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In this mode, we let WebRTC Audio Processing figure-out the delay. This is useful when the latency reported by the stack cannot be trusted. Note that in this mode, the leaking of echo during packet lost is much worst. It is recommanded to use PLC (e.g. spanplc, or opus built-in plc). In this mode, we don't do any synchronization. Instead, we simply process all the available reverse stream data as it comes. |
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| gstwebrtcdsp.cpp | ||
| gstwebrtcdsp.h | ||
| gstwebrtcechoprobe.cpp | ||
| gstwebrtcechoprobe.h | ||
| Makefile.am | ||