Low-Coverage-Reason: EXPERIMENTAL_CODE Code is behind field trial that will only be used for testing.
Bug: webrtc:13322
Change-Id: Ie306be808381b3a20b4e0d58349927bf3524018a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335840
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41608}
This is a step towards sending audio timestamps from Meet in iOS.
Next step is to enable sending the audio timestamps (in harmony).
After enable absolute-capture-time header extension in harmony, the receiving participants will be able to store E2E audio latency and A/V sync metrics.
Bug: webrtc:13609
Change-Id: I797c1ed0035625ed065307314ac34c932c5abe7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334720
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41574}
This allows using different encoder and decoder implementations in a test. For example, to encode with SW encoder and to decode with HW decoder or vice versa.
Bug: webrtc:14852
Change-Id: Ic100cba2158fb6311b84a54a0831f2a0dcff9270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335300
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@{#41571}
CGDisplayStreamCreate is an deprecated API. It was believed that the use
of it was disabled on Sonoma through the setting allow_iosurface = false
[1], which causes the thumbnails to be created by the API CGDisplayCreateImage.
This API is not marked as deprecated at the moment.
However, although the thumbnails are created through CGDisplayCreateImage,
CGDisplayStreamCreate() is still called and runs in the background.
This makes the capture chip appear.
No capture chip appears if this CL is landed and the ScreenCaptureKit
thumbnail capturer is enabled,
--enable-features="ScreenCaptureKitMac,ScreenCaptureKitStreamPickerSonoma,ThumbnailCapturerMac:capture_mode/sc_screenshot_manager"
[1] https://chromium-review.googlesource.com/c/chromium/src/+/4892397
Bug: chromium:1486851
Change-Id: I3422efffc57dcb3e8965f19a5eca7f2a95d62da1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334721
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41563}
Tracking keyframe packets is a useless optimization that kicked in when the nack list is full (1000 packets).
Bug: none
Change-Id: I134ecb4d51131718e5bb8775847fbde18f262ef9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334645
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41547}
This CL addresses a crash we started seeing in M121 where a
function is being called on loss_based_bandwidth_estimator_v2_
without checking whether it is enabled (it's not) which leads
to absl::optional<> throwing since config_ is not valid.
Bug: chromium:1518852
Change-Id: Iffef1051fe7988046e33a709ce281aebefd2bcd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334103
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41538}
in favor of stock StartsWith and HasSubstr matchers provided by gmock.
BUG=None
Change-Id: Ib7e9a0ac73d506c349b8ec102dd4236767077d61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41531}
Bug: webrtc:15719
Change-Id: I7daf8ee5b90fbe9f1246f1d99211ffa0d8a19f73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330780
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41503}
rtc::TaskQueue is a wrapper of TaskQueueBase providing no extra functionality in this case
Bug: webrtc:14169
Change-Id: I5eb27a5dbb16f6097a9c71c2633c807808e50c05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333800
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41501}
Ensure top_active_prio_level_ is set to -1 in MaybeUpdateTopPrioLevel if
last packet is purged.
Bug: webrtc:15740
Change-Id: I81df9ee084de89f79b8ab79db8ce52fe1e20738a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333883
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41498}
This field trial was added 5 years ago in
https://webrtc-review.googlesource.com/c/src/+/111883
probably as a safe guard, but looks never used.
Bug: webrtc:11503
Change-Id: Ia9544b652b25fad4c614d66fe020f3d994c96505
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41490}
This is a reland of commit 63d03f586bb668f72113b61030ec0930aa192010
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}
Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41467}
Adds separate priorities for audio and video retranmission.
Done by adding an original type to RtpPacketToSend.
Add possiblity to set TTL for audio nack, video nack and video packet separately.
Oldest packet for these types are dropped when a new packet of that type is pushed to the pacer, or when the pacer switch current priority type to that priority.
Effect is that:
-pacer queue does not grow unlimited for these types if a TTL has been set.
-an old packet is not sent.
Bug: webrtc:15740
Change-Id: I38718bc570aebca54eacbded69824905f3694f41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331823
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41414}
Fix the unintended disabling of RTP retransmissions for cloned encoded
frames, caused by passing an infinite "expected_retransmission_time".
Instead use a constant 10ms for now. For frames encoded locally, this is
set from an estimate of the RTT, but we currently don't have access to
that here (TODO added to pipe it through)
If an integration is cloning and then sending frames it received, it's
almost certainly resending received media to other peers on a local
network, so 10ms is a fair upperbound.
Tested locally with Chrome on Mac, configuring packet drops & observing
on chrome://webrtc-internals that retransmission packets are now sent.
Bug: chromium:1512631
Change-Id: I2483415dc7e0079f8a7b66f6607f4907698514c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41405}
which come from the a=fmtp:<pt> lines in the SDP and were used as either
std::map<std::string, std:string>
with three aliases,
cricket::CodecParameterMap
SdpAudioFormat::Parameters
SdpVideoFormat::Parameters
Use webrtc::CodecParameterMap in all places.
BUG=None
Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
After https://webrtc-review.googlesource.com/c/src/+/329141, best candidate can still be less than acked rate if not_increase_if_inherent_loss_less_than_average_loss, or the selected candidate is 95% of current estimate. This cl/ is ensure the previous cl works as intended. And add unit test.
Bug: webrtc:12707
Change-Id: Ie5683ca8ea51f6d80c4c59cbf08c22e8b24c0cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329441
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41298}
Move logging of decode failure from VCMGenericDecoder to VideoReceiveStream2 where remote SSRC is always known. Log frame details such as size and resolution which help to identify this frame in bitstream dump.
Bug: b/309132190
Change-Id: Ibe50799e448ffdc19f9857cc1625cfde0d7aa7a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328821
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41276}
Move some logic from PacketBuffer to NetEqImpl.
Bug: webrtc:13322
Change-Id: I88b1e55c0cd69700730d9ed41be04fcf1effa03f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328861
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41270}
These flags were never experimented or launched.
Bug: webrtc:12707
Change-Id: Iefedeade52fdcf7f978894c4bf837261810f41bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329080
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41265}
PacingController per default use a burst interval of 40ms. The behaviour can still be overriden by using the method SetSendBurstInterval.
Bug: chromium:1354491
Change-Id: Ie3513109e88e9832dff47380c482ed6d943a2f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41254}
This means that RtpPacketHistory::PaddingMode::kRecentLargePacket is
used per default.
Bug: webrtc:15201, b/284281602
Change-Id: If8feb66105a9b1e13ae4cb28a44a74c8839b72e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327602
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41215}
