Multiple derived classes duplcated that code, and one more fixture
can benefit from the same direct access to avoid saving reference to port allocator
Cleaned includes and build dependencies on the way, in particular left single build target that contains peer_connection_wrapper
Bug: webrtc:42232556
Change-Id: Ieb3d5449f3a0285230847716e33fb3b2d1b47882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376300
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43847}
and make it less opus-specific.
BUG=None
Change-Id: I6fe2975ba6e45a3758fedc5b950de90e8d9df362
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375436
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43846}
This is to fix build error when we set use_libcxx_modules=true in
chromium.
Bug: chromium:40263312
Change-Id: I7dd87e43823f33f70c6c8e15f4c64df7d231f021
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376003
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/main@{#43845}
Useful for when creating aggregate stats objects, seems like an
oversight that we don't have it already.
Bug: None
Change-Id: Ied36afd2122464a81c7d725825353f9af59fe632
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376220
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43844}
This was needed because Android builds depended on chromium base which is not the case anymore.
Bug: None
Change-Id: I0de26ff61fe105c68a6d60def291c542f4b65a70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376240
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43843}
std::aligned_storage is deprecated in C++23, see linked bug. The only
webrtc usage (VoidUnion::inline_storage) employed the default Align
parameter (i.e. std::aligned_storage<Len>, not std::aligned_storage<Len,
Alignment>), which is defined as the largest alignment <= Len.
This patch replaces the usage with a char buffer of same size and
alignment alignof(std::max_align_t). In theory, this might increase
alignment and use more memory. In practice, local testing in my
machines showed no behavior change, since Len =
kInlineStorageWords * sizeof(uintptr_t) = 32 > 16 (maximum alignment
on linux x86_64) > 8 (maximum alignment on mac arm64).
Bug: chromium:388068052
Change-Id: If08b69f7a5ff7b716a66c8703e31eb5d4de14431
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373800
Commit-Queue: Victor Vianna <victorvianna@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Victor Vianna <victorvianna@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43842}
Adds test coverage for the following mitigations (need both to pass):
1. https://webrtc-review.googlesource.com/c/src/+/375847
2. https://webrtc-review.googlesource.com/c/src/+/376022
Like the comment says, this is neither mandated by the spec or
guaranteed, but when the number of codecs is quite small like it is in
this test it will be true for all RTX codecs.
Bug: webrtc:360058654
Change-Id: Ib73cea59d06a62390dd039eb2dc04677d6178460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375865
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43841}
Some methods used a weak peer_connection reference directly,
supposedly causing crashes.
Now, both peer_connection and delegate are captured as
strong references and validated before use for consistency.
Bug: webrtc:393263500
Change-Id: I0ab39acf7097d4c8082d05749b37b6d4998f9aa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375880
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#43839}
Strong reference must be validated when created from weak reference,
otherwise crashes are possible and actually observed via Crashlytics.
Bug: None
Change-Id: I68355080ba3b20119c913a9c7e27edc07b5447cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375901
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43837}
It make sense to initialize dictionary with capacity to avoid
reallocation during inserting values to dictionary.
Bug: None
Change-Id: Ic3e73b95a36a39dafb89e57d49e89424eb131b4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375960
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#43836}
Changes the order of payload type assignment to keep the rtx_pt =
primary_pt+1 pattern (even if not required by the specification)
On https://webrtc.github.io/samples/src/content/peerconnection/pc1/ this
changes PTs as follows:
M132: m=video 9 UDP/TLS/RTP/SAVPF 96 97 102 103
104 105 106 107 108 109 127 125 39 40 45 46 98 99 100 101 112 113 114
(121 more lines) mid=1
M134: m=video 9 UDP/TLS/RTP/SAVPF 96 109 99 118 100 119 101 120 103 121
104 122 37 123 40 127 97 114 98 115 107 44 108 (121 more lines) mid=1
M134 with this patch: m=video 9 UDP/TLS/RTP/SAVPF 96 97 107 108 109 114
115 116 117 118 119 120 37 121 40 124 98 99 100 101 127 42 43 (121 more
lines) mid=1
Note that this pushes red and ulpfec into lower range but those codecs
are not widely used and it is possible to get them back into the upper
range e.g. by using setCodecPreferences to disable H264 (where ulpfec is
not supported)
BUG=chromium:391132280
Change-Id: I892f00a2f276728d16c37e8ba5f76d01f6322a7d
No-Iwyu: single include missing, keep it more merge-able
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375847
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43833}
This is a reland of commit e20fbb00d0e0219b710da24664e81a10b12c703a
which failed due to CHECK failing on monitor_id in
DxgiDuplicatorController::GetDeviceScaleFactor(). This has now been
fixed in this CL by removing the check for greater than 0 and instead
returning std::nullopt.
