Cases above 100 ms have been observed on mac; use 60 seconds as
an offset.
Bug: webrtc:380712819
Change-Id: I52a085cb196472188bb5493276a1b32524717c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369881
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43473}
Previous computation assumed that local clock is UTC. It isn't.
Adding integration test for abs-capture stats.
Bug: webrtc:380712819
Change-Id: I054d61984cbd017b7ad04ab13e5a687eab89db69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43465}
It works in the same way as the first packet received callback and can be used for latency measurements.
One important detail is that RTCP and probe packets are excluded from triggering the callback.
Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
When it waits for only one frame, the test is flaky.
When it waits for two frames, it is not.
# Relying on triviality for confidence due to purple bots atm,
# see b/367211396
NOTRY=True
NOPRESUBMIT=True
Bug: webrtc:367205682, webrtc:42220900
Change-Id: I14963b7a86961f438fd511aba8f29525e1f19750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362583
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43025}
- Modify munger to take (mutable)
std::unique_ptr<SessionDescriptionInterface> rather than
cricket::SessionDescription (that latter is embedded in the former)
- For all pranswer test cases, do a final SetRemoteDescription(kAnswer) and
check that signaling_state == stable
Add new test cases:
1) A test case that only applies it as prAnswer on caller (callee is stable)
2) A test case that "scramble" sdb between prAnswer and Anser.
Bug: None
Change-Id: Ifedd92ade01ae781a2e59d0569133c486c7093fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42891}
since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
These are aliases for cricket::Codec.
Also remove internal usage
Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
This is a reland of commit 59f3b35013a29f8c73a46fa6fd06aadc96aad892
Landing after taking out the Chrome usages.
Original change's description:
> Take out Fuchsia-only SDES-enabling parameters
>
> This does not remove all traces of SDES - we still need to delete
> the cricket::CryptoParams struct and all code that uses it.
>
> Bug: webrtc:11066, chromium:804275
> Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41634}
Bug: webrtc:11066, chromium:804275
Change-Id: I31414dfb6a0ecfa7b6fd91c68603cfd6146869d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337260
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41660}
This reverts commit 59f3b35013a29f8c73a46fa6fd06aadc96aad892.
Broke WebRTC into Chrome rolls:
https://chromium-review.googlesource.com/c/chromium/src/+/5248171?tab=checks
/../third_party/blink/renderer/modules/peerconnection/rtc_peer_connection_handler.cc:216:18: error: no member named 'enable_dtls_srtp' in 'webrtc::PeerConnectionInterface::RTCConfiguration'
216 | configuration->enable_dtls_srtp = dtls_srtp_key_agreement;
| ~~~~~~~~~~~~~ ^
Original change's description:
> Take out Fuchsia-only SDES-enabling parameters
>
> This does not remove all traces of SDES - we still need to delete
> the cricket::CryptoParams struct and all code that uses it.
>
> Bug: webrtc:11066, chromium:804275
> Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41634}
Bug: webrtc:11066, chromium:804275
Change-Id: I2c2114873091e0c662977a6ef5723e6447166a65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337181
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41643}
This does not remove all traces of SDES - we still need to delete
the cricket::CryptoParams struct and all code that uses it.
Bug: webrtc:11066, chromium:804275
Change-Id: I811c8d40da7f4af714d53376f24cd53332a15945
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336780
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41634}
This is a follow up to a previous CL that removed direct dependency on
the `cricket::` string globals.
Bug: none
Change-Id: I4d839a36739fc4694ce81b72ee036e83dae580df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41623}
Same method, different name. Unblocks replacing RTCStatsMember<T> with
absl::optional<T>.
Bug: webrtc:15164
Change-Id: I251dd44d3b0f9576b3b68915fe0406d1b3381e5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41573}
To be submitted after downstream usage has been removed, but no earlier than December 1, 2023.
Bug: webrtc:12598
Change-Id: Id9acbac591c48c0c5883fe8f06cf6a68471b70f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323004
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41290}
This is done as a preparation to move RtcEventLog ownership into Environment where destruction happens later, when all users of the Environment are deleted.
