The -ldl flag was previously set globally, it's now set for the libs
that use it.
Also rationalize the difference between HAVE_DLOPEN and HAVE_DLFCN_H.
Differential Revision: https://phabricator.services.mozilla.com/D203594
TSAN detects a race between destroying the scoped race checker in StartCapture
and creating the scoped race checker in CaptureProcess, because once the capture
thread has been created (and given its run function) there is no synchronization
before the run function starts to run.
This patch avoids the race by destroying the StartCapture scoped race checker
before creating the capture thread.
Differential Revision: https://phabricator.services.mozilla.com/D207203
Upstream commit: https://webrtc.googlesource.com/src/+/45e49ef5371ed67ee3278244248133bf9783d65c
[M123] Fix handling of rejected m-lines without transport description
A fingerprint should not be required for m-lines which are rejected.
BUG=chromium:326493639,webrtc:11066
(cherry picked from commit 845d6bef52ec08dfd9c87d3eff5ae5c07c3fe55d)
Change-Id: I7428c91a144ca46650e13d72868f160652a98339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340322
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Original-Commit-Position: refs/heads/main@{#41794}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341023
Cr-Commit-Position: refs/branch-heads/6312@{#2}
Cr-Branched-From: 0355f455a48b141a8277442825ec776a74d66fb7-refs/heads/main@{#41763}
Upstream commit: https://webrtc.googlesource.com/src/+/0355f455a48b141a8277442825ec776a74d66fb7
Use Environment propagated through android sdk
This way VP8Decoder and DecoderFallback would use propagated instead of global field trials.
Bug: webrtc:15791, webrtc:10335
Change-Id: I5ad5fae38f5b9379bc6376334562c154fbc56e39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340040
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41763}
Upstream commit: https://webrtc.googlesource.com/src/+/0ba663c245d336997dc912ba11dd3add0c310e0c
Change a few uses of Candidate::type() to Candidate::type_name()
Switch to type_name() for things like logging since `type()` will
change to returning an enumeration value.
The functional change that this has is that log statements and
Connection::ToString() (used for logging) will contain "host"
instead of "local" and "srflx" instead of "stun".
Bug: webrtc:15846
Change-Id: I35c50d026e4578a25d51765d59c6f2e01b850c94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41757}
Upstream commit: https://webrtc.googlesource.com/src/+/052bc3af924057df8e67e31893968f35f3c4ac30
Field trial to control SVC frame dropping mode in libvpx VP9 encoder
Example: "WebRTC-LibvpxVp9Encoder-SvcFrameDropConfig/Enabled,layer_drop_mode:1,max_consec_drop:7/"
It is only possible to enable LAYER_DROP (layer_drop_mode=1) for now. All other modes are ignored. Max consecutive frame drops (max_consec_drop) value from the field is always applied if the field trial is enabled.
LAYER_DROP requires flexible mode (is_flexible_mode_=true) which can be enabled by means of WebRTC-Vp9InterLayerPred: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=976
Bug: webrtc:15827, b/320629637
Change-Id: I9c4d4838b11547e608d863198b109cb1485902d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335041
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41755}
Upstream commit: https://webrtc.googlesource.com/src/+/1b52d5641e062bf56f1f7b40fd980b8b9e9f2679
Fix generate_buildbot_json and switch to ios_runtime_cache_17_4.
When running it, even without changes at HEAD I got:
```
KeyError: 'ios_runtime_cache_17_0'
```
Bug: b/325441006
Change-Id: I7ea236ccc1f7439d7750208260b01d7636db4ae5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339842
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41751}
Upstream commit: https://webrtc.googlesource.com/src/+/62cbdcea050529a5cd18d5e82dc3c8f6997b09fc
Allow getDisplayMedia capture HDR monitor.
The code uses IDXGIOutput1::DuplicateOutput for screen capture and
it allows only DXGI_FORMAT_B8G8R8A8_UNORM texture format, which
works on most monitor cases except HDR monitor.
HDR mointor returns type of DXGI_FORMAT_R16G16B16A16_FLOAT.
These two types of DXGI_FORMAT_B8G8R8A8_UNORM and
DXGI_FORMAT_R16G16B16A16_FLOAT are all formats that DuplicateOutput
returns based on Windows OS team.
The fix is to add allowed format of DXGI_FORMAT_R16G16B16A16_FLOAT.
Manually repro the issue and validated the fix.
Bug: chromium:40787684
Change-Id: I0a7be38b14a06261d631d2db172f12725edbbf1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339621
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41749}
Upstream commit: https://webrtc.googlesource.com/src/+/7e0bd7aaaf52a6a4bd6c3f84c107071cd2827299
Reland "Add HEVC support for h264_packet_buffer."
