Commit Graph

1582 Commits

Author SHA1 Message Date
35fc1537af Fallback to software decoders on consequtive decode errors on key-frames
Bug: webrtc:11575
Change-Id: I09be17ab5155e9f610c8f7c451ca52d7d65e24d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175222
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31295}
2020-05-18 09:57:04 +00:00
fd5ae7f959 Pass datachannel priority in DC open messages
This adds priority to the API configuration of datachannels,
and passes the value in the OPEN message.

It does not yet influence SCTP prioritization of messages.

Bug: chromium:1083227
Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31287}
2020-05-17 10:57:27 +00:00
576ad5d510 Make TransformableVideoFrameInterface::GetMetadata pure virtual.
GetMetadata() has been implemented downstream and can be made pure
virtual.

Bug: chromium:1069295
Change-Id: I62a3be6106552d2d82d8c413c6f523d31626b0d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175001
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31281}
2020-05-15 17:46:20 +00:00
6866dc7806 Reland "Make sure that "current" rtc::Thread instances are always current for TaskQueueBase."
This reverts commit 28685dc08cb34f756f9200519fba3222ba3a66f2.

Reason for revert: Speculative reland after looking into downstream
failures. It's possible that carryover state from unrelated tests
running in parallel was causing failures.

Original change's description:
> Revert "Make sure that "current" rtc::Thread instances are always current for TaskQueueBase."
> 
> This reverts commit 46b3bc6c24c233fe41a2401ce6e8eb8204a2d5a8.
> 
> Reason for revert: Speculative revert. Breaks downstream project
> 
> Original change's description:
> > Make sure that "current" rtc::Thread instances are always current for TaskQueueBase.
> > 
> > This is a necessary part of fulfilling the TaskQueueBase
> > interface. If a thread does not register as the current TQ, yet offers
> > the TQ interface, TQ 'current' checks will not work as expected and
> > code that relies them (TaskQueueBase::Current() and IsCurrent())
> > will run in unexpected ways.
> > 
> > Bug: webrtc:11572
> > Change-Id: Iab747bc474e74e6ce4f9e914cfd5b0578b19d19c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175080
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31254}
> 
> TBR=mbonadei@webrtc.org,tommi@webrtc.org
> 
> Change-Id: I69ff3355f0ec447b25604bd95fdacbdb4d4f3f27
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175104
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31259}

TBR=mbonadei@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org

# Not skipping CQ checks because this is a reland.

Bug: webrtc:11572
Change-Id: I00c82d99af8e05851769e09cb682b5b73895a6f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175133
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31273}
2020-05-15 10:20:03 +00:00
bb13f38a9e Add MockTransformableVideoFrame to api/test/.
Define MockTransformableVideoFrame in api/test to be use in Blink
tests.

Bug: chromium:1069295
Change-Id: I960d8228eddbdf8263b731814a00fd02d0a6c54e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175136
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31272}
2020-05-15 09:50:19 +00:00
9b7232a68c Set up a new rtc::Thread instance per test.
Several tests leave pending tasks behind after executing, which may
affect the state of subsequent tests. This CL isolates each test in
the sense that a dedicated Thread instance is created per test and
then pending tasks are flushed and the Thread instance deleted.

Down the line we may want to improve on this and flag those tests
that leave pending tasks/timers etc.

Change-Id: Ibaf3719a9974c57ac2169edca0e2a06a9ea6c78f
Bug: webrtc:11574
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175132
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31268}
2020-05-15 09:13:02 +00:00
42748d8e22 In rtc_base/ and api/ replace mock macros with unified MOCK_METHOD macro
Bug: webrtc:11564
Change-Id: I8f4171490df39c683276f9d401a4083962ee409a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175130
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31266}
2020-05-14 19:33:26 +00:00
cdc89b4d14 Add GetMetadata() to TransformableVideoFrameInterface API.
Define VideoHeaderMetadata, containing a subset of the metadata in RTP
video header, and expose it the TransformableVideoFrameInterface, to
enable web application to compute additional data according to their own
logic, and eventually remove GetAdditionalData() from the interface.

