api/video_codecs/video_encoder.h - src - Git at Google

 /*
  *  Copyright (c) 2014 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_CODECS_VIDEO_ENCODER_H_
 #define API_VIDEO_CODECS_VIDEO_ENCODER_H_

 #include <limits>
 #include <memory>
 #include <string>
 #include <vector>

 #include "absl/container/inlined_vector.h"
 #include "absl/types/optional.h"
 #include "api/fec_controller_override.h"
 #include "api/units/data_rate.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_codec_constants.h"
 #include "api/video/video_frame.h"
 #include "api/video_codecs/video_codec.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/system/rtc_export.h"

 namespace webrtc {

 class RTPFragmentationHeader;
 // TODO(pbos): Expose these through a public (root) header or change these APIs.
 struct CodecSpecificInfo;

 constexpr int kDefaultMinPixelsPerFrame = 320 * 180;

 class EncodedImageCallback {
  public:
   virtual ~EncodedImageCallback() {}

   struct Result {
     enum Error {
       OK,

       // Failed to send the packet.
       ERROR_SEND_FAILED,
     };

     explicit Result(Error error) : error(error) {}
     Result(Error error, uint32_t frame_id) : error(error), frame_id(frame_id) {}

     Error error;

     // Frame ID assigned to the frame. The frame ID should be the same as the ID
     // seen by the receiver for this frame. RTP timestamp of the frame is used
     // as frame ID when RTP is used to send video. Must be used only when
     // error=OK.
     uint32_t frame_id = 0;

     // Tells the encoder that the next frame is should be dropped.
     bool drop_next_frame = false;
   };

   // Used to signal the encoder about reason a frame is dropped.
   // kDroppedByMediaOptimizations - dropped by MediaOptimizations (for rate
   // limiting purposes).
   // kDroppedByEncoder - dropped by encoder's internal rate limiter.
   enum class DropReason : uint8_t {
     kDroppedByMediaOptimizations,
     kDroppedByEncoder
   };

   // Callback function which is called when an image has been encoded.
   virtual Result OnEncodedImage(
       const EncodedImage& encoded_image,
       const CodecSpecificInfo* codec_specific_info,
       const RTPFragmentationHeader* fragmentation) = 0;

   virtual void OnDroppedFrame(DropReason reason) {}
 };

 class RTC_EXPORT VideoEncoder {
  public:
   struct QpThresholds {
     QpThresholds(int l, int h) : low(l), high(h) {}
     QpThresholds() : low(-1), high(-1) {}
     int low;
     int high;
   };

   // Quality scaling is enabled if thresholds are provided.
   struct RTC_EXPORT ScalingSettings {
    private:
     // Private magic type for kOff, implicitly convertible to
     // ScalingSettings.
     struct KOff {};

    public:
     // TODO(nisse): Would be nicer if kOff were a constant ScalingSettings
     // rather than a magic value. However, absl::optional is not trivially copy
     // constructible, and hence a constant ScalingSettings needs a static
     // initializer, which is strongly discouraged in Chrome. We can hopefully
     // fix this when we switch to absl::optional or std::optional.
     static constexpr KOff kOff = {};

     ScalingSettings(int low, int high);
     ScalingSettings(int low, int high, int min_pixels);
     ScalingSettings(const ScalingSettings&);
     ScalingSettings(KOff);  // NOLINT(runtime/explicit)
     ~ScalingSettings();

     absl::optional<QpThresholds> thresholds;

     // We will never ask for a resolution lower than this.
     // TODO(kthelgason): Lower this limit when better testing
     // on MediaCodec and fallback implementations are in place.
     // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7206
     int min_pixels_per_frame = kDefaultMinPixelsPerFrame;

    private:
     // Private constructor; to get an object without thresholds, use
     // the magic constant ScalingSettings::kOff.
     ScalingSettings();
   };

