When an IRAP frame was present in the Aggregation Packet,
the control flow was incorrectly transferred to SPS parsing
due to ABSL_FALLTHROUGH_INTENDED within the IRAP case statement,
resulting in a parsing error and generating a warning log.
A break statement has been introduced to prevent this fallthrough.
Bug: webrtc:13485
Change-Id: I523fbf548f14b31eae7c41f607fe33572f094aac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346381
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#42132}
fuchsia.ui.display.singleton
We previously used 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:64206
Test: applied changes manually to local chromium repo's third_party/webrtc directory and compiled
Change-Id: If3c7fbd641ebd3b3333e7e5f1126f8f3ae3b97e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322780
Commit-Queue: Caroline Liu <carolineliu@google.com>
Reviewed-by: Emircan Uysaler <emircan@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42104}
That forces external field trials, thus VideoCoding will be able to remove dependency on the global field trials string through FieldTrialBasedConfig class.
Bug: webrtc:10335
Change-Id: I6d22a7d20a4433801a0086b0863cda78e91f4f60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347646
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42092}
This reverts commit 501c4f37bf.
Patch set 1 contains pure reland.
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
Bug: webrtc:14928
Change-Id: I6a8660da20ac54237f04a29461e03b31bd988bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347643
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42086}
This reverts commit 33cc83595a.
Reason for revert: Perf bots showed that this cl cause a change in metrics. It looks like it is for the better, but we want this to be behind a field trial.
Original change's description:
> Ignore allocated bitrate during initial exponential BWE.
>
> The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
> That is the, initial probe will try to probe up to the max configured bitrate.
>
> ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
> continue up to the max configured bitrate, regardless of of the max
> allocated bitrate.
>
> Bug: webrtc:14928
> Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42049}
Bug: webrtc:14928
Change-Id: I56ba58560b6857b6069552c02df822691f7af64d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347622
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42081}
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
continue up to the max configured bitrate, regardless of of the max
allocated bitrate.
Bug: webrtc:14928
Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42049}
The width and height of mapped_buffer must match the d_w and d_h members
of frame_to_encode_, which is passed to aom_codec_encode().
Bug: b:330482827
Change-Id: I85d8c82133768685565f165eafc893c42dc40b12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345807
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Wan-Teh Chang <wtc@google.com>
Cr-Commit-Position: refs/heads/main@{#42036}
Instead of relying on the global field trial string
Bug: webrtc:10335
Change-Id: I491be089ffc725fd28483edf10eae4ae5d17d651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346263
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42021}
This hard-codes the behavior to mode 3 with a threshold of 0.5 like was
already done by FetchPreEchoConfiguration.
Bug: webrtc:14205
Change-Id: I48d47a77c9df0001460788b504524203417f9647
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345483
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42015}
This CL partly restores the changes that were introduced in
https://webrtc-review.googlesource.com/c/src/+/344681
The predefined SdpVideoFormat for AV1 causes some backwards
compatibility issues with downstream projects that are using
the preliminary codec name AV1X.
Bug: b/333007070
Change-Id: I2d4df241d47b399b0012e6095dd6c2445e60e2c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345941
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42011}
This changes the libvpx VP9 encoder to generate the scalability mode based on the current encoding parameters when using layer activation.
Tested: Ran with L3T3_KEY reduced to L2T3_KEY and L1T3 due to bandwidth or layer activation. Added unit tests.
Bug: webrtc:15892
Change-Id: Iaedca4ea5fc3a692996666ceaf0d6aa03fb058a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344760
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42007}
The predefined SdpVideoFormats were not used everywhere,
which caused a discrepancy between send/receive capabilities
for AV1. This CL solves the immediate problems by making sure
send/receive capabilities for AV1 are reported the same way.
Fixed: chromium:331565934
Change-Id: I073091b7b5f987c7f434c17276fd84047ec723c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41991}
This CL updates H26xPacketBuffer to store and prepend SPS and PPS for
H.264 bitstreams when IDR only keyframe is allowed.
