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}

api/BUILD.gn

@@ -454,6 +454,7 @@ rtc_library("rtp_parameters") {
     "../rtc_base:checks",
     "../rtc_base:stringutils",
     "../rtc_base/system:rtc_export",
+    "video:resolution",
     "video_codecs:scalability_mode",
   ]
   absl_deps = [

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 @@ struct RTC_EXPORT RtpEncodingParameters {
   // 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 @@ struct RTC_EXPORT RtpEncodingParameters {
            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);

api/video/BUILD.gn

@@ -131,6 +131,11 @@ rtc_source_set("render_resolution") {
   public = [ "render_resolution.h" ]
 }
 
+rtc_source_set("resolution") {
+  visibility = [ "*" ]
+  public = [ "resolution.h" ]
+}
+
 rtc_library("encoded_image") {
   visibility = [ "*" ]
   sources = [

api/video/recordable_encoded_frame.h

@@ -24,6 +24,7 @@ namespace webrtc {
 class RecordableEncodedFrame {
  public:
   // Encoded resolution in pixels
+  // TODO(bugs.webrtc.org/12114) : remove in favor of Resolution.
   struct EncodedResolution {
     bool empty() const { return width == 0 && height == 0; }
 

api/video/render_resolution.h

@@ -13,6 +13,7 @@
 
 namespace webrtc {
 
+// TODO(bugs.webrtc.org/12114) : remove in favor of Resolution.
 class RenderResolution {
  public:
   constexpr RenderResolution() = default;

api/video/resolution.h

@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_RESOLUTION_H_
+#define API_VIDEO_RESOLUTION_H_
+
+namespace webrtc {
+
+// A struct representing a video resolution in pixels.
+struct Resolution {
+  int width = 0;
+  int height = 0;
+};
+
+inline bool operator==(const Resolution& lhs, const Resolution& rhs) {
+  return lhs.width == rhs.width && lhs.height == rhs.height;
+}
+
+inline bool operator!=(const Resolution& lhs, const Resolution& rhs) {
+  return !(lhs == rhs);
+}
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_RESOLUTION_H_

api/video/video_source_interface.h

@@ -80,6 +80,24 @@ struct RTC_EXPORT VideoSinkWants {
   // Note that the `resolutions` can change while frames are in flight and
   // should only be used as a hint when constructing the webrtc::VideoFrame.
   std::vector<FrameSize> resolutions;
+
+  // This is the resolution requested by the user using RtpEncodingParameters.
+  absl::optional<FrameSize> requested_resolution;
+
+  // `active` : is (any) of the layers/sink(s) active.
+  bool is_active = false;
+
+  // This sub-struct contains information computed by VideoBroadcaster
+  // that aggregates several VideoSinkWants (and sends them to
+  // AdaptedVideoTrackSource).
+  struct Aggregates {
+    // `active_without_requested_resolution` is set by VideoBroadcaster
+    // when aggregating sink wants if there exists any sink (encoder) that is
+    // active but has not set the `requested_resolution`, i.e is relying on
+    // OnOutputFormatRequest to handle encode resolution.
+    bool any_active_without_requested_resolution = false;
+  };
+  absl::optional<Aggregates> aggregates;
 };
 
 inline bool operator==(const VideoSinkWants::FrameSize& a,
@@ -87,6 +105,11 @@ inline bool operator==(const VideoSinkWants::FrameSize& a,
   return a.width == b.width && a.height == b.height;
 }
 
+inline bool operator!=(const VideoSinkWants::FrameSize& a,
+                       const VideoSinkWants::FrameSize& b) {
+  return !(a == b);
+}
+
 template <typename VideoFrameT>
 class VideoSourceInterface {
  public:

api/video_codecs/BUILD.gn

@@ -84,6 +84,7 @@ rtc_library("video_codecs_api") {
     "../units:data_rate",
     "../video:encoded_image",
     "../video:render_resolution",
+    "../video:resolution",
     "../video:video_bitrate_allocation",
     "../video:video_codec_constants",
     "../video:video_frame",

api/video_codecs/video_encoder_config.h

@@ -18,6 +18,7 @@
 
 #include "absl/types/optional.h"
 #include "api/scoped_refptr.h"
+#include "api/video/resolution.h"
 #include "api/video_codecs/scalability_mode.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_codec.h"
@@ -32,10 +33,11 @@ struct VideoStream {
   VideoStream(const VideoStream& other);
   std::string ToString() const;
 
-  // Width in pixels.
+  // Width/Height in pixels.
+  // This is the actual width and height used to configure encoder,
+  // which might be less than `requested_resolution` due to adaptation
+  // or due to the source providing smaller frames than requested.
   size_t width;
-
-  // Height in pixels.
   size_t height;
 
   // Frame rate in fps.
@@ -69,6 +71,17 @@ struct VideoStream {
 
   // If this stream is enabled by the user, or not.
   bool active;
+
+  // An optional user supplied max_frame_resolution
+  // than can be set independently of (adapted) VideoSource.
+  // This value is set from RtpEncodingParameters::requested_resolution
+  // (i.e. used for signaling app-level settings).
+  //
+  // The actual encode resolution is in `width` and `height`,
+  // which can be lower than requested_resolution,
+  // e.g. if source only provides lower resolution or
+  // if resource adaptation is active.
+  absl::optional<Resolution> requested_resolution;
 };
 
 class VideoEncoderConfig {