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.h b/webrtc/modules/video_coding/frame_object.h
index b0d0d12..5b299e6 100644
--- a/webrtc/modules/video_coding/frame_object.h
+++ b/webrtc/modules/video_coding/frame_object.h
@@ -23,9 +23,19 @@
static const uint8_t kMaxFrameReferences = 5;
FrameObject();
+ virtual ~FrameObject() {}
virtual bool GetBitstream(uint8_t* destination) const = 0;
- virtual ~FrameObject() {}
+
+ // The capture timestamp of this frame.
+ virtual uint32_t Timestamp() const = 0;
+
+ // When this frame was received.
+ virtual int64_t ReceivedTime() const = 0;
+
+ // When this frame should be rendered.
+ virtual int64_t RenderTime() const = 0;
+
// The tuple (|picture_id|, |spatial_layer|) uniquely identifies a frame
// object. For codec types that don't necessarily have picture ids they
@@ -49,7 +59,8 @@
uint16_t first_seq_num,
uint16_t last_seq_num,
size_t frame_size,
- int times_nacked);
+ int times_nacked,
+ int64_t received_time);
~RtpFrameObject();
uint16_t first_seq_num() const;
@@ -58,6 +69,9 @@
enum FrameType frame_type() const;
VideoCodecType codec_type() const;
bool GetBitstream(uint8_t* destination) const override;
+ uint32_t Timestamp() const override;
+ int64_t ReceivedTime() const override;
+ int64_t RenderTime() const override;
RTPVideoTypeHeader* GetCodecHeader() const;
private:
@@ -66,6 +80,8 @@
VideoCodecType codec_type_;
uint16_t first_seq_num_;
uint16_t last_seq_num_;
+ uint32_t timestamp_;
+ int64_t received_time_;
// Equal to times nacked of the packet with the highet times nacked
// belonging to this frame.