webrtc/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/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;

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 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 = 320 * 180;

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

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

    EncoderInfo();
    EncoderInfo(const EncoderInfo&);

    ~EncoderInfo();

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

    // 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];
  };

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

  virtual ~VideoEncoder() {}

  // Initialize the encoder with the information from the codecSettings
  //
  // Input:
  //          - codec_settings    : Codec settings
  //          - 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
  virtual int32_t InitEncode(const VideoCodec* codec_settings,
                             int32_t number_of_cores,
                             size_t max_payload_size) = 0;

  // 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);
  // TODO(bugs.webrtc.org/10379): Deprecated. Delete, and make above method pure
  // virtual, as soon as downstream applications are updated.
  virtual int32_t Encode(const VideoFrame& frame,
                         const CodecSpecificInfo* codec_specific_info,
                         const std::vector<VideoFrameType>* frame_types);

  // Inform the encoder about the new target bit rate.
  //
  // Input:
  //          - bitrate         : New target bit rate
  //          - framerate       : The target frame rate
  //
  // Return value                : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
  virtual int32_t SetRates(uint32_t bitrate, uint32_t framerate);

  // Default fallback: Just use the sum of bitrates as the single target rate.
  // TODO(sprang): Remove this default implementation when we remove SetRates().
  virtual int32_t SetRateAllocation(const VideoBitrateAllocation& allocation,
                                    uint32_t framerate);

  // 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);

  // 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_