Bug: webrtc:15656
Change-Id: I2a72c74f1fabb1e25c5200aa47a5d61e4b3d9cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41272}
and unify algorithms a bit more.
BUG=webrtc:15214
Change-Id: Ie9903f3e56d25b1dc026367e8ae6817275faa07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328442
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@{#41244}
StunServer is updated to ensure registring for receiving packet from the socket is happening on the same thread as where the packets are recevied.
Bug: webrtc:15368, webrtc:11943
Change-Id: I94cc3a47278d5489de7f170c8d43015d1551c437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328120
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41219}
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
This is required for ReportTransportStats since iterating over the
transceiver list from the network thread is not safe.
Bug: chromium:1446274, webrtc:12692
Change-Id: I7c514df9f029112c4b1da85826af91217850fb26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40197}
Track stats are roughly equal in size as the RTP stream stats which
are the largest objects making up the majority of the RTCStatsReport
size and scales with meeting size. Deleting track/stream reduces the
size in approximately half which should reduce performance overhead
and unblock code simplifications.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4517530
# Relevant bots already passed
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: Ib7bdb84c10459b42b829228d11876498e5227312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40129}
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.
The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.
BUG=webrtc:14973
Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
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@{#39497}
I don't quite understand why this is flaking but I beleive it is a
test-only problem, see description in https://crbug.com/webrtc/14947
how I have trouble understanding if "frames received" is measured
correctly.
Bug: webrtc:14947, webrtc:14909
Change-Id: I667306b7cd33687645ad6a9294364330075434ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295700
Reviewed-by: Markus Handell <handellm@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39433}
This adds a test that sets the required feedback mechanisms
to get NACK configured for video, connects, and then sets packet
loss to 100%.
The expected result is that the receiver sends NACK; this will cause
the test to set packet loss to 0%, so the next NACK sent should get
to the sender and cause retransmission.
This is explicating a problematic case in splitting media channel.
Bug: webrtc:13931
Change-Id: I0c23c4a89953976454d84b0211f0a7545bbb717a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293720
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39412}
in favor of the variant with (or returning) a Timestamp object.
BUG=webrtc:14813,webrtc:13756
Change-Id: I7b40f48f640a8be40a134b380a7a1b99cc99913b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294287
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39366}
This test verifies perkj's fixes in https://crbug.com/webrtc/14817.
I ran the test 6000 times locally and it didn't fail once.
Bug: webrtc:14817
Change-Id: I3f78f3ae2ca09b328cbfa12a89ad228d3de899c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294522
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39365}
This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
Original change's description:
> Ensure RTCRtpSenders are always created with one encoding
>
> It is possible to have AddTransceiver calls with an empty array
> of encodings or AddTrack calls. In both cases, before negotiation,
> the sender's encodings array would be empty and it was not possible
> to update any value.
>
> Now, a default entry should be created in those cases, allowing to
> update the configuration before negotiation.
>
> Bug: webrtc:10567
> Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> Auto-Submit: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39126}
Bug: webrtc:10567
Change-Id: I558a95f7b587302b5e95f6ec26d1eb1fedf3dbed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39150}
This reverts commit fc5d627cef71f906e921476c2e6b1e01d07732fe.
Reason for revert: Breaks upstream WPT tests
Original change's description:
> Reland "Ensure RTCRtpSenders are always created with one encoding"
>
> This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
>
> Original change's description:
> > Ensure RTCRtpSenders are always created with one encoding
> >
> > It is possible to have AddTransceiver calls with an empty array
> > of encodings or AddTrack calls. In both cases, before negotiation,
> > the sender's encodings array would be empty and it was not possible
> > to update any value.
> >
> > Now, a default entry should be created in those cases, allowing to
> > update the configuration before negotiation.
> >
> > Bug: webrtc:10567
> > Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> > Auto-Submit: Florent Castelli <orphis@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39126}
>
> Bug: webrtc:10567
> Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39145}
Bug: webrtc:10567
Change-Id: If9b5adb5debb7c87a15662a8d0f232405a0e8136
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291221
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39147}
This is a reland of commit b8023690d9f0e150cfe698cd68b477903ac66178
Original change's description:
> Ensure RTCRtpSenders are always created with one encoding
>
> It is possible to have AddTransceiver calls with an empty array
> of encodings or AddTrack calls. In both cases, before negotiation,
> the sender's encodings array would be empty and it was not possible
> to update any value.
