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}
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}
This is a reland of commit 050ffefd854f8a57071992238723259e9ae0d85a
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
NOTRY=true
Bug: b/322132132
Change-Id: I25dd34b9ba59ea8502e47b4c89cd111430636e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41736}
This reverts commit 050ffefd854f8a57071992238723259e9ae0d85a.
Reason for revert: Breaks downstream project.
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
Bug: b/322132132
Change-Id: I24d0a4e71a43ac192485f1af208563a51d919865
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41735}
This CL extends logging related to HW->SW fallbacks on the encoder
side in WebRTC. The goal is to make it easier to track down the
different steps taken when setting up the video encoder and why/when
HW encoding fails.
Current logs are added on several lines which makes regexp searching
difficult. This CL adds all related information on one line instead.
Three new search tags are also added VSE (VideoStreamEncoder), VESFW
(VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
It has been verified that these added logs also show up in WebRTC
logs in Meet.
Logs from the GPU process are not included due to the sandboxed
nature which makes it much more complex to add to the native
WebRTC log. I think that these simple logs will provide value as is.
Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
Bug: b/322132132
Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41733}
This reverts commit 1cce1d7ddcbde3a3648007b5a131bd0c2638724b.
Reason for revert: Breaks WPTs
Original change's description:
> Make setCodecPreferences only look at receive codecs
>
> which is what is noted in JSEP:
> https://www.rfc-editor.org/rfc/rfc8829.html#name-setcodecpreferences
>
> Some W3C spec modifications are required since the W3C specification
> currently takes into account send codecs as well.
>
> Spec issue:
> https://github.com/w3c/webrtc-pc/issues/2888
> Spec PR:
> https://github.com/w3c/webrtc-pc/pull/2926
>
> setCodecPreferences continues to modify the codecs in an offer.
>
> Also rename RtpSender::SetCodecPreferences to RtpSender::SetSendCodecs for consistent semantics.
>
> BUG=webrtc:15396
>
> Change-Id: I1e8fbe77cb2670575578a777ed1336567a1e4031
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41719}
Bug: webrtc:15396
Change-Id: I7b545e91f820c3affc39841c6e93939eac75c363
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41725}
by creating an id collision and letting UsedIds resolve it.
Also avoid id=15 which is forbidden by
https://www.rfc-editor.org/rfc/rfc8285#section-4.2
so might cause interop issues in offers to implementations
not supporting two-byte extensions.
BUG=webrtc:15378
Change-Id: I27926f065f8e396257294da7acf2be9802169805
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41696}
https://webrtc-review.googlesource.com/c/src/+/324282 moved default QP to media/base/media_constants.h. Dependent projects have been switched to the new constant.
Bug: webrtc:14852
Change-Id: Ic547a6b08490151d45543b68d4ed4b9da3a1629f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324820
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41671}
Instead, always use VideoSendStream::Start.
VideoSendStream::StartPerRtpStream was used for controlling if
individual rtp stream for a RtpEncodingParameter should be able to send RTP packets. It was not used for controlling the actual encoder layers.
With this change RtpEncodingParameter.active still controls actual encoder layers but it does not control if RTP packets can be sent or not.
The cleanup is done to simplify code and in the future allow sending
probe packet on a RtpTransceiver that allows sending, regardless of the
RtpEncodingParameter.active flag.
Bug: webrtc:14928
Change-Id: I896c055ed4de76db58d76f452147c29783f77ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335042
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@{#41619}
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.
One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.
Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
Add ulpfec and flexfec to list of resiliency mechanisms taken
into account and in general exclude Comfort Noise (CN) from media
codecs.
Also introduce RtpCodecCapability::IsMediaCodec & ::IsResiliencyCodec
behaving like the MediaCodec methods.
BUG=webrtc:15396
Change-Id: I79041898928190bfdd33a06d8f6975d7556c46b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330424
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41485}
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}
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}
instead of throwing an error when trying to pick a send codec.
BUG=webrtc:15145,webrtc:4957
Change-Id: I056b145c093348576e1aeaf5def50d5414f2de70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330122
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41360}
This adds neccessary checks for SDP negotiation with HEVC.
Test: Manually apply the CL on Chromium and enable HEVC HW encoder,
and add HEVC profiles in rtc video decoder/encoder factory, H265 is
negotiated in SDP with correct FMTP lines added.
Bug: webrtc:13485
Change-Id: I5557b20b646cc96c5acb578521204fe10df0dcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330202
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#41357}
These functions had dummy implementations, but were not virtual.
The need for those functions seems to be lost in time.
Bug: None
Change-Id: I66dcac4a92f9993d82031f943f2f9ae767156b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330422
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41336}
This reverts commit 1ae700a9233ed647e1b4080c0fcb48f61a0cca0a.
Reason for revert: Potential root cause of crbug.com/1504351
Original change's description:
> Make Codec::Matches also consider packetization
>
> If it's not considered it can lead to payload IDs erroneously being
> reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
>
> Bug: webrtc:15473
> Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41153}
Bug: webrtc:15473 chromium:1504351
Change-Id: I87fb671d76c3b17beb65124603cc040bb9bf4fa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329201
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41285}
This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation.
This poison extends and thus replaces default task queue poison.
Bug: webrtc:15656
Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41247}
If it's not considered it can lead to payload IDs erroneously being
reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
Bug: webrtc:15473
Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41153}
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
instead of requiring to pass in call_factory and media_engine
webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies
Bug: webrtc:15574
Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41049}
after dependencies adopted the RtpMediaContentDescription which
this is currently aliased to.
Also move definition of AudioCodecs and VideoCodecs to the place
where codecs are defined.
BUG=webrtc:15214
Change-Id: I9b0456e1c69c8b23e0cc7665a59baae268872d9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325021
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41020}
kDefaultQpMax=56 was defined in multiple places. Move it to media_constants and split it into two: VPx/AV1 and H26x values. H26x value is set to 51 which is the max bitstream QP value for H264/5.
This CL is expected to be a no-op because:
1. VideoCodec::qpMax value has not changed for VP8/9 and AV1.
2. VideoCodec::qpMax is currently not used by OpenH264 wrapper (wiring it up is out-of-scope of this CL).
3. Previous default qpMax=56 exceeded the max value for H26x (=51). External HW H26x encoders likely clamped it and used 51.
Bug: webrtc:14852
Change-Id: I1d795e695dac5c78e86ed829b24281e61066f668
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324282
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40997}
In some very high-bandwidth application there have been observations of
packet loss in the socket implementation (not on the network itself) due
to large bursts of packets arriving. Allocating too big buffers can of
course lead to issue as well, so this flag is intended to find a good
tradeoff.
Bug: webrtc:15585
Change-Id: I63eccb1a9f34d852d80c286fc27bffd17818f0ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324021
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40963}
Because of our asymmetrical codec situation, it's possible to have
send only codecs that we cannot negotiate even with ourselves.
This means that we should not have a DCHECK, but just a plain error.
Bug: webrtc:15064
Change-Id: I0c170e5c7f356197bcb04bcecb8259c344423ccb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323183
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40939}
This makes it consistent with how things are done in webrtc_video_engine.cc
This will improve the JS code by not having to initialize an audio
track every time frames need to be sent over, especially from another
peer connection in case of encoded transforms.
Bug: chromium:1477192
Change-Id: I3f938ad812ff377599a3799d4c2d2cd85149189e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322702
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#40917}
Convert most field trials used in PCLF tests.
Change-Id: I26c0c4b1164bb0870aae1a488942cde888cb459d
Bug: webrtc:10335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322703
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40909}
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp
Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
