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}
diff --git a/api/rtp_parameters.h b/api/rtp_parameters.h
index e311577..0d3c9df 100644
--- a/api/rtp_parameters.h
+++ b/api/rtp_parameters.h
@@ -23,6 +23,7 @@
#include "api/media_types.h"
#include "api/priority.h"
#include "api/rtp_transceiver_direction.h"
+#include "api/video/resolution.h"
#include "api/video_codecs/scalability_mode.h"
#include "rtc_base/system/rtc_export.h"
@@ -502,6 +503,24 @@
// https://w3c.github.io/webrtc-svc/#rtcrtpencodingparameters
absl::optional<std::string> scalability_mode;
+ // Requested encode resolution.
+ //
+ // This field provides an alternative to `scale_resolution_down_by`
+ // that is not dependent on the video source.
+ //
+ // When setting requested_resolution it is not necessary to adapt the
+ // video source using OnOutputFormatRequest, since the VideoStreamEncoder
+ // will apply downscaling if necessary. requested_resolution will also be
+ // propagated to the video source, this allows downscaling earlier in the
+ // pipeline which can be beneficial if the source is consumed by multiple
+ // encoders, but is not strictly necessary.
+ //
+ // The `requested_resolution` is subject to resource adaptation.
+ //
+ // It is an error to set both `requested_resolution` and
+ // `scale_resolution_down_by`.
+ absl::optional<Resolution> requested_resolution;
+
// For an RtpSender, set to true to cause this encoding to be encoded and
// sent, and false for it not to be encoded and sent. This allows control
// across multiple encodings of a sender for turning simulcast layers on and
@@ -527,7 +546,8 @@
num_temporal_layers == o.num_temporal_layers &&
scale_resolution_down_by == o.scale_resolution_down_by &&
active == o.active && rid == o.rid &&
- adaptive_ptime == o.adaptive_ptime;
+ adaptive_ptime == o.adaptive_ptime &&
+ requested_resolution == o.requested_resolution;
}
bool operator!=(const RtpEncodingParameters& o) const {
return !(*this == o);