This reverts commit I41cae74605fecf454900a958776b95607ccf3745
Reason for revert: Needed in order to cherry pick this revert into M93,
in order to fix crbug.com/1236202.
Original change description:
> If a bundle is established, then per JSEP, the offerer is required to
> include the new track in the bundle, and per BUNDLE, the answerer has
> to either accept the track as part of the bundle or reject the track;
> it cannot move it out of the group. So we will never need the transport.
>
> Bug: webrtc:12837
> Change-Id: I41cae74605fecf454900a958776b95607ccf3745
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221601
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34290}
TBR=hta@webrtc.org
Bug: webrtc:12837, chromium:1236202
Change-Id: Ie59e2ad5168e6829eefa67b1031b8f400ed66507
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227822
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34669}
This reverts commit 704a834f685eb96c9fcf891ca345557bef4d138a.
Reason for revert: Reverting this in order to revert
https://webrtc-review.googlesource.com/c/src/+/221601, so we can
merge that revert to M93.
Original change's description:
> Reland "Fix bug where we assume new m= sections will always be bundled."
>
> This is a reland of d2b885fd91909f1b17fb11292a8c989d5d883b22, after
> making sure transports that are just being kept alive in case of
> rollback don't contribute to connection state, which broke a WPT.
>
> Original change's description:
> > Fix bug where we assume new m= sections will always be bundled.
> >
> > A recent change [1] assumes that all new m= sections will share the
> > first BUNDLE group (if one already exists), which avoids generating
> > ICE candidates that are ultimately unnecessary. This is fine for JSEP
> > endpoints, but it breaks the following scenarios for non-JSEP endpoints:
> >
> > * Remote offer adding a new m= section that's not part of any BUNDLE
> > group.
> > * Remote offer adding an m= section to the second BUNDLE group.
> >
> > The latter is specifically problematic for any application that wants
> > to bundle all audio streams in one group and all video streams in
> > another group when using Unified Plan SDP, to replicate the behavior of
> > using Plan B without bundling. It may try to add a video stream only
> > for WebRTC to bundle it with audio.
> >
> > This is fixed by doing some minor re-factoring, having BundleManager
> > update the bundle groups at offer time.
> >
> > Also:
> > * Added some additional validation for multiple bundle groups in a
> > subsequent offer, since that now becomes relevant.
> > * Improved rollback support, because now rolling back an offer may need
> > to not only remove mid->transport mappings but alter them.
> >
> > [1]: https://webrtc-review.googlesource.com/c/src/+/221601
> >
> > Bug: webrtc:12906, webrtc:12999
> > Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34544}
>
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I68bf988b1918dd2d51de76e53e4fd696fea5a09b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227120
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34596}
TBR=hta@webrtc.org
Bug: webrtc:12906, webrtc:12999
Change-Id: I129d9eb3b9831317fa24b0263db191027246cb99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227821
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34666}
Remove private members that are no longer used or always have same value
Use less allocations
Bug: None
Change-Id: I5430c2356f0039212baf8b248b92775e8852ce1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227765
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34665}
With this turned on, packets will be sequence number after the pacing
stage rather that during packetization.
This avoids a race where packets may be sent out of order, and paves
the way for the ability to cull packets from the pacer queue without
causing sequence number gaps.
For now, the feature is off by default. Follow-ups will enable it for
video and audio separately.
Bug: webrtc:11340, webrtc:12470
Change-Id: I6d411d8c85b9047e3e9b05ff4c2c3ed97c579aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208584
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34661}
Schedule the frames to be decoded based on the pacing delay from the
last decode scheduled time. In the current implementation, multiple
threads and different functions in same thread can call
MaxWaitingTime(), thereby increasing the wait time each time the
function is called. Instead of returning the wait time for a future
frame based on the number of times the function is called, return the
wait time only for the next frame to be decoded. Threads can call the
function repeatedly to check the waiting time for next frame and wake
up and then go back to waiting if an encoded frame is not available.
Change-Id: I00886c1619599f94bde5d5eb87405572e435bd73
Bug: chromium:1237402
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226502
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34660}
The race can happen when an encoder thread is packetizing a video frame
and is calling RTPSender::AssignSequenceNumber() while the RtpRtcp
module is calling GeneratePadding() and querying
PacketSequencer::CanSendPaddingOnMediaSsrc().
The solution for now is to simply not call
PacketSequencer::CanSendPaddingOnMediaSsrc() from the RtpRtcp module,
as that parameter will be ignored anyway - RTPSender will query that
method internally while holding the send lock.
Once deferred sequencing is implemented, the
can_send_padding_on_media_ssrc parameter can be populated safely since
it is then always called on the pacer thread.
Bug: webrtc:11340, webrtc:12470
Change-Id: I9e90808166453d0e29746df89044e1d3bdffa286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227767
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34655}
When layers are activated/deactivated via UpdateActiveSimulcastLayers,
the flag wasn't being updated. This resulted in calls to Stop() getting
ignored after an implicit start via activating layers.
Bug: chromium:1234779
Change-Id: I4a72e624874526d27d3e97d6903112367c5e77fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227700
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34654}
This CL fixes "variable length array folded to constant array as an
extension" by switching the constant variable declaration to an enum,
which is compile-time constant.
Bug: None
Change-Id: Iff8cd22e6320a19439fc3f1ae2fb7126210e080b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227642
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34651}
RTCIceCandidate.nativeCandidate returns a unique_ptr that
can be null. As with the previous CL, this is used without checking
whether it is null or not, so it should be fixed.
Bug: None
Change-Id: I70a84f7a2a9a9d47b8cefa198204f9b6d63a7c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227620
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34649}
There are two problems with setLocalDescription / setRemoteDescription
in ObjC SDK.
First, RTCSessionDescription.nativeDescription returns a raw
nullableSessionDescriptionInterface pointer, where sLD/sRD are calling
Clone() method unconditionally, so it might crash.
Second, unnecessary sLD/sRD calls Clone() of the raw pointer and
does not delete it, so this pointer will leak.
To solve these problems, I changed the return type of nativeDescription to
std::unique_ptr and removed the call to Clone() method.
Bug: webrtc:13022, webrtc:13035
Change-Id: Icbb87dda62d3a11af47ec74621cf64b8a6c05228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227380
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/master@{#34647}
fill the audio level of the recovery packets from the main packet.
While not exact, this should be close enough. Without this,
the audio level in getStats() will be filled but the audio level
in getSynchronizationSources() will be empty.
In chrome this is easy to test, the audio level graph on
https://webrtc.github.io/samples/src/content/peerconnection/audio/
will be empty all the time prior to this fix.
BUG=webrtc:11640
Change-Id: Ia1e61fd1852445239021a76d08032120a92d83aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226840
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34635}
Update framerate for top spatial layer instead of per timestamp (to ensure all simulcast layers are updated).
Bug: webrtc:13037
Change-Id: I4fa423dee40d74aee22a87855207b885f0536e25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227344
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34634}
Some hardware H.264 encoders does not place average QP delta in
slice_qp_delta field. Adding an optional flag in EncoderInfo to notify
quality scaler about this.
Bug: webrtc:12942
Change-Id: I3ee29c5ae9bd7bb34d26eba7e6bede3798ca44b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34627}
