Drop dependency on MemoryStream and the complex Stream interface.
Bug: None
Change-Id: I2226324b10ddbf5606e27bfecb82efdd25929163
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213145
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33583}
This reverts commit c184047fef005b86a6dd76f03b0eb5ec01de3c5c.
Reason for revert: Breaks the WebRTC->Chromium roll:
ERROR Unresolved dependencies.
//third_party/webrtc/test/fuzzers:vp9_encoder_references_fuzzer(//build/toolchain/win:win_clang_x64)
needs //third_party/webrtc/modules/video_coding:mock_libvpx_interface(//build/toolchain/win:win_clang_x64)
We need to add tryjob to catch these. The fix is to make
//third_party/webrtc/modules/video_coding:mock_libvpx_interface
visible in built_with_chromium builds by moving the target
out of this "if" https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/modules/video_coding/BUILD.gn;l=615;drc=3889de1c4c7ae56ec742fb9ee0ad89657f638169.
Original change's description:
> Add fuzzer to validate libvpx vp9 encoder wrapper
>
> Fix simulcast svc controller to reuse dropped frame configuration,
> same as full svc and k-svc controllers do.
> This fuzzer reminded the issue was still there.
>
> Bug: webrtc:11999
> Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33568}
TBR=danilchap@webrtc.org,sprang@webrtc.org
Change-Id: I1676986308c6d37ff168467ff2099155e8895452
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212973
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33573}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
Bug: webrtc:11999
Change-Id: I74156bd743124723562e99deb48de5b5018a81d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33568}
This reverts commit 2072b87261a6505a88561bdeab3e7405d7038eaa.
Reason for revert: Causing test failure.
Original change's description:
> Reland "[Battery]: Delay start of TaskQueuePacedSender." Take 2
>
> This is a reland of 89cb65ed663a9000b9f7c90a78039bd85731e9ae
> ... and f28aade91dcc2cb8f590dc1379ac7ab5c1981909
>
> Reason for revert: crashes due to uninitialized pacing_bitrate_
> crbug.com/1190547
> Apparently pacer() is sometimes being used before EnsureStarted()
> Fix: Instead of delaying first call to SetPacingRates(),
> this CL no-ops MaybeProcessPackets() until EnsureStarted()
> is called for the first time.
>
> Original change's description:
> > [Battery]: Delay start of TaskQueuePacedSender.
> >
> > To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
> > only upon RtpTransportControllerSend::EnsureStarted().
> >
> > More specifically, the repeating task happens in
> > TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
> > task_queue_.PostDelayedTask().
> >
> > Bug: chromium:1152887
> > Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
> > Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33421}
>
> Bug: chromium:1152887
> Change-Id: I9aba4882a64bbee7d97ace9059dea8a24c144f93
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#33554}
TBR=hbos@webrtc.org,sprang@webrtc.org,etiennep@chromium.org
Change-Id: I430fd31c7602702c8ec44b9e38e68266abba8854
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1152887
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212965
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33559}
This is a reland of 89cb65ed663a9000b9f7c90a78039bd85731e9ae
... and f28aade91dcc2cb8f590dc1379ac7ab5c1981909
Reason for revert: crashes due to uninitialized pacing_bitrate_
crbug.com/1190547
Apparently pacer() is sometimes being used before EnsureStarted()
Fix: Instead of delaying first call to SetPacingRates(),
this CL no-ops MaybeProcessPackets() until EnsureStarted()
is called for the first time.
Original change's description:
> [Battery]: Delay start of TaskQueuePacedSender.
>
> To avoid unnecessary repeating tasks, TaskQueuePacedSender is started
> only upon RtpTransportControllerSend::EnsureStarted().
>
> More specifically, the repeating task happens in
> TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self
> task_queue_.PostDelayedTask().
