EncoderStreamFactory has two code paths for creating a stream: the
"simulcast path" and the "default path". Only the former cares about
encoding paramter's maxBitrate. The latter assumes that
`encoder_config.max_bitrate_bps` already encompasses the maxBitrate of
the first encoding, but this is not always the case.
As of M113, when scalability mode is specified, {active,inactive} does
not count as simulcast stream but as a default stream represented by
encoding[0].
The problem is that `encoder_config.max_bitrate_bps` only includes
`encodings[0].max_bitrate_bps` when `encodings.size() == 1` which isn't
the case here.
This CL fixes the problem by making the "create default stream" code
path look at the first encoding's maxBitrate and remove existing
assumptions that `encoder_config.max_bitrate_bps` encompasses
`encodings[0].max_bitrate_bps`. This is a step in the right direction
since we're trying to remove all special cases and have encodings map
1:1 with SSRCs, so the "max bps of entire stream" should indeed be a
separate limit than the per-encoding limits and it was confusing that
sometimes it included and sometimes it excluded encoding[0]'s limit.
This issue did not happen in {inactive,active} since that code path
counts as "simulcast stream", so "default stream" is only ever
applicable for index 0.
TESTED=Simulcast Playground, see https://crbug.com/1455962.
Bug: chromium:1455962
Change-Id: I7c44925b780623b5979751e8959e972293648a3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313282
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40482}
The following can lead to ODR violations with symbols present in the
app and in the test module:
gn path out/Perf //:webrtc_perf_tests_module //sdk:helpers_objc
//:webrtc_perf_tests_module --[public]-->
//:webrtc_perf_tests_module_loadable_module --[private]-->
//test:google_test_runner_objc --[private]-->
//test:test_support_objc --[private]-->
//sdk:helpers_objc
After this CL:
gn path out/Debug/ //:webrtc_perf_tests_module //sdk:helpers_objc
No non-data paths found between these two targets.
Bug: b/292472934
Change-Id: If8a6ecab9b34bea0f52fe91b3404d1afeca685fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313520
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40481}
ReconfigureEncoder() is supposed to recreate the send stream when
switching between legacy and standard API paths to ensure that the
upper and lower layers agree on the number of streams that exist
(legacy = 3 encodings but 1 stream, standard = same as encodings).
This successfully happened when going from standard to legacy but due
to a bug in the condition this did not happen when going from legacy to
standard because `scalability_mode_used` is always false here (even
though the standard path does use a scalability mode).
As a consequence, SetRtpParameters()'s call to UpdateSendState()
resulted in a DCHECK-crash. In release builds we still avoid IOOB
because active_modules.size() < rtp_streams.size() but to avoid mistakes
like this happening again in the future, the DCHECK is promoted to a
CHECK.
The fix is to remove the scalability mode condition which didn't make
sense anyway - changing scalability mode does not require recreation but
recreation is necessary when number of streams change, whether or not
scalability mode changed.
TESTED = Using Simulcast Playground and switching back and forth
between standard and legacy and changing scalability modes and
confirming from stats, see https://crbug.com/1467455.
Bug: chromium:1467455
Change-Id: Ide29742972ba83f2e0a11f135ab9b39c39d4eb49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313280
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40477}
This CL also adds the prefix RTC_TESTING to `ios_internal_pure_release_bot_arm64` in order to avoid ODR
violations.
Bug: b/292472934
Change-Id: If63020e679c8670b4c797217eb38fc8c2954d422
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313240
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40476}
remove some of the templating around the Codec-derived types and
use more modern C++ loops.
BUG=webrtc:15214
Change-Id: I2710628741deca647e7ae88f5966ec7c7f12669a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311260
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40475}
since this may contain sensitive data, just like the address.
BUG=None
Change-Id: I3faa1512a15467cd5cc4bcbcaebadb736f1bae07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313040
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40473}
This CL does 2 things:
- Change the DCHECK for payload_type_frequency to a CHECK (so that
this error will be a crash not a divide-by-zero)
- Change the replay helper that was used by the fuzzer to set the
frequency of the packets to the video value (90K).
Bug: chromium:1466826
Change-Id: I39941f250b1782b36a3bcddfd347a016591466ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312700
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40468}
It was discovered that if libvpx reported a scalability mode in getStats
(e.g. L3T3_KEY) and we then changed encoder implementation to an
RTCVideoEncoder (such as MediaFoundationVideoEncodeAccelerator),
getStats continued to report the old scalability mode value.
This CL makes sure to clear the scalability mode on encoder
implementation change or if the `codec_info` is missing.
We should update MediaFoundation to report L1T1 as well, but in the
meantime we should clear any old scalability modes values when the
implementation changes (if the scalability mode is not known it is
better to report nothing than to report an old misleading value).