Original change's description:
> Get DeviceScaleFactor for the captured monitor/screen
>
> Accesses DeviceScaleFactor using the windows API
> GetScaleFactorForMonitor and adds it to the DesktopFrame. In a follow-up
> CL, this value is propagated to
> DesktopCaptureDevice::Core::OnCaptureResult where it is added to the
> frame metadata.
>
> In a follow-up CL, add RegisterScaleChangeEvent to get notified whenever
> the device scale factor changes.
>
> Design doc: go/expose-captured-surface-resolution
>
> Bug: chromium:383946052
> Change-Id: I363af33c569419d95ddf31a0cc2f9cecf6fb0c7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374344
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mark Foltz <mfoltz@chromium.org>
> Commit-Queue: Palak Agarwal <agpalak@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43827}
Bug: chromium:383946052
Change-Id: Iffcc889b9a560695302f71623e3e929ecb2489fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376061
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43832}
This CL focuses on addressing the remaining TODO items in the
`//rtc_base/bounded_inline_vector_impl.h` file, mainly involving the two
functions `void DefaultInitializeElements` and `void MoveElements`.
In the `void DefaultInitializeElements` function, the placement new usage is
adopted. When using placement new, manual construction operations on objects are
usually required, and a fair amount of logic needs to be written to ensure
correct object construction. In contrast, the
`std::uninitialized_default_construct_n()` function only requires passing in the
starting address of the memory and the number of objects to be constructed, and
it can automatically construct objects in batches.
In the `void MoveElements` function, the original implementation also uses the
placement new usage. When using placement new to move objects, manual handling
of the moving operation for each object is required, and usually, `std::move`
needs to be combined to achieve the object's move semantics. However, the
`std::uninitialized_move_n()` function can automatically complete the object
moving process just by passing in specific parameters.
To improve the code's simplicity, it is considered to use
`std::uninitialized_default_construct_n()` and `std::uninitialized_move_n()` to
replace the original two placement new usages.
Fixed: webrtc:392037564
Change-Id: If8edcd9ac8a4d85be0ce0a23f6433d7f91b39ad3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375182
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Ho Cheung <hocheung@chromium.org>
Cr-Commit-Position: refs/heads/main@{#43831}
This reverts commit e20fbb00d0e0219b710da24664e81a10b12c703a.
Reason for revert: Breaks WebRTC roll to Chromium, see:
https://chromium-review.googlesource.com/c/chromium/src/+/6218060
Example of error: https://ci.chromium.org/ui/p/chrome/builders/ci/win-arm64-rel-ready/51821/test-results?sortby=&groupby=
Original change's description:
> Get DeviceScaleFactor for the captured monitor/screen
>
> Accesses DeviceScaleFactor using the windows API
> GetScaleFactorForMonitor and adds it to the DesktopFrame. In a follow-up
> CL, this value is propagated to
> DesktopCaptureDevice::Core::OnCaptureResult where it is added to the
> frame metadata.
>
> In a follow-up CL, add RegisterScaleChangeEvent to get notified whenever
> the device scale factor changes.
