/*
 *  Copyright 2013 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.
 */

package org.webrtc;

import androidx.annotation.Nullable;
import java.lang.Double;
import java.lang.String;
import java.util.List;
import java.util.Map;
import org.webrtc.MediaStreamTrack;

/**
 * The parameters for an {@code RtpSender}, as defined in
 * http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface.
 *
 * Note: These structures use nullable Integer/etc. types because in the
 * future, they may be used to construct ORTC RtpSender/RtpReceivers, in
 * which case "null" will be used to represent "choose the implementation
 * default value".
 */
public class RtpParameters {
  /** Java version of webrtc::DegradationPreference - the ordering must be kept in sync. */
  public enum DegradationPreference {
    /**
    * Maintain framerate and resolution regardless of video quality. Frames may be dropped
    * before encoding if necessary not to overuse network and encoder resources.
    */
    MAINTAIN_FRAMERATE_AND_RESOLUTION,
    /** Degrade resolution in order to maintain framerate. */
    MAINTAIN_FRAMERATE,
    /** Degrade framerate in order to maintain resolution. */
    MAINTAIN_RESOLUTION,
    /** Degrade a balance of framerate and resolution. */
    BALANCED,
    // TODO(webrtc:450044904): Switch downstream projects to MAINTAIN_FRAMERATE_AND_RESOLUTION
    // and remove DISABLED.
    /** Does not degrade resolution or framerate. */
    DISABLED;

    @CalledByNative
    static DegradationPreference fromNativeIndex(int nativeIndex) {
      return values()[nativeIndex];
    }
  }

  public static class Encoding {
    // If non-null, this represents the RID that identifies this encoding layer.
    // RIDs are used to identify layers in simulcast.
    @Nullable public String rid;
    // Set to true to cause this encoding to be sent, and false for it not to
    // be sent.
    public boolean active = true;
    // The relative bitrate priority of this encoding. Currently this is
    // implemented for the entire RTP sender by using the value of the first
    // encoding parameter.
    // See: https://w3c.github.io/webrtc-priority/#enumdef-rtcprioritytype
    // "very-low" = 0.5
    // "low" = 1.0
    // "medium" = 2.0
    // "high" = 4.0
    public double bitratePriority = 1.0;
    // The relative DiffServ Code Point priority for this encoding, allowing
    // packets to be marked relatively higher or lower without affecting
    // bandwidth allocations.
    @Priority public int networkPriority = Priority.LOW;
    // If non-null, this represents the Transport Independent Application
    // Specific maximum bandwidth defined in RFC3890. If null, there is no
    // maximum bitrate.
    @Nullable public Integer maxBitrateBps;
    // The minimum bitrate in bps for video.
    @Nullable public Integer minBitrateBps;
    // The max framerate in fps for video.
    @Nullable public Integer maxFramerate;
    // The number of temporal layers for video.
    @Nullable public Integer numTemporalLayers;
    // If non-null, scale the width and height down by this factor for video. If null,
    // implementation default scaling factor will be used.
    @Nullable public Double scaleResolutionDownBy;
    // SSRC to be used by this encoding.
    // Can't be changed between getParameters/setParameters.
    public Long ssrc;
    // Set to true to allow dynamic frame length changes for audio:
    // https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime
    public boolean adaptiveAudioPacketTime;

    // This constructor is useful for creating simulcast layers.
    public Encoding(String rid, boolean active, Double scaleResolutionDownBy) {
      this.rid = rid;
      this.active = active;
      this.scaleResolutionDownBy = scaleResolutionDownBy;
    }

    @CalledByNative
    Encoding(String rid, boolean active, double bitratePriority, @Priority int networkPriority,
        Integer maxBitrateBps, Integer minBitrateBps, Integer maxFramerate,
        Integer numTemporalLayers, Double scaleResolutionDownBy, Long ssrc,
        boolean adaptiveAudioPacketTime) {
      this.rid = rid;
      this.active = active;
      this.bitratePriority = bitratePriority;
      this.networkPriority = networkPriority;
      this.maxBitrateBps = maxBitrateBps;
      this.minBitrateBps = minBitrateBps;
      this.maxFramerate = maxFramerate;
      this.numTemporalLayers = numTemporalLayers;
      this.scaleResolutionDownBy = scaleResolutionDownBy;
      this.ssrc = ssrc;
      this.adaptiveAudioPacketTime = adaptiveAudioPacketTime;
    }

    @Nullable
    @CalledByNative
    String getRid() {
      return rid;
    }

    @CalledByNative
    boolean getActive() {
      return active;
    }

    @CalledByNative
    double getBitratePriority() {
      return bitratePriority;
    }

    @CalledByNative
    @Priority
    int getNetworkPriority() {
      return networkPriority;
    }