Bug: webrtc:13485
Change-Id: Ic1edc623dff568d54d3ce29b42dd8eab3312f5cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342225
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41986}
This is a reland of commit 7ae48c452a with updated RtpVp9RefFinder
RtpVp9RefFinder relied on the fact that frames with (inter_pic_predicted=true && inter_layer_predicted=true) were marked as keyframes. Since this is not the case anymore, the related code paths in RtpVp9RefFinder have been deleted.
Calculation of gof_info_[] index for non-keyframes has been updated to account for that fact it is now possible to received multiple T0 frames belonging to the same temporal unit (we don't need to do "unwrapped_tl0 - 1" in this case).
Original change's description:
> Mark frames with inter_layer_predicted=true as delta frames
>
> As it is currently implemented, the VP9 depacketizer decides packet's frame type based on p_bit ("Inter-picture predicted layer frame"). p_bit is set to 0 for upper spatial layer frames of keyframe since they do not have temporal refs. This results in marking packets of upper spatial layer frames, and, eventually these frames, of SVC keyframes as "keyframe" while they are in fact delta frames.
>
> Normally spatial layer frames are merged into a superframe and the superframe is passed to decoder. But passing individual layers to a single decoder instance is a valid scenario too and is used in downstream projects. In this case, an upper layer frame marked as keyframe may cause decoder reset [2] and break decoding.
>
> This CL changes frame type decision logic in the VP9 depacketizer such that only packets with both P and D (inter-layer predicted) bits unset are considered as keyframe packets.
>
> When spatial layer frames are merged into a superframe in CombineAndDeleteFrames [1], frame type of the superframe is inferred from the lowest spatial layer frame.
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/frame_helpers.cc;l=53
>
> [2] https://source.corp.google.com/piper///depot/google3/third_party/webrtc/files/stable/webrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc;l=209
>
> Bug: webrtc:15827
> Change-Id: Idc3445636f0eae0192dac998876fedec48628560
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343342
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41939}
Bug: webrtc:15827
Change-Id: Ic69b94989919cf6d353bceea85d0eba63bc500ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344144
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41985}
With this cl
RtpTransportControllerSend::OnAddPacket is instead directly invoked from PacketRouter::SendPacket instead of going via RTP module.
Transport sequence numbers are instead of directly written to header
extension, added to RtpPacketToSendMetaData and written to the extenion
by RTP module.
This is to allow transport sequence numbers without actually sending
them in an extension.
Bug: webrtc:15368
Change-Id: Idd03e02a4257dfc4d0f1898b2803345975d7dad2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344720
Reviewed-by: Erik Språng <sprang@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41974}
This reverts commit d427e83a15.
Reason for revert: Flaky test fixed.
Refactor FrameCandenceAdapter to keep track of input frame rate. This fixes an issue where frame rate is calculated too low if congestion window drop a frame.
Also a field trial WebRTC-FrameCadenceAdapter-UseVideoFrameTimestamp is added to control if VideoFrame timestamp should be used or local clock when calculating frame rate.
Uma is recorded to tell if input frame timestamp is monotonically increasing.
Bug: webrtc:10481, webrtc:15887, webrtc:15893
Change-Id: I76268aa0991dbc99c1b881fb251a76aa54ff2673
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344561
Reviewed-by: Erik Språng <sprang@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41972}
This reverts commit 784af1f42e.
Reason for revert: Seems like test test_support_unittests
ResolutionAdaptsToAvailableBandwidth is flaky with this cl.
Original change's description:
> FrameCadenceAdapter keep track of Input framerate
>
> Refactor FrameCandenceAdapter to keep track of input frame rate.
>
> Also a field trial WebRTC-FrameCadenceAdapter-UseVideoFrameTimestamp is added to control if VideoFrame timestamp should be used or local clock when calculating frame rate.
> Uma is recorded to tell if input frame timestamp is monotonically increasing.
