The `FrameToRender` function is considered a part of WebRTC's API so it cannot just be removed all at once. Since it is a pure virtual function it needs some preparation for the deprecation. This CL implements a default implementation. It will now be possible to not implement the function, but it will kill the process in that case.
Bug: webrtc:358039777
Change-Id: Ia83c63ab035abda76beb30ba98b23f9cc835a6a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365500
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43235}
This interface allows to delegate construction of AudioProcessing to
the PeerConnectionFactory where it can provide propagated field trials
Bug: webrtc:369904700
Change-Id: Ie05cd771e4a869fa5f43173e127256800ae0727f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43233}
which shows that a DtlsSrtpTransport can send and receive
from the SrtpTransport which extracts the key from its DTLS transport.
The SrtpTransport takes its keys from the DtlsSrtpTransport which
(by the way of encryption and decryption) ensures both sides agree
on the keys to use
BUG=webrtc:357776213
Change-Id: I605c6ae660eab5a53bef69bcf84d7e70a34d7be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365274
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43231}
The split shows that some places don't need it at all. Most other
places will depend on both send and receive stream targets.
Bug: webrtc:373151158
Change-Id: I788136a2ee84180c16345a7929b7f7bf3f97507b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43230}
std::optional<T>::emplace() without a value is broken
on clang++ with gnu libstdc++. this workarounds the bug
by using the assignment operator instead.
https://gcc.gnu.org/bugzilla/show_bug.cgi?id=101227
Bug: None
Change-Id: I6fd096ff4d632259e6eab776e318c1d7b15e4bd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43229}
This is a reland of commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d
with more backward compat which also fixes the off-by-one issue which caused wrong SRTP keys to be extracted.
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: I5d43dc23f90ef630834fb400751979fcc5e18203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43225}
This is in preparation for making a matcher that checks the parameters
when all payload types come from the same number space.
Bug: webrtc:360058654
Change-Id: Ibcf4fee8d882eb0fa7f83faf0278bc6757761e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365361
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43223}
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}
