UpdateActiveSimulcastLayers has been blocking
WebRtcVideoChannel::SetSend which may be called quite frequently during
negotiations. This CL changes UpdateActiveSimulcastLayers to not
synchronize with the transport's task queue to wait for the changes to
get applied.
This synchronization is quite costly, but so too are other remaining
things in VideoSendStream, so we should aim to get rid of the
`thread_sync_event_` in VideoSendStream.
Bug: webrtc:12840, webrtc:12854
Change-Id: Idb48d29b6b8382881c7c1e6f1d0f5e708dbca30f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221203
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34228}
* Make VideoSendStream and VideoSendStreamImpl construction non-blocking.
* Move ownership of the rtp video sender to VideoSendStream.
* Most state is constructed in initializer lists.
* More state is now const (including VideoSendStreamImpl ptr)
* Adding thread checks to classes that appear to have had a race before
E.g. RtpTransportControllerSend. The change in threading now actually
fixes an issue we weren't aware of.
* Moved from using weak_ptr to safety flag and made some PostTask calls
cancellable that could potentially have been problematic. Initalizing
the flag without thread synchronization is also simpler.
This should speed up renegotiation significantly when there are
multiple channels. A follow-up change will improve SetSend as well
which is another costly step during renegotiation.
Bug: webrtc:12840
Change-Id: If4b28da5a085643ce132c7cfcf80a62cd1a625c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221105
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34224}
Also renaming 'worker_queue_' variables to 'rtp_transport_queue' to
avoid confusion with the worker thread.
Bug: webrtc:12840
Change-Id: Ia647a9a5ed8fdc59929f5b7ac222328ccd129a18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221140
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34217}
This removes the two step initialization and explicit circular
dependency between the sender and the observer that complicates
construction and making members const that should be.
Moving forward the encoder feedback instance will move to a different
class, so this CL is one part of making that change possible.
Also removing an unnecessary mutex and replacing with a checker.
Bug: webrtc:12840
Change-Id: I21694806b122592de0cd1e1d96f241d339a0860f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221108
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34214}
This variable is not used, always set to false but complicates
things for `keyframe_generation_requested_` as setting keyframe_needed
requires keyframe_generation_requested_ to be read synchronously from
what soon will be a different thread than where SetAndGetRecordingState
is called on.
Bug: webrtc:11993
Change-Id: I25675d9b70c9ec96a2542e7cf5480c835ea984eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220840
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34188}
Specifying guards for functions and member variables. Also updating
a few places for VideoReceiveStream2 accordingly.
Bug: webrtc:11993
Change-Id: I2d13b009ec9853c6b2d90b08af555ecdd2b1ced6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220765
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34174}
This is in preparation for actually doing this initialization
differently in the Call class. This CL takes the registration
steps that are inherently network thread associated and makes
them separate from the ctor/dtor.
Inject Call* instead of worker_thread(), which will simplify upcoming
work that needs to access the network_thread() as well.
This is related to:
https://webrtc-review.googlesource.com/c/src/+/220608https://webrtc-review.googlesource.com/c/src/+/220609
Bug: webrtc:11993
Change-Id: I72769fd61de84967d9a645750c40d01660a2716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220764
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34172}
Reland of commit a743303211b89bbcf4cea438ee797bbbc7b59e80
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
- Mark FindHeaderExtensionByUri without filter argument as deprecated
Bug: webrtc:11713
Change-Id: I52a5ade1b94bc01d1c2a35cb56023684fcaf9982
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34129}
This also changes the meaning of |estimated_capture_clock_offset| in
|absolute_capture_time_| to become a remote to capturer clock offset.
Bug: chromium:1056230, webrtc:10739
Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#34067}
its content is duplicated in the report_block_data member
Bug: webrtc:10678
Change-Id: I89421ae4ab5f727a233161924372105e222ed404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219080
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34039}
Historically the PacketBuffer used a callback for assembled frames, and because of that RtpPacketInfos were piped through it even though they didn't have anything to do with the PacketBuffer.
With this CL RtpPacketInfos are stored in the RtpVideoStreamReceiver(2) instead.
Bug: webrtc:12579
Change-Id: Ia6285b59e135910eee7234b89b23425bb0fc0d2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215320
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33980}
This reverts commit 793bac569fdf1be16cbf24d7871d20d00bbec81b.
Reason for revert: rare compilation error fixed
Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c4c8af608017c76f75bf27e4c3d602e.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> > a) Handles for detached threads are simply closed where applicable.
> > b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
This reverts commit c89fdd716c4c8af608017c76f75bf27e4c3d602e.
Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
Original change's description:
> Refactor the PlatformThread API.
>
> PlatformThread's API is using old style function pointers, causes
> casting, is unintuitive and forces artificial call sequences, and
> is additionally possible to misuse in release mode.
>
> Fix this by an API face lift:
> 1. The class is turned into a handle, which can be empty.
> 2. The only way of getting a non-empty PlatformThread is by calling
> SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> code reader.
> 3. Handles can be Finalized, which works differently for joinable and
> detached threads:
> a) Handles for detached threads are simply closed where applicable.
> b) Joinable threads are joined before handles are closed.
> 4. The destructor finalizes handles. No explicit call is needed.
>
> Fixed: webrtc:12727
> Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33923}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=handellm@webrtc.org
Bug: webrtc:12727
Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33936}
Previously this value was rounded up to a millisecond value.
This change is complementary to another change:
https://webrtc-review.googlesource.com/c/src/+/216398
Bug: webrtc:12722
Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33928}
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.
Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
a) Handles for detached threads are simply closed where applicable.
b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.
Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.
Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
This reverts commit 48a4d33719390b7bcaf8445a1581a00825f67bfb.