>
> Bug: webrtc:10481, webrtc:15887
> Change-Id: I6d698e9f9dcfe8c023d2d35371435c47f70102b0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342760
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41967}
Bug: webrtc:10481, webrtc:15887
Change-Id: Id9672764768f2f40f8e711e990ad8ac18c28efcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344560
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41969}
Refactor FrameCandenceAdapter to keep track of input frame rate.
Also a field trial WebRTC-FrameCadenceAdapter-UseVideoFrameTimestamp is added to control if VideoFrame timestamp should be used or local clock when calculating frame rate.
Uma is recorded to tell if input frame timestamp is monotonically increasing.
Bug: webrtc:10481, webrtc:15887
Change-Id: I6d698e9f9dcfe8c023d2d35371435c47f70102b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342760
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41967}
This reverts commit 802dd5bdbd.
First patch set is pure reland.
New patch set adds field trial flag.
Original change's description:
Stop exponential probing if 2xmax allocated bitrate lower than BWE.
Without this, if max allocated bitrate is lowered while exponential probing is ongoing, a new probe can be sent at a rate of the new low max allocated bitrate which may cause BWE to decrease.
Bug: webrtc:14928
Change-Id: I35c341bbaced800d9931657d62c73a17a3279b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344440
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41965}
When refactoring the WGC capture path, the check of a closed source
had been placed at a level where the notification of a closed source
was left without being detected since the error message was never
provided to the main WgcCapturerWin::CaptureFrame() which sends the
error message up to the client.
This trivial change ensures that WgcCapturerWin::CaptureFrame() returns
with DesktopCapturer::Result::ERROR_PERMANENT as soon as
WgcCaptureSession::OnItemClosed() has been triggered and it will
ensure that the WGC capture session stops and that any attached
MediaStreamTrack will signal its onended event as expected.
Bug: chromium:330863510
Change-Id: I57e44df417c33efa0595fc277cac5429cf539b26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344420
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41963}
The purpose is to be able to create a RtpPacketSendInfo from Pacing and RtpPacketSendInfo only.
This allow further refactoring where we directly in PacketRouter can notify BWE and early loss detection that a packet will be sent.
RtpPacketSendInfo::From is mostly added to be able to test conversion.
Bug: webrtc:15368
Change-Id: I5ebe2dc91d2eedf2c86e62c3f9738437082a49e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343766
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41961}
This reverts commit 5a4ce03101.
Reason for revert: Breaks tests in downstream project.
Original change's description:
> Stop exponential probing if 2xmax allocated bitrate lower than BWE.
>
> Without this, if max allocated bitrate is lowered while exponential probing is ongoing, a new probe can be sent at a rate of the new low max allocated bitrate which may cause BWE to decrease.
>
> Bug: webrtc:14928
> Change-Id: Id8e314740c2403d3b801d28f634dbc718f77c16e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343384
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41929}
Bug: webrtc:14928
Change-Id: I0d48b37bfb8684fd490f5685e510b438a83254b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343900
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41958}
This reverts commit 7ae48c452a.
Reason for revert: breaks RtpVp9RefFinder
Original change's description:
> Mark frames with inter_layer_predicted=true as delta frames
>
> As it is currently implemented, the VP9 depacketizer decides packet's frame type based on p_bit ("Inter-picture predicted layer frame"). p_bit is set to 0 for upper spatial layer frames of keyframe since they do not have temporal refs. This results in marking packets of upper spatial layer frames, and, eventually these frames, of SVC keyframes as "keyframe" while they are in fact delta frames.
>
> Normally spatial layer frames are merged into a superframe and the superframe is passed to decoder. But passing individual layers to a single decoder instance is a valid scenario too and is used in downstream projects. In this case, an upper layer frame marked as keyframe may cause decoder reset [2] and break decoding.
>
> This CL changes frame type decision logic in the VP9 depacketizer such that only packets with both P and D (inter-layer predicted) bits unset are considered as keyframe packets.
>
> When spatial layer frames are merged into a superframe in CombineAndDeleteFrames [1], frame type of the superframe is inferred from the lowest spatial layer frame.
