roc-recv - Man Page
receive real-time audio
Synopsis
roc-recv Options
Description
Receive audio streams from remote senders and write them to an audio device or file.
Options
- -h, --help
Print help and exit
- -V, --version
Print version and exit
- -v, --verbose
Increase verbosity level (may be used multiple times)
- -L, --list-supported
list supported schemes and formats
- -o,--output=IO_URI
Output file or device URI
- --output-format=FILE_FORMAT
Force output file format
- --backup=IO_URI
Backup file or device URI (if set, used when there are no sessions)
- --backup-format=FILE_FORMAT
Force backup file format
- -s,--source=ENDPOINT_URI
Local source endpoint
- -r,--repair=ENDPOINT_URI
Local repair endpoint
- -c,--control=ENDPOINT_URI
Local control endpoint
- --miface=MIFACE
IPv4 or IPv6 address of the network interface on which to join the multicast group
- --sess-latency=STRING
Session target latency, TIME units
- --min-latency=STRING
Session minimum latency, TIME units
- --max-latency=STRING
Session maximum latency, TIME units
- --io-latency=STRING
Playback target latency, TIME units
- --np-timeout=STRING
Session no playback timeout, TIME units
- --bp-timeout=STRING
Session broken playback timeout, TIME units
- --bp-window=STRING
Session breakage detection window, TIME units
- --packet-limit=INT
Maximum packet size, in bytes
- --frame-limit=INT
Maximum internal frame size, in bytes
- --frame-length=TIME
Duration of the internal frames, TIME units
- --rate=INT
Override output sample rate, Hz
- --no-resampling
Disable resampling (default=off)
- --resampler-backend=ENUM
Resampler backend (possible values="default", "builtin", "speex" default=`default')
- --resampler-profile=ENUM
Resampler profile (possible values="low", "medium", "high" default=`medium')
- -1, --oneshot
Exit when last connected client disconnects (default=off)
- --poisoning
Enable uninitialized memory poisoning (default=off)
- --profiling
Enable self profiling (default=off)
- --beeping
Enable beeping on packet loss (default=off)
- --color=ENUM
Set colored logging mode for stderr output (possible values="auto", "always", "never" default=`auto')
Endpoint URI
--source, --repair, and --control options define network endpoints on which to receive the traffic.
ENDPOINT_URI should have the following form:
protocol://host[:port][/path][?query]
Examples:
- rtsp://localhost:123/path?query
- rtp+rs8m://localhost:123
- rtp://0.0.0.0:123
- rtp://[::1]:123
- rtcp://0.0.0.0:123
The list of supported protocols can be retrieved using --list-supported option.
The host field should be either FQDN (domain name), or IPv4 address, or IPv6 address in square brackets. It may be 0.0.0.0 (for IPv4) or [::] (for IPv6) to bind endpoint to all network interfaces.
The port field can be omitted if the protocol defines standard port. Otherwise, it is mandatory. It may be set to zero to bind endpoint to a radomly chosen ephemeral port.
The path and query fields are allowed only for protocols that support them, e.g. for RTSP.
If FEC is enabled on sender, a pair of a source and repair endpoints should be provided. The two endpoints should use compatible protocols, e.g. rtp+rs8m:// for source endpoint, and rs8m:// for repair endpoint. If FEC is disabled, a single source endpoint should be provided.
Supported source and repair protocols:
- source rtp://, repair none (bare RTP without FEC)
- source rtp+rs8m://, repair rs8m:// (RTP with Reed-Solomon FEC)
- source rtp+ldpc://, repair ldpc:// (RTP with LDPC-Staircase FEC)
In addition, it is recommended to provide control endpoint. It is used to exchange non-media information used to identify session, carry feedback, etc. If no control endpoint is provided, session operates in reduced fallback mode, which may be less robust and may not support all features.
Supported control protocols:
- rtcp://
IO URI
--output and --backup options require a device or file URI in one of the following forms:
- DEVICE_TYPE://DEVICE_NAME -- audio device
- DEVICE_TYPE://default -- default audio device for given device type
- file:///ABS/PATH -- absolute file path
- file://localhost/ABS/PATH -- absolute file path (alternative form; only "localhost" host is supported)
- file:/ABS/PATH -- absolute file path (alternative form)
- file:REL/PATH -- relative file path
- file://- -- stdout
- file:- -- stdout (alternative form)
Examples:
- pulse://default
- pulse://alsa_output.pci-0000_00_1f.3.analog-stereo
- alsa://hw:1,0
- file:///home/user/test.wav
- file://localhost/home/user/test.wav
- file:/home/user/test.wav
- file:./test.wav
- file:-
The list of supported schemes and file formats can be retrieved using --list-supported option.
If the --output is omitted, the default driver and device are selected. If the --backup is omitted, no backup source is used.
The --output-format and --backup-format options can be used to force the output or backup file format. If the option is omitted, the file format is auto-detected. The option is always required when the output or backup is stdout or stdin.