Bug: chromium:1069295
Change-Id: Id85b494a72cfd8bdd4c0614844b9f0ffae98c956
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174822
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31265}
2020-05-14 19:26:55 +00:00
28685dc08c Revert "Make sure that "current" rtc::Thread instances are always current for TaskQueueBase."
This reverts commit 46b3bc6c24c233fe41a2401ce6e8eb8204a2d5a8.

Reason for revert: Speculative revert. Breaks downstream project

Original change's description:
> Make sure that "current" rtc::Thread instances are always current for TaskQueueBase.
> 
> This is a necessary part of fulfilling the TaskQueueBase
> interface. If a thread does not register as the current TQ, yet offers
> the TQ interface, TQ 'current' checks will not work as expected and
> code that relies them (TaskQueueBase::Current() and IsCurrent())
> will run in unexpected ways.
> 
> Bug: webrtc:11572
> Change-Id: Iab747bc474e74e6ce4f9e914cfd5b0578b19d19c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175080
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31254}

TBR=mbonadei@webrtc.org,tommi@webrtc.org

Change-Id: I69ff3355f0ec447b25604bd95fdacbdb4d4f3f27
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175104
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31259}
2020-05-14 13:55:22 +00:00
46b3bc6c24 Make sure that "current" rtc::Thread instances are always current for TaskQueueBase.
This is a necessary part of fulfilling the TaskQueueBase
interface. If a thread does not register as the current TQ, yet offers
the TQ interface, TQ 'current' checks will not work as expected and
code that relies them (TaskQueueBase::Current() and IsCurrent())
will run in unexpected ways.

Bug: webrtc:11572
Change-Id: Iab747bc474e74e6ce4f9e914cfd5b0578b19d19c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31254}
2020-05-14 12:40:42 +00:00
f3319816ad Separate capturing device index from VideoConfig
The last step of the pc framework tests migration.

Bug: webrtc:11534
Change-Id: I344c443b6d21422ef418315b7e5a6cb26ae3473d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174741
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31232}
2020-05-13 09:16:40 +00:00
b856dc1556 Remove VideoGeneratorType from pc framework test api.
VideoGeneratorType wasn't deleted in https://webrtc-review.googlesource.com/c/src/+/174541

Bug: webrtc:11534
Change-Id: I3e631240dc0b28a53e62b65e3dd094b5773fac2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174721
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31228}
2020-05-12 22:24:36 +00:00
b0bd0708d6 Surface ResidualEchoDetector creation to API
This allows users to inject the residual echo detector, as a step toward making it an optional part of compilation.

Bug: webrtc:11292, webrtc:11539
Change-Id: I7fcc8dbaced67a82851cd6cdcbc115eb01c21fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174040
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31222}
2020-05-12 10:56:18 +00:00
f9ed56b656 Add ability to set custom RtpEncodingParameters for each simulcast stream in PC framework
Bug: webrtc:11557
Change-Id: I9f44728ff9178cd9c7dbe4cbcd639d610a981015
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174754
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31218}
2020-05-11 20:46:30 +00:00
cc57b935cd Make video quality analyzer compatible with real SFU in the network
Bug: webrtc:11557
Change-Id: I8ab1fb0896e267f30856a45df6099bd9aae9bc03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174801
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31216}
2020-05-11 18:54:33 +00:00
baa2c836ba Introduce ability to set peer name for PC level tests
Add peer's name to params and use it for logging and metrics naming
for whole peer related metrics.

