Added various timestamps to FrameObject.
Added various timestamps to the FrameObject class which are needed to calculate
the jitter delay.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2124943002
Cr-Commit-Position: refs/heads/master@{#13434}
diff --git a/webrtc/modules/video_coding/frame_object.cc b/webrtc/modules/video_coding/frame_object.cc
index c33fdf7..fe3b2b0 100644
--- a/webrtc/modules/video_coding/frame_object.cc
+++ b/webrtc/modules/video_coding/frame_object.cc
@@ -26,10 +26,12 @@
uint16_t first_seq_num,
uint16_t last_seq_num,
size_t frame_size,
- int times_nacked)
+ int times_nacked,
+ int64_t received_time)
: packet_buffer_(packet_buffer),
first_seq_num_(first_seq_num),
last_seq_num_(last_seq_num),
+ received_time_(received_time),
times_nacked_(times_nacked) {
size = frame_size;
VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num);
@@ -84,6 +86,18 @@
return packet_buffer_->GetBitstream(*this, destination);
}
+uint32_t RtpFrameObject::Timestamp() const {
+ return timestamp_;
+}
+
+int64_t RtpFrameObject::ReceivedTime() const {
+ return received_time_;
+}
+
+int64_t RtpFrameObject::RenderTime() const {
+ return _renderTimeMs;
+}
+
RTPVideoTypeHeader* RtpFrameObject::GetCodecHeader() const {
VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
if (!packet)