A follow up cl/ is to remove passing upper link capacity from goog_cc to loss_based_bwe_v2.
Bug: webrtc:12707
Change-Id: I45af8ca6e8ba185700d0b7eb57004d2b61edeb9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40780}
This reverts commit 0f87b3853554ee5d4e92e487a5165b57771b6742.
This is not needed with the macOS 14 SDK, which has the fix, and which
was landed in https://crrev.com/c/4875713.
Bug: chromium:1484363, chromium:1431897
Change-Id: I1e019ce71b90333d5d1333a3cf8bb510a3dbd212
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320820
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Auto-Submit: Avi Drissman <avi@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40777}
which can happen if the remote end reuses a mid.
BUG=webrtc:15471
Change-Id: I38da7dced712400002bc61d616e481a1255aa896
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319460
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40776}
It's possible that a peer can signal the same payload with multiple
packetization options. As such, we shouldn't try to fall back to default
packetization until we have considered all the alternatives.
Bug: webrtc:15473
Change-Id: I21772b4d8c53819d1c3105988551ebdbea0df045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320241
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40775}
allowing for a lot of de-templating
BUG=webrtc:15214
Change-Id: Ibe1a5f5d704564566f24c496822a4308ba23c4dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319160
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40774}
When the specified device was not found in GetDevicesInfo,
SetPlayoutDevice/SetRecordingDevice will never return a (-1) error.
Bug: None
Change-Id: I9ac71cf72f7876c1c54ee593f184aa4007dba22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320500
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40768}
Start to save local capture clock offset for video. This is part of a effort to add End 2 End metric on Android.
Bug: None
Change-Id: Icd6e567faf66f1dc200d8661344708356bda470b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320300
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#40764}
which allows exposing more granular errors from CreateOffer/CreateAnswer
BUG=webrtc:15499
Change-Id: If72a84515e220d1e7ca739318bf0b6e8a662f60e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40763}
which is not a generic SDP serializer but only deals with the
simulcast SDP.
BUG=None
Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40761}
If there happens to be an asymmetry between local and remote codecs we
shouldn't validate that there's a 1:1 packetization mapping for every
codec. It's sufficient to check that there's at least one matching
packetization per codec.
Bug: webrtc:15473
Change-Id: Ib4fc8fdd54bb4dccf96f0c802746c848e2deed83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320440
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40760}
Change-Id: I118817fe9fc4d4e674268743ac7d6d2773d366de
Bug: webrtc:15496
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320260
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#40756}
and the associated UMA metrics after rollout in M116 stable.
BUG=webrtc:14782
Change-Id: Ib2e0f96e8aa0c1ffbf48aea30f93195aa8b44bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40754}
Use the new Converts function added in webrtc-review.googlesource.com/c/src/+/320080. Later this will also be added to video.
This change is part of an effort to get Glass 2 Glass metrics. This particular change is not needed, but I intend to add this code to video, and thinks it's nice if the code for video and audio looks the same.
Bug: None
Change-Id: I04caff0dbef1cd4f391bbaa4f8bdee0e66043888
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320281
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#40753}
which is forbidden by
https://tools.ietf.org/html/rfc5761#section-4
BUG=webrtc:12197
Change-Id: I6227f01e7dcbca3f5871a2e4a8cea3c4db0b16cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319120
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40752}
The purpose is to not allow an initial low link capacity estimate to reduce the current estimate.
Only delay overuse detection , low probe results or a loss event can
reduce the estimate.
Bug: webrtc:14392
Change-Id: Ib1618347f2c7681e3bd65d85ee687dec3cd67c97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320380
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40751}
The sequence number is generally not used for the estimation,
but may be used as a tie-breaker when ordering packet feedbacks.
Bug: b/299667054
Change-Id: I52a5145c889c8f6924838667cc267b1cd9565f7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320240
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40749}
The FrameCadenceAdapter starts a delayed task to request a
new refresh frame on receiving frame drop. However, the
resulting RepeatingTaskHandle was not Stop()ed on destruction,
leading to UAF.
Fixed: chromium:1478944
Change-Id: Iba441420953e989cfc7fcfd2f358b5b30f375786
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320200
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40747}
In change https://webrtc-review.googlesource.com/c/src/+/319961, I changed a error. Also the same code will be added for video to enable Glass 2 Glass metric for Android. To me it make sense to add this method, and then change the audio code and video code to use it.
Bug: None
Change-Id: Id5d38c3bb8266213a93e67ceb82e88d65f29de53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#40745}
logging the configuration, in particular the content type which
together with RTP configuration information like the ssrcs helps differentiating between encoders.
BUG=None
Change-Id: I1b4b2ec2bffea338cc73c3a9c6a3f775d8f1c26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319560
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40744}
Update RtpPacketizerH264::PacketizeStapA to use
single_packet_reduction_len when all fragments end up in one H.264
packet.
Previous code was using first_packet_reduction_len +
last_packet_reduction_len for this case, which can cause an occasional
RTC_CHECK crash in RtpPacketizerH264::NextAggregatePacket due to
exceeding the available payload capacity of an RTP packet.
Bug: webrtc:15477
Change-Id: Iba1564a6a29618bef22f19d82aba938420994b23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319645
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40737}
rtc::strcpyn second param should be the size of the destination buffer,
not the size of the source string. The result is that the final character
(usually a trailing directory path separator) is lost during the copy.
This has been masked because FormFileName helpfully adds a trailing path
separator if one is missing.
BUG=webrtc:15441
Change-Id: I992e69cad86a7e8bc2057ec629063f34c75fe75f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317502
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40736}
