Reland with fixes to undefined behavior.
Define new optional struct in VideoFrame to signal that the frame is a
changed part of a whole picture and add a flag to signal that partial
update may be issued by the VideoFrame source.
Also, fix too strict assumptions in FrameBuffers PasteFrom methods.
Also, add ability to set a new buffer in video frame.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/120405
Bug: webrtc:10152
Change-Id: I85790dfc7cec2f23abfe9d6cd18dc76a0c343bc0
Reviewed-on: https://webrtc-review.googlesource.com/c/120780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26493}
This reverts commit 8a21e1c9c95c9b9b570c84bdfeda0315ede9dc29.
Reason for revert: breaks buildbots
Original change's description:
> Partial frame capture API part 1
>
> Define new optional struct in VideoFrame to signal that the frame is a
> changed part of a whole picture and add a flag to signal that partial
> update may be issued by the VideoFrame source.
>
> Also, fix too strict assumptions in FrameBuffers PasteFrom methods.
> Also, add ability to set a new buffer in video frame.
>
>
> Bug: webrtc:10152
> Change-Id: Ie0da418fd60bc7a34334329292e0b860ec388788
> Reviewed-on: https://webrtc-review.googlesource.com/c/120405
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26489}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
Change-Id: Ibf61f28e529a444882962b984474d4849bb44e4b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10152
Reviewed-on: https://webrtc-review.googlesource.com/c/120760
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26490}
Define new optional struct in VideoFrame to signal that the frame is a
changed part of a whole picture and add a flag to signal that partial
update may be issued by the VideoFrame source.
Also, fix too strict assumptions in FrameBuffers PasteFrom methods.
Also, add ability to set a new buffer in video frame.
Bug: webrtc:10152
Change-Id: Ie0da418fd60bc7a34334329292e0b860ec388788
Reviewed-on: https://webrtc-review.googlesource.com/c/120405
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26489}
This is useful for internal calculations in bitrate control code as we
can skip conversion constants.
DataRate Example(TimeDelta time, DataSize size) {
double time_seconds = time.seconds<double>();
double size_bytes = size.bytes<double>();
double rate_bytes_per_sec = size_bytes/time_seconds;
return DataRate::bytes_per_sec(std::max(0.0,rate_bytes_per_sec));
}
Bug: webrtc:9709
Change-Id: I8eefed578b6e8eee67fc36af723216407e0d0323
Reviewed-on: https://webrtc-review.googlesource.com/c/120720
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26488}
Calculate packet exit times "just in time" rather than at send time.
This allows changing bandwidth with packets in the queue being reflected
correctly.
Bug: webrtc:10265
Change-Id: I5a38663def4d2bfee64164f9ae62bc61277064bb
Reviewed-on: https://webrtc-review.googlesource.com/c/120403
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26487}
Moved from RtpSender to RtpSenderVideo, since currently the
PlayoutDelay extension is used for video only, and configured via
RTPVideoHeader.
Bug: webrtc:7135
Change-Id: Idfcc90cefea83e40a4e79164d7914cdcd50e41fe
Reviewed-on: https://webrtc-review.googlesource.com/c/120357
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26484}
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:10270
Change-Id: I316f9788adac954c52b0f9230881b872c54a7ac9
Reviewed-on: https://webrtc-review.googlesource.com/c/120348
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26482}
Currently, the RtpTransport checks that the packet is either RTP or
RTCP. However, the RTCP check does not verify that the packet is a valid RTP,
and therefore invalid RTCP packets were allowed in the RtpTransport::OnReadPacket.
This change makes sure that the test for RTCP header (IsRtcpPacket) checks that it has the valid RTP version (2).
So far if the packet had the second byte that looked like
RTCP, it would ignore the first byte.
Bug: None
Change-Id: I5d07d497b9ef609c74b6e507c5f3e19e4bf10194
Reviewed-on: https://webrtc-review.googlesource.com/c/120646
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26480}
Will focus on delivering model based controller instead.
Bug: webrtc:9718
Change-Id: I5df82424469c577f3c170758e0db64e3e1aa7705
Reviewed-on: https://webrtc-review.googlesource.com/c/120607
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26478}
Now that WebRTC requires OpenSSL 1.1.0 as minimum, some bits can be
removed. The simpler versioning API is shared between BoringSSL and
OpenSSL 1.1.0, and there are some remnants of the threading callbacks
that can be removed.