Bug: webrtc:11479
Change-Id: Ia7e3fc4839c90a958d66910614515ac02a96e389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174752
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31215}
2020-05-11 18:47:03 +00:00
6efc14b33d VideoTrackSourceInterface: make some newly introduced methods pure virtual.
Bug: webrtc:11114
Change-Id: Ic4d3835ae84b6a652c49f30a9c275870bbf3dacf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174440
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31211}
2020-05-11 12:28:32 +00:00
fc11519c92 Cleanup mocks in api/test
Modernise functions to unified MOCK_METHOD macro,
delete few deprecated functions on the way.
add one missing function (in MockEncodedImageCallback)
Remove proxy mock function (in MockVideoBitrateAllocatorFactory)

Remove default constructors and destructors

Bug: None
Change-Id: Ibebb0d9e3c9be5877649af7bde8b87222ddf04fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174751
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31195}
2020-05-08 20:01:03 +00:00
435fb9ad06 Remove screen_share_config from the VideoConfig.
After the migration of the pc framework tests (https://webrtc-review.googlesource.com/c/src/+/174023), having "absl::optional<ScreenShareConfig> screen_share_config" field in VideoConfig became redundant. Replaced it with VideoTrackInterface::ContentHint content_hint field.

Bug: webrtc:11534
Change-Id: Ibf4b1c8daed95ef02111fe952171f11e290905d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174702
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31187}
2020-05-08 08:56:13 +00:00
04e1bab1b3 Replaces OverheadObserver with simple getter.
This interface has a couple of issues. Primarily for me, it makes it
difficult work with the paced sender as we need to either temporarily
release a lock or force a thread-handover in order to avoid a cyclic
lock order.

For video in particular, its behavior is also falky since header sizes
can vary not only form frame to frame, but from packet to packet within
a frame (e.g. TimingInfo extension is only on the last packet, if set).
On bitrate allocation, the last reported value is picked, leading to
timing issues affecting the bitrate set.

This CL removes the callback interface and instead we simply poll the
RTP module for a packet overhead. This consists of an expected overhead
based on which non-volatile header extensions are registered (so for
instance AbsoluteCaptureTime is disregarded since it's only populated
once per second). The overhead estimation is a little less accurate but
instead simpler and deterministic.

Bug: webrtc:10809
Change-Id: I2c3d3fcca6ad35704c4c1b6b9e0a39227aada1ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173704
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31185}
2020-05-07 17:33:45 +00:00
b63331bb8f Cleanup mocks for Video (en|de)coder factories
In particular remove proxy mocks in favor of lambdas and Return(ByMove(...))

Bug: None
Change-Id: If6b79601437e82a7116479d128d538e965622fab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174701
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31179}
2020-05-07 11:58:50 +00:00
1e83d34fc1 Remove pc level test framework redundant code.
After the migration to passing frame video source implementation directly, part of the peer connection framework code became redundant. Removing screen_share_config and capturing_device_index from the VideoConfig is to be done in later reviews.

Bug: webrtc:11534
Change-Id: I7a8ea85d26d00fb5bfe7ec0d2facef9c44a0f749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174541
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31178}
2020-05-07 09:23:29 +00:00
42c59525b1 Create default frame generator in the AddVideoConfig method.
Bug: webrtc:11534
Change-Id: I5f8e6009ac48be99180574ab3ac835005f67cf58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174540
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31176}
2020-05-06 21:01:29 +00:00
81be4217b8 Remove FrameTransformerInterface functions using EncodedFrame.
Replaced by the function versions using TransformableFrameInterface
downstream.

Bug: webrtc:11380
Change-Id: Ia4aef84dd76b542ba33287aff6c9151448ed5be6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171864
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31170}
2020-05-06 07:26:44 +00:00
a0ff50c031 Reland "Improve outbound-rtp statistics for simulcast"
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.

Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".

Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}

TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com

# Not skipping CQ checks because this is a reland.

Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 20:22:19 +00:00
c0df5fc25b VoIP API implementation on top of AudioIngress/Egress
This is one last CL that includes the rest of VoIP API implementation.

Bug: webrtc:11251
Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31168}
2020-05-05 19:55:29 +00:00
c064467b32 Pass frame generator to the AddVideoConfig method in the pc framework tests.
Bug: webrtc:11534
Change-Id: Id68feca50611f412897ddef3d43b811a224b200f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174023
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31167}
2020-05-05 17:20:25 +00:00
9a925c9ce3 Revert "Improve outbound-rtp statistics for simulcast"
This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.

