7222 Commits

Author SHA1 Message Date
Jan Grulich
3aa47cfd30 PipeWire camera: get max FPS for each format when specified as list
In many cases, the framerate can be specified as list of possible values
and in that case, we would end up with max FPS to be set to 0 as this
case was not handled.

Bug: webrtc:42225999
Change-Id: I036af6db1da3309b1310b754504369e8fe392d09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362961
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43057}
2024-09-20 06:35:22 +00:00
Joachim Reiersen
bba1a2e476 Propagate Environment to RtpPacketHistory
Passing Environment instead of Clock into this class simplifies some plumbing for downstream consumers that need to read field trials within this class.

Bug: webrtc:362762208
Change-Id: Ia501e9f7f1d91a8115a2f71fb005dd35146db172
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362535
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43048}
2024-09-19 09:39:07 +00:00
Hanna Silen
54903b407f Delete transient suppression code
Transient suppression is no longer used in audio processing after
https://webrtc-review.googlesource.com/c/src/+/355880.

Bug: webrtc:357281131
Change-Id: Iec5e9ddc300dfdda2dbb82066d12e1129e3cb1df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362840
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43045}
2024-09-18 16:52:10 +00:00
Takuto Ikuta
b08a045e92 fix missing deps for proto compile actions
We need to have imported proto as proto_data_sources in BUILD.gn to
run the action remotely without workaround config in siso.

Bug: b/366137880
Change-Id: I053774f00b761520a8a85154e386da3edb8f39b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362680
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Takuto Ikuta <tikuta@google.com>
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/main@{#43040}
2024-09-18 05:38:30 +00:00
Danil Chapovalov
52ea2c3d2a Propagate FieldTrialsView to query WebRTC-StableTargetRate field trial
Bug: webrtc:42220378
Change-Id: Ie2a2c3eccc36c98f09176eb6f4c5f06ded9f516f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362701
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43036}
2024-09-17 14:24:41 +00:00
Lionel Koenig
098c128a15 Explicitly use the Opus DTX encoder state.
Use the DTX state from inside the Opus encoder instead of trying to
mimic the logic outside.

Bug: None
Change-Id: I852044fee261a5b7f9255c557a27adfd0b1701bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43034}
2024-09-17 13:28:45 +00:00
Danil Chapovalov
a1ed306293 Cleanup unused members in RtpRtcp::Configuration
They are now passed as part of the Environment

Bug: webrtc:362762208
Change-Id: I02868e9f41533a546f62fe30fdc6f3a7708eb346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362084
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43032}
2024-09-17 12:02:19 +00:00
Henrik Lundin
8487d3248b Remove all use of AcmReceiver from WebRTC
The class itself and its unit test remains, for now, but will be removed
later.

Bug: webrtc:14867
Change-Id: I36cec8fca7913663f63c53622ed2760e5e048c2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362580
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43023}
2024-09-16 08:49:25 +00:00
Lionel Koenig
ec38238af7 Ensure the AudioCodingModule is reset when sending is stopped.
Bug: webrtc:42226041
Change-Id: Ife3548bda3042a7447b7c50f48f023a2bc0bc443
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362103
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43017}
2024-09-12 22:47:11 +00:00
henrika
dfd8f570cd Adds a WebRTC.DesktopCapture.Win.WgcDirtyRegionSupport UMA for diagnostic purposes.
Checks if the DirtyRegionMode property is present in GraphicsCaptureSession and logs a boolean histogram with the result.
Detecting support for this property means that the WGC API supports
dirty regions and it can be utilized to improve the capture
performance and the existing zero-herz support.

See also https://issues.chromium.org/issues/347991512 for more details
on how to detect support for dirty regions in WGC.

Bug: chromium:40259177
Change-Id: Ia316c4ece54bd93cfef1fa23c199675c64143f76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362240
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43015}
2024-09-12 19:41:44 +00:00
Jeremy Leconte
1bd331f102 Ensure <netinet/in.h> is included by using rtc_base/ip_address.h.
Change-Id: I1b48275ef458bcd579d027b879240c702975ab56
Bug: b/236227627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#43001}
2024-09-11 08:11:44 +00:00
Sergey Silkin
84273f56d9 Specify max number of consecutive drops using time units
AV1E_SET_MAX_CONSEC_FRAME_DROP_MS_CBR was added in https://aomedia-review.googlesource.com/c/aom/+/192402. It allows to configure max number of consecutive frame drops using time units. Use it instead of AV1E_SET_MAX_CONSEC_FRAME_DROP_CBR.