   // Bitrate limits for resolution.
   struct ResolutionBitrateLimits {
     ResolutionBitrateLimits(int frame_size_pixels,
                             int min_start_bitrate_bps,
                             int min_bitrate_bps,
                             int max_bitrate_bps)
         : frame_size_pixels(frame_size_pixels),
           min_start_bitrate_bps(min_start_bitrate_bps),
           min_bitrate_bps(min_bitrate_bps),
           max_bitrate_bps(max_bitrate_bps) {}
     // Size of video frame, in pixels, the bitrate thresholds are intended for.
     int frame_size_pixels = 0;
     // Recommended minimum bitrate to start encoding.
     int min_start_bitrate_bps = 0;
     // Recommended minimum bitrate.
     int min_bitrate_bps = 0;
     // Recommended maximum bitrate.
     int max_bitrate_bps = 0;

     bool operator==(const ResolutionBitrateLimits& rhs) const;
     bool operator!=(const ResolutionBitrateLimits& rhs) const {
       return !(*this == rhs);
     }
   };

   // Struct containing metadata about the encoder implementing this interface.
   struct RTC_EXPORT EncoderInfo {
     static constexpr uint8_t kMaxFramerateFraction =
         std::numeric_limits<uint8_t>::max();

     EncoderInfo();
     EncoderInfo(const EncoderInfo&);

     ~EncoderInfo();

     std::string ToString() const;
     bool operator==(const EncoderInfo& rhs) const;
     bool operator!=(const EncoderInfo& rhs) const { return !(*this == rhs); }

     // Any encoder implementation wishing to use the WebRTC provided
     // quality scaler must populate this field.
     ScalingSettings scaling_settings;

     // The width and height of the incoming video frames should be divisible
     // by |requested_resolution_alignment|. If they are not, the encoder may
     // drop the incoming frame.
     // 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 requested_resolution_alignment;

     // If true, encoder supports working with a native handle (e.g. texture
     // handle for hw codecs) rather than requiring a raw I420 buffer.
     bool supports_native_handle;

     // The name of this particular encoder implementation, e.g. "libvpx".
     std::string implementation_name;

     // If this field is true, the encoder rate controller must perform
     // well even in difficult situations, and produce close to the specified
     // target bitrate seen over a reasonable time window, drop frames if
     // necessary in order to keep the rate correct, and react quickly to
     // changing bitrate targets. If this method returns true, we disable the
     // frame dropper in the media optimization module and rely entirely on the
     // encoder to produce media at a bitrate that closely matches the target.
     // Any overshooting may result in delay buildup. If this method returns
     // false (default behavior), the media opt frame dropper will drop input
     // frames if it suspect encoder misbehavior. Misbehavior is common,
     // especially in hardware codecs. Disable media opt at your own risk.
     bool has_trusted_rate_controller;

     // If this field is true, the encoder uses hardware support and different
     // thresholds will be used in CPU adaptation.
     bool is_hardware_accelerated;

     // If this field is true, the encoder uses internal camera sources, meaning
     // that it does not require/expect frames to be delivered via
     // webrtc::VideoEncoder::Encode.
     // Internal source encoders are deprecated and support for them will be
     // phased out.
     bool has_internal_source;

     // For each spatial layer (simulcast stream or SVC layer), represented as an
     // element in |fps_allocation| a vector indicates how many temporal layers
     // the encoder is using for that spatial layer.
     // For each spatial/temporal layer pair, the frame rate fraction is given as
     // an 8bit unsigned integer where 0 = 0% and 255 = 100%.
     //
     // If the vector is empty for a given spatial layer, it indicates that frame
     // rates are not defined and we can't count on any specific frame rate to be
     // generated. Likely this indicates Vp8TemporalLayersType::kBitrateDynamic.
     //
     // The encoder may update this on a per-frame basis in response to both
     // internal and external signals.
     //
     // Spatial layers are treated independently, but temporal layers are
     // cumulative. For instance, if:
     //   fps_allocation[0][0] = kFullFramerate / 2;
     //   fps_allocation[0][1] = kFullFramerate;
     // Then half of the frames are in the base layer and half is in TL1, but
     // since TL1 is assumed to depend on the base layer, the frame rate is
     // indicated as the full 100% for the top layer.
     //
     // Defaults to a single spatial layer containing a single temporal layer
     // with a 100% frame rate fraction.
     absl::InlinedVector<uint8_t, kMaxTemporalStreams>
         fps_allocation[kMaxSpatialLayers];