This is a reland of commit a2655449ee310704ee2053fd6d43a5ab7002b755
This CL guards H265 header behind RTC_ENABLE_H265.
Original change's description:
> Add HEVC support for h264_packet_buffer.
>
> Renamed to h26x_packet_buffer as it also supports HEVC now. For HEVC,
> start code is added by depacktizer, and remote endpoint must send
> sequence and picture information in-band.
>
> Co-authored-by: Qiujiao Wu <qiujiao.wu@intel.com>
>
> Bug: webrtc:13485
> Change-Id: I321cb223357d8d15da95cec80ec0c3a43c26734e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333863
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41739}
Bug: webrtc:13485
Change-Id: I478e0ab88adcef34100670a90b12251ab3c9b623
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339822
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41748}
Upstream commit: https://webrtc.googlesource.com/src/+/2eee89e9044f51263a9635febbb923af73925439
Cleanup webrtc::Environment propagation through java wrappers
Force and thus guarantee VideoDecoder created through java wrappers get access to the webrtc::Environment
Bug: webrtc:15791
Change-Id: I3f145937c0b914c8e34b24e1ecc55da756551069
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338441
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41745}
Upstream commit: https://webrtc.googlesource.com/src/+/45242adc4cbda3e56d949d92727122abaa3d6fd0
Add field trial property alloc_current_bwe_limit
The new field trial can be used to ensure probes are limited by the current BWE and does not automatically send a probe at the new max rate.
Also removes unused
FieldTrialFlag allocation_allow_further_probing;
FieldTrialParameter<DataRate> allocation_probe_max;
Bug: webrtc:14928
Change-Id: I0d5c350c0231ca0600033ad8211dca0574104201
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339840
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41744}
We already cherry-picked this when we vendored a2655449ee.
Upstream commit: https://webrtc.googlesource.com/src/+/611f21d0d4c994b791db6ea90f484c83237f63f2
Revert "Add HEVC support for h264_packet_buffer."
This reverts commit a2655449ee310704ee2053fd6d43a5ab7002b755.
Reason for revert: H265 tests must be hidden behind RTC_ENABLE_H265.
Original change's description:
> Add HEVC support for h264_packet_buffer.
>
> Renamed to h26x_packet_buffer as it also supports HEVC now. For HEVC,
> start code is added by depacktizer, and remote endpoint must send
> sequence and picture information in-band.
>
> Co-authored-by: Qiujiao Wu <qiujiao.wu@intel.com>
>
> Bug: webrtc:13485
> Change-Id: I321cb223357d8d15da95cec80ec0c3a43c26734e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333863
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41739}
Bug: webrtc:13485
Change-Id: I64660d73ef0d790b25622ce882aab3db63facf26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339861
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41742}
Upstream commit: https://webrtc.googlesource.com/src/+/f7b22c66ff7ef9ea9f4688812cbbab2fd48cf098
Add Candidate::type_name()
Candidate::type() is currently how the name of the type is fetched,
but that getter returns a non-standard type name.
Instead, I'm adding a new getter, type_name(), will follow up with
updating dependent code that needs the string, to use type_name (and
adapt to potential dependency on "local" or "stun") and then switch
type() to be enum based.
Also adding a test file for Candidate with a couple of basic tests to
start with.
Bug: webrtc:15846
Change-Id: I9b78b2405a9f962a3c07eaa8e72a79854c6f5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41740}
Essentially a no-op since we're going to see this change
reverted when we vendor in 611f21d0d4.
Upstream commit: https://webrtc.googlesource.com/src/+/a2655449ee310704ee2053fd6d43a5ab7002b755
Add HEVC support for h264_packet_buffer.
Renamed to h26x_packet_buffer as it also supports HEVC now. For HEVC,
start code is added by depacktizer, and remote endpoint must send
sequence and picture information in-band.
Co-authored-by: Qiujiao Wu <qiujiao.wu@intel.com>
Bug: webrtc:13485
Change-Id: I321cb223357d8d15da95cec80ec0c3a43c26734e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333863
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41739}
Upstream commit: https://webrtc.googlesource.com/src/+/4efc830e53da5b83eb1f06ac6eac0a0d8a8664a4
Provide test output path with `OutputPathWithRandomDirectory` 1/n
First commit in a series of commits to wire up the test output path utility that adds a random directory in the path, for problematic tests that run in concurrent execution environments.
