piixishstream 1.0.3 → 1.0.4
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
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@@ -75,7 +75,7 @@ export class WebRtcConnWrapper {
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const { clockRate } = rtpConfig;
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if (this.mediaConnection.daveReady)
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frame = this.mediaConnection.daveSession.encryptOpus(frame);
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-
this._audioTrack?.sendMessageBinary(frame);
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try { this._audioTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
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rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
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}
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sendVideoFrame(frame, frametime) {
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@@ -118,7 +118,7 @@ export class WebRtcConnWrapper {
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}
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frame = this.mediaConnection.daveSession.encrypt(1 /* MediaType.VIDEO */, daveCodec, frame);
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}
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this._videoTrack?.sendMessageBinary(frame);
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try { this._videoTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
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rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
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}
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setPacketizer(videoCodec) {
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package/package.json
CHANGED
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@@ -1,6 +1,6 @@
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{
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"name": "piixishstream",
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"version": "1.0.
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"version": "1.0.4",
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"description": "Experiment for making video streaming work for discord selfbots",
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"main": "./dist/index.js",
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"exports": "./dist/index.js",
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@@ -80,4 +80,4 @@
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"node-datachannel"
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]
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}
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-
}
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}
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@@ -108,7 +108,7 @@ export class WebRtcConnWrapper {
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const { clockRate } = rtpConfig;
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if (this.mediaConnection.daveReady)
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frame = this.mediaConnection.daveSession!.encryptOpus(frame);
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-
this._audioTrack?.sendMessageBinary(frame);
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try { this._audioTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
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rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
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}
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@@ -154,7 +154,7 @@ export class WebRtcConnWrapper {
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frame,
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);
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}
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-
this._videoTrack?.sendMessageBinary(frame);
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try { this._videoTrack?.sendMessageBinary(frame); } catch (e) { /* ignore SRTP/WebRTC drop errors */ }
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rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
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}
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