     // Recommended bitrate limits for different resolutions.
     std::vector<ResolutionBitrateLimits> resolution_bitrate_limits;

     // Obtains the limits from |resolution_bitrate_limits| that best matches the
     // |frame_size_pixels|.
     absl::optional<ResolutionBitrateLimits>
     GetEncoderBitrateLimitsForResolution(int frame_size_pixels) const;

     // If true, this encoder has internal support for generating simulcast
     // streams. Otherwise, an adapter class will be needed.
     // Even if true, the config provided to InitEncode() might not be supported,
     // in such case the encoder should return
     // WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED.
     bool supports_simulcast;
   };

   struct RTC_EXPORT RateControlParameters {
     RateControlParameters();
     RateControlParameters(const VideoBitrateAllocation& bitrate,
                           double framerate_fps);
     RateControlParameters(const VideoBitrateAllocation& bitrate,
                           double framerate_fps,
                           DataRate bandwidth_allocation);
     virtual ~RateControlParameters();

     // Target bitrate, per spatial/temporal layer.
     // A target bitrate of 0bps indicates a layer should not be encoded at all.
     VideoBitrateAllocation bitrate;
     // Target framerate, in fps. A value <= 0.0 is invalid and should be
     // interpreted as framerate target not available. In this case the encoder
     // should fall back to the max framerate specified in |codec_settings| of
     // the last InitEncode() call.
     double framerate_fps;
     // The network bandwidth available for video. This is at least
     // |bitrate.get_sum_bps()|, but may be higher if the application is not
     // network constrained.
     DataRate bandwidth_allocation;

     bool operator==(const RateControlParameters& rhs) const;
     bool operator!=(const RateControlParameters& rhs) const;
   };

   struct LossNotification {
     // The timestamp of the last decodable frame *prior* to the last received.
     // (The last received - described below - might itself be decodable or not.)
     uint32_t timestamp_of_last_decodable;
     // The timestamp of the last received frame.
     uint32_t timestamp_of_last_received;
     // Describes whether the dependencies of the last received frame were
     // all decodable.
     // |false| if some dependencies were undecodable, |true| if all dependencies
     // were decodable, and |nullopt| if the dependencies are unknown.
     absl::optional<bool> dependencies_of_last_received_decodable;
     // Describes whether the received frame was decodable.
     // |false| if some dependency was undecodable or if some packet belonging
     // to the last received frame was missed.
     // |true| if all dependencies were decodable and all packets belonging
     // to the last received frame were received.
     // |nullopt| if no packet belonging to the last frame was missed, but the
     // last packet in the frame was not yet received.
     absl::optional<bool> last_received_decodable;
   };

   // Negotiated capabilities which the VideoEncoder may expect the other
   // side to use.
   struct Capabilities {
     explicit Capabilities(bool loss_notification)
         : loss_notification(loss_notification) {}
     bool loss_notification;
   };

   struct Settings {
     Settings(const Capabilities& capabilities,
              int number_of_cores,
              size_t max_payload_size)
         : capabilities(capabilities),
           number_of_cores(number_of_cores),
           max_payload_size(max_payload_size) {}