>
> Bug: chromium:1152887
> Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560
> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33421}
Bug: chromium:1152887
Change-Id: I9aba4882a64bbee7d97ace9059dea8a24c144f93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33554}
This API should allow existing factories to be used unmodified, but
offers a new API that documents ownership better and does not use
sigslot.
Bug: webrtc:12598
Change-Id: I0f68371059cd4a18ab07b87fc0e7526dcc0ac669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212609
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33553}
At the point where an EncodedImage is reported to the
EncoderBitrateAdjuster (in order to estimate utilization), the image
data has been cleared so the size is 0 - meaning the esimtated
utilization is 0 so pushback is in effect only applied at the
beginning before an estimate is available.
This CL fixes that by explicitly using spatial/temporal id and size in
bytes, rather than passing along the EncodedImage proxy.
It is unclear when this broke, but the regression seems rather old.
This CL will affect the encoded bitrate (and thus indirectly BWE
ramp-up rate), but should avoid exessive delay at low bitrates.
Perf bots will likely trigger alerts, this is expected.
In case there are undesired side-effects, we can entirely disable the
adjuster using existing field-trials.
Bug: webrtc:12606
Change-Id: I936c2045f554696d8b4bb518eee6871ffc12c47d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212900
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33550}
Deleting obsolete stats. Spec: https://www.w3.org/TR/webrtc-stats/
1. RTCInbound/OutboundRtpStats.isRemote: No longer useful with remote stream stats
2. RTCIceCandidateStats.deleted: This field was obsoleted because if the ICE candidate is deleted it no longer appears in getStats()
I also marked as many other obsoleted stats possible according to spec. I am not as confident to delete them but feel free to comment to let me know if anything is off / can be deleted.
Bug: webrtc:12583
Change-Id: I688d0076270f85caa86256349753e5f0e0a44931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33549}
Follow-up CL to VP8 and VP9 encoders taking care of mapping.
Context again:
This CL is part of Optimized Scaling efforts. In Chromium, the native
frame buffer is getting an optimized CropAndScale() implementation. To
support HW accelerated scaling, returning pre-scaled images and skipping
unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale
instead of libyuv::XXXXScale and 2) only map buffers it actually intends
to encode.
In this CL, VideoStreamEncoder no longer calls GetMappedFrameBuffer() on
behalf of the encoders, since the encoders are now able to either do the
mapping or performs ToI420() anyway.
- Tests for old VSE behaviors are updated to test the new behavior (i.e.
that native frames are pretty much always forwarded).
- The "having to call ToI420() twice" workaround to Android bug
https://crbug.com/webrtc/12602 is added to H264 and AV1 encoders.
Bug: webrtc:12469
Change-Id: Ibdc2e138d4782a140f433c8330950e61b9829f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211940
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#33548}
`RTCInboundRtpStreamStats.lastPacketReceivedTimestamp` must be a time
value in milliseconds with Unix epoch as time origin (see
bugs.webrtc.org/12605#c4).
This change fixes both audio and video `RTCInboundRtpStreamStats` stats.
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12605
Change-Id: I68157fcf01a5933f3d4e5d3918b4a9d3fbd64f16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212865
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33547}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
In refactoring CL https://webrtc-review.googlesource.com/c/src/+/210340,
the RTCRemoteInboundRtpStreamStats hierarchy was updated to inherit from
RTCReceivedRtpStreamStats but we forgot to update the
WEBRTC_RTCSTATS_IMPL() macro to say that RTCReceivedRtpStreamStats is
the parent. As a consequence, RTCReceivedRtpStreamStats's members
(jitter and packetsLost) were not included when iterating over all
members of RTCRemoteInboundRtpStreamStats, which means these two merics
stopped being exposed to JavaScript in Chromium.
There is sadly no way to safe-guard against this, but the fix is simple.
TBR=hta@webrtc.org,meetwudi@gmail.com
Bug: webrtc:12532
Change-Id: I0179dad6eaa592ee36cfe48978f2fc22133b8f45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212866
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33543}