Bug: webrtc:351644568
Change-Id: I73265d5258d681926eb5b65e32c2a61b26c310ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360842
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42995}
2024-09-10 13:16:34 +00:00
Danil Chapovalov
0acbb7745f Pass Environment into RtcpSender
To remove usage of RtcpConfiguration fields that are passed through Environment

Bug: webrtc:362762208
Change-Id: I1a0f218efe6a893c31ef2272cf2379c66fb7b205
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361746
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42984}
2024-09-09 13:44:21 +00:00
Ilya Nikolaevskiy
363dc19f9d SimulcastToSvcConverter: Allow not setting scalability mode on frame
Bug: webrtc:347737882
Change-Id: I61e5a7a538bf43a9377fc9e3b8d399754232a2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362081
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42983}
2024-09-09 13:33:52 +00:00
Danil Chapovalov
02113a2169 Pass Environment into RtcpReceiver
to avoid relying on the global field trials.

Bug: webrtc:362762208
Change-Id: I94e96f0a3f16cfd64f7deb4deb4aaa924ac1bba8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361865
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42982}
2024-09-09 11:36:29 +00:00
Sergey Silkin
26146bbce0 Add support for screencast with temporal layering to SvcRateAllocator
SvcRateAllocator assumed no temporal layering for screencast content and allocated all bitrate to base temporal layer. Now it distributes bitrate to spatial and temporal layers (if configured) no matter of content type.

Bug: webrtc:351644568, b/364190191
Change-Id: I445f0157d2c14cad033648693dc0564ae97023e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362080
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42979}
2024-09-09 11:12:33 +00:00
Fanny Linderborg
6f64ae1ff5 Extract corruption detection message to its own target
Bug: webrtc:358039777
Change-Id: I6bc064aaba4c5b7f9b55215414e70e55eb0e0f64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361864
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42977}
2024-09-06 13:32:35 +00:00
Henrik Lundin
c9aaf11985 Remove use of AcmReceiver in ChannelReceive
ChannelReceive is now owning and interfacing with NetEq directly.
A new ResamplerHelper is added to acm_resampler.cc/.h, to do the
audio resampling that was previously done inside AcmReceiver.

AcmReceiver still remains, since it is used in other places for now.

Bug: webrtc:14867
Change-Id: If3eb6415e06b9b5e729d393713f3fccb31b0570f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361820
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42974}
2024-09-06 12:47:36 +00:00
Björn Terelius
c7da857813 Fix lint issues in pacing/
(Mostly include-what-you-use.)

Bug: webrtc:42226242
Change-Id: I3717cccb24ac4f0a5443995d1d355561a5a54c8d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361601
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42971}
2024-09-06 11:28:43 +00:00
Danil Chapovalov
e922cd1262 Use Environment instead of Clock in ModuleRtpRtcp and its RTP subcomponents
Bug: webrtc:362762208
Change-Id: I35af5cf3ed48e2c738c12df2ed9117a640ed0ff7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361720
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42966}
2024-09-05 15:58:43 +00:00
Ilya Nikolaevskiy
5ac7495701 Prepare to use SimulcastToSvcConverter in chromium
Allow moving the class, add required RTC_EXPORTs

Bug: webrtc:347737882
Change-Id: Iac14e6f62adfa13ff1e757918a2f92009f5be36f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361760
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42962}
2024-09-05 12:20:57 +00:00
Jakob Ivarsson
6255a7f3a0 Avoid negative timestamp in SourceTracker.
Bug: b/364184684
Change-Id: If03cd697fed05c24549b9ef80bbaf9f11b47d8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361640
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42959}
2024-09-05 10:43:37 +00:00
Jakob Ivarsson
010c189f76 Move thread handling from source tracker.
This makes it simpler to use in more contexts.

Bug: b/364184684
Change-Id: I1b08ebd24e51ba1b3f85261eed503a78cd006fd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361480
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42956}
2024-09-05 08:45:11 +00:00
Danil Chapovalov
af8f6264ca Use Environment instead of Clock in ModuleRtpRtcp2 and its RTP subcomponents
Bug: webrtc:362762208
Change-Id: Ie9bbb7f3b505acd8aab1b8552ba64e09a5a1bddf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361481
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42949}
2024-09-04 20:39:39 +00:00
Fanny Linderborg
dac0805955 Add FrameInstrumentationData to RTPVideoHeader and CodecSpecificInfo
Bug: webrtc:358039777
Change-Id: If2659240047e1935f7666266bff25ed86a6a234c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361420
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42940}
2024-09-04 07:21:02 +00:00
yazdan0a
55a59337c8 Minor format to extrapolate local time
- Removing unnecessary else {} blocks for better readability.
- Consistent naming of: timestamp_diff with explicit typecast.