     Capabilities capabilities;
     int number_of_cores;
     size_t max_payload_size;
   };

   static VideoCodecVP8 GetDefaultVp8Settings();
   static VideoCodecVP9 GetDefaultVp9Settings();
   static VideoCodecH264 GetDefaultH264Settings();

   virtual ~VideoEncoder() {}

   // Set a FecControllerOverride, through which the encoder may override
   // decisions made by FecController.
   // TODO(bugs.webrtc.org/10769): Update downstream, then make pure-virtual.
   virtual void SetFecControllerOverride(
       FecControllerOverride* fec_controller_override);

   // Initialize the encoder with the information from the codecSettings
   //
   // Input:
   //          - codec_settings    : Codec settings
   //          - settings          : Settings affecting the encoding itself.
   // Input for deprecated version:
   //          - number_of_cores   : Number of cores available for the encoder
   //          - max_payload_size  : The maximum size each payload is allowed
   //                                to have. Usually MTU - overhead.
   //
   // Return value                  : Set bit rate if OK
   //                                 <0 - Errors:
   //                                  WEBRTC_VIDEO_CODEC_ERR_PARAMETER
   //                                  WEBRTC_VIDEO_CODEC_ERR_SIZE
   //                                  WEBRTC_VIDEO_CODEC_MEMORY
   //                                  WEBRTC_VIDEO_CODEC_ERROR
   // TODO(bugs.webrtc.org/10720): After updating downstream projects and posting
   // an announcement to discuss-webrtc, remove the three-parameters variant
   // and make the two-parameters variant pure-virtual.
   /* RTC_DEPRECATED */ virtual int32_t InitEncode(
       const VideoCodec* codec_settings,
       int32_t number_of_cores,
       size_t max_payload_size);
   virtual int InitEncode(const VideoCodec* codec_settings,
                          const VideoEncoder::Settings& settings);

   // Register an encode complete callback object.
   //
   // Input:
   //          - callback         : Callback object which handles encoded images.
   //
   // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
   virtual int32_t RegisterEncodeCompleteCallback(
       EncodedImageCallback* callback) = 0;

   // Free encoder memory.
   // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
   virtual int32_t Release() = 0;

   // Encode an I420 image (as a part of a video stream). The encoded image
   // will be returned to the user through the encode complete callback.
   //
   // Input:
   //          - frame             : Image to be encoded
   //          - frame_types       : Frame type to be generated by the encoder.
   //
   // Return value                 : WEBRTC_VIDEO_CODEC_OK if OK
   //                                <0 - Errors:
   //                                  WEBRTC_VIDEO_CODEC_ERR_PARAMETER
   //                                  WEBRTC_VIDEO_CODEC_MEMORY
   //                                  WEBRTC_VIDEO_CODEC_ERROR
   virtual int32_t Encode(const VideoFrame& frame,
                          const std::vector<VideoFrameType>* frame_types) = 0;

   // Sets rate control parameters: bitrate, framerate, etc. These settings are
   // instantaneous (i.e. not moving averages) and should apply from now until
   // the next call to SetRates().
   virtual void SetRates(const RateControlParameters& parameters) = 0;

   // Inform the encoder when the packet loss rate changes.
   //
   // Input:   - packet_loss_rate  : The packet loss rate (0.0 to 1.0).
   virtual void OnPacketLossRateUpdate(float packet_loss_rate);

   // Inform the encoder when the round trip time changes.
   //
   // Input:   - rtt_ms            : The new RTT, in milliseconds.
   virtual void OnRttUpdate(int64_t rtt_ms);

   // Called when a loss notification is received.
   virtual void OnLossNotification(const LossNotification& loss_notification);

   // Returns meta-data about the encoder, such as implementation name.
   // The output of this method may change during runtime. For instance if a
   // hardware encoder fails, it may fall back to doing software encoding using
   // an implementation with different characteristics.
   virtual EncoderInfo GetEncoderInfo() const;
 };
 }  // namespace webrtc
 #endif  // API_VIDEO_CODECS_VIDEO_ENCODER_H_
	/*
	* Copyright (c) 2014 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_CODECS_VIDEO_ENCODER_H_
	#define API_VIDEO_CODECS_VIDEO_ENCODER_H_