Bug: webrtc:15833
Change-Id: I5e5b3940007be773d77dbbfc953efa810e4e3ea9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339522
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41738}
Upstream commit: https://webrtc.googlesource.com/src/+/414c94290af044991c7e2d1aadbec9dafb9ee64e
Reland "Extends WebRTC logs for software encoder fallback"
This is a reland of commit 050ffefd854f8a57071992238723259e9ae0d85a
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
NOTRY=true
Bug: b/322132132
Change-Id: I25dd34b9ba59ea8502e47b4c89cd111430636e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41736}
We already cherry-picked this when we vendored 050ffefd85.
Upstream commit: https://webrtc.googlesource.com/src/+/23c32da48ae3f5f146dbd60b8b7fe4416f3750b4
Revert "Extends WebRTC logs for software encoder fallback"
This reverts commit 050ffefd854f8a57071992238723259e9ae0d85a.
Reason for revert: Breaks downstream project.
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
Bug: b/322132132
Change-Id: I24d0a4e71a43ac192485f1af208563a51d919865
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41735}
Essentially a no-op since we're going to see this change
reverted when we vendor in 23c32da48a.
Upstream commit: https://webrtc.googlesource.com/src/+/050ffefd854f8a57071992238723259e9ae0d85a
Extends WebRTC logs for software encoder fallback
This CL extends logging related to HW->SW fallbacks on the encoder
side in WebRTC. The goal is to make it easier to track down the
different steps taken when setting up the video encoder and why/when
HW encoding fails.
Current logs are added on several lines which makes regexp searching
difficult. This CL adds all related information on one line instead.
Three new search tags are also added VSE (VideoStreamEncoder), VESFW
(VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
It has been verified that these added logs also show up in WebRTC
logs in Meet.
Logs from the GPU process are not included due to the sandboxed
nature which makes it much more complex to add to the native
WebRTC log. I think that these simple logs will provide value as is.
Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
Bug: b/322132132
Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41733}
Upstream commit: https://webrtc.googlesource.com/src/+/2bd4129e910683af471eefad320c755af4376e4a
Set scoped field trials in encode/decode test
Since not all codecs read field trials from the environment yet.
Bug: webrtc:14852
Change-Id: Ia2477c41d09dabf91f47c59eb3139d6d6a711548
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339380
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41731}
Upstream commit: https://webrtc.googlesource.com/src/+/94c3328b6138f75d0f027e04b7d9570cc479324f
Provide unified solution for dir name randomization in tests
This approach actually wraps the unique identifier generation into the
function that provides the output path for a test.
This way we don't need to add `CreateRandomUuid()` everywhere that we
have `test::OutputPath` and instead just rename to
`test::OutputPathRandomDir`
Bug: webrtc:15833
Change-Id: Ic9b69b5b599727f07b2906569a84a40edeecd1a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41730}
Upstream commit: https://webrtc.googlesource.com/src/+/14d7d2d845827fbae56d054805e1a4878875046f
Add an option to allow pacing at loss based estimate when network bandwidth is loss limited.
Add a small clean up in LossBasedBandwidthEstimatorV2ReadyForUse since IsReady() includes IsEnabled().
Bug: webrtc:12707
Change-Id: I20dfeb2ab31e7724041f89af9f312211a3ae3d23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339521
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41727}
We already cherry-picked this when we vendored 1cce1d7ddc.
Upstream commit: https://webrtc.googlesource.com/src/+/1e7a6f3b6a8eee7efcb129eec10fe734d718ebc8
Revert "Make setCodecPreferences only look at receive codecs"
This reverts commit 1cce1d7ddcbde3a3648007b5a131bd0c2638724b.
Reason for revert: Breaks WPTs
Original change's description:
> Make setCodecPreferences only look at receive codecs
>
> which is what is noted in JSEP:
> https://www.rfc-editor.org/rfc/rfc8829.html#name-setcodecpreferences
>
> Some W3C spec modifications are required since the W3C specification
> currently takes into account send codecs as well.
>
> Spec issue:
> https://github.com/w3c/webrtc-pc/issues/2888
> Spec PR:
> https://github.com/w3c/webrtc-pc/pull/2926
>
> setCodecPreferences continues to modify the codecs in an offer.
>
> Also rename RtpSender::SetCodecPreferences to RtpSender::SetSendCodecs for consistent semantics.
>
> BUG=webrtc:15396
>
> Change-Id: I1e8fbe77cb2670575578a777ed1336567a1e4031
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41719}
Bug: webrtc:15396
Change-Id: I7b545e91f820c3affc39841c6e93939eac75c363
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41725}
