With this changes users can calculate the corruption score on two frames e.g. in test scenarios where one has access to the input and output file.
Bug: webrtc:358039777
Change-Id: Id864010115aa040284ec09b42d0279ccb45960b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364161
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#43222}
Use the same code in PayloadTypePicker as in Codec.Matches()
Bug: webrtc:360058654
Change-Id: I549ed24860648cfdb6a173a19773daf01db827b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365102
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43217}
This would allow to remove abseil visibility exceptions for WebRTC targets built with chromium
Bug: None
Change-Id: I63c1052f3d5b626d51bfa7209445c317bea5f970
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43215}
This CL is a pure move; later CLs will try to increase consistency
between the functions.
Bug: webrtc:360058654
Change-Id: I6662b3d35f8e2dab60c2778a4755454fe3029fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43210}
Some TODOs with an old from where added in https://webrtc-review.googlesource.com/c/src/+/363946.
This CL updates the TODO comments to the current form.
Bug: None
Change-Id: Id61dca5a0f4d705f4dfe74f6523dae3e357d49ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365140
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43209}
This reverts commit bcb19c00ba8ab1788ba3c08f28ee1b23e0cc77b9.
Reason for revert: speculative revert
Original change's description:
> Allow sending to separate payload types for each simulcast index.
>
> This change is for mixed-codec simulcast.
>
> By obtaining the payload type via RtpConfig::GetStreamConfig(),
> the correct payload type can be retrieved regardless of whether
> RtpConfig::stream_configs is initialized or not.
>
> Bug: webrtc:362277533
> Change-Id: I6b2a1ae66356b20a832565ce6729c3ce9e73a161
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364760
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43197}
Bug: webrtc:362277533
Change-Id: I50ac1fa0d9963bf9796f8604542aef5cec653493
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365161
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43208}
Add an environment clock timestamp to SenderReportStats and make it visible in rtc_stats_collector.cc. This make it possible to use the pc->GetConfiguration().stats_timestamp_with_environment_clock() flag to decide which timestamp to use when creating a RTCRemoteOutboundRtpStreamStats object.
This CL is the third (and possible the last) of a series of CLs that aim to replace the UTC timestamps in RTCStats objects to Environment clock timestamps. The other CLs where https://webrtc-review.googlesource.com/c/src/+/363946 and https://webrtc-review.googlesource.com/c/src/+/364782.
When Chromium and Google internal uses of RTCStats are updated to set the stats_timestamp_with_environment_clock configuration, the flag can be deleted.
Bug: chromium:369369568
Change-Id: Ic0b07d7b012505267bd6516f19a9ba90df4cafab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365001
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43206}
This reverts commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d.
Reason for revert: breaks downstream compilation
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I03ffcda3d6821718f355b243ce78a9c54b4036f3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365062
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43202}
in the PRESUBMIT script.
Bug: None
Change-Id: I8471d174b87fe2a558e3bbab049943829d1a3daa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365000
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43199}
and simplify the interface used as this is only used for exporting
SRTP keys and passing arcane OpenSSL arguments around does not make
much sense.
BUG=webrtc:357776213
Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43198}
This change is for mixed-codec simulcast.
By obtaining the payload type via RtpConfig::GetStreamConfig(),
the correct payload type can be retrieved regardless of whether
RtpConfig::stream_configs is initialized or not.
Bug: webrtc:362277533
Change-Id: I6b2a1ae66356b20a832565ce6729c3ce9e73a161
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364760
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43197}
This is to make the name consistent with the other methods in the
interface and additionally to in the future not have a function that has
the same name as the `FrameToRender` struct.
Bug: webrtc:358039777
Change-Id: Iac727d93ab9e020a073477bd33d0f67f9983a0aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364961
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43195}
I missed one timestamp in https://webrtc-review.googlesource.com/c/src/+/363946, meaning that the config flag that was added do not yet work for all timestamps in RTCStats objects. The RTCRemoteOutboundRtpStreamStats still has UTC timestamps even if the config flag is set.
I will solve this by saving both an UTC (existing) and env (to be added) timestamp, and then let rtc_stats_collector choose timestamp based on the value of the config flag (just like RTCRemoteInboundRtpStreamStats is done in the 363946 commit).