BUG=None

Change-Id: I35161ffed245737c789336316f0cfb6821b12349
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361060
Reviewed-by: Florent Castelli <orphis@webrtc.org>
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@{#42939}
2024-09-04 07:19:15 +00:00
Björn Terelius
0c2cd6264f Fix lint issues in congestion_controller.
Bug: None
Change-Id: I178c26044d5d678259181a9290c621bf64d3d7a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361361
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42935}
2024-09-03 17:55:07 +00:00
Ilya Nikolaevskiy
e432503389 Rewrite simulcast config to equivalent SVC for vp9 simulcast
This allows to utilize libvpx optimizations considerably improving performance.
The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.

This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.

Bug: webrtc:347737882
Change-Id: I03bc27c920787a7305a9775e6341e26904592fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360280
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42931}
2024-09-03 12:10:33 +00:00
Danil Chapovalov
fb7c3065b2 Run include cleaner on subset of modules/rtp_rtcp
Bug: webrtc:362762208, webrtc:42226242
Change-Id: Iaa28c21346380c634ef983b02b370c1523e4ef36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361300
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42930}
2024-09-03 12:08:19 +00:00
Jan Grulich
3881cb65cf PipeWire camera: make member variable with the PipeWire status updated
We keep information about the PipeWire camera status as a member of the
PipeWire session, but it's never updated and remains in uninitialized
state. Make sure it gets updated once PipeWire is initialized or when it
fails. There is currently no use for this member variable, but there is
a plan to use it so I'm rather keeping it instead of removing it.

Bug: webrtc:42225999
Change-Id: If409761b148be8f0724fd9ab7a1ed4cf0e459503
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360922
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42926}
2024-09-03 10:44:32 +00:00
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Danil Chapovalov
164b3b3fce Introduce ModuleRtpRtcpImpl factory that accepts Environment
ModuleRtpRtcpImpl and ModuleRtpRtcpImpl2 share certain components, RtcpReceiver in particular.
To always have Environment in RtcpReceiver both legacy and new module need to propagate it.

No-Iwyu: suggests too many changes, better address them separately.
Bug: webrtc:362762208
Change-Id: I2c885f57e24f135229fb7cd9781126d663017b3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361142
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42908}
2024-09-02 10:24:12 +00:00
henrika
5a92ddb2b9 Updates review date in ADM g3doc.
Bug: b/362683981
Change-Id: I6e78c93abd0fe9be7f427120b2912a59e480dd75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42905}
2024-09-02 09:11:02 +00:00
Sergey Silkin
0b4b5b0ae8 Use AV1E_SET_AUTO_TILES
Delegate control over number of times to encoder using AV1E_SET_AUTO_TILES that was added in https://aomedia-review.googlesource.com/c/aom/+/191102.

Bug: webrtc:351644568
Change-Id: I87ed11734e907c7f6c6508ac7389c84ececf5b21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361140
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42903}
2024-09-02 08:54:07 +00:00
Danil Chapovalov
d385af56c3 Introduce ModuleRtpRtcpImpl2 constructor that accepts Environment
And checks similar fields in Configuration struct are not set.
Migrate rtp_rtcp to use new constructor.

Bug: webrtc:362762208
Change-Id: I2385439c169a7432d174c72ca57ecb0ca639d864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361100
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42896}
2024-08-30 12:05:27 +00:00
Sergey Silkin
058972f84e Make LAYER_DROP and max_consec_drop=2 to be default settings
Based on the results of the experiment (b/335129329).

Bug: webrtc:15827, b/320629637, b/335129329, chromium:329396373
Change-Id: I1599f4c1be79ee3385aac1ff345168982c8278f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42895}
2024-08-30 10:10:09 +00:00
Jakob Ivarsson
b5f4006589 Inject field trials in NetEqTest instead of setting global.
We can avoid using the global now that field trials from Environment are
used in NetEq. This allows running multiple instances in parallel with
different settings.

Bug: webrtc:42220378
Change-Id: Icff8539e3ae9b61c86bb393d9a313e786e032b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359720
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42894}
2024-08-30 09:11:50 +00:00
Emil Vardar
2c637aa16f Register filter loop parameters' start position in VP9 frame header.
This is needed in order to create corruptions (by altering the filter loop params) to test the corruption detection algorithm.