Reason for revert: Breaks googRtt in legacy getStats API

Original change's description:
> Improve outbound-rtp statistics for simulcast
> 
> Bug: webrtc:9547
> Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Cr-Commit-Position: refs/heads/master@{#31097}

TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:9547
Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31165}
2020-05-05 13:38:51 +00:00
49f574b3b3 Delete EncodedImage methods buffer(), set_buffer() and mutable_data()
Bug: webrtc:9378
Change-Id: Iab21fe537f03a5cd130d8435cd94520952e693a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168494
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31164}
2020-05-05 09:11:40 +00:00
dad6a940e1 Add helper frame generator factories for the pc framework tests.
Bug: webrtc:11534
Change-Id: I569fb9e78aa38f0a17f4e4a261dd93c4b8ba9ca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174340
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31162}
2020-05-04 18:56:22 +00:00
b5a013815f Rename done() into condition(), because it is actually condition in TimeController API
Bug: None
Change-Id: Ia3a742d1d2ad1238223f4da7ae843a8d22108ec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174060
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31144}
2020-04-29 10:29:09 +00:00
b261118156 Fix a typo for decoder naming
Bug: None
Change-Id: I1e1e7fe1d3efb6e7f302d7633673418b5de7212c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173940
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31135}
2020-04-27 08:03:47 +00:00
90ecee1ed9 Make AudioEncoder::GetFrameLengthRange() pure virtual.
In order for WebRTC to be able to include packet overhead in its
bitrate calculations, the AudioEncoder::GetFrameLengthRange()
function must be implemented by all audio encoders. Making this
member function pure virtual as per the following PSA:

https://groups.google.com/forum/#!topic/discuss-webrtc/qscwYr38je0

Bug: webrtc:11427
Change-Id: I30d297ef05f57453bfc257624729559057cad118
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171517
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31127}
2020-04-24 09:22:57 +00:00
cda577fd59 Enable simulcast statistics
Bug: webrtc:9547
Change-Id: I8b2920dacfac0085449a797f2831b86e2e5e65b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173749
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31126}
2020-04-24 08:32:13 +00:00
e110a44628 Delete uri for the Generic Frame Descriptor v1
Bug: webrtc:11358
Change-Id: I0c3c3a7f682f172b92dcdcbc6c13d353e1e48ada
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173747
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31124}
2020-04-23 12:44:03 +00:00
da6cda839d Improve outbound-rtp statistics for simulcast
Bug: webrtc:9547
Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31097}
2020-04-17 11:28:00 +00:00
ce0a11d5f9 Unify AdaptationReason and AdaptReason enums.
Moves the unified AdaptationReason to the api/ folder.

Bug: webrtc:11392
Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 13:33:49 +00:00
dff792591f Remove VideoStreamEncoderObserver::AdaptationReason::kNone
Replaces this with 2 methods instead, adding clarity.

ClearAdaptationStats
- Resets the adaptations statistics to 0. This is done,
when the degredation is reset, for example when the preference
is changed to/from BALANCED.

UpdateAdaptationMaskingSettings
- Updates the settings for adaptation statistics reporting.
This way we don't report quality adaptations if quality scaling
is not enabled (same for resolution/fps scaling).

The adaptation counting inside the SendStatisticsProxy is
now done in a struct that counts the totals, and then masks
out these counts based on the adaptation settings. The
MaskedAdaptationSteps uses optionals to hide the values we
shoudn't report, while the AdaptationSteps always hold the real
totals.

All tests have been updated to use the Reset/Clear method as needed.

Now that AdaptationCounters and AdaptSteps use the same structure,
AdaptationCounters was moved to api/video and replaces AdaptSteps.

The AdaptReason enum is also redundant now, and will be removed
in a follow-up CL.

