Index: webrtc/api/video/video_timing.h |
diff --git a/webrtc/api/video/video_timing.h b/webrtc/api/video/video_timing.h |
index a44a8ef68d8d8fe6433fe80f7676cbb495e32471..34aeefcecd86e347cb0b4f0e1287bcc4c8d5e0eb 100644 |
--- a/webrtc/api/video/video_timing.h |
+++ b/webrtc/api/video/video_timing.h |
@@ -12,14 +12,17 @@ |
#define WEBRTC_API_VIDEO_VIDEO_TIMING_H_ |
#include <stdint.h> |
-#include <limits> |
+ |
+#include <string> |
+ |
#include "webrtc/base/checks.h" |
#include "webrtc/base/safe_conversions.h" |
namespace webrtc { |
-// Video timing timstamps in ms counted from capture_time_ms of a frame. |
-struct VideoTiming { |
+// Video timing timestamps in ms counted from capture_time_ms of a frame. |
+// This structure represents data sent in video-timing RTP header extension. |
+struct VideoSendTiming { |
static const uint8_t kEncodeStartDeltaIdx = 0; |
static const uint8_t kEncodeFinishDeltaIdx = 1; |
static const uint8_t kPacketizationFinishDeltaIdx = 2; |
@@ -45,6 +48,39 @@ struct VideoTiming { |
bool is_timing_frame; |
}; |
+// Used to report precise timings of a 'timing frames'. Contains all important |
+// timestamps for a lifetime of that specific frame. Reported as a string via |
+// GetStats(). Only frame which took the longest between two GetStats calls is |
+// reported. |
+struct TimingFrameInfo { |
+ TimingFrameInfo(); |
+ |
+ // Returns end-to-end delay of a frame, if sender and receiver timestamps are |
+ // synchronized, -1 otherwise. |
+ int64_t EndToEndDelay() const; |
hbos
2017/06/27 14:08:30
Generally speaking, is there any way to know if th
ilnik
2017/06/27 15:23:56
We know when timestamps are synchronised, because
hbos
2017/07/06 13:27:29
Is this synchronization method and the video-timin
ilnik
2017/07/06 14:30:56
I don't know about synchronization method standard
|
+ |
+ // Returns true if current frame took longer to process than |other| frame. |
+ // If other frame's clocks are not synchronized, current frame is always |
+ // preferred. |
+ bool IsLongerThan(const TimingFrameInfo& other) const; |
+ |
+ std::string ToString() const; |
+ |
+ uint32_t rtp_timestamp; // Identifier of a frame. |
+ int64_t capture_time_ms; |
hbos
2017/06/27 14:08:30
Are these relative to session or relative to unix
ilnik
2017/06/27 15:23:56
These are in some local monotonous clock. Added cl
|
+ int64_t encode_start_ms; |
+ int64_t encode_finish_ms; |
+ int64_t packetization_finish_ms; |
+ int64_t pacer_exit_ms; |
+ int64_t network_timestamp_ms; // In network RTP processor timestamps: |
+ int64_t network2_timestamp_ms; // Meaning is application specific. |
hbos
2017/06/27 14:08:30
nit: If comment spans more than one line add it to
ilnik
2017/06/27 15:23:56
Done.
|
+ int64_t receive_start_ms; // First received packet time. |
+ int64_t receive_finish_ms; // Last received packet time. |
+ int64_t decode_start_ms; |
+ int64_t decode_finish_ms; |
+ int64_t render_time_ms; // Inferred smooth render time. |
hbos
2017/06/27 14:08:31
What does "Inferred smooth render time" mean?
ilnik
2017/06/27 15:23:56
WebRTC provides intended render time for each fram
|
+}; |
+ |
} // namespace webrtc |
#endif // WEBRTC_API_VIDEO_VIDEO_TIMING_H_ |