Bug: chromium:1426440
Change-Id: I1b5f324c4d29a00a6c73404cbee0faa2ae9cd843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40467}
Make sure the callback is reset when tearing down the PipeWireSession
and that there is no concurrent access to it, which can potentially lead
to a crash.
Bug: webrtc:15386
Change-Id: I0b09002fe0479dc1cd946c80684bcc5d8754d54a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311546
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#40464}
test.webrtc.org is gone and webrtc-internals got some updates which make
it more clear which dump is used
BUG=None
No-Try: true
Change-Id: I040e54398ced78148345804a4ab4922f67de133d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312360
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40463}
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.
Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
This reverts commit 726992d7a4722b8a283d797d04432d0c6335ca96.
Reason for revert: Relanding with original errors fixed (tested by building the patch locally against Chromium)
This change no longer attempts to migrate the display size protocol from fuchsia.ui.scenic.Scenic/GetDisplayInfo to fuchsia.ui.display.singleton.Info/GetMetrics because the latter API was introduced in Fuchsia API 12, which is not yet supported in Chrome (hence some of the build errors causing the revert).
Original change's description:
> Revert "[fuchsia] remove Scenic and GFX dependencies in DesktopCapturer"
>
> This reverts commit fe5be2eb4ff8dccd96257fb8cbf32500c636c358.
>
> Reason for revert: This breaks the WebRTC roll into Chromium:
>
> - https://chromium-review.googlesource.com/c/chromium/src/+/4688561
> - https://ci.chromium.org/ui/p/chromium/builders/try/fuchsia-binary-size/399140/overview
>
> Error:
>
> [4273/4389] CXX obj/third_party/webrtc/modules/desktop_capture/desktop_capture/screen_capturer_fuchsia.o
> FAILED: obj/third_party/webrtc/modules/desktop_capture/desktop_capture/screen_capturer_fuchsia.o
> ../../buildtools/reclient/rewrapper -cfg=../../buildtools/reclient_cfgs/chromium-browser-clang/rewra...(too long)
> ../../third_party/webrtc/modules/desktop_capture/screen_capturer_fuchsia.cc:59:10: error: use of undeclared identifier 'capturer'
> 59 | return capturer(new ScreenCapturerFuchsia());
> | ^
> ../../third_party/webrtc/modules/desktop_capture/screen_capturer_fuchsia.cc:199:36: error: no type named 'InfoSyncPtr' in namespace 'fuchsia::ui::display::singleton'
>
> Original change's description:
> > [fuchsia] remove Scenic and GFX dependencies in DesktopCapturer
> >
> > We previously used:
> > - fuchsia.ui.scenic.Scenic/UsesFlatland to determine whether to use
> > Flatland; from now on it should always be the case, so this check is
> > no longer necessary.
> > - fuchsia.ui.scenic.Scenic/GetDisplayInfo to get
> > fuchsia.ui.gfx.DisplayInfo. This has been migrated to
> > fuchsia.ui.display.singleton.Info/GetMetrics and
> > fuchsia.ui.display.singleton.Metrics.
> >
> > Bug: fuchsia:100303
> > Change-Id: I147da9ffdf0ca49e1c5bde5d188e434fc660becc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311860
> > Reviewed-by: Emircan Uysaler <emircan@google.com>
> > Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> > Commit-Queue: Caroline Liu <carolineliu@google.com>
> > Cr-Commit-Position: refs/heads/main@{#40432}
>
> Bug: fuchsia:100303, b/291393959
> Change-Id: Iae70e568a8c9819e40e48069af8cea0d4ef2b6c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311801
> 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@{#40436}
Bug: fuchsia:100303, b/291393959
Change-Id: Icb7074ac86c1804ab2bdf809ea1496539ee2bf80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312000
Commit-Queue: Caroline Liu <carolineliu@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40452}
With the intent to migrate all usages of the RateStatistics and RateTracker to these two new classes and thus encourage strong types over raw ints
Bug: webrtc:13756
Change-Id: I6d98024e903e75c41b2929509f601bb32d15259d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40451}
The existing equality check did not always work since content_type
is sometimes overloaded with extra internal information such as simulcast layer index. Fix by using the videocontenttypehelpers::IsScreenshare helper method.
Bug: webrtc:15381
Change-Id: I2fe84e7f036ea2c223e4fa6dd58af1c4c0bcfbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312261
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40448}
The old Android ADM was removed in https://webrtc-review.googlesource.com/c/src/+/271841.
This change resulted in a NULL as result when asking for a
kDummyAudio ADM on Android.
The small change below should ensure that a dummy ADM can be
created on Android as well.
Bug: webrtc:7452, b/291275589
Change-Id: I2c995ce6ba9a4117e3e39596546b133fe1c49204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311946
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40440}
