Current rate statistic tracker has assumption, the tracking window will
always be full after first filled up. This assumption looks not always
true. One example is the input_framerate_ tracker inside
video_stream_sender.cc which is used for setup frame droper and encoder.
Whenever there is a gap in video stream, like mute/unmute,
pacer pause/unpause etc. The fps detected from the rate_statistics
becomes samples_filled_partial_window / full_window_size, which could
be extremely low for a while. This creates a misalignment between the
fps we told encoder/frame dropper, and the real fps we fed into them,
which causes short-term serious overshot and very bad experience on
delay, avsync, congestion etc. This may also depends on how fast
encoder could react to the gap between set fps and real fps, but
libvpx and openh264 at least cannot handle this well.
So propose a fix to update first timestamp after tracker window
drained. This will give more accurate fps estimate similar based on
active window after sample gets drained
Bug: webrtc:13403
Change-Id: I96792c11091fe8bfa63e669f4360a3b3e95593e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35447}
Since RateStatistics is in practice always used with increasing
timestamps, and is often sparesely populated, replace the pre-allocated
ring buffer with a simple deque where each element tracks which time it
represents.
Bug: webrtc:11600
Change-Id: I866d7cfa607228c35452f0f19575825d2e694f75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175906
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31344}
This avoids integer overflows when size_t is 32 bits, and conforms
to style guide recommendations to avoid unsigned integers.
Also add tests for overflow on RateStatistics accumulator.
Bug: webrtc:11247
Change-Id: Ifa0db567f41bbcf3ec46d89ab888f2ed9d03f3f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163991
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30268}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'rtc_base'
Then manually fix where Optional was used without rtc prefix (patchset#3)
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I825f80cc8089747876ba6316d9e7c30e05716974
Reviewed-on: https://webrtc-review.googlesource.com/84585
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23700}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=sprang@webrtc.org
Bug: None
Change-Id: I50d25d6174486928963c2e98455587a8a9f0bee6
Reviewed-on: https://webrtc-review.googlesource.com/23616
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20930}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/rtc_base/rate_statistics.cc (Browse further)