R=hbos@webrtc.org

Bug: webrtc:11392
Change-Id: Iaed6488581325d341a056b5bbf76a01c19d6c282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171685
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31083}
2020-04-16 13:27:50 +00:00
f0684b5a8a Remove NetEq::InsertPacket deprecated method.
Bug: webrtc:10198
Change-Id: Ia789524c459982705a5d0fc92b216e0b5a084952
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173463
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31069}
2020-04-14 18:07:47 +00:00
cc34441554 Remove deprecated RtpPacketInfo::RtpPacketInfo.
Bug: webrtc:10739
Change-Id: Iceda881ffa0645d8e1519c2b1a62c840ffa6a93f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173468
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31060}
2020-04-14 10:59:44 +00:00
fdabfbc334 [InsertableStreams] Pass ssrc on TransformedFrameCallback registration.
Add new methods in the FrameTransformerInterfaces, passing the ssrc on
registering the transformed frame callback, to associate separate frame
transformer sinks for each ssrc. Same for unregister.

Bug: chromium:1065838
Change-Id: I8a406815e9d0cce5199f9df06c286d8b10d75b4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173183
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31047}
2020-04-10 18:00:26 +00:00
e156287855 AEC3: Remove deprecated parameter
Bug: webrtc:8671
Change-Id: Ia9bcfef9d626729b79fdcce5e8df82bf020dc9af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173321
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31042}
2020-04-09 12:25:05 +00:00
8b844f21e1 AEC3: Remove parameters for the legacy filter naming
Bug: webrtc:8671
Change-Id: Ia5f8e33b9646e2b922428a72364cbbca47091579
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173092
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31030}
2020-04-08 07:34:08 +00:00
c70b1028d4 Move AdaptationCounters from video/ to api/
- Rename AdaptationCounters to VideoAdaptationCounters
- Move VideoAdaptationCounters to the api/ folder
- Move related tests to api/test/ folder
- Remove VideoAdaptationCounters::operator-

Bug: webrtc:11392
Change-Id: I0de2537e9c8dd9cf29a2ecceee00f92a5b155c83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172920
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31006}
2020-04-06 13:27:28 +00:00
06d3559b79 Replace std::string::find() == 0 with absl::StartsWith (part 2).
This CL has been generated using clang-tidy [1] except for changes to
BUILD.gn files.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/abseil-string-find-startswith.html

Bug: None
Change-Id: Ibf75601065a53bde28623b8eef57bec067235640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172586
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30984}
2020-04-02 14:38:30 +00:00
93be66cdaa Calculate video padding for vp9 in the same way as for vp8
Bug: webrtc:11476
Change-Id: I8d7b5aac91868e10061605cc5043226ee916cc09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172722
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30982}
2020-04-02 13:49:10 +00:00
486232025b Transform received audio frames in ChannelReceive.
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
No-Try: True
Change-Id: I1a7ef9fd8130936176b5a4f78ad835cba52666d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171873
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30961}
2020-04-01 11:23:00 +00:00
c24b6b7815 Introduce TransformableFrameInterface.
Add a new frame interface to be used by frame transformers in Insertable
Streams. TransformableFrameInterface will replace
video_coding::EncodedFrame in a follow up CL, once downstream
dependecies are updated to use the new interface.

Until the functions using video_coding::EncodedFrame are removed from
the API, the video sender and receiver frame transformer delegates call
both function versions to avoid breaking tests downstream.

The TransformableFrameInterface will be used for both audio and video
frame transformers in follow-up CLs.

Bug: webrtc:11380
Change-Id: I9389a8549c156e13b1d8c938ff51eaa69c502f33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171863
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30941}
2020-03-30 13:35:26 +00:00
3e98368ec5 Reland "Distinguish between send and receive codecs"
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.

Reason for revert: The CL has been improved with the following changes,
  - Fixed negotiation of send/receive only clients.
  - Handles the implicit assumption that any H264 decoder also can
    decode H264 constraint baseline.

Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}

Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
2020-03-29 21:03:27 +00:00
fb4351b085 Enforce "comprehension-required" STUN rules.
If a STUN attribute is in the "comprehension-required" range
(0x0000-0x7FFF), and the implementation does not recognize it, this
should be treated as an error (as per RFC5389), with different behavior
depending on the type of the message received.

Bug: webrtc:9063
Change-Id: Ic31b0cdd3c26772c21d770b44fe4ee4a1b47030a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/64500
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30925}
2020-03-28 02:07:49 +00:00