Bug: None
Change-Id: I0c82815b21b1eb0be3e12cf6ad52bf6082dfea7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143798
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28392}
This fixes a bug where NACK mode was not properly enabled
due to missing send side configuration.
Bug: webrtc:9510
Change-Id: I318fdf44f17e57d30589115a452f6a64f81ee973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143781
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28391}
When provided, these thresholds will be used instead of WebRTC default
limits specified in DropDueToSize() and GetMaxDefaultVideoBitrateKbps().
Bug: none
Change-Id: Ida45ea832041963b8b8475d69114b5c60a172fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142170
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28390}
Remove expectation in PacketRouter tests for exact number const accessors are called
Bug: None
Change-Id: I79c08f0c802b0c863adb133819d32e0b9203e721
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143799
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28387}
This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the video side with the spec-compliant `SourceTracker`-implementation.
The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.
Bug: webrtc:10545
Change-Id: I895b5790280ac94c1501801d226c643633c67349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143177
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28386}
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28385}
Using relative paths for JNI includes is causing build failures in chromium.
WebRTC already uses full include paths for generated JNI headers, so this CL
just removes the "jni_package" parameter from WebRTC generate_jni() targets
and removes the "jni/" portion of includes. The "jni_package" variable will be
removed from the generate_jni() template shortly.
To fix includes:
find . -name *.cc -exec sed -i -E 's@(#include.+generated.+jni)/jni/(.+_jni.h)@\1/\2@' {} \;
See https://groups.google.com/a/chromium.org/forum/?#!topic/java/MEovGrAwbqI
for discussion on naming scheme.
No-Try: True
TBR: kwiberg@webrtc.org
Bug: chromium:964169
Change-Id: I758c1b41bf6f5005587e55b82f14065fe251baad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143521
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28380}
Passing --stats_file_ref to frame_analyzer (which does not support
this flag anymore!) became an error with the switch to absl flags.
Bug: webrtc:10616
Change-Id: Ifc34001eafd9a92234ec1d12c3004d9f51a65f22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143783
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28370}
This reverts commit 87977dd06e702ed517f26235c12e37bd927527c7.
Reason for revert: Breaks downstream project
Original change's description:
> Change buffer level filter to store current level in number of samples.
>
> The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
>
> Bug: webrtc:10736
> Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28368}
TBR=henrik.lundin@webrtc.org,minyue@webrtc.org,jakobi@webrtc.org
Change-Id: I3900c9f6071fce51d13fb3b7c886157304d7a5c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143786
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28369}
The buffer level should not be converted back and forth between samples and packets in case of variable packet lengths.
Bug: webrtc:10736
Change-Id: Ia08dcfac3d8104dc79fbad0704a5f6f12a050a01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142178
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28368}
instead of using components that rely on GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Icf7d1758b7f3ff6277b6a6d1b152715f0ab50969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142800
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28367}
* Adds capture to decode time.
* Calculating PSNR only for delivered frames, keeping the old PSNR
value including freezes as a separate field.
* Calculates end to end delay only for delivered frames.
* Adds Count member for stats collectors.
* Minor cleanups.
Bug: webrtc:10365
Change-Id: Iaa7b1f0666a10764a513eecd1a08b9b6e76f3bc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142812
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28355}
Per discussions at https://crbug.com/webrtc/10753, the
remote-outbound-rtp.ssrc is supposed to reflect the SSRC of the RTP
media stream (i.e. outbound-rtp.ssrc) and not the sender that the
corresponding RTCP report block was transmitted on.
Bug: webrtc:10753
Change-Id: Id88f5fdbe6397ba81a46f0ef430bd6f08e66b145
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143484
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28354}
RtpPacketSender interface will be removed when downstream projects have
been updated.
Bug: webrtc:10633
Change-Id: Ie127b9814f39bd213d00ded0f7b98380f2f01084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143175
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28350}
This is a reland of fa79081dca9faa8322943641352d9d2fd1b1b445
It crashed due to inability to handle small timestamps in probe
estimator. This was fixed by moving history window check to avoid
subtracting from the timestamp.
Original change's description:
> Cleanup of RTP references in GoogCC implementation.
>
> As the send time congestion controller now has been removed,
> we don't need the RTP related constructs anymore.
>
> Bug: webrtc:9510
> Change-Id: I02c059ed8ae907ab4672d183c5639ad459b581aa
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142221
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28330}
Bug: webrtc:9510
Change-Id: I3bf91222068e4fbb6aa159bfeb7a73e00bb6a0d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143165
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28347}
This CL also improves test coverage and fixes an issue where the
(until now) unused code path for useful padding did not respect the
lower bound packet sizes.
Bug: webrtc:8975
Change-Id: I065745ca7ac9f7098d796c6a015cd96f052ee94f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142801
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28343}
