Audio Player

Stopped
⚙ Firmware Settings
Source
Audio
80 %
In L
R
Out L
R
-60-48-36-24-120
Advanced
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Status
State Stopped Source — Codec — Bitrate — Stream Title — Buffer delay — Uptime — Reconnects —
Missed pkts — Max delay evt. —

Counters reset on Apply & Restart.

Parameter Reference

Source — Type
HTTP / Web Radio
Connects to an HTTP(S) audio stream — web radio stations, Icecast or Shoutcast servers, and any direct MP3/AAC/Opus URL.
RTP Stream
Receives a Real-time Transport Protocol (RTP) audio stream. Used for point-to-point or multicast audio distribution, typically over a local area network.
BRTP
Barix extension to RTP. The device initiates the connection toward the server, which allows reception through firewalls and NAT without any port-forwarding configuration. The server then streams unicast RTP back to the device.
Source — HTTP
Stream URL
Full URL of the audio stream, including protocol, host, port and path.
Examples:
http://radio.example.com:8000/stream
https://live.example.com/audio.mp3
Source — RTP
IP Address
Source address of the RTP stream. Use a multicast address (e.g. 239.x.x.x) for multicast distribution, or a unicast IP for point-to-point delivery.
Port
UDP port number the stream is transmitted on. Must match the sender configuration. Default is 5004.
Payload — Static
Select when the codec is known in advance. Choose the matching codec from the list. The device will force-decode the stream with the selected codec.
Payload — Dynamic (SDP)
Select when a Session Description Protocol file is available. The SDP document describes the codec, sample rate, and other stream parameters automatically.
Codec
Audio codec used by the RTP sender. Must match the encoder on the sending side exactly, otherwise playback will produce noise or silence.
SDP Content
The Session Description Protocol document for the stream. Paste it directly into the text area, or use Load file to import a .sdp file from your computer.
Source — BRTP
Server IP
IP address of the BRTP server that will send the audio stream.
Server Port
UDP port the BRTP server listens on. Request and keepalive packets are sent to this port, and the server also sends the RTP stream from this port back to the device. Default is 5004.
Local Receive Port
The UDP port the device binds to on its network interface. BRTP request packets are sent from this port, so the server knows exactly where to send the RTP stream back. The same port is where the incoming stream arrives. Change this only if 5004 is already in use by another service on this device.
Password
Optional access credential. If the BRTP server requires authentication, enter the password here. Leave blank if the server has no password.
Payload (Static / Dynamic)
Same as for RTP. Static selects a fixed codec; Dynamic uses an SDP file to describe the stream format. With Dynamic, the SDP port is automatically rewritten internally — paste the original SDP as-is.
Source — RIST
Mode — Caller
The device initiates the connection to a remote RIST sender. Provide the sender's IP address and port. Equivalent to pulling a stream.
Mode — Listener
The device waits for a remote RIST sender to connect to it. The sender must be configured to push to this device's IP and port. Optionally specify a bind address to restrict which network interface accepts the connection.
Sender IP / Bind Address
In Caller mode: the IP address of the RIST sender to connect to. In Listener mode: the local interface address to bind to (leave blank to accept on all interfaces).
Port
UDP port number for the RIST session. In Caller mode this is the sender's port; in Listener mode this is the local port the device listens on. Default is 5000.
Profile — Simple
RIST Simple Profile (profile 0). Provides reliable transport via NACK-based retransmission. No authentication or encryption. Suitable for trusted networks.
Profile — Main
RIST Main Profile (profile 1). Adds DTLS-based authentication and optional AES encryption on top of the Simple Profile retransmission mechanism.
Encryption
AES encryption for the RIST session (Main Profile only). Choose AES-128 or AES-256 and provide a shared secret that matches the sender configuration.
Secret Passphrase
The pre-shared secret for RIST Main Profile authentication and encryption. Must match exactly on both sender and receiver sides.
Payload (Static / Dynamic)
Same as for RTP. RIST carries RTP packets internally. Static forces a known codec; Dynamic uses an SDP file to describe the stream format automatically.
Source — SRT
Mode — Caller
The device connects to a remote SRT listener. Provide the remote host IP and port. Equivalent to pulling a stream from a server.
Mode — Listener
The device waits for a remote SRT caller to connect to it. Only a port is needed. The sender must target this device's IP on the configured port.
Mode — Rendezvous
Both sides connect to each other simultaneously on the same port. Useful for NAT traversal where neither side can act as a pure listener. Provide the remote host IP and a shared port.
Remote Host
IP address or hostname of the remote SRT sender (Caller and Rendezvous modes). Not used in Listener mode.
Port
UDP port for the SRT session. In Caller mode this is the remote port; in Listener mode it is the local port the device binds to; in Rendezvous mode both sides use the same port. Default is 4900.
Latency
SRT receiver buffer latency in milliseconds. Higher values give the protocol more time to recover lost packets via retransmission, reducing dropouts at the cost of increased end-to-end delay. Minimum recommended value is twice the one-way network delay. Default is 120 ms.
Stream ID
Optional identifier sent to the SRT server during the handshake. Used by some servers (e.g. Haivision, Wowza) for stream routing or access control. Leave blank if the server does not require it.
Encryption
AES encryption for the SRT session. Requires a passphrase of 10–79 characters shared with the sender. AES-128 uses a 16-byte key; AES-192 uses 24 bytes; AES-256 uses 32 bytes. Both sides must use the same key length and passphrase.
Passphrase
The shared secret for SRT encryption. Must be between 10 and 79 characters and match the sender exactly.
Payload (Static / Dynamic)
SRT carries RTP packets as its payload. Static forces a known codec; Dynamic uses an SDP file to describe the stream format automatically.
Status — RIST Statistics
Quality
Instant stream quality score reported by libRIST (0–100%). Calculated from the ratio of received to expected packets in the last measurement window. Values below 100% indicate packet loss or late arrivals.
RTT / Avg RTT
Current and average round-trip time in milliseconds between this device and the RIST sender. High or increasing RTT indicates network congestion. The retransmission window is sized to the RTT, so high RTT requires a larger buffer.
Avg Buffer
Average time packets spend in the RIST receive buffer before being released for playback. Should stay well below the configured output buffer duration. A rising value may indicate increasing network jitter.
Lost (total)
Cumulative number of packets that were never recovered despite retransmission requests. Each lost packet results in a brief audio glitch. A non-zero value indicates a network path with insufficient reliability for the current buffer settings.
Recovered (total)
Cumulative number of packets that were successfully recovered via RIST retransmission (NACK). A non-zero recovered count with zero lost is normal — it means the ARQ mechanism is working correctly and compensating for network impairments.
Status — SRT Statistics
RTT
Round-trip time in milliseconds between this device and the SRT peer. SRT uses RTT to calculate the retransmission timeout. High RTT relative to the configured Latency means less time is available for packet recovery before the playout deadline.
Recv Rate
Cumulative average receive throughput in Mbit/s. Should be close to the stream's nominal bitrate. A significantly lower value may indicate network congestion, packet loss, or the sender limiting output.
Lost (total)
Cumulative number of packets declared lost — not recovered within the latency window. Each causes an audio artefact. Persistent losses suggest the RTT exceeds the latency budget, or the network loss rate is too high for SRT to compensate.
Dropped (total)
Cumulative number of packets dropped at the receiver because they arrived after their playout deadline. Dropping avoids a growing delay but causes audio gaps. If non-zero, increase the Latency setting to give packets more time to arrive.
Retransmit (total)
Cumulative number of packets received that were retransmissions requested by this device. A non-zero count with zero lost means SRT ARQ is successfully recovering impaired packets within the latency window.
Belated (total)
Cumulative number of packets that arrived after their scheduled playout time but were still accepted (within a tolerance). A high belated count with rising RTT suggests the latency setting needs to be increased.
Audio
Volume
Output playback level from 0 to 100%. Changes take effect immediately without restarting the player — no need to click Apply & Restart.
VU Meter — In
Pre-volume audio level as received from the source, before the volume control is applied. Active only while the stream is playing. Useful to verify the stream is arriving even when volume is set to zero.
VU Meter — Out
Post-volume audio level sent to the physical audio output. Scales with the Volume slider. This is what you would measure at the analog output.
Advanced
Output Buffer (ms)
How much audio is buffered in memory before playback begins, in milliseconds. A larger buffer gives the player more time to absorb network jitter and packet loss, producing smoother playback — but it increases the end-to-end latency by the same amount.