>
> Design doc: go/expose-captured-surface-resolution
>
> Bug: chromium:383946052
> Change-Id: I363af33c569419d95ddf31a0cc2f9cecf6fb0c7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374344
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mark Foltz <mfoltz@chromium.org>
> Commit-Queue: Palak Agarwal <agpalak@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43827}
Bug: chromium:383946052
Change-Id: I3065b278939ca0e686ee6da0f0721082bc0c99e8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375902
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43829}
Accesses DeviceScaleFactor using the windows API
GetScaleFactorForMonitor and adds it to the DesktopFrame. In a follow-up
CL, this value is propagated to
DesktopCaptureDevice::Core::OnCaptureResult where it is added to the
frame metadata.
In a follow-up CL, add RegisterScaleChangeEvent to get notified whenever
the device scale factor changes.
Design doc: go/expose-captured-surface-resolution
Bug: chromium:383946052
Change-Id: I363af33c569419d95ddf31a0cc2f9cecf6fb0c7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374344
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43827}
This change expands on https://webrtc-review.googlesource.com/c/src/+/374420 to cover the error produced when copying microphone samples.
Change-Id: I7aa58c9c9ac175d5f4cfdb60bbd3f16334c03c1b
Bug: webrtc:390314937
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375540
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43826}
This reverts commit 4de5839c1117e5bb96148c8575a74a69bde02768.
Reason for revert: Bug fixed + DCHECK added
Original change's description:
> Revert "Move piggybacking controller from P2PTC to DTLS transport"
>
> This reverts commit 29e639e0a495a537c610182ab9b04aed8cf10426.
>
> Reason for revert: found bug accessing variable after it has been moved.
>
> Original change's description:
> > Move piggybacking controller from P2PTC to DTLS transport
> >
> > The DTLS-STUN piggybacking controller is associated with both the DTLS
> > transport and the ICE transport (P2PTransportChannel). It turned out to
> > be more closely associated with the DTLS transport and requires less
> > plumbing when moved there.
> >
> > The config option to enable the feature remains as part of the ICE
> > transport config since the ICE transport does not know its "upstream"
> > DTLS transport and hence can not query the config from it.
> >
> > BUG=webrtc:367395350
> >
> > Change-Id: Iafd5abd8b65855bcf32bf840414d96513d8e6300
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375283
> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43823}
>
> Bug: webrtc:367395350
> Change-Id: I2d83de8890b0aa230dd9e21cb5ce2eb03c8d3564
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375861
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43824}
Bug: webrtc:367395350
Change-Id: I4b4acccf15de565736b072ca2de88a1551a6378e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375862
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43825}
This reverts commit 29e639e0a495a537c610182ab9b04aed8cf10426.
Reason for revert: found bug accessing variable after it has been moved.
Original change's description:
> Move piggybacking controller from P2PTC to DTLS transport
>
> The DTLS-STUN piggybacking controller is associated with both the DTLS
> transport and the ICE transport (P2PTransportChannel). It turned out to
> be more closely associated with the DTLS transport and requires less
> plumbing when moved there.
>
> The config option to enable the feature remains as part of the ICE
> transport config since the ICE transport does not know its "upstream"
> DTLS transport and hence can not query the config from it.
>
> BUG=webrtc:367395350
>
> Change-Id: Iafd5abd8b65855bcf32bf840414d96513d8e6300
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375283
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43823}
Bug: webrtc:367395350
Change-Id: I2d83de8890b0aa230dd9e21cb5ce2eb03c8d3564
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375861
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43824}
The DTLS-STUN piggybacking controller is associated with both the DTLS
transport and the ICE transport (P2PTransportChannel). It turned out to
be more closely associated with the DTLS transport and requires less
plumbing when moved there.
The config option to enable the feature remains as part of the ICE
transport config since the ICE transport does not know its "upstream"
DTLS transport and hence can not query the config from it.
BUG=webrtc:367395350
Change-Id: Iafd5abd8b65855bcf32bf840414d96513d8e6300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375283
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43823}
When capturing only one display, it is unnecessary to use
DoDuplicateAll, which allocates bitmap image memory for a rectangle
encompassing all screens and captures all displays. In this case, I
switch to DoDuplicateOne.
