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Merge pull request jitsi#22 from jitsi/rtp-extensions
Implements rtp extensions natively (does not use libjitsi).
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| @@ -0,0 +1,104 @@ | ||
| /* | ||
| * Copyright @ 2019-Present 8x8, Inc | ||
| * | ||
| * Licensed under the Apache License, Version 2.0 (the "License"); | ||
| * you may not use this file except in compliance with the License. | ||
| * You may obtain a copy of the License at | ||
| * | ||
| * http://www.apache.org/licenses/LICENSE-2.0 | ||
| * | ||
| * Unless required by applicable law or agreed to in writing, software | ||
| * distributed under the License is distributed on an "AS IS" BASIS, | ||
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | ||
| * See the License for the specific language governing permissions and | ||
| * limitations under the License. | ||
| */ | ||
| package org.jitsi.nlj.rtp | ||
|
|
||
| /** | ||
| * Represents a signalled RTP header extension. | ||
| * | ||
| * @author Boris Grozev | ||
| */ | ||
| data class RtpExtension( | ||
| /** | ||
| * The ID that was signalled. | ||
| */ | ||
| val id: Byte, | ||
| /** | ||
| * The type. | ||
| */ | ||
| val type: RtpExtensionType) | ||
|
|
||
| /** | ||
| * Represents an RTP header extension type. | ||
| * | ||
| * @author Boris Grozev | ||
| */ | ||
| enum class RtpExtensionType (val uri: String) { | ||
| /** | ||
| * The URN identifying the RTP extension that allows mixers to send to | ||
| * conference participants the audio levels of all contributing sources. | ||
| * Defined in RFC6465. | ||
| */ | ||
| CSRC_AUDIO_LEVEL("urn:ietf:params:rtp-hdrext:csrc-audio-level"), | ||
|
|
||
| /** | ||
| * The URN identifying the RTP extension that allows clients to send to | ||
| * conference mixers the audio level of their packet payload. Defined in | ||
| * RFC6464. | ||
| */ | ||
| SSRC_AUDIO_LEVEL("urn:ietf:params:rtp-hdrext:ssrc-audio-level"), | ||
|
|
||
| /** | ||
| * The URN identifying the abs-send-time RTP extension. | ||
| * Defined at | ||
| * {@link "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"} | ||
| */ | ||
| ABS_SEND_TIME("http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"), | ||
|
|
||
| /** | ||
| * The URN which identifies the framemarking RTP extension defined at | ||
| * {@link "https://tools.ietf.org/html/draft-ietf-avtext-framemarking-03"} | ||
| */ | ||
| FRAME_MARKING("http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07"), | ||
|
|
||
| /** | ||
| * The URN which identifies the Original Header Block RTP extension defined | ||
| * in {@link "https://tools.ietf.org/html/draft-ietf-perc-double-02"}. | ||
| */ | ||
| ORIGINAL_HEADER_BLOCK("urn:ietf:params:rtp-hdrext:ohb"), | ||
|
|
||
| /** | ||
| * The URN which identifies the Transport-Wide Congestion Control RTP | ||
| * extension. | ||
| */ | ||
| TRANSPORT_CC("http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"), | ||
|
|
||
| /** | ||
| * The URN which identifies the rtp-stream-id extensions | ||
| * in {@link "https://tools.ietf.org/html/draft-ietf-mmusic-rid-10"}. | ||
| */ | ||
| RTP_STREAM_ID("urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"), | ||
|
|
||
| /** | ||
| * The URN which identifies the transmission time-offset extensions | ||
| * in {@link "https://tools.ietf.org/html/rfc5450"}. | ||
| */ | ||
| TOF("urn:ietf:params:rtp-hdrext:toffset"), | ||
|
|
||
| /** | ||
| * The URN which identifies the RTP Header Extension for Video Content Type. | ||
| */ | ||
| VIDEO_CONTENT_TYPE("http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"); | ||
|
|
||
| companion object { | ||
| fun createFrom(value: String): RtpExtensionType? { | ||
| return try { | ||
| RtpExtensionType.valueOf(value.toUpperCase()) | ||
| } catch (e: IllegalArgumentException) { | ||
| return null | ||
| } | ||
| } | ||
| } | ||
| } |
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