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Added various timestamps to FrameObject.

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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}
This commit is contained in:
philipel
2016-07-11 08:46:29 -07:00
committed by Commit bot
parent 65a001fe30
commit b4d31085b4
6 changed files with 67 additions and 16 deletions
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16
webrtc/modules/video_coding/frame_object.cc
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@ -26,10 +26,12 @@ RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
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 @@ bool RtpFrameObject::GetBitstream(uint8_t* destination) const {
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)