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/frame_helpers.cc;l=53
>
> [2] https://source.corp.google.com/piper///depot/google3/third_party/webrtc/files/stable/webrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc;l=209
>
> Bug: webrtc:15827
> Change-Id: Idc3445636f0eae0192dac998876fedec48628560
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343342
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41939}
Bug: webrtc:15827
Change-Id: I697a057b8b3e88c07499f77c42f014da43cf1dc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343763
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41953}
The purpose of these new methods are to allow creating a RTP packet with
sequence numbers that
can be inspected and is ensured to be sent if SendPacket is invoked.
virtual bool CanSendPacket(const RtpPacketToSend& packet) const = 0;
virtual void AssignSequenceNumber(RtpPacketToSend& packet) = 0;
virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& pacing_info) = 0;
Bug: webrtc:15368
Change-Id: I671e737575e15328e796aa98761a4d540c5812d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343785
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41951}
RtpPacketToSend::transport_sequence_number
packed_id is set to be 64 bit to align with rtc::PacketOptions.
packet_id is only set to RtpPacketToSend::transport_sequence_number if
TransportSequenceNumber header extension is not used in order to not
change current behaviour.
Bug: webrtc:15368
Change-Id: Ia532714226421422bdb292f8dd34b175560e9dc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41950}
And move writing of the header extension from PacketRouter to
RtpSenderEgress::SendPacket.
Bug: webrtc:15368
Change-Id: Ieb18af4bc20115bf02d37e1f9a815a5c120975a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343786
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41949}
Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead.
The old fields are preserved for compatibility with downstream projects, but will be removed in the future.
Bug: webrtc:15788
Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41947}
As it is currently implemented, the VP9 depacketizer decides packet's frame type based on p_bit ("Inter-picture predicted layer frame"). p_bit is set to 0 for upper spatial layer frames of keyframe since they do not have temporal refs. This results in marking packets of upper spatial layer frames, and, eventually these frames, of SVC keyframes as "keyframe" while they are in fact delta frames.
Normally spatial layer frames are merged into a superframe and the superframe is passed to decoder. But passing individual layers to a single decoder instance is a valid scenario too and is used in downstream projects. In this case, an upper layer frame marked as keyframe may cause decoder reset [2] and break decoding.
This CL changes frame type decision logic in the VP9 depacketizer such that only packets with both P and D (inter-layer predicted) bits unset are considered as keyframe packets.
When spatial layer frames are merged into a superframe in CombineAndDeleteFrames [1], frame type of the superframe is inferred from the lowest spatial layer frame.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/frame_helpers.cc;l=53
[2] https://source.corp.google.com/piper///depot/google3/third_party/webrtc/files/stable/webrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc;l=209
Bug: webrtc:15827
Change-Id: Idc3445636f0eae0192dac998876fedec48628560
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343342
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41939}
FFmpeg has removed this field and usage of it in chromium must be
removed before the ffmpeg dependency is updated. The chromium media
change can be found here:
https://chromium-review.googlesource.com/c/chromium/src/+/5384308
The usage of the field in webrtc seems only to be for sanity checking,
so it should be just safe to remove entirely, since webrtc does not
expect re-ordering at all.
Bug: chromium:330573128
Change-Id: I9c5854ec82c3ad2d55374ea4eaa0c571437f8267
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343840
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ted (Chromium) Meyer <tmathmeyer@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41935}
Without this, if max allocated bitrate is lowered while exponential probing is ongoing, a new probe can be sent at a rate of the new low max allocated bitrate which may cause BWE to decrease.
Bug: webrtc:14928
Change-Id: Id8e314740c2403d3b801d28f634dbc718f77c16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343384
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41929}
Webrtc is build with FFmpeg sources on defined in the include path
through the -I flag, so they should be included this way instead. This
would otherwise cause a conflict when the chromium ffmpeg sources move
from third_party/ffmpeg/* to third_party/ffmpeg/src/*
BUG: chromium:329282834
Change-Id: Id8f7e91446bdc536db77e74388a73e51f5111ffc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342820
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ted (Chromium) Meyer <tmathmeyer@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41899}
This seems to confuse perfetto, and the data ends up on its own track
and the end event is just ignored. As it was invalid, I am assuming it
is not used, and can be simply removed.
