webrtc/api/video/video_source_interface.h
Jonas Oreland 0deda15c96 Reland "RtpEncodingParameters::request_resolution patch 1"
This reverts commit b625101da8.

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 ef7359e679.
>
> 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}
2022-09-23 11:48:19 +00:00

130 lines
5.3 KiB
C++

/*
* Copyright (c) 2016 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_VIDEO_SOURCE_INTERFACE_H_
#define API_VIDEO_VIDEO_SOURCE_INTERFACE_H_
#include <limits>
#include <vector>
#include "absl/types/optional.h"
#include "api/video/video_sink_interface.h"
#include "rtc_base/system/rtc_export.h"
namespace rtc {
// VideoSinkWants is used for notifying the source of properties a video frame
// should have when it is delivered to a certain sink.
struct RTC_EXPORT VideoSinkWants {
struct FrameSize {
FrameSize(int width, int height) : width(width), height(height) {}
FrameSize(const FrameSize&) = default;
~FrameSize() = default;
int width;
int height;
};
VideoSinkWants();
VideoSinkWants(const VideoSinkWants&);
~VideoSinkWants();
// Tells the source whether the sink wants frames with rotation applied.
// By default, any rotation must be applied by the sink.
bool rotation_applied = false;
// Tells the source that the sink only wants black frames.
bool black_frames = false;
// Tells the source the maximum number of pixels the sink wants.
int max_pixel_count = std::numeric_limits<int>::max();
// Tells the source the desired number of pixels the sinks wants. This will
// typically be used when stepping the resolution up again when conditions
// have improved after an earlier downgrade. The source should select the
// closest resolution to this pixel count, but if max_pixel_count is set, it
// still sets the absolute upper bound.
absl::optional<int> target_pixel_count;
// Tells the source the maximum framerate the sink wants.
int max_framerate_fps = std::numeric_limits<int>::max();
// Tells the source that the sink wants width and height of the video frames
// to be divisible by `resolution_alignment`.
// For example: With I420, this value would be a multiple of 2.
// Note that this field is unrelated to any horizontal or vertical stride
// requirements the encoder has on the incoming video frame buffers.
int resolution_alignment = 1;
// The resolutions that sink is configured to consume. If the sink is an
// encoder this is what the encoder is configured to encode. In singlecast we
// only encode one resolution, but in simulcast and SVC this can mean multiple
// resolutions per frame.
//
// The sink is always configured to consume a subset of the
// webrtc::VideoFrame's resolution. In the case of encoding, we usually encode
// at webrtc::VideoFrame's resolution but this may not always be the case due
// to scaleResolutionDownBy or turning off simulcast or SVC layers.
//
// For example, we may capture at 720p and due to adaptation (e.g. applying
// `max_pixel_count` constraints) create webrtc::VideoFrames of size 480p, but
// if we do scaleResolutionDownBy:2 then the only resolution we end up
// encoding is 240p. In this case we still need to provide webrtc::VideoFrames
// of size 480p but we can optimize internal buffers for 240p, avoiding
// downsampling to 480p if possible.
//
// 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,
const VideoSinkWants::FrameSize& b) {
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:
virtual ~VideoSourceInterface() = default;
virtual void AddOrUpdateSink(VideoSinkInterface<VideoFrameT>* sink,
const VideoSinkWants& wants) = 0;
// RemoveSink must guarantee that at the time the method returns,
// there is no current and no future calls to VideoSinkInterface::OnFrame.
virtual void RemoveSink(VideoSinkInterface<VideoFrameT>* sink) = 0;
// Request underlying source to capture a new frame.
// TODO(crbug/1255737): make pure virtual once downstream projects adapt.
virtual void RequestRefreshFrame() {}
};
} // namespace rtc
#endif // API_VIDEO_VIDEO_SOURCE_INTERFACE_H_