/*
* 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_