type Port = string | u16
type InputRtpVideoOptions = {
codec?: "h264";
rtp_payload_type?: u8;
}
type InputRtpVideoOptions = {
codec?: "h264";
}
codec - (default="h264") Video codec.
@@ -37,11 +37,9 @@ "h264" - H264 video.rtp_payload_type - (default=96) Value of payload type field in received RTP packets.
-Packets with different payload type won't be treated as video and included in composing. Values should be in [0, 64] or [96, 255]. Values in range [65, 95] can't be used. For more information, see RFC Packets with different payload type won't be treated as video and included in composing.type InputRtpAudioOptions = {
codec?: "opus";
sample_rate: u32;
channels: "mono" | "stereo";
rtp_payload_type?: u8;
forward_error_correction?: bool;
}
type InputRtpAudioOptions = {
codec?: "opus";
forward_error_correction?: bool;
}
codec - (default="opus") Audio codec.
@@ -49,15 +47,6 @@ "opus" - Opus audio.sample_rate - Sample rate. If the specified sample rate doesn't match real sample rate, audio won't be mixed properly.channels - Audio channels.
-"mono" - Mono audio (single channel)."stereo" - Stereo audio (two channels).rtp_payload_type - (default=97) Value of payload type field in received RTP packets.
-Packets with different payload type won't be treated as audio and included in mixing. Values should be in range [0, 64] or [96, 255]. Values in range [65, 95] can't be used. For more information, check out RFC.forward_error_correction - (default=false) Specifies whether the stream uses forward error correction. It's specific for Opus codec. For more information, check out RFC.type Port = string | u16
type InputRtpVideoOptions = {
codec?: "h264";
rtp_payload_type?: u8;
}
type InputRtpVideoOptions = {
codec?: "h264";
}
codec - (default="h264") Video codec.
@@ -39,11 +39,9 @@ "h264" - H264 video.rtp_payload_type - (default=96) Value of payload type field in received RTP packets.
-Packets with different payload type won't be treated as video and included in composing. Values should be in [0, 64] or [96, 255]. Values in range [65, 95] can't be used. For more information, see RFC Packets with different payload type won't be treated as video and included in composing.type InputRtpAudioOptions = {
codec?: "opus";
sample_rate: u32;
channels: "mono" | "stereo";
rtp_payload_type?: u8;
forward_error_correction?: bool;
}
type InputRtpAudioOptions = {
codec?: "opus";
forward_error_correction?: bool;
}
codec - (default="opus") Audio codec.
@@ -51,15 +49,6 @@ "opus" - Opus audio.sample_rate - Sample rate. If the specified sample rate doesn't match real sample rate, audio won't be mixed properly.channels - Audio channels.
-"mono" - Mono audio (single channel)."stereo" - Stereo audio (two channels).rtp_payload_type - (default=97) Value of payload type field in received RTP packets.
-Packets with different payload type won't be treated as audio and included in mixing. Values should be in range [0, 64] or [96, 255]. Values in range [65, 95] can't be used. For more information, check out RFC.forward_error_correction - (default=false) Specifies whether the stream uses forward error correction. It's specific for Opus codec. For more information, check out RFC.