modified lib-jitsi-meet dev repo
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LocalSdpMunger.js 9.0KB

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  1. /* global __filename */
  2. import { getLogger } from 'jitsi-meet-logger';
  3. import * as MediaType from '../../service/RTC/MediaType';
  4. import { SdpTransformWrap } from './SdpTransformUtil';
  5. const logger = getLogger(__filename);
  6. /**
  7. * Fakes local SDP exposed to {@link JingleSessionPC} through the local
  8. * description getter. Modifies the SDP, so that it will contain muted local
  9. * video tracks description, even though their underlying {MediaStreamTrack}s
  10. * are no longer in the WebRTC peerconnection. That prevents from SSRC updates
  11. * being sent to Jicofo/remote peer and prevents sRD/sLD cycle on the remote
  12. * side.
  13. */
  14. export default class LocalSdpMunger {
  15. /**
  16. * Creates new <tt>LocalSdpMunger</tt> instance.
  17. *
  18. * @param {TraceablePeerConnection} tpc
  19. */
  20. constructor(tpc) {
  21. this.tpc = tpc;
  22. }
  23. /**
  24. * Makes sure that muted local video tracks associated with the parent
  25. * {@link TraceablePeerConnection} are described in the local SDP. It's done
  26. * in order to prevent from sending 'source-remove'/'source-add' Jingle
  27. * notifications when local video track is muted (<tt>MediaStream</tt> is
  28. * removed from the peerconnection).
  29. *
  30. * NOTE 1 video track is assumed
  31. *
  32. * @param {SdpTransformWrap} transformer the transformer instance which will
  33. * be used to process the SDP.
  34. * @return {boolean} <tt>true</tt> if there were any modifications to
  35. * the SDP wrapped by <tt>transformer</tt>.
  36. * @private
  37. */
  38. _addMutedLocalVideoTracksToSDP(transformer) {
  39. // Go over each video tracks and check if the SDP has to be changed
  40. const localVideos = this.tpc.getLocalTracks(MediaType.VIDEO);
  41. if (!localVideos.length) {
  42. return false;
  43. } else if (localVideos.length !== 1) {
  44. logger.error(
  45. `${this.tpc} there is more than 1 video track ! `
  46. + 'Strange things may happen !', localVideos);
  47. }
  48. const videoMLine = transformer.selectMedia('video');
  49. if (!videoMLine) {
  50. logger.debug(
  51. `${this.tpc} unable to hack local video track SDP`
  52. + '- no "video" media');
  53. return false;
  54. }
  55. let modified = false;
  56. for (const videoTrack of localVideos) {
  57. const muted = videoTrack.isMuted();
  58. const mediaStream = videoTrack.getOriginalStream();
  59. // During the mute/unmute operation there are periods of time when
  60. // the track's underlying MediaStream is not added yet to
  61. // the PeerConnection. The SDP needs to be munged in such case.
  62. const isInPeerConnection
  63. = mediaStream && this.tpc.isMediaStreamInPc(mediaStream);
  64. const shouldFakeSdp = muted || !isInPeerConnection;
  65. if (!shouldFakeSdp) {
  66. continue; // eslint-disable-line no-continue
  67. }
  68. // Inject removed SSRCs
  69. const requiredSSRCs
  70. = this.tpc.isSimulcastOn()
  71. ? this.tpc.simulcast.ssrcCache
  72. : [ this.tpc.sdpConsistency.cachedPrimarySsrc ];
  73. if (!requiredSSRCs.length) {
  74. logger.error(`No SSRCs stored for: ${videoTrack} in ${this.tpc}`);
  75. continue; // eslint-disable-line no-continue
  76. }
  77. modified = true;
  78. // We need to fake sendrecv.
  79. // NOTE the SDP produced here goes only to Jicofo and is never set
  80. // as localDescription. That's why
  81. // TraceablePeerConnection.mediaTransferActive is ignored here.
  82. videoMLine.direction = 'sendrecv';
  83. // Check if the recvonly has MSID
  84. const primarySSRC = requiredSSRCs[0];
  85. // FIXME The cname could come from the stream, but may turn out to
  86. // be too complex. It is fine to come up with any value, as long as
  87. // we only care about the actual SSRC values when deciding whether
  88. // or not an update should be sent.
