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samples/GstAudioVideoReceiver.c

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@@ -68,7 +68,7 @@ VOID onGstAudioFrameReady(UINT64 customData, PFrame pFrame)
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GST_BUFFER_DTS(buffer) = presentationTsIncrement;
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GST_BUFFER_PTS(buffer) = presentationTsIncrement;
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GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(pFrame->size, GST_SECOND, DEFAULT_AUDIO_OPUS_BYTE_RATE);
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// TODO: check for other codecs once the pipelines are added
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if (gst_buffer_fill(buffer, 0, pFrame->frameData, pFrame->size) != pFrame->size) {

samples/kvsWebrtcClientMasterGstSample.c

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@@ -263,7 +263,6 @@ PVOID sendGstreamerAudioVideo(PVOID args)
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"queue leaky=2 max-size-buffers=400 ! audioconvert ! audioresample ! opusenc ! "
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"audio/x-opus,rate=48000,channels=2 ! appsink sync=TRUE emit-signals=TRUE name=appsink-audio",
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&error);
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}
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// TODO: test and add more such combinations
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break;

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