This reverts commit d427e83a15ad2950095ce1d352cc7e11eaf6cad3.
Reason for revert: Flaky test fixed.
Refactor FrameCandenceAdapter to keep track of input frame rate. This fixes an issue where frame rate is calculated too low if congestion window drop a frame.
Also a field trial WebRTC-FrameCadenceAdapter-UseVideoFrameTimestamp is added to control if VideoFrame timestamp should be used or local clock when calculating frame rate.
Uma is recorded to tell if input frame timestamp is monotonically increasing.
Bug: webrtc:10481, webrtc:15887, webrtc:15893
Change-Id: I76268aa0991dbc99c1b881fb251a76aa54ff2673
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344561
Reviewed-by: Erik Språng <sprang@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41972}
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.
The old fields are preserved for compatibility with downstream projects, but will be removed in the future.
Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
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@{#41947}
This code was extracted to make the next following CL easier to review.
This CL simply exposes the getters, setters and callbacks to set the
buffered amount low threshold on a specific SCTP stream. It will be
used in a follow-up CL, but is just boilerplate.
Bug: chromium:40072842
Change-Id: Iccd72208b369ddc252cc5886f6446b9c2ceeb0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41943}
And thus require Environment to be propagated to this test helper
Bug: webrtc:15860
Change-Id: Ia4796d7a6a8e6f5dcb947899617df43e991419e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343181
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41910}
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.
Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.
This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.
This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.
Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
The DcSctpTransport will soon use field trials to conditionally enable
some options.
And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.
Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
To prepare for a new AudioLevel struct to be added to the public WebRTC API, rename the internal RTP extension reader/writer class to AudioLevelExtension. A temporary alias is provided to avoid breaking downstream projects.
Bug: webrtc:15788
Change-Id: Ie231668f25932fd9b539229114128b1d0b949a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339887
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@{#41787}
First commit in a series of commits to wire up the test output path utility that adds a random directory in the path, for problematic tests that run in concurrent execution environments.
Bug: webrtc:15833
Change-Id: I5e5b3940007be773d77dbbfc953efa810e4e3ea9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339522
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41738}
This approach actually wraps the unique identifier generation into the
function that provides the output path for a test.
This way we don't need to add `CreateRandomUuid()` everywhere that we
have `test::OutputPath` and instead just rename to
`test::OutputPathRandomDir`
Bug: webrtc:15833
Change-Id: Ic9b69b5b599727f07b2906569a84a40edeecd1a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41730}
Providing unique identifiers for files and directories created as part
of unit tests.
Bug: webrtc:15833
Change-Id: If2835c362c47a111aa99b0e3c6ad6a33be061978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41704}
Test that BWE proving can be started without sending audio or video.
Bug: webrtc:14928
Change-Id: Ie55cb2de774f0c3b497b2636e7a6f5eac58d36a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337322
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41703}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
This completes the breakup of the rtc_p2p target.
Remaining cleanup is to delete the rtc_p2p target and make clients
depend on the base targets.
Bug: webrtc:15796
Change-Id: I67bbeee9abf0bb663283ec3420a9a00bd3a2436a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338340
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41683}
This takes the rest of the .cc files out of the rtc_p2p build
target, leaving only one entangled target to clean up.
Bug: webrtc:15796
Change-Id: I4312b70ffe96a8affc1a02456ac466eea05dd44c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41676}
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/
Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
Instead, always use VideoSendStream::Start.
VideoSendStream::StartPerRtpStream was used for controlling if
individual rtp stream for a RtpEncodingParameter should be able to send RTP packets. It was not used for controlling the actual encoder layers.
With this change RtpEncodingParameter.active still controls actual encoder layers but it does not control if RTP packets can be sent or not.
The cleanup is done to simplify code and in the future allow sending
probe packet on a RtpTransceiver that allows sending, regardless of the
RtpEncodingParameter.active flag.
Bug: webrtc:14928
Change-Id: I896c055ed4de76db58d76f452147c29783f77ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335042
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41619}
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.
One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.
Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb
Previously reverted due to an importer issue (b/320646178) and later a
dependency on RTCStatsMember<T>::ValueToString().
In this reland, we add Attribute::ToString() but we don't delete the
RTCStatsMember<T> stringifier methods, allowing downstream to migrate
before they are deleted.
Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}
Bug: webrtc:15164
Change-Id: I281ccf5b23d8f194b5ce00186a32846c757b46fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41575}
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}
This reverts commit 55cdc29b9d7259d17ccc281855dd21adc51ca957.
Reason for revert: Breaks downstream project.
Original change's description:
> Reland "[Stats] Attribute::ToString(), to replace member ValueToString/ToJson."
>
> This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb
>
> Original change's description:
> > [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
> >
> > Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> > Attribute::ToString().
> >
> > The difference between "ToString" and "ToJson" is that the "ToJson"
> > version converts 64-bit integers and doubles to floating points with no
> > more than ~15 digits of precision as to not exceed JSON's precision
> > limitations. So only in edge cases of really large numbers or numbers
> > with a silly number of digits will the two methods produce different
> > results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> > as opposed to "{foo:123}".
> >
> > Going forward we see no reason to maintain two different string
> > converted paths that are this similar, so we only implement one
> > Attribute::ToString() method which does what "ToJson" did.
> >
> > In the next CL we can delete RTCStatsMember<T>.
> >
> > Bug: webrtc:15164
> > Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Evan Shrubsole <eshr@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#41544}
>
> Bug: webrtc:15164
> Change-Id: If34509ebf3d7c0291442ae11596e7c2d3978fb64
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335240
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41566}
Bug: webrtc:15164
Change-Id: I5819811237a6dbd85a8c738ca0180039fc705909
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335280
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41567}
This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb
Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}
Bug: webrtc:15164
Change-Id: If34509ebf3d7c0291442ae11596e7c2d3978fb64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335240
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41566}
This is a reland of commit 9e4a97bb02663604b02e219b9d501a8dd91b5614
Original change's description:
> [Stats] Add value_or() and migrate from ValueOrDefault().
>
> Yet another prerequisite for replacing RTCStatsMember<T> with
> absl::optional<T>, but this looks like the last one.
>
> Bug: webrtc:15164
> Change-Id: I2cde51e8c8c951f71b48ccd45e07146091a99616
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334647
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41541}
Bug: webrtc:15164
Change-Id: I5fdba499383e5d9efe0a1dcef6bf6c2e0a812857
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335102
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41564}