Before adding the new env_ timestamp I want to make this change. I rename the existing timestamp to show what epoch it uses (NTP or UTC). This will later make it clear which timestamp is which.
So this CL will make no logical change, just renaming members.
I only need to rename the last_sender_report_timestamp_ms, but opted to rename the remote timestamp as well, to be consistent with the naming convention I add in this CL.
Bug: chromium:369369568
Change-Id: Icfe7cf274995b39799e1478a1bb8cdf5134f0b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364782
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43194}
According to RFC 6904 a header extension MAY be offered both encrypted and unecrypted. In the case when encryption is enabled the encrypted version SHOULD be used and vice versa. However, this is under the assumption that both peers actually offer the same extension header both encrypted and unecrypted. With this PR we tighten the negotiation rules to the encryption option SHOULD be the same both in the sender and receiver in order to not drop the extension. Especially, see test `TestOfferAnswerPreferEncryptedRtpHeaderExtensionsWhenEncryptionEnabled` and `TestOfferAnswerPreferEncryptedRtpHeaderExtensionsWhenEncryptionDisabled`.
Bug: chromium:40623740
Change-Id: I68c65a776fcf7be97aaf60a797594c4361a06800
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363940
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43191}
To have a single way of describing how to log a custom type: AbslStringify
Bug: None
Change-Id: I6a4a6db455685be01bff1b6eeddc121b4ea51b77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364901
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43189}
This is a reland of commit 01f91c81f7660be842fa44e96bf804a8b2402f47
Original change's description:
> Add a FrameToRender argument struct as input to FrameToRender
>
> This is to make it easier to add new arguments to the method in the
> future. We will remove the already existing method accordingly to WebRTCs deprecation rules.
>
> Bug: webrtc:358039777
> Change-Id: Id0706de5216fbd0182cac80ebfccfc4a6a055ee8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364642
> Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43181}
Bug: webrtc:358039777
Change-Id: I404bb9660d9f4436c0658814fd3ac7d74e483f0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364900
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43188}
This makes the delay manager interface significantly simpler and easier to expose.
Bug: None
Change-Id: Ie3d37c3b869eb17ca421a76e9d1af8f0a1a36ee5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364781
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43186}
More test coverage for previously fixed bug
https://crbug.com/webrtc/369654168.
Two tests are added:
1. LibvpxVp9Encoder unit test that 4:2:1 720p can be reconfigured to
singlecast (which is what happens for encodings[0] in the bug).
2. Integration test that 4:2:1 720p can change to 180p,360p,540p.
This is the exact same test as was added in [1] but using
requested_resolution instead of scale_resolution_down_by.
[1] https://webrtc-review.googlesource.com/c/src/+/363941
Bug: webrtc:369654168
Change-Id: I83456b9254c1c6f647586d340d0fe5864b5515c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364200
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43185}
This reverts commit 01f91c81f7660be842fa44e96bf804a8b2402f47.
Reason for revert: break downstream projects.
Original change's description:
> Add a FrameToRender argument struct as input to FrameToRender
>
> This is to make it easier to add new arguments to the method in the
> future. We will remove the already existing method accordingly to WebRTCs deprecation rules.
>
> Bug: webrtc:358039777
> Change-Id: Id0706de5216fbd0182cac80ebfccfc4a6a055ee8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364642
> Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43181}
Bug: webrtc:358039777
Change-Id: Id59633023a428fb63aadeb266421b09040e590bb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364841
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43184}
This is to make it easier to add new arguments to the method in the
future. We will remove the already existing method accordingly to WebRTCs deprecation rules.
Bug: webrtc:358039777
Change-Id: Id0706de5216fbd0182cac80ebfccfc4a6a055ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364642
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43181}
The goal is to be able to write the rtc::Buffer by another utility
(like rtc::ByteBufferWriter) and pass it into EncodedImageBuffer
without memcpy.
Bug: webrtc:42223344
Change-Id: Ieda55e77a36636e8cdff6ad6b7d078de0aeafec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364243
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43179}
The field trial is just a kill-switch and is enabled by default.
No need to test with and without it.
Bug: chromium:371233788
Change-Id: I1b21670761284d974319aa7adaa3af60863b23ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364780
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43177}
