Expose adaptive_ptime from Android SDK.
Bug: None Change-Id: Ideec24a0561efef83387f9b9605a5b68371fefa3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215228 Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com> Reviewed-by: Xavier Lepaul <xalep@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33768}
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@ -79,6 +79,9 @@ public class RtpParameters {
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// SSRC to be used by this encoding.
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// Can't be changed between getParameters/setParameters.
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public Long ssrc;
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// Set to true to allow dynamic frame length changes for audio:
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// https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime
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public boolean adaptiveAudioPacketTime;
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// This constructor is useful for creating simulcast layers.
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public Encoding(String rid, boolean active, Double scaleResolutionDownBy) {
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@ -90,7 +93,8 @@ public class RtpParameters {
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@CalledByNative("Encoding")
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Encoding(String rid, boolean active, double bitratePriority, @Priority int networkPriority,
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Integer maxBitrateBps, Integer minBitrateBps, Integer maxFramerate,
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Integer numTemporalLayers, Double scaleResolutionDownBy, Long ssrc) {
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Integer numTemporalLayers, Double scaleResolutionDownBy, Long ssrc,
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boolean adaptiveAudioPacketTime) {
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this.rid = rid;
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this.active = active;
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this.bitratePriority = bitratePriority;
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@ -101,6 +105,7 @@ public class RtpParameters {
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this.numTemporalLayers = numTemporalLayers;
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this.scaleResolutionDownBy = scaleResolutionDownBy;
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this.ssrc = ssrc;
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this.adaptiveAudioPacketTime = adaptiveAudioPacketTime;
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}
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@Nullable
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@ -159,6 +164,11 @@ public class RtpParameters {
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Long getSsrc() {
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return ssrc;
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}
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@CalledByNative("Encoding")
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boolean getAdaptivePTime() {
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return adaptiveAudioPacketTime;
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}
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}
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public static class Codec {
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