>
> Now, a default entry should be created in those cases, allowing to
> update the configuration before negotiation.
>
> Bug: webrtc:10567
> Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> Auto-Submit: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39126}
Bug: webrtc:10567
Change-Id: I2d52fa5b1d7cfdc9dce279fcf9faf1e0129c9008
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291140
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39145}
This reverts commit b8023690d9f0e150cfe698cd68b477903ac66178.
Reason for revert: Breaking WPT tests in Chrome. Example build https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/1263191/overview
Original change's description:
> Ensure RTCRtpSenders are always created with one encoding
>
> It is possible to have AddTransceiver calls with an empty array
> of encodings or AddTrack calls. In both cases, before negotiation,
> the sender's encodings array would be empty and it was not possible
> to update any value.
>
> Now, a default entry should be created in those cases, allowing to
> update the configuration before negotiation.
>
> Bug: webrtc:10567
> Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
> Auto-Submit: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39126}
Bug: webrtc:10567
Change-Id: Ib8931b38182251baac616540788a02a5fafaf670
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291120
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39132}
It is possible to have AddTransceiver calls with an empty array
of encodings or AddTrack calls. In both cases, before negotiation,
the sender's encodings array would be empty and it was not possible
to update any value.
Now, a default entry should be created in those cases, allowing to
update the configuration before negotiation.
Bug: webrtc:10567
Change-Id: I1271e2965e1a97c1e472451e0ab8867fc24f6c2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290994
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39126}
Unsignaled SSRCs are only applicable for the receiver case (not sender).
This CL updates the receievr's GetParameters() and GetSources() methods
to lookup parameters/sources by the current SSRC (whether or not it was
signaled) instead of only looking at the signaled SSRC.
To clarify that the `ssrc_` variable inside the [Audio/Video]RtpReceiver
is the signaled ssrc (and not set if the current ssrc is unsignaled),
we rename this variable to `signaled_ssrc_`.
By the looks of it, other APIs like setting volume or packetizers also
have a dependency on the assumptions that the SSRC is signaled. We will
not address that in this CL, but this CL makes that more clear.
Bug: webrtc:14811
Change-Id: I32c93d264ab441ade23a4078639744d25b791742
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290573
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39051}
This unblocks the deletion of this deprecated stats object.
Bug: webrtc:14175
Change-Id: I850c028fc9556a36191909afa3d635a7e6b65b69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288582
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38983}
This includes the stats dictionaries that have been made obsolete in
the spec and whose IDs are prefixed "DEPRECATED_":
- RTCMediaStreamTrackStats
- RTCMediaStreamStats
There is an ongoing experiment to unship these stats dictionaries in
Chrome (https://crbug.com/1374215). Marking then as [[deprecated]] helps
alert other dependencies that these classes are deprecated.
In the meantime, the "DEPRECATED_RTCMediaStreamTrackStats" prefix makes
it possible to use the deprecated classes.
# Unrelated infra failures
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: I498d370294058a628278e1e5b027cd12e24ad31a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38439}
Now that SDES is (largely) removed, this is no longer useful.
Bug: chromium:1365484
Change-Id: I3e626a7d5d83130a70958851de3df0aa27616bd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38278}
https://w3c.github.io/webrtc-stats/#guidelines-for-getstats-results-caching-throttling
"When the state of the RTCPeerConnection visibly changes as a result of an API call, a promise resolving or an event firing, subsequent new getStats() calls must return up-to-date dictionaries for the affected objects."
BUG=webrtc:14190
Change-Id: I4560be22795f30e0369d573bda0100e490efb57b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265870
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37255}
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
This is verifying the theory that the fix on bug 12592 also fixed
bug 3608.
Bug: webrtc:3608, webrtc:12592
Change-Id: Ia1f5ba5ebdc9a839034092351c970c3b6a159fa2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264829
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37113}