#rtc_fixit
Bug: webrtc:15867
Change-Id: I31a814f6c2147c3ce534726bf9046a79369b9eb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342761
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41896}
This seems to confuse perfetto, and the data ends up on its own track
and the end event is just ignored. As it was invalid, I am assuming it
is not used, and can be simply removed.
#rtc_fixit
Bug: webrtc:15867
Change-Id: I77e59adcd35c51911474446a5f92505bf6b860f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342780
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41892}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
There's an AV1 encoder speed setting 11 that is supposed to be used
for screen sharing content.
Bug: chromium:328598314
Change-Id: Id97898554a740eb1684d03c782c718c19f4c95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342201
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41874}
We want to copy device id to _lastUsedDeviceName variable, but we use
length of display name instead of length of device id, which might be
longer than expected and we end up reading beyond the source string.
Bug: webrtc:15853
Change-Id: Id278ed7e361ead85475910adec18b9db51e6890b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341521
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41844}
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
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@{#41833}
Announce that we support SPA_DATA_DmaBuf and tell PipeWire not to map
memory for us so we can handle it ourself, similar like we do in case of
screen sharing. This fixes an issue when a camera is already in use by
gstreamer (pipewiresrc), where DMABufs are used, and we try to share
same camera and get no content, as PipeWire doesn't want to mmap DMABuf
memory for us and we get NULL data pointers.
Firefox bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1876895
Bug: webrtc:15654
Change-Id: I788d8d12b2fcd5588329d7265e45b479f74bb628
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338921
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41826}
Marking capturer as failed will indicate consumers will not be getting
any new frames by sending back ERROR_PERMANENT and let them know that
screencast can be stopped from their side. This will make screencast to
stop when a window we share is closed or when screencast is closed from
system tray.
Bug: chromium:40276865
Change-Id: Ia2c13461bd3126cab9c4838b8aa6840578562e9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339560
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41817}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
which needs to be added to the remote codecs a=fmtp:
This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes.
This parameter allows for large-scale experimentation and A/B testing
whether the new behavior has advantages. It is to be considered
transitional and may be removed again in the future.
BUG=webrtc:10107
Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41805}
To prepare for a new AudioLevel struct to be added to the public WebRTC API, rename the internal RTP extension reader/writer class to AudioLevelExtension. A temporary alias is provided to avoid breaking downstream projects.
Bug: webrtc:15788
Change-Id: Ie231668f25932fd9b539229114128b1d0b949a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339887
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41787}
Same can be achieved by having multiple Parse functions in the same
RtpDependencyDescriptorExtension trait
Bug: None
Change-Id: I4eab0001d1ffff631a9d70fafde13e51f5c6ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340320
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41786}
It used up to 3 threads for QVGA on Android before. This change disables Android-specific code path in NumberOfThreads() and uses the generic settings, which configure 1 thread for resolutions <=VGA, instead. The change is guarded by a killswitch.
For reference, frame encode time for VGA 512kbps using 1 thread on Pixel 2 (7 years old device; SD835) is ~5.5ms: https://chromeperf.appspot.com/report?sid=6e80c701ef6ff0d008a299fb122a16f0d2600ddfcd9981d3d75cd722c92b2869
Bug: webrtc:15828, b/316494683
Change-Id: I0e9571ede64c6cb77d529d21ccb0310ccb8bfdaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337601
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41770}
Mark FU packets with type between kBlaWLp and kRsvIrapVcl23 as key frames.
This behavior aligns with AP and single NALU.
Bug: webrtc:13485
Change-Id: I51762e89ebb4829b50524d9f5476f2d5d9c093f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338860
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41764}
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}
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}
This is a reland of commit a2655449ee
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}
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}