Reason for reland:
Relanding the original change but without the modification for
VideoSendStream::GetStats. Essentially there's a TODO there to fix
the downstream issue, which seems to be benign.
Original change's description:
> Revert "Remove Invoke from VideoChannel::FillBitrateInfo."
>
> This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.
>
> Reason for revert: Speculative revert (breaks downstream project).
>
> Original change's description:
> > Remove Invoke from VideoChannel::FillBitrateInfo.
> >
> > The method is relied upon by StatsCollector where it was called from the
> > signaling thread in a loop. Now there's at most one invoke (not N).
> >
> > Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> > VideoSendStream. Updating all related tests that fetched stats from
> > the wrong context.
> >
> > Bug: webrtc:12726
> > Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33894}
>
> TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12726
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33898}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12726
Change-Id: I41cce3b11a29905cde982c22e82b9b1f5a98e654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217222
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33902}
This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.
Reason for revert: Speculative revert (breaks downstream project).
Original change's description:
> Remove Invoke from VideoChannel::FillBitrateInfo.
>
> The method is relied upon by StatsCollector where it was called from the
> signaling thread in a loop. Now there's at most one invoke (not N).
>
> Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> VideoSendStream. Updating all related tests that fetched stats from
> the wrong context.
>
> Bug: webrtc:12726
> Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33894}
TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33898}
The method is relied upon by StatsCollector where it was called from the
signaling thread in a loop. Now there's at most one invoke (not N).
Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
VideoSendStream. Updating all related tests that fetched stats from
the wrong context.
Bug: webrtc:12726
Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33894}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
* Adds a OnPacketSent callback to MediaChannel, which matches with
MediaChannel::NetworkInterface::SendPacket.
* Moves the OnPacketSent handling to the media channel implementations
(video/voice) and removes the PeerConnection/SdpOfferAnswerHandler
layer from the call path.
* Call::OnSentPacket is called directly from the channels on the network
thread. This eliminates a PostTask to the worker thread for every
audio/video network packet.
* Remove sigslot dependency from MediaChannel (and derived).
Bug: webrtc:11993
Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33777}
In RtpVideoStreamReceiver2 it can be protected by the `worker_task_checker_` instead.
Bug: webrtc:12579
Change-Id: I4f7d64f16172139eddc7a3e07d1dbbf338beaf2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215224
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33734}
This reverts commit a743303211b89bbcf4cea438ee797bbbc7b59e80.
Reason for revert: Breaks downstream tests that attempt to call FindHeaderExtensionByUri with 2 arguments. Could you keep the old 2-argument method declaration and just forward the call to the new 3-argument method with a suitable no-op filter?
Original change's description:
> Fix RTP header extension encryption
>
> Previously, RTP header extensions with encryption had been filtered
> if the encryption had been activated (not the other way around) which
> was likely an unintended logic inversion.
>
> In addition, it ensures that encrypted RTP header extensions are only
> negotiated if RTP header extension encryption is turned on. Formerly,
> which extensions had been negotiated depended on the order in which
> they were inserted, regardless of whether or not header encryption was
> actually enabled, leading to no extensions being sent on the wire.
>
> Further changes:
>
> - If RTP header encryption enabled, prefer encrypted extensions over
> non-encrypted extensions
> - Add most extensions to list of extensions supported for encryption
> - Discard encrypted extensions in a session description in case encryption
> is not supported for that extension
>
> Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
> into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
> header extensions will prevent any RTP packets being sent/received.
>
> Bug: webrtc:11713
> Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33723}
TBR=deadbeef@webrtc.org,terelius@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com
Change-Id: I7df6b0fa611c6496dccdfb09a65ff33ae4a52b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33727}
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.
Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
As of
1e4d4fdf88
we no longer expect an InitEncode on deativation of a layer.
Bug: webrtc:12540
Change-Id: I10d447d90d1019258f662caf7f6e649d63d6927a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215076
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33709}
VideoStreamEncoderTest: Remove unneeded set_timestamp_rtp in CreateFrame methods (the timestamp is set based on ntp_time_ms in VideoStreamEncoder::OnFrame).
Bug: none
Change-Id: I6b5531a9ac21cde5dac54df6de9b9d43261e90c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214488
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33683}
This change adds support for emitting encoded frames
for recording when the decoder can't easily read out
encoded width and height as is the case for AV1 streams,
in which case the information is buried in OBUs. Downstream
project relies on resolution information being present for key
frames. With the change, VideoReceiveStream2 infers the
resolution from decoded frames, and supplies it in the
RecordableEncodedFrames.
Bug: chromium:1191972
Change-Id: I07beda6526206c80a732976e8e19d3581489b8fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214126
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33662}
provides an implementation of the rtx-time parameter from
https://tools.ietf.org/html/rfc4588#section-8
that determines the maximum time a receiver waits for a frame
before sending a PLI.
BUG=webrtc:12420
Change-Id: Iff20d92c806989cd4d56fe330d105b3dd127ed24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33627}
At the point where an EncodedImage is reported to the
EncoderBitrateAdjuster (in order to estimate utilization), the image
data has been cleared so the size is 0 - meaning the esimtated
utilization is 0 so pushback is in effect only applied at the
beginning before an estimate is available.
This CL fixes that by explicitly using spatial/temporal id and size in
bytes, rather than passing along the EncodedImage proxy.
It is unclear when this broke, but the regression seems rather old.
This CL will affect the encoded bitrate (and thus indirectly BWE
ramp-up rate), but should avoid exessive delay at low bitrates.
Perf bots will likely trigger alerts, this is expected.
In case there are undesired side-effects, we can entirely disable the
adjuster using existing field-trials.
Bug: webrtc:12606
Change-Id: I936c2045f554696d8b4bb518eee6871ffc12c47d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212900
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33550}
