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/*
* Copyright (c) 2018 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_STREAM_ENCODER_INTERFACE_H_
#define API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_
#include <vector>
#include "api/adaptation/resource.h"
#include "api/fec_controller_override.h"
#include "api/rtp_parameters.h" // For DegradationPreference.
#include "api/scoped_refptr.h"
#include "api/units/data_rate.h"
#include "api/video/video_bitrate_allocator.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_config.h"
namespace webrtc {
// This interface represents a class responsible for creating and driving the
// encoder(s) for a single video stream. It is also responsible for adaptation
// decisions related to video quality, requesting reduced frame rate or
// resolution from the VideoSource when needed.
// TODO(bugs.webrtc.org/8830): This interface is under development. Changes
// under consideration include:
//
// 1. Taking out responsibility for adaptation decisions, instead only reporting
// per-frame measurements to the decision maker.
//
// 2. Moving responsibility for simulcast and for software fallback into this
// class.
class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
public:
// Interface for receiving encoded video frames and notifications about
// configuration changes.
class EncoderSink : public EncodedImageCallback {
public:
virtual void OnEncoderConfigurationChanged(
std::vector<VideoStream> streams,
bool is_svc,
VideoEncoderConfig::ContentType content_type,
int min_transmit_bitrate_bps) = 0;
};
// If the resource is overusing, the VideoStreamEncoder will try to reduce
// resolution or frame rate until no resource is overusing.
// TODO(https://crbug.com/webrtc/11565): When the ResourceAdaptationProcessor
// is moved to Call this method could be deleted altogether in favor of
// Call-level APIs only.
virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
virtual std::vector<rtc::scoped_refptr<Resource>>
GetAdaptationResources() = 0;
// Sets the source that will provide video frames to the VideoStreamEncoder's
// OnFrame method. |degradation_preference| control whether or not resolution
// or frame rate may be reduced. The VideoStreamEncoder registers itself with
// |source|, and signals adaptation decisions to the source in the form of
// VideoSinkWants.
// TODO(nisse): When adaptation logic is extracted from this class,
// it no longer needs to know the source.
virtual void SetSource(
rtc::VideoSourceInterface<VideoFrame>* source,
const DegradationPreference& degradation_preference) = 0;
// Sets the |sink| that gets the encoded frames. |rotation_applied| means
// that the source must support rotation. Only set |rotation_applied| if the
// remote side does not support the rotation extension.
virtual void SetSink(EncoderSink* sink, bool rotation_applied) = 0;
// Sets an initial bitrate, later overriden by OnBitrateUpdated. Mainly
// affects the resolution of the initial key frame: If incoming frames are
// larger than reasonable for the start bitrate, and scaling is enabled,
// VideoStreamEncoder asks the source to scale down and drops a few initial
// frames.
// TODO(nisse): This is a poor interface, and mixes bandwidth estimation and
// codec configuration in an undesired way. For the actual send bandwidth, we
// should always be somewhat conservative, but we may nevertheless want to let
// the application configure a more optimistic quality for the initial
// resolution. Should be replaced by a construction time setting.
virtual void SetStartBitrate(int start_bitrate_bps) = 0;
// Request a key frame. Used for signalling from the remote receiver.
virtual void SendKeyFrame() = 0;
// Inform the encoder that a loss has occurred.
virtual void OnLossNotification(
const VideoEncoder::LossNotification& loss_notification) = 0;
// Set the currently estimated network properties. A |target_bitrate|
// of zero pauses the encoder.
// |stable_target_bitrate| is a filtered version of |target_bitrate|. It is
// always less or equal to it. It can be used to avoid rapid changes of
// expensive encoding settings, such as resolution.
// |link_allocation| is the bandwidth available for this video stream on the
// network link. It is always at least |target_bitrate| but may be higher
// if we are not network constrained.
virtual void OnBitrateUpdated(DataRate target_bitrate,
DataRate stable_target_bitrate,
DataRate link_allocation,
uint8_t fraction_lost,
int64_t round_trip_time_ms,
double cwnd_reduce_ratio) = 0;
// Register observer for the bitrate allocation between the temporal
// and spatial layers.
virtual void SetBitrateAllocationObserver(
VideoBitrateAllocationObserver* bitrate_observer) = 0;
// Set a FecControllerOverride, through which the encoder may override
// decisions made by FecController.
virtual void SetFecControllerOverride(
FecControllerOverride* fec_controller_override) = 0;
// Creates and configures an encoder with the given |config|. The
// |max_data_payload_length| is used to support single NAL unit
// packetization for H.264.
virtual void ConfigureEncoder(VideoEncoderConfig config,
size_t max_data_payload_length) = 0;
// Permanently stop encoding. After this method has returned, it is
// guaranteed that no encoded frames will be delivered to the sink.
virtual void Stop() = 0;
};
} // namespace webrtc
#endif // API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_