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Author SHA1 Message Date
Rezaul Barbhuiya
82c2248511 Calculate pacing delay based on decode start time 
Schedule the frames to be decoded based on the pacing delay from the
last decode scheduled time. In the current implementation, multiple
threads and different functions in same thread can call
MaxWaitingTime(), thereby increasing the wait time each time the
function is called. Instead of returning the wait time for a future
frame based on the number of times the function is called, return the
wait time only for the next frame to be decoded. Threads can call the
function repeatedly to check the waiting time for next frame and wake
up and then go back to waiting if an encoded frame is not available.

Change-Id: I00886c1619599f94bde5d5eb87405572e435bd73
Bug: chromium:1237402
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226502
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34660}
 2021-08-06 11:25:17 +00:00
Johannes Kron
985905d42d Add fieldtrial to enable minimum pacing of video frames 
If the RTP header extension playout-delay is used and set
to min=0, max>=0, frames are scheduled to be decoded as
soon as possible. There's a risk that too many frames are
sent to the decoder at once, which may cause problems
further down in the video pipeline.

This CL adds the fieldtrial WebRTC-ZeroPlayoutDelay with
the parameter min_pacing that determines the minimum
pacing interval between two frames scheduled for
decoding.

Bug: None
Change-Id: I471f7718761cfce9789b3aa8adea3e8a16ecb2fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223742
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34387}
 2021-06-29 19:37:42 +00:00
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Johannes Kron
bfd343b9be Add totalDecodeTime to RTCInboundRTPStreamStats 
Pull request to WebRTC stats specification:
https://github.com/w3c/webrtc-stats/pull/450

Bug: webrtc:10775
Change-Id: Id032cb324724329fee284ebc84595b9c39208ab8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144035
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28440}
 2019-07-02 08:28:06 +00:00
Åsa Persson
8368d1a031 Remove unused functions in VCMTiming. 
Remove VCMTiming::EnoughTimeToDecode, VCMTiming::ResetDecodeTime.

Make VCMTiming::StopDecodeTimer void (always returning zero).

Update ReceiverTiming.WrapAround test to insert timestamp that wraps.

Bug: none
Change-Id: I85a8bfd6be18371810b638284b4af73a46894be7
Reviewed-on: https://webrtc-review.googlesource.com/36060
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21660}
 2018-01-17 14:47:10 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/ 
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}
 2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/. 
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/modules/video_coding/timing_unittest.cc (Browse further)