  89. const primaryCname = `injected-${primarySSRC}`;
  90. for (const ssrcNum of requiredSSRCs) {
  91. // Remove old attributes
  92. videoMLine.removeSSRC(ssrcNum);
  93. // Inject
  94. videoMLine.addSSRCAttribute({
  95. id: ssrcNum,
  96. attribute: 'cname',
  97. value: primaryCname
  98. });
  99. videoMLine.addSSRCAttribute({
  100. id: ssrcNum,
  101. attribute: 'msid',
  102. value: videoTrack.storedMSID
  103. });
  104. }
  105. if (requiredSSRCs.length > 1) {
  106. const group = {
  107. ssrcs: requiredSSRCs.join(' '),
  108. semantics: 'SIM'
  109. };
  110. if (!videoMLine.findGroup(group.semantics, group.ssrcs)) {
  111. // Inject the group
  112. videoMLine.addSSRCGroup(group);
  113. }
  114. }
  115. // Insert RTX
  116. // FIXME in P2P RTX is used by Chrome regardless of config option
  117. // status. Because of that 'source-remove'/'source-add'
  118. // notifications are still sent to remove/add RTX SSRC and FID group
  119. if (!this.tpc.options.disableRtx) {
  120. this.tpc.rtxModifier.modifyRtxSsrcs2(videoMLine);
  121. }
  122. }
  123. return modified;
  124. }
  125. /**
  126. * Modifies 'cname', 'msid', 'label' and 'mslabel' by appending
  127. * the id of {@link LocalSdpMunger#tpc} at the end, preceding by a dash
  128. * sign.
  129. *
  130. * @param {MLineWrap} mediaSection - The media part (audio or video) of the
  131. * session description which will be modified in place.
  132. * @returns {void}
  133. * @private
  134. */
  135. _transformMediaIdentifiers(mediaSection) {
  136. const pcId = this.tpc.id;
  137. for (const ssrcLine of mediaSection.ssrcs) {
  138. switch (ssrcLine.attribute) {
  139. case 'cname':
  140. case 'label':
  141. case 'mslabel':
  142. ssrcLine.value = ssrcLine.value && `${ssrcLine.value}-${pcId}`;
  143. break;
  144. case 'msid': {
  145. if (ssrcLine.value) {
  146. const streamAndTrackIDs = ssrcLine.value.split(' ');
  147. if (streamAndTrackIDs.length === 2) {
  148. const streamId = streamAndTrackIDs[0];
  149. const trackId = streamAndTrackIDs[1];
  150. ssrcLine.value
  151. = `${streamId}-${pcId} ${trackId}-${pcId}`;
  152. } else {
  153. logger.warn(
  154. 'Unable to munge local MSID'
  155. + `- weird format detected: ${ssrcLine.value}`);
  156. }
  157. }
  158. break;
  159. }
  160. }
  161. }
  162. }
  163. /**
  164. * Maybe modifies local description to fake local video tracks SDP when
  165. * those are muted.
  166. *
  167. * @param {object} desc the WebRTC SDP object instance for the local
  168. * description.
  169. * @returns {RTCSessionDescription}
  170. */
  171. maybeAddMutedLocalVideoTracksToSDP(desc) {
  172. if (!desc) {
  173. throw new Error('No local description passed in.');
  174. }
  175. const transformer = new SdpTransformWrap(desc.sdp);
  176. if (this._addMutedLocalVideoTracksToSDP(transformer)) {
  177. return new RTCSessionDescription({
  178. type: desc.type,
  179. sdp: transformer.toRawSDP()
  180. });
  181. }
  182. return desc;
  183. }
  184. /**
  185. * This transformation will make sure that stream identifiers are unique
  186. * across all of the local PeerConnections even if the same stream is used
  187. * by multiple instances at the same time.
  188. * Each PeerConnection assigns different SSRCs to the same local
  189. * MediaStream, but the MSID remains the same as it's used to identify
  190. * the stream by the WebRTC backend. The transformation will append
  191. * {@link TraceablePeerConnection#id} at the end of each stream's identifier
  192. * ("cname", "msid", "label" and "mslabel").
  193. *
  194. * @param {RTCSessionDescription} sessionDesc - The local session
  195. * description (this instance remains unchanged).
  196. * @return {RTCSessionDescription} - Transformed local session description
  197. * (a modified copy of the one given as the input).
  198. */
  199. transformStreamIdentifiers(sessionDesc) {
  200. // FIXME similar check is probably duplicated in all other transformers
  201. if (!sessionDesc || !sessionDesc.sdp || !sessionDesc.type) {
  202. return sessionDesc;
  203. }
  204. const transformer = new SdpTransformWrap(sessionDesc.sdp);
  205. const audioMLine = transformer.selectMedia('audio');
  206. if (audioMLine) {
  207. this._transformMediaIdentifiers(audioMLine);
  208. }
  209. const videoMLine = transformer.selectMedia('video');
  210. if (videoMLine) {
  211. this._transformMediaIdentifiers(videoMLine);
  212. }
  213. return new RTCSessionDescription({
  214. type: sessionDesc.type,
  215. sdp: transformer.toRawSDP()
  216. });
  217. }
  218. }