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}
The feauture was added in https://webrtc-review.googlesource.com/c/src/+/291119 in order to ensure RTX packet is part of BWE even before the RTP stream is known.
However, it cause an issue if media is signaled with an SSRC that has this RTX SSRC.
Since BWE is now notified about received packets before demuxing to the correct receive stream, it is not necessary to demux RTX packets before the media SSRC is known.
Note that WebRTC require at least one negotiated SSRC/MID before RTCP feedback can be sent. Ie, for BWE to work, at least one media SSRC must be known after this cl. It can either be unsignaled or signaled.
BWE tested with BweRampupWithInitialProbeTest.
Bug: webrtc:14795, webrtc:14817, b/320258158
Change-Id: Icf2c67bedc352720bf846b9ee38d509346af36f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41785}
Opening the visibility of a few methods from this utils class to allow
it to be used by other implementations of `AudioDeviceModule`.
Also updating a few methods, like adding new audio device types from
recent Android SDKs, and updating the definition of an emulator.
Bug: b/287409066
Change-Id: I1473fa0342252347ce92ee2319380ebb14e9885b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339905
Commit-Queue: Zoé Lepaul <xalep@webrtc.org>
Reviewed-by: Ranveer Aggarwal <ranvr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41778}
To ensure CreateVideoDecoder is only used as a fallback when Create is not implemented,
and thus make it safer to migrate VideoDecoderFactory implementations to Create.
Bug: webrtc:15791
Change-Id: Ifb15cf1d303348949ba51a3bb4c91b855a06627f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339841
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41777}
Move code from P2PTransportChannel to Candidate, where we set the
foundation value for remote prflx candidates.
Bug: none
Change-Id: I7dbcb85bca35dca7297136b0706092dd8d2b153c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339902
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41774}
and when the different codec comparison methods are applied.
No functional changes.
BUG=webrtc:15847
Change-Id: I583c6a42869a80d3a920b9caf18e2a18431c5b94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339700
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@{#41772}
until the code duplication can be removed which requires breaking
up the circular dependency.
BUG=webrtc:15847
Change-Id: Icc5f27dfcda26b1fcf16b19f79005d8b52fb6af3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339903
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@{#41771}
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}
This should be a no-op since RTX is only supported for video which
has one channel and uses a clockrate of 90000.
Parameters are not compared for RTX since the RTX capabilities do not
include the associated payload type (apt).
BUG=webrtc:15847
Change-Id: Ibe6677135ecc56cdc5f3d3ccdc2e680dd449f66f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41769}
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}
This way VP8Decoder and DecoderFallback would use propagated instead of global field trials.
Bug: webrtc:15791, webrtc:10335
Change-Id: I5ad5fae38f5b9379bc6376334562c154fbc56e39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340040
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41763}
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-sctp-zero-checksum/06/
The previous implementation was for version 00, and since then changes
have been made. The chunk that is used to negotiate this capability has
now grown to include an additional property - the sender's alternate
error detection method.
Bug: webrtc:14997
Change-Id: I78043d187b79f40bbadbcba02eae6eedf54f30f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336380
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41759}
Switch to type_name() for things like logging since `type()` will
change to returning an enumeration value.
The functional change that this has is that log statements and
Connection::ToString() (used for logging) will contain "host"
instead of "local" and "srflx" instead of "stun".
Bug: webrtc:15846
Change-Id: I35c50d026e4578a25d51765d59c6f2e01b850c94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41757}
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}
When running it, even without changes at HEAD I got:
```
KeyError: 'ios_runtime_cache_17_0'
```
Bug: b/325441006
Change-Id: I7ea236ccc1f7439d7750208260b01d7636db4ae5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339842
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41751}
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 a2655449ee310704ee2053fd6d43a5ab7002b755
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}
Force and thus guarantee VideoDecoder created through java wrappers get access to the webrtc::Environment
Bug: webrtc:15791
Change-Id: I3f145937c0b914c8e34b24e1ecc55da756551069
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338441
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41745}
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}
This is an obscure error that was found by a fuzzer.
Bug: webrtc:15845
Change-Id: I3509fa264a3af6f0f5e8e6b75a8b7dcd8fb0da1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339681
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41743}