	#include <limits>
	#include <memory>
	#include <string>
	#include <vector>

	#include "absl/container/inlined_vector.h"
	#include "absl/types/optional.h"
	#include "api/fec_controller_override.h"
	#include "api/units/data_rate.h"
	#include "api/video/encoded_image.h"
	#include "api/video/video_bitrate_allocation.h"
	#include "api/video/video_codec_constants.h"
	#include "api/video/video_frame.h"
	#include "api/video_codecs/video_codec.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/system/rtc_export.h"

	namespace webrtc {

	class RTPFragmentationHeader;
	// TODO(pbos): Expose these through a public (root) header or change these APIs.
	struct CodecSpecificInfo;

	constexpr int kDefaultMinPixelsPerFrame = 320 * 180;

	class EncodedImageCallback {
	public:
	virtual ~EncodedImageCallback() {}

	struct Result {
	enum Error {
	OK,

	// Failed to send the packet.
	ERROR_SEND_FAILED,
	};

	explicit Result(Error error) : error(error) {}
	Result(Error error, uint32_t frame_id) : error(error), frame_id(frame_id) {}

	Error error;

	// Frame ID assigned to the frame. The frame ID should be the same as the ID
	// seen by the receiver for this frame. RTP timestamp of the frame is used
	// as frame ID when RTP is used to send video. Must be used only when
	// error=OK.
	uint32_t frame_id = 0;

	// Tells the encoder that the next frame is should be dropped.
	bool drop_next_frame = false;
	};

	// Used to signal the encoder about reason a frame is dropped.
	// kDroppedByMediaOptimizations - dropped by MediaOptimizations (for rate
	// limiting purposes).
	// kDroppedByEncoder - dropped by encoder's internal rate limiter.
	enum class DropReason : uint8_t {
	kDroppedByMediaOptimizations,
	kDroppedByEncoder
	};

	// Callback function which is called when an image has been encoded.
	virtual Result OnEncodedImage(
	const EncodedImage& encoded_image,
	const CodecSpecificInfo* codec_specific_info,
	const RTPFragmentationHeader* fragmentation) = 0;

	virtual void OnDroppedFrame(DropReason reason) {}
	};

	class RTC_EXPORT VideoEncoder {
	public:
	struct QpThresholds {
	QpThresholds(int l, int h) : low(l), high(h) {}
	QpThresholds() : low(-1), high(-1) {}
	int low;
	int high;
	};

	// Quality scaling is enabled if thresholds are provided.
	struct RTC_EXPORT ScalingSettings {
	private:
	// Private magic type for kOff, implicitly convertible to
	// ScalingSettings.
	struct KOff {};

	public:
	// TODO(nisse): Would be nicer if kOff were a constant ScalingSettings
	// rather than a magic value. However, absl::optional is not trivially copy
	// constructible, and hence a constant ScalingSettings needs a static
	// initializer, which is strongly discouraged in Chrome. We can hopefully
	// fix this when we switch to absl::optional or std::optional.
	static constexpr KOff kOff = {};

	ScalingSettings(int low, int high);
	ScalingSettings(int low, int high, int min_pixels);
	ScalingSettings(const ScalingSettings&);
	ScalingSettings(KOff); // NOLINT(runtime/explicit)
	~ScalingSettings();

	absl::optional<QpThresholds> thresholds;

	// We will never ask for a resolution lower than this.
	// TODO(kthelgason): Lower this limit when better testing
	// on MediaCodec and fallback implementations are in place.
	// See https://bugs.chromium.org/p/webrtc/issues/detail?id=7206
	int min_pixels_per_frame = kDefaultMinPixelsPerFrame;

	private:
	// Private constructor; to get an object without thresholds, use
	// the magic constant ScalingSettings::kOff.
	ScalingSettings();
	};

