| /* | |
| * 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("DegradationPreference") | |
| 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") | |
| 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("Encoding") | |
| String getRid() { | |
| return rid; | |
| } | |
| @CalledByNative("Encoding") | |
| boolean getActive() { | |
| return active; | |
| } | |
| @CalledByNative("Encoding") | |
| double getBitratePriority() { | |
| return bitratePriority; | |
| } | |
| @CalledByNative("Encoding") | |
| @Priority | |
| int getNetworkPriority() { | |
| return networkPriority; | |
| } | |
| @Nullable | |
| @CalledByNative("Encoding") | |
| Integer getMaxBitrateBps() { | |
| return maxBitrateBps; | |
| } | |
| @Nullable | |
| @CalledByNative("Encoding") | |
| Integer getMinBitrateBps() { | |
| return minBitrateBps; | |
| } | |
| @Nullable | |
| @CalledByNative("Encoding") | |
| Integer getMaxFramerate() { | |
| return maxFramerate; | |
| } | |
| @Nullable | |
| @CalledByNative("Encoding") | |
| Integer getNumTemporalLayers() { | |
| return numTemporalLayers; | |
| } | |
| @Nullable | |
| @CalledByNative("Encoding") | |
| Double getScaleResolutionDownBy() { | |
| return scaleResolutionDownBy; | |
| } | |
| @CalledByNative("Encoding") | |
| Long getSsrc() { | |
| return ssrc; | |
| } | |
| @CalledByNative("Encoding") | |
| 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") | |
| 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("Codec") | |
| int getPayloadType() { | |
| return payloadType; | |
| } | |
| @CalledByNative("Codec") | |
| String getName() { | |
| return name; | |
| } | |
| @CalledByNative("Codec") | |
| MediaStreamTrack.MediaType getKind() { | |
| return kind; | |
| } | |
| @CalledByNative("Codec") | |
| Integer getClockRate() { | |
| return clockRate; | |
| } | |
| @CalledByNative("Codec") | |
| Integer getNumChannels() { | |
| return numChannels; | |
| } | |
| @CalledByNative("Codec") | |
| Map 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") | |
| Rtcp(String cname, boolean reducedSize) { | |
| this.cname = cname; | |
| this.reducedSize = reducedSize; | |
| } | |
| @CalledByNative("Rtcp") | |
| public String getCname() { | |
| return cname; | |
| } | |
| @CalledByNative("Rtcp") | |
| 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") | |
| HeaderExtension(String uri, int id, boolean encrypted) { | |
| this.uri = uri; | |
| this.id = id; | |
| this.encrypted = encrypted; | |
| } | |
| @CalledByNative("HeaderExtension") | |
| public String getUri() { | |
| return uri; | |
| } | |
| @CalledByNative("HeaderExtension") | |
| public int getId() { | |
| return id; | |
| } | |
| @CalledByNative("HeaderExtension") | |
| 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; | |
| } | |
| } |