Reland "RtpEncodingParameters::request_resolution patch 1"
This reverts commit b625101da8d798c936cfd695505a5514644158b0. Reason for revert: Found problem that was specific how configuration is handled for VP9. A 1-line change in webrtc_video_engine.cc line 3715. Thanks Rasmus and great that this was tested! Original change's description: > Revert "RtpEncodingParameters::request_resolution patch 1" > > This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2. > > Reason for revert: Breaks downstream test > > Original change's description: > > RtpEncodingParameters::request_resolution patch 1 > > > > This patch adds RtpEncodingParameters::request_resolution > > with documentation and plumming. No behaviour is changed yet. > > > > Bug: webrtc:14451 > > Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262 > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Commit-Queue: Jonas Oreland <jonaso@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38172} > > Bug: webrtc:14451 > Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541 > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org> > Owners-Override: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38176} Bug: webrtc:14451 Change-Id: Ica9b74180bce22d09bf289126bb5ac137bf9eb70 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276543 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38178}
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WebRTC LUCI CQ
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@ -23,6 +23,7 @@
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#include "api/media_types.h"
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#include "api/priority.h"
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#include "api/rtp_transceiver_direction.h"
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#include "api/video/resolution.h"
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#include "api/video_codecs/scalability_mode.h"
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#include "rtc_base/system/rtc_export.h"
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@ -502,6 +503,24 @@ struct RTC_EXPORT RtpEncodingParameters {
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// https://w3c.github.io/webrtc-svc/#rtcrtpencodingparameters
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absl::optional<std::string> scalability_mode;
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// Requested encode resolution.
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//
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// This field provides an alternative to `scale_resolution_down_by`
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// that is not dependent on the video source.
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//
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// When setting requested_resolution it is not necessary to adapt the
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// video source using OnOutputFormatRequest, since the VideoStreamEncoder
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// will apply downscaling if necessary. requested_resolution will also be
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// propagated to the video source, this allows downscaling earlier in the
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// pipeline which can be beneficial if the source is consumed by multiple
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// encoders, but is not strictly necessary.
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//
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// The `requested_resolution` is subject to resource adaptation.
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//
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// It is an error to set both `requested_resolution` and
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// `scale_resolution_down_by`.
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absl::optional<Resolution> requested_resolution;
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// For an RtpSender, set to true to cause this encoding to be encoded and
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// sent, and false for it not to be encoded and sent. This allows control
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// across multiple encodings of a sender for turning simulcast layers on and
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@ -527,7 +546,8 @@ struct RTC_EXPORT RtpEncodingParameters {
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num_temporal_layers == o.num_temporal_layers &&
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scale_resolution_down_by == o.scale_resolution_down_by &&
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active == o.active && rid == o.rid &&
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adaptive_ptime == o.adaptive_ptime;
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adaptive_ptime == o.adaptive_ptime &&
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requested_resolution == o.requested_resolution;
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}
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bool operator!=(const RtpEncodingParameters& o) const {
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return !(*this == o);
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