	// Bitrate limits for resolution.
	struct ResolutionBitrateLimits {
	ResolutionBitrateLimits(int frame_size_pixels,
	int min_start_bitrate_bps,
	int min_bitrate_bps,
	int max_bitrate_bps)
	: frame_size_pixels(frame_size_pixels),
	min_start_bitrate_bps(min_start_bitrate_bps),
	min_bitrate_bps(min_bitrate_bps),
	max_bitrate_bps(max_bitrate_bps) {}
	// Size of video frame, in pixels, the bitrate thresholds are intended for.
	int frame_size_pixels = 0;
	// Recommended minimum bitrate to start encoding.
	int min_start_bitrate_bps = 0;
	// Recommended minimum bitrate.
	int min_bitrate_bps = 0;
	// Recommended maximum bitrate.
	int max_bitrate_bps = 0;

	bool operator==(const ResolutionBitrateLimits& rhs) const;
	bool operator!=(const ResolutionBitrateLimits& rhs) const {
	return !(*this == rhs);
	}
	};

	// Struct containing metadata about the encoder implementing this interface.
	struct RTC_EXPORT EncoderInfo {
	static constexpr uint8_t kMaxFramerateFraction =
	std::numeric_limits<uint8_t>::max();

	EncoderInfo();
	EncoderInfo(const EncoderInfo&);

	~EncoderInfo();

	std::string ToString() const;
	bool operator==(const EncoderInfo& rhs) const;
	bool operator!=(const EncoderInfo& rhs) const { return !(*this == rhs); }

	// Any encoder implementation wishing to use the WebRTC provided
	// quality scaler must populate this field.
	ScalingSettings scaling_settings;

	// The width and height of the incoming video frames should be divisible
	// by \|requested_resolution_alignment\|. If they are not, the encoder may
	// drop the incoming frame.
	// 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 requested_resolution_alignment;

	// If true, encoder supports working with a native handle (e.g. texture
	// handle for hw codecs) rather than requiring a raw I420 buffer.
	bool supports_native_handle;

	// The name of this particular encoder implementation, e.g. "libvpx".
	std::string implementation_name;

	// If this field is true, the encoder rate controller must perform
	// well even in difficult situations, and produce close to the specified
	// target bitrate seen over a reasonable time window, drop frames if
	// necessary in order to keep the rate correct, and react quickly to
	// changing bitrate targets. If this method returns true, we disable the
	// frame dropper in the media optimization module and rely entirely on the
	// encoder to produce media at a bitrate that closely matches the target.
	// Any overshooting may result in delay buildup. If this method returns
	// false (default behavior), the media opt frame dropper will drop input
	// frames if it suspect encoder misbehavior. Misbehavior is common,
	// especially in hardware codecs. Disable media opt at your own risk.
	bool has_trusted_rate_controller;

	// If this field is true, the encoder uses hardware support and different
	// thresholds will be used in CPU adaptation.
	bool is_hardware_accelerated;

	// If this field is true, the encoder uses internal camera sources, meaning
	// that it does not require/expect frames to be delivered via
	// webrtc::VideoEncoder::Encode.
	// Internal source encoders are deprecated and support for them will be
	// phased out.
	bool has_internal_source;