The path component of the provided URI is percent-decoded. For convenience, unencoded characters are allowed as well, except that % should be always encoded as %25.
For example, the file named /foo/bar%/[baz] may be specified using either of the following URIs: file:///foo%2Fbar%25%2F%5Bbaz%5D and file:///foo/bar%25/[baz].
Multicast interface
If --miface option is present, it defines an IP address of the network interface on which to join the multicast group. If not present, no multicast group should be joined.
It's not possible to receive multicast traffic without joining a multicast group. The user should either provide multicast interface, or join the group manually using foreign tools.
MIFACE should be an IP address of the network interface on which to join the multicast group. It may be 0.0.0.0 (for IPv4) or :: (for IPv6) to join the multicast group on all available interfaces.
Multiple slots
Multiple sets of endpoints can be specified to retrieve media from multiple addresses.
Such endpoint sets are called slots. All slots should have the same set of endpoint types (source, repair, etc) and should use the same protocols for them. All slots should also have their own multicast interface option, if it's used.
Backup audio
If --backup option is given, it defines input audio device or file which will be played when there are no connected sessions. If it's not given, silence is played instead.
Backup file is restarted from the beginning each time when the last session disconnect. The playback of of the backup file is automatically looped.
Time units
- TIME should have one of the following forms:
123ns, 123us, 123ms, 123s, 123m, 123h
Examples
Endpoint examples
Bind one bare RTP endpoint on all IPv4 interfaces:
$ roc-recv -vv -s rtp://0.0.0.0:10001
Bind two source and repair endpoints to all IPv4 interfaces (but not IPv6):
$ roc-recv -vv -s rtp+rs8m://0.0.0.0:10001 -r rs8m://0.0.0.0:10002
Bind two source and repair endpoints to all IPv6 interfaces (but not IPv4):
$ roc-recv -vv -s rtp+rs8m://[::]:10001 -r rs8m://[::]:10002
Bind two source and repair endpoints to a particular network interface:
$ roc-recv -vv -s rtp+rs8m://192.168.0.3:10001 -r rs8m://192.168.0.3:10002
Bind three source, repair, and control endpoints:
$ roc-recv -vv \ -s rtp+rs8m://192.168.0.3:10001 -r rs8m://192.168.0.3:10002 -c rtcp://192.168.0.3:10003
Bind two source and repair endpoints to a particular multicast address and join to a multicast group on a particular network interface:
$ roc-recv -vv -s rtp+rs8m://225.1.2.3:10001 -r rs8m://225.1.2.3:10002 --miface 192.168.0.3
Bind two sets of source, repair, and control endpoints:
$ roc-recv -vv \ -s rtp+rs8m://192.168.0.3:10001 -r rs8m://192.168.0.3:10002 -c rtcp://192.168.0.3:10003 \ -s rtp+rs8m://198.214.0.7:10001 -r rs8m://198.214.0.7:10002 -c rtcp://198.214.0.7:10003
I/O examples
Output to the default ALSA device:
$ roc-recv -vv -s rtp://0.0.0.0:10001 -o alsa://default
Output to a specific PulseAudio device:
$ roc-recv -vv -s rtp://0.0.0.0:10001 -o pulse://alsa_input.pci-0000_00_1f.3.analog-stereo
Output to a file in WAV format (guess format by extension):
$ roc-recv -vv -s rtp://0.0.0.0:10001 -o file:./output.wav
Output to a file in WAV format (specify format manually):
$ roc-recv -vv -s rtp://0.0.0.0:10001 -o file:./output --output-format wav
Output to stdout in WAV format:
$ roc-recv -vv -s rtp://0.0.0.0:10001 -o file:- --output-format wav >./output.wav
Output to a file in WAV format (absolute path):
$ roc-recv -vv -s rtp://0.0.0.0:10001 -o file:///home/user/output.wav
Specify backup file:
$ roc-recv -vv -s rtp://0.0.0.0:10001 --backup file:./backup.wav
Tuning examples
Force a specific rate on the output device:
$ roc-recv -vv -s rtp://0.0.0.0:10001 --rate=44100
Select the LDPC-Staircase FEC scheme:
$ roc-recv -vv -s rtp+ldpc://0.0.0.0:10001 -r ldpc://0.0.0.0:10002
Select higher session latency and timeouts:
$ roc-recv -vv -s rtp://0.0.0.0:10001 \ --sess-latency=5s --min-latency=-1s --max-latency=10s --np-timeout=10s --bp-timeout=10s
Select higher I/O latency:
$ roc-recv -vv -s rtp://0.0.0.0:10001 \ --io-latency=200ms
Select resampler profile:
$ roc-recv -vv -s rtp://0.0.0.0:10001 \ --resampler-profile=high
See Also
roc-send(1), and the Roc web site at https://roc-streaming.org/
Bugs
Please report any bugs found via GitHub (https://github.com/roc-streaming/roc-toolkit/).
Authors
See authors page on the website for a list of maintainers and contributors.
Copyright
2023, Roc Streaming authors