc
@@ -36,7 +36,7 @@
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 @@
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),