	// For each spatial layer (simulcast stream or SVC layer), represented as an
	// element in \|fps_allocation\| a vector indicates how many temporal layers
	// the encoder is using for that spatial layer.
	// For each spatial/temporal layer pair, the frame rate fraction is given as
	// an 8bit unsigned integer where 0 = 0% and 255 = 100%.
	//
	// If the vector is empty for a given spatial layer, it indicates that frame
	// rates are not defined and we can't count on any specific frame rate to be
	// generated. Likely this indicates Vp8TemporalLayersType::kBitrateDynamic.
	//
	// The encoder may update this on a per-frame basis in response to both
	// internal and external signals.
	//
	// Spatial layers are treated independently, but temporal layers are
	// cumulative. For instance, if:
	// fps_allocation[0][0] = kFullFramerate / 2;
	// fps_allocation[0][1] = kFullFramerate;
	// Then half of the frames are in the base layer and half is in TL1, but
	// since TL1 is assumed to depend on the base layer, the frame rate is
	// indicated as the full 100% for the top layer.
	//
	// Defaults to a single spatial layer containing a single temporal layer
	// with a 100% frame rate fraction.
	absl::InlinedVector<uint8_t, kMaxTemporalStreams>
	fps_allocation[kMaxSpatialLayers];

	// Recommended bitrate limits for different resolutions.
	std::vector<ResolutionBitrateLimits> resolution_bitrate_limits;

	// Obtains the limits from \|resolution_bitrate_limits\| that best matches the
	// \|frame_size_pixels\|.
	absl::optional<ResolutionBitrateLimits>
	GetEncoderBitrateLimitsForResolution(int frame_size_pixels) const;

	// If true, this encoder has internal support for generating simulcast
	// streams. Otherwise, an adapter class will be needed.
	// Even if true, the config provided to InitEncode() might not be supported,
	// in such case the encoder should return
	// WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED.
	bool supports_simulcast;
	};

	struct RTC_EXPORT RateControlParameters {
	RateControlParameters();
	RateControlParameters(const VideoBitrateAllocation& bitrate,
	double framerate_fps);
	RateControlParameters(const VideoBitrateAllocation& bitrate,
	double framerate_fps,
	DataRate bandwidth_allocation);
	virtual ~RateControlParameters();

	// Target bitrate, per spatial/temporal layer.
	// A target bitrate of 0bps indicates a layer should not be encoded at all.
	VideoBitrateAllocation bitrate;
	// Target framerate, in fps. A value <= 0.0 is invalid and should be
	// interpreted as framerate target not available. In this case the encoder
	// should fall back to the max framerate specified in \|codec_settings\| of
	// the last InitEncode() call.
	double framerate_fps;
	// The network bandwidth available for video. This is at least
	// \|bitrate.get_sum_bps()\|, but may be higher if the application is not
	// network constrained.
	DataRate bandwidth_allocation;

	bool operator==(const RateControlParameters& rhs) const;
	bool operator!=(const RateControlParameters& rhs) const;
	};

	struct LossNotification {
	// The timestamp of the last decodable frame prior to the last received.
	// (The last received - described below - might itself be decodable or not.)
	uint32_t timestamp_of_last_decodable;
	// The timestamp of the last received frame.
	uint32_t timestamp_of_last_received;
	// Describes whether the dependencies of the last received frame were
	// all decodable.
	// \|false\| if some dependencies were undecodable, \|true\| if all dependencies
	// were decodable, and \|nullopt\| if the dependencies are unknown.
	absl::optional<bool> dependencies_of_last_received_decodable;
	// Describes whether the received frame was decodable.
	// \|false\| if some dependency was undecodable or if some packet belonging
	// to the last received frame was missed.
	// \|true\| if all dependencies were decodable and all packets belonging
	// to the last received frame were received.
	// \|nullopt\| if no packet belonging to the last frame was missed, but the
	// last packet in the frame was not yet received.
	absl::optional<bool> last_received_decodable;
	};

	// Negotiated capabilities which the VideoEncoder may expect the other
	// side to use.
	struct Capabilities {
	explicit Capabilities(bool loss_notification)
	: loss_notification(loss_notification) {}
	bool loss_notification;
	};

	struct Settings {
	Settings(const Capabilities& capabilities,
	int number_of_cores,
	size_t max_payload_size)
	: capabilities(capabilities),
	number_of_cores(number_of_cores),
	max_payload_size(max_payload_size) {}