Bug: webrtc:358039777
Change-Id: Ib26e9c0187b79c13b9862898625742def4091b91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42890}
2024-08-30 07:17:11 +00:00
Fanny Linderborg
2f91bdceee Declare corruption detection URI in RtpExtension
R=sprang@webrtc.org

Bug: webrtc:358039777
Change-Id: I9c66794b8a622bef5505f3a4a7252a0e7a989813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42887}
2024-08-29 19:41:16 +00:00
Andrew Grieve
6ea1c96325 Fix license metadata for spl_sqrt_floor, portaudio, sigslot
Bug: b/361140175
Change-Id: I35e76039608fa5094c04ace5f3ad1dba868ccb85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360900
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Andrew Grieve <agrieve@google.com>
Cr-Commit-Position: refs/heads/main@{#42885}
2024-08-29 19:11:29 +00:00
Jakob Ivarsson
04cc4ce2f2 Deprecate NetEq::GetDecoderFormat and remove implementation.
Bug: None
Change-Id: I9c90b41ee528984d1a3cd1632565c6dc1598e4d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360920
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42881}
2024-08-29 10:47:29 +00:00
Jakob Ivarsson
44df591447 Use NetEq::GetCurrentDecoderFormat in AcmReceiver.
This replaces the payload type tracking in AcmReceiver with the one in
NetEq and should be a noop.

Bug: None
Change-Id: Iaf124b5e56a646f994b5c2af65d349ede550b7fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360840
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42875}
2024-08-28 17:33:36 +00:00
Jakob Ivarsson
32dd2ed745 Improve NetEq simulation frame size estimation.
This fixes an edge case where the frame size changes for a DTX packet.
We should avoid having the frame size larger than the timestamp gap.

Bug: None
Change-Id: I0a384cfb06f5aebc1654c1e3d127541fd24e05c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360722
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42873}
2024-08-28 13:09:23 +00:00
Jakob Ivarsson
b6046aece2 Add NetEq API to get info about the current decoder.
This is currenly tracked in both AcmReceiver and NetEq. Adding this API
enables us to have it in just one place.

Bug: None
Change-Id: Ia537f87f36b0aedf19c00a57bd6cec4425a49df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360743
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42872}
2024-08-28 12:50:50 +00:00
Erik Språng
c1a0d233d0 Update explainer text for corruption detection header extension.
Bug: webrtc:358039777
Change-Id: I6a1cffc2a5797d154bfecb50c60b4c05d4943426
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42862}
2024-08-27 08:34:31 +00:00
Fanny Linderborg
fd6f4b4e51 Add the corruption detection extension to RTPExtensionType
Bug: webrtc:358039777
Change-Id: Ib825593e5c37beb0cba3190c1d3bdcf1c9d957cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360144
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42861}
2024-08-27 08:27:20 +00:00
Christoffer Jansson
ad1775634f Re-enable ApiCallDurationTest
Bug: b/260832909
Change-Id: I2a3e7651fbeba7a33a52a7139496b631cd2a446a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285469
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42860}
2024-08-27 08:24:47 +00:00
Danil Chapovalov
c54c85fe8f Attach Mid/Rid RTP header extension to pure padding packets
same as they attached to other packets.
Otherwise there is risk that ssrc will be acked after few initial pure padding packets are sent, before remote endpoint seen any mid or rid attached.

Bug: b/361257385
Change-Id: I695b379221debe2518ad33d13d65620877f0b2a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42851}
2024-08-26 16:00:16 +00:00
Benjamin Williams
ab009c27b4 Refactor WebRTC self assignments in if clauses
This change refactors existing self-assignments within if clauses across
the WebRTC codebase.

*Why:*

- Bug Prevention: Assignments within conditionals are frequently
  unintended errors, often mistaken for equality checks.

- Clearer Code: Separating assignments from conditionals improves code
  readability and reduces the risk of misinterpretation.

Change-Id: I199dc26a35ceca109a2ac569b446811314dfdf0b
Bug: chromium:361594695
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360460
Reviewed-by: Chuck Hays <haysc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42850}
2024-08-26 15:56:43 +00:00
Fan Zhou
8771cf470d Allow gap on packet buffer fix with GFD
The current solution does not work for GFD since GFD is only parsed from the first packet of the frame. As a result, to access the generic information, we have to check every packet when traversing the packet buffer to find the first packet of frame. This fix is necessary to ensure temporal scaling works correctly with GFD.

Bug: webrtc:42225186
Change-Id: Iadda4ec690deab62c32eb6101583e6a6d75cfeaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344840
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42836}
2024-08-23 09:41:14 +00:00