    @Nullable
    @CalledByNative
    Integer getMaxBitrateBps() {
      return maxBitrateBps;
    }

    @Nullable
    @CalledByNative
    Integer getMinBitrateBps() {
      return minBitrateBps;
    }

    @Nullable
    @CalledByNative
    Integer getMaxFramerate() {
      return maxFramerate;
    }

    @Nullable
    @CalledByNative
    Integer getNumTemporalLayers() {
      return numTemporalLayers;
    }

    @Nullable
    @CalledByNative
    Double getScaleResolutionDownBy() {
      return scaleResolutionDownBy;
    }

    @CalledByNative
    Long getSsrc() {
      return ssrc;
    }

    @CalledByNative
    boolean getAdaptivePTime() {
      return adaptiveAudioPacketTime;
    }
  }

  public static class Codec {
    // Payload type used to identify this codec in RTP packets.
    public int payloadType;
    // Name used to identify the codec. Equivalent to MIME subtype.
    public String name;
    // The media type of this codec. Equivalent to MIME top-level type.
    MediaStreamTrack.MediaType kind;
    // Clock rate in Hertz.
    public Integer clockRate;
    // The number of audio channels used. Set to null for video codecs.
    public Integer numChannels;
    // The "format specific parameters" field from the "a=fmtp" line in the SDP
    public Map<String, String> parameters;

    @CalledByNative
    Codec(int payloadType, String name, MediaStreamTrack.MediaType kind, Integer clockRate,
        Integer numChannels, Map<String, String> parameters) {
      this.payloadType = payloadType;
      this.name = name;
      this.kind = kind;
      this.clockRate = clockRate;
      this.numChannels = numChannels;
      this.parameters = parameters;
    }

    @CalledByNative
    int getPayloadType() {
      return payloadType;
    }

    @CalledByNative
    String getName() {
      return name;
    }

    @CalledByNative
    MediaStreamTrack.MediaType getKind() {
      return kind;
    }

    @CalledByNative
    Integer getClockRate() {
      return clockRate;
    }

    @CalledByNative
    Integer getNumChannels() {
      return numChannels;
    }

    @CalledByNative
    Map<String, String> getParameters() {
      return parameters;
    }
  }

  public static class Rtcp {
    /** The Canonical Name used by RTCP */
    private final String cname;
    /** Whether reduced size RTCP is configured or compound RTCP */
    private final boolean reducedSize;

    @CalledByNative
    Rtcp(String cname, boolean reducedSize) {
      this.cname = cname;
      this.reducedSize = reducedSize;
    }

    @CalledByNative
    public String getCname() {
      return cname;
    }

    @CalledByNative
    public boolean getReducedSize() {
      return reducedSize;
    }
  }

  public static class HeaderExtension {
    /** The URI of the RTP header extension, as defined in RFC5285. */
    private final String uri;
    /** The value put in the RTP packet to identify the header extension. */
    private final int id;
    /** Whether the header extension is encrypted or not. */
    private final boolean encrypted;

    @CalledByNative
    HeaderExtension(String uri, int id, boolean encrypted) {
      this.uri = uri;
      this.id = id;
      this.encrypted = encrypted;
    }

    @CalledByNative
    public String getUri() {
      return uri;
    }

    @CalledByNative
    public int getId() {
      return id;
    }

    @CalledByNative
    public boolean getEncrypted() {
      return encrypted;
    }
  }

  public final String transactionId;

  /**
   * When bandwidth is constrained and the RtpSender needs to choose between degrading resolution or
   * degrading framerate, degradationPreference indicates which is preferred.
   */
  @Nullable public DegradationPreference degradationPreference;

  private final Rtcp rtcp;

  private final List<HeaderExtension> headerExtensions;

  public final List<Encoding> encodings;

  public final List<Codec> codecs;

  @CalledByNative
  RtpParameters(String transactionId, DegradationPreference degradationPreference, Rtcp rtcp,
      List<HeaderExtension> headerExtensions, List<Encoding> encodings, List<Codec> codecs) {
    this.transactionId = transactionId;
    this.degradationPreference = degradationPreference;
    this.rtcp = rtcp;
    this.headerExtensions = headerExtensions;
    this.encodings = encodings;
    this.codecs = codecs;
  }

  @CalledByNative
  String getTransactionId() {
    return transactionId;
  }

  @CalledByNative
  DegradationPreference getDegradationPreference() {
    return degradationPreference;
  }

  @CalledByNative
  public Rtcp getRtcp() {
    return rtcp;
  }

  @CalledByNative
  public List<HeaderExtension> getHeaderExtensions() {
    return headerExtensions;
  }

  @CalledByNative
  List<Encoding> getEncodings() {
    return encodings;
  }

  @CalledByNative
  List<Codec> getCodecs() {
    return codecs;
  }
}