	Capabilities capabilities;
	int number_of_cores;
	size_t max_payload_size;
	};

	static VideoCodecVP8 GetDefaultVp8Settings();
	static VideoCodecVP9 GetDefaultVp9Settings();
	static VideoCodecH264 GetDefaultH264Settings();

	virtual ~VideoEncoder() {}

	// Set a FecControllerOverride, through which the encoder may override
	// decisions made by FecController.
	// TODO(bugs.webrtc.org/10769): Update downstream, then make pure-virtual.
	virtual void SetFecControllerOverride(
	FecControllerOverride* fec_controller_override);

	// Initialize the encoder with the information from the codecSettings
	//
	// Input:
	// - codec_settings : Codec settings
	// - settings : Settings affecting the encoding itself.
	// Input for deprecated version:
	// - number_of_cores : Number of cores available for the encoder
	// - max_payload_size : The maximum size each payload is allowed
	// to have. Usually MTU - overhead.
	//
	// Return value : Set bit rate if OK
	// <0 - Errors:
	// WEBRTC_VIDEO_CODEC_ERR_PARAMETER
	// WEBRTC_VIDEO_CODEC_ERR_SIZE
	// WEBRTC_VIDEO_CODEC_MEMORY
	// WEBRTC_VIDEO_CODEC_ERROR
	// TODO(bugs.webrtc.org/10720): After updating downstream projects and posting
	// an announcement to discuss-webrtc, remove the three-parameters variant
	// and make the two-parameters variant pure-virtual.
	/* RTC_DEPRECATED */ virtual int32_t InitEncode(
	const VideoCodec* codec_settings,
	int32_t number_of_cores,
	size_t max_payload_size);
	virtual int InitEncode(const VideoCodec* codec_settings,
	const VideoEncoder::Settings& settings);

	// Register an encode complete callback object.
	//
	// Input:
	// - callback : Callback object which handles encoded images.
	//
	// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
	virtual int32_t RegisterEncodeCompleteCallback(
	EncodedImageCallback* callback) = 0;

	// Free encoder memory.
	// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
	virtual int32_t Release() = 0;

	// Encode an I420 image (as a part of a video stream). The encoded image
	// will be returned to the user through the encode complete callback.
	//
	// Input:
	// - frame : Image to be encoded
	// - frame_types : Frame type to be generated by the encoder.
	//
	// Return value : WEBRTC_VIDEO_CODEC_OK if OK
	// <0 - Errors:
	// WEBRTC_VIDEO_CODEC_ERR_PARAMETER
	// WEBRTC_VIDEO_CODEC_MEMORY
	// WEBRTC_VIDEO_CODEC_ERROR
	virtual int32_t Encode(const VideoFrame& frame,
	const std::vector<VideoFrameType>* frame_types) = 0;

	// Sets rate control parameters: bitrate, framerate, etc. These settings are
	// instantaneous (i.e. not moving averages) and should apply from now until
	// the next call to SetRates().
	virtual void SetRates(const RateControlParameters& parameters) = 0;

	// Inform the encoder when the packet loss rate changes.
	//
	// Input: - packet_loss_rate : The packet loss rate (0.0 to 1.0).
	virtual void OnPacketLossRateUpdate(float packet_loss_rate);

	// Inform the encoder when the round trip time changes.
	//
	// Input: - rtt_ms : The new RTT, in milliseconds.
	virtual void OnRttUpdate(int64_t rtt_ms);

	// Called when a loss notification is received.
	virtual void OnLossNotification(const LossNotification& loss_notification);

	// Returns meta-data about the encoder, such as implementation name.
	// The output of this method may change during runtime. For instance if a
	// hardware encoder fails, it may fall back to doing software encoding using
	// an implementation with different characteristics.
	virtual EncoderInfo GetEncoderInfo() const;
	};
	} // namespace webrtc
	#endif // API_VIDEO_CODECS_VIDEO_ENCODER_H_