Index: webrtc/api/video/video_timing.cc |
diff --git a/webrtc/api/video/video_timing.cc b/webrtc/api/video/video_timing.cc |
new file mode 100644 |
index 0000000000000000000000000000000000000000..a7d392367dc9c8205b0eefa322fba482c94cf4d9 |
--- /dev/null |
+++ b/webrtc/api/video/video_timing.cc |
@@ -0,0 +1,59 @@ |
+/* |
+ * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
+ * |
+ * Use of this source code is governed by a BSD-style license |
+ * that can be found in the LICENSE file in the root of the source |
+ * tree. An additional intellectual property rights grant can be found |
+ * in the file PATENTS. All contributing project authors may |
+ * be found in the AUTHORS file in the root of the source tree. |
+ */ |
+ |
+#ifndef WEBRTC_API_VIDEO_VIDEO_TIMING_CC_ |
+#define WEBRTC_API_VIDEO_VIDEO_TIMING_CC_ |
+ |
+#include "webrtc/api/video/video_timing.h" |
+ |
+#include <sstream> |
+ |
Taylor Brandstetter
2017/06/25 20:48:03
nit: Extra newline
ilnik
2017/06/26 08:46:38
Done.
|
+ |
+namespace webrtc { |
+ |
+TimingFrameInfo::TimingFrameInfo() |
+ : rtp_timestamp(0), |
+ capture_time_ms(-1), |
+ encode_start_ms(-1), |
+ encode_finish_ms(-1), |
+ packetization_finish_ms(-1), |
+ pacer_exit_ms(-1), |
+ network_timestamp_ms(-1), |
+ network2_timestamp_ms(-1), |
+ receive_start_ms(-1), |
+ receive_finish_ms(-1), |
+ decode_start_ms(-1), |
+ decode_finish_ms(-1), |
+ render_time_ms(-1) {} |
Taylor Brandstetter
2017/06/25 20:48:03
Can we use rtc::Optional instead of using -1 as a
ilnik
2017/06/26 08:46:38
Since some default values will have to be reported
Taylor Brandstetter
2017/06/27 07:01:38
That seems like an implementation detail of the To
ilnik
2017/06/27 15:23:56
Acknowledged.
|
+ |
+int64_t TimingFrameInfo::EndToEndDelay() const { |
+ return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1; |
+} |
+ |
+bool TimingFrameInfo::IsLongerThan(const TimingFrameInfo& other) const { |
+ int64_t other_delay = other.EndToEndDelay(); |
+ return other_delay == -1 || EndToEndDelay() > other_delay; |
+} |
+ |
+std::string TimingFrameInfo::ToString() const { |
+ std::stringstream out; |
+ out << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms |
+ << ',' << encode_finish_ms << ',' << packetization_finish_ms << ',' |
+ << pacer_exit_ms << ',' << network_timestamp_ms << ',' |
+ << network2_timestamp_ms << ',' << receive_start_ms << ',' |
+ << receive_finish_ms << ',' << decode_start_ms << ',' |
+ << decode_finish_ms << ',' << render_time_ms; |
+ return out.str(); |
+} |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_API_VIDEO_VIDEO_TIMING_CC_ |
+ |