danilchap@webrtc.org suggested to add a converter for NtpTime <-> UTC Timestamp for in https://webrtc-review.googlesource.com/c/src/+/365641.
This CL add a NtpTime -> UTC Timestamp in Clock, and change code to start to use the new function.
Bug: None
Change-Id: If4af6cb8e31c1731692edfb8358e67b7a43226a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366001
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43293}
This fixes the bug where scaleResolutionDownTo must be specified even
on inactive encodings (scaleResolutionDownTo is the JavaScript name for
what is called requested_resolution inside WebRTC).
Bug: chromium:375048792
Change-Id: I3206ef7de09eaba24a5b4305d888ec4904617e58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366522
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43292}
These interfaces were deprecated 6 years ago and have not been
functional for a long time. It is safe to remove them.
Bug: None
Change-Id: Icbc85758551a96b8da10fbf16a1771c3641d1ac3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366500
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43291}
Otherwise, the captured variables are immutable, so a subsequent
std::move() silently degrades to a copy.
-- Add missing () for consistency with other no-arg lambda
captures.
Bug: webrtc:374845009
Change-Id: I205589ff8047446918a45203a22620846b271187
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43289}
The const-ref result of .str() must be copied into the returned
value, whereas the result of .Release() can be moved.
Bug: webrtc:374845009
Change-Id: I3abc98be30ce9947127c7664f5ffa6846b772ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43288}
Mentioned reasons for re-implementing it do not apply when simplest form of aliasing is used.
Bug: webrtc:42225969
Change-Id: If8b51f173faf4c66cde74413f4fbc7c72ad87323
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366460
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43285}
Similar to CompactNtpRttToTimeDelta but can return negative values.
Bug: webrtc:42225697
Change-Id: Iea97502ea73eb6240f42c2040cdc576e51298704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366422
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43284}
For now, old file forward include api/transport/ecn_marking.h
Done in preparation for more usage of this enum when handling received
RFC8888 feedback.
Bug: webrtc:42225697
Change-Id: I022c2b7f1e7f986b24aa32b8911ad67c6640a5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366440
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43282}
CongestrionControllerGenerator tracks received packets per SSRC.
Lost packets are included in rtcp:CongestionControlFeedback::Packets()
This is done in order to be able to track lost packets between
feedback packets.
Bug: webrtc:42225697
Change-Id: Ib47d9b55c3d150cb98a44a4f3997cfcfe6c5fbb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43274}
This would simplify migrating from PeerConnectionFactoryDependencies::audio_processing
for users who use own implementation of the AudioProcessing
Bug: webrtc:369904700
Change-Id: Id05f7280fd01a3e8fd4953f1b24b2467335ab065
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43273}
This is similar to MatchesRtpCodec but not an exact match of parameters, unspecified parameters are treated as default. Use IsSameRtpCodec for comparison when codec is configured via encodings.
Bug: b:299588022
Change-Id: I0ea800e50af6f5666e3e867a928e15b0aa044635
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365142
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43272}
In the `MaybeProcess` method, a new variable `time_until_cc_rep` is introduced
to track the time until congestion control feedback generation is processed.
The minimum of this value and the times until RBE and transport sequence number
feedback processing are calculated.
Co-authored-by: Shridhar Majali <smajali@nvidia.com>
Bug: webrtc:42225697
Change-Id: I44173062d8f8f84bf7e7791e05578c0ffc4fd017
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365273
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43271}
Directly use RtpPacketToSend instead of RtpPacketSendInfo
For RFC8888 we will need to match SSRC and RTP sequence number with the transport sequence number that only exist on the sending side.
For retransmitted packets, RTX, RtpPacketSendInfo contain original SSRC and original sequence number.
Bug: webrtc:42225697
Change-Id: Iafa5d851ca5c51c85e4607ed4c1919d96da6084a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366000
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43264}
and use uint8_t instead of unsigned char. Follow-up from
https://webrtc-review.googlesource.com/c/src/+/365274
BUG=webrtc:357776213
Change-Id: Ibc97e5cc85316ba69b4133b7f3c42e3afbdd7abd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365540
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43263}
Indirect input deps for imported proto is now handled by deps in
proto_library template, so we don't need to use proto_data_sources
anymore after https://crrev.com/c/5919027.
To remove proto_data_sources from proto_library template, let me clean
up proto_data_sources usages from this repository.
Bug: chromium:366137880
Change-Id: I288a30004f7d622be502477a0567b00d19432e89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366060
Auto-Submit: Takuto Ikuta <tikuta@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43261}
with the exception of the legacy stats collector unittest
BUG=None
Change-Id: I1ef28ab2052b1194ec788fa69606418d42d5a433
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366020
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43258}
is provided by the encoder.
Note that the number of temporal streams is hardcoded to kMaxTemporalStreams (4).
Bug: b/369617423
Change-Id: I05204bc1aebc9f344d59add7b097f3e653950444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365741
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Brennan Waters <brennanw@google.com>
Cr-Commit-Position: refs/heads/main@{#43257}
This would allow users of the voip engine to migrate away from the AudioProcessingBuilder
Bug: webrtc:369904700
Change-Id: Ie4f6f4579e185ff6366333a3f37e6aaa23b892b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365920
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43255}
This adds a Call-based test, that sets up video-pipeline with a VP8
encoder and the corruption detection header extension configured.
It then verifies that the corruption likelihood metrics are populated
in the receive stream stats.
Bug: webrtc:358039777
Change-Id: Ide005459a801778de4238e786f13efc8c3245f3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365860
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43254}
It would allow to use EncodedImageBufferInterface with gtest container matchers.
Bug: None
Change-Id: Iae37d1a019e044a4ec583c32e8141fe0758e60ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365501
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43253}
This allows this type to meet the requirements of e.g.
std::ranges::range, which is necessary for it to work with the std::span
range constructor, or the "non-legacy" constructor for Chromium's
base::span.
Bug: chromium:364987728
Change-Id: I6cb2b9c6d849c97e304719140dcb967a9e2c254c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365780
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43251}
Compiling webrtc with `-Werror=unused-parameters` is failling duo to
those parameters.
Also, it shouldn't harm us to put those in comment for code readability as
well.
NOTE: This time I made sure to iterate over the C files in the
audio_processing folder and compile them using gcc.
On the original CL that was reverted - that failed with the same error
Danil mentioned. This time it seems fine.
I'll make sure to run the same script on the rest of my CLs for sanity
Bug: webrtc:370878648
Change-Id: I83cea3a08777e21d26a95bcad503a2d1b74566eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364537
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43249}
We should use the Timestamp type, rather then int64, to store timestamps. In https://webrtc-review.googlesource.com/c/src/+/365001/ an additional int64 timestamp was added (last_sender_report_timestamp_ms).
This CL fixes the new timestamp, as well as other similar timestamps in MediaReceiverInfo (last_sender_report_utc_timestamp_ms and last_sender_report_remote_utc_timestamp_ms).
Bug: webrtc:372393493
Change-Id: I0e473730e85a69ec595b421e2c3db920364008eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43248}
