use the helper functions
srtp_crypto_policy_set_from_profile_for_rtp
and
srtp_crypto_policy_set_from_profile_for_rtcp
provided by libsrtp to initialize the rtp and rtcp policies.
BUG=None
Change-Id: Ib1560c0fc1c06d9e79c1f871b028555b3b4d66d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33399}
This is consistent with other Notify methods in this class, which
handle callbacks from java using blocking invokes to the network
thread.
This eliminates the use of the deprecated AsyncInvoker class.
Bug: webrtc:12339
Change-Id: Ib2d19b37b8f669df5b97e89d720f6eb6fc9e5517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209181
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33398}
documents why it is safe to not follow libsrtp's advice
to ensure additional SRTP_MAX_TRAILER_LEN bytes are available
when calling srtp_protect (and similar srtcp functions).
BUG=None
Change-Id: I504645d21553160f06133fd8bb3ee79e178247da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209064
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33396}
This reverts commit 83be84bb74133343358bba22e4e5106ecc385721.
Reason for revert: Suspect of crbug.com/1185276
Original change's description:
> Reland "Enable quality scaling when allowed"
>
> This reverts commit 609b524dd3ff36719b5c4470b85d37dcdadfb1f8.
>
> Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
>
> Original change's description:
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020
> Bug: webrtc:12511
> Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33385}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: I7004014c5936176f8c125aeb55da91ce095b266e
TBR: ssilkin@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209708
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33393}
This reverts commit a88fe7be146b9b85575504d4d5193c007f2e3de4.
Reason for revert: Breaks downstream test, still investigating.
Original change's description:
> Fix race between destroying SctpTransport and receiving notification on timer thread.
>
> This gets rid of the SctpTransportMap::Retrieve method and forces
> everything to go through PostToTransportThread, which behaves safely
> with relation to the transport's destruction.
>
> Bug: webrtc:12467
> Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33364}
TBR=nisse@webrtc.org
Bug: webrtc:12467
Change-Id: Ib5d815a2cbca4feb25f360bff7ed62c02d1910a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209820
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33386}
This reverts commit 609b524dd3ff36719b5c4470b85d37dcdadfb1f8.
Reason for revert: Disable QualityScalingAllowed_QualityScalingEnabled on iOS.
Original change's description:
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Ia0923e5a62acdfdeb06f9aad5d670be8a0f8d746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209643
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33385}
Some of the PCF and PC methods are actually return nil, but was by
default annotated as nonnull via NS_ASSUME_NONNULL_BEGIN.
Bug: None
No-Presubmit: True
Change-Id: Ib8b9263452a61241c9e7a16c1807d87bd597c093
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209180
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33384}
This reverts commit 752cbaba907de077e5f1b24a232e71feb479dccb.
Reason for revert: The test VideoStreamEncoderTest.QualityScalingAllowed_QualityScalingEnabled seems to fail on iOS.
Original change's description:
> Enable quality scaling when allowed
>
> Before this CL quality scaling was conditioned on scaling settings
> provided by encoder. That should not be a requirement since encoder
> may not be aware of quality scaling which is a WebRTC feature. In M90
> chromium HW encoders do not provide scaling settings (chromium:1179020).
> The default scaling settings provided by these encoders are not correct
> (b/181537172).
>
> This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
> is set to true in singlecast with normal video feed (not screen sharing)
> mode. If quality scaling is allowed it is enabled no matter whether
> scaling settings are present in encoder info or not. Setting from
> QualityScalingExperiment are used in case if not provided by encoder.
>
> Bug: chromium:1179020, webrtc:12511
> Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33373}
Bug: chromium:1179020
Bug: webrtc:12511
Change-Id: Icabf2d9a034d359f79491f2c37f1044f17a7445d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209641
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33381}
It is now possible to use AV1 encoder and decoder on iOS and test
them in apps like AppRTCMobile.
Bug: None
Change-Id: Ifae221020e5abf3809010676862eecd9ffeec5e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208400
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33378}
The access to |_timestampMap| was guarded by a lock but
not the access to the data pointer stored in |_timestampMap|.
There was a potential race condition if new data was added
in VCMGenericDecoder::Decode() while the data pointer
retrieved from _timestampMap.Pop() was being used in
VCMDecodedFrameCallback::Decoded().
This CL moves the storage of data to within |_timestampMap|,
instead of being a pointer so that it's guarded by the same
lock.
Bug: webrtc:11229
Change-Id: I3f2afb568ed724db5719d508a73de402c4531dec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209361
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33374}
Before this CL quality scaling was conditioned on scaling settings
provided by encoder. That should not be a requirement since encoder
may not be aware of quality scaling which is a WebRTC feature. In M90
chromium HW encoders do not provide scaling settings (chromium:1179020).
The default scaling settings provided by these encoders are not correct
(b/181537172).
This CL adds is_quality_scaling_allowed to VideoEncoderConfig. The flag
is set to true in singlecast with normal video feed (not screen sharing)
mode. If quality scaling is allowed it is enabled no matter whether
scaling settings are present in encoder info or not. Setting from
QualityScalingExperiment are used in case if not provided by encoder.
Bug: chromium:1179020, webrtc:12511
Change-Id: I83d55319ce4b9f4fb143187ced94a16a778b4de3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209184
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33373}
Check if codec was successfully created and exit from RunTest if not
before creating VideoProcessor.
Bug: none
Change-Id: Ia6d7171650dbc9824fb78f4a8e2851f755cfd63b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209362
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33372}
The new verification makes verification a function on a message.
It also stores the password used in the request message, so that
it is easily accessible when verifying the response.
Bug: chromium:1177125
Change-Id: I505df4b54214643a28a6b292c4e2262b9d97b097
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209060
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33366}
This gets rid of the SctpTransportMap::Retrieve method and forces
everything to go through PostToTransportThread, which behaves safely
with relation to the transport's destruction.
Bug: webrtc:12467
Change-Id: Id4a723c2c985be2a368d2cc5c5e62deb04c509ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33364}