Bug: none
Change-Id: I2078ca9c444b1f1efa9e4b235eb4e6037865d8fb
Reviewed-on: https://webrtc-review.googlesource.com/c/120261
Commit-Queue: David Benjamin <davidben@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26475}
There's now one const method PlayoutDelayToSend to produce the delay
values to insert into outgoing packets, and two update methods,
OnSentPacket, and OnReceivedAck, to observe outgoing packets and acks,
respectively.
Bug: webrtc:7135
Change-Id: I07498c30f0de87ae0113f7e2eb6357a091a1f0af
Reviewed-on: https://webrtc-review.googlesource.com/c/120603
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26474}
And remove custom expiration_time for release bots because currently
it equals to default value
Bug: webrtc:10047
Change-Id: Ife7fd154237575e3d43f7be814e1156624166dab
Reviewed-on: https://webrtc-review.googlesource.com/c/120604
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26471}
This prepares for making the Clock interface fully mutable.
Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.
Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
TestPeer represent single participant in the call and will own most
required for call objects.
TestPeer::CreateTestPeer is responsible for full setup of TestPeer and
allow to correctly inject media analyzers into call.
Bug: webrtc:10138
Change-Id: Ide7062004b0dc113b9c05181d8144797a3cc27a8
Reviewed-on: https://webrtc-review.googlesource.com/c/119941
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26464}
Reland with fixes. Previous iteration affected media bitrate in bunch of tests.
Always use real VideoStreamsFactory in full stack tests
Because quality scaling is enabled now in full stack test, correct
factory should be used to compute actual resolution.
Also, since analyzed stream may be disabled completely now, change how
analyzer considers the test finished --- count captured frames and
stop if required amount of frames is captured and no new comparison were made.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/118687
Bug: webrtc:10204
Change-Id: Id1d9066add185d56fe3cb6856b700d350576c6b2
Reviewed-on: https://webrtc-review.googlesource.com/c/119950
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26460}
Enables downstream projects to use the existing fake ice transport implementation, without taking dependency on gunit
Bug: None
Change-Id: I78bac9d40aa6e12b55e86f0460bcd98d85c7f214
Reviewed-on: https://webrtc-review.googlesource.com/c/120445
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26456}
This simplifies the design by making simulated network more self
sufficient. It also prepares for removing network node specific
configuration (The behavior implementation should be responsible
for handling any configuration.)
Bug: webrtc:9510
Change-Id: I218d70c0359774d9891178fbd8b1bbc729cbad92
Reviewed-on: https://webrtc-review.googlesource.com/c/120346
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26450}
Support varies by codec, especially in the simulcast case, but using
the EncoderSimulcastProxy codec should fix this.
Bug: webrtc:10069
Change-Id: Idb6a5f400ffda1cdb139004f540961a9cf85d224
Reviewed-on: https://webrtc-review.googlesource.com/c/119400
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26449}
FrameConfig is not specific to temporal layers. Anything that
can control referenced/updated buffers could potentially use it.
Bug: webrtc:10259
Change-Id: I04ed177ee884693798c3b69e35fd4255ce1e9062
Reviewed-on: https://webrtc-review.googlesource.com/c/120355
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26448}
Made path from NetEq to AudioTransport ready for many-channel audio.
If there is one stream, we can handle anything that fits in an
AudioFrame. For many streams, the current limit is 6.
Some multi-channel combinations are not supported: e.g. if we get
stereo audio and attempt to play out 6 channels.
Changes:
* AudioFrameOperations - replaced the MonoTo* and *ToMono methods by
UpmixChannels & DownmixChannels.
* AudioMixer: removed DCHECKs for <= 2 channels and tweaked the mixing
algorithm to handle many channels.
Bug: webrtc:8649
Change-Id: Ib83e16d463694e35658caa09c27849e853d508fb
Reviewed-on: https://webrtc-review.googlesource.com/c/106040
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26446}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/111750.
This time we don't use the multistream decoder unless we have to.
(Which is when #channels >2). Pros: don't make downstream projects
crash due to used up stack space, a few % more efficiency for the
typical case (because multistream adds some overhead). Cons: Messy
C-code with "union" types and #define MACROs, probably more
maintenance.
Bug: webrtc:8649
Change-Id: I4253a5e0c382f67ac7c6731dc6602a31e6779e63
Reviewed-on: https://webrtc-review.googlesource.com/c/120049
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26445}