I have added an int parameter to GetNumFrameCaptured and
EnsureFrameCaptured to distinguish which display's skip behavior is
currently being executed.
After the modification, when capturing a single monitor in a
multi-monitor environment, only the bitmap image memory of the size of
the captured monitor will be allocated, rather than for all monitors.
Bug: webrtc:391914255
Change-Id: Iee56796c50282beaf1dbeef47f5937fe7fe94a05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375181
Reviewed-by: Joe Downing <joedow@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#43822}
NetworkStateEstimator is not used by WebRTC from receive side.
ReceiveSidesCongestionController::SetTransportOverhead is not needed either since NetworkStateEstimator is removed.
Note, CongestionControlFeedbackGenerator is used with RFC 8888 only and feedback frequency will be refactored in later cl.
Bug: webrtc:42220808
Change-Id: I08980aa19117e1de7a9b7896d05d07715dd9f962
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375460
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43821}
This will cause automation to not pick it up when searching for
headers.
Bug: webrtc:42226155
Change-Id: I4e93cd4eca13af32f76201df784b20a80ac9baed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375581
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43816}
There can be error log each frame when maximum playout delay sent with the frame exceed delay derived from the av-sync.
In such scenario prefer to limit the playout delay by the one attached to the received frame.
Bug: b/390043766
Change-Id: Ia57969df72f7a649e5a9280d5bb29986f5ea14b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374284
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43814}
Extend DtlsRestart test to also
test with Dtls13 (and add variants
that tests caller/callee active).
BUG=webrtc:383141571
Change-Id: Ib8b48653d4ad3cb2f5d66d6e28fc9ab54064d804
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375620
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43813}
This reverts commit a97304ca03c2aeb4267dc1bd794c50aa8bdb9a69.
Reason for revert: performance tests still rely in on global field trials to configure PC created by this test fixture
Original change's description:
> Cleanup usage of the global field trials in the PeerConnectionE2EQualityTest helper
>
> Bug: webrtc:42220378
> Change-Id: I3dc1a71c043ef506b6d592673b04e49f4a022d17
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374901
> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43803}
Bug: webrtc:42220378, webrtc:392672060
Change-Id: Ide265c1284f9d53c0b652ed5e144dfb0a532f87a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375621
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Commit-Queue: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43812}
This test was only testing codec vendor functionality.
Bug: webrtc:360058654
Change-Id: I5763e766a44f6bb1542c4281b1d6c177a52c8c74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375600
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43811}
Since Xrandr depends on Xrender, it needs to be explicitely listed before, otherwise linkers may not find the Xrender symbols.
Bug: None
Change-Id: Ifb1e82f63e1fc1645979c14b65e3beab06637cb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375428
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli SE <fcastelli@nvidia.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43808}
This flip default behavior for webrtc users that create packetizers without help of RtpSenderVideo class.
Bug: webrtc:42226301
Change-Id: I42fe696039334672b7d0b0ed1f87a52c3f6bf5ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374883
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43807}
The new class "CodecVendor" is intended to handle all logic dealing
with codecs. This CL is a no-behavior-change CL, later CLs will
change the logic.
Bug: webrtc:360058654
Change-Id: I44e76f0e0bd364eeb7d4506f3e01e9e00e2843a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375500
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43806}
Whenever encoding info change, this ToString() method is called for some
LS_INFO logging inside video_stream_encoder.cc. Apparently the char
buffer used for constructing this string is not large enough because I
can get WebRTC to crash in a demo page that gets and sets a lot of
parameters.
By changing to rtc::StringBuilder, we don't have to make assumptions
about how long the string can get at runtime.
Bug: None
Change-Id: I32695523282143a301c0e13e06082d55bd2796b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375520
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43805}
instead use the standard API to get the rollover counter and
determine the extended sequence number which is the basis for the packet index.
See https://github.com/cisco/libsrtp/issues/738 and
https://github.com/cisco/libsrtp/issues/721
BUG=webrtc:357776213
Change-Id: I90c5a4a538f56132158aa48db8700187fcdb47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371960
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43802}