Must be at least 40 ms and at least 25 % greater than the Ready Threshold. Range: 40 ms – 60 000 ms.

Recommended values: 5000–10000 ms for internet streams, 500–2000 ms for stable LAN sources.
Ready Threshold (ms)
The minimum amount of audio that must be buffered before the player starts playing. A higher value reduces the chance of a buffer underrun at startup, at the cost of a longer initial delay.

Must always be less than the Output Buffer and no more than 80 % of it. Range: 20 ms – 10 000 ms.
Actions
Stop
Immediately halts playback. The player stays stopped until manually restarted. Configuration is not changed.
Apply & Restart
Saves the current form values to disk and restarts the player with the new configuration. Required after changing the source, codec, or buffer settings.
Status
State
Current player state:
Playing — stream is active and audio is playing.
Connecting — establishing the stream for the first time.
Reconnecting — stream was interrupted; player is retrying automatically.
Stopped — player was manually stopped.
Not configured — no source URL or address has been set.
Audio device busy — the ALSA audio device is held by another process; the player will retry automatically.
Error — unrecoverable connection error; player will retry after a delay.
Source
The URL or address the player is currently connected to.
Codec
Audio codec detected in the incoming stream (e.g. MP3, AAC, PCM_ALAW). Detected automatically from the stream when it connects. Shows — until the stream is established.
Bitrate
Encoded bitrate of the incoming audio stream in kb/s, as reported by the decoder. Shows — if the stream does not carry bitrate information (common with uncompressed PCM or certain RTP payloads).
Buffer Delay
Current audio buffer depth in milliseconds. This is the end-to-end latency from the sender to the listener's speakers. It increases toward the Output Buffer value and resets on reconnect.
Uptime
How long the current stream connection has been continuously active without interruption.
Reconnects
Number of times the player has automatically reconnected since the last Apply & Restart. A non-zero value indicates the stream has been interrupted at least once.
Missed Packets
Number of RTP packets that arrived out of sequence or were lost entirely. Increments during the session. A non-zero value indicates network packet loss.
Max Delay Events
Number of times the RTP jitter buffer reached its maximum depth. Indicates episodes of high network latency or congestion.