Provided by: roc-toolkit-tools_0.3.0+dfsg-7ubuntu2_amd64 bug

NAME

       roc-send - send real-time audio

SYNOPSIS

       roc-send OPTIONS

DESCRIPTION

       Read audio stream from an audio device or file and send it to remote receiver.

   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

       -i,--input=IO_URI
              Input file or device URI

       --input-format=FILE_FORMAT
              Force input file format

       -s,--source=ENDPOINT_URI
              Remote source endpoint

       -r,--repair=ENDPOINT_URI
              Remote repair endpoint

       -c,--control=ENDPOINT_URI
              Remote control endpoint

       --reuseaddr
              enable SO_REUSEADDR when binding sockets

       --io-latency=STRING
              Recording target latency, TIME units

       --nbsrc=INT
              Number of source packets in FEC block

       --nbrpr=INT
              Number of repair packets in FEC block

       --packet-length=STRING
              Outgoing packet length, 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 input sample rate, Hz

       --resampler-backend=ENUM
              Resampler  backend   (possible  values="default",  "builtin",  "speex",  "speexdec"
              default=`default')

       --resampler-profile=ENUM
              Resampler profile  (possible values="low", "medium", "high" default=`medium')

       --interleaving
              Enable packet interleaving  (default=off)

       --profiling
              Enable self profiling  (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 to which to send the
       traffic.

       ENDPOINT_URI should have the following form:

       protocol://host[:port][/path][?query]

       Examples:

       • rtsp://localhost:123/path?queryrtp+rs8m://localhost:123rtp://127.0.0.1:123rtp://[::1]:123rtcp://10.9.8.3:123

       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.

       The  port  field  can  be  omitted if the protocol defines standard port. Otherwise, it is
       mandatory.

       The path and query fields are allowed only for protocols that support them, e.g. for RTSP.

       If FEC is enabled, 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
       --input option requires 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://- -- stdin

       • file:- -- stdin (alternative form)

       Examples:

       • pulse://defaultpulse://alsa_input.pci-0000_00_1f.3.analog-stereoalsa://hw:1,0file:///home/user/test.wavfile://localhost/home/user/test.wavfile:/home/user/test.wavfile:./test.wavfile:-

       The  list  of  supported  schemes and file formats can be retrieved using --list-supported
       option.

       If the --input is omitted, the default driver and device are selected.

       The --input-format option can be used to force the input file format. If  it  is  omitted,
       the file format is auto-detected. This option is always required when the input is 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].

   Multiple slots
       Multiple sets of endpoints can be specified to send media to 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.

   SO_REUSEADDR
       If --reuseaddr option is provided, SO_REUSEADDR socket option  will  be  enabled  for  all
       sockets.

       For  TCP,  it  allows immediately reusing recently closed socket in TIME_WAIT state, which
       may be useful you want to be able to restart server quickly.

       For UDP, it allows multiple processes to bind to the same address, which may be useful  if
       you're using systemd socket activation.

       Regardless of the option, SO_REUSEADDR is always disabled when binding to ephemeral port.

   Time units
       TIME should have one of the following forms:
              123ns, 123us, 123ms, 123s, 123m, 123h

EXAMPLES

   Endpoint examples
       Send file to receiver with one bare RTP endpoint:

          $ roc-send -vv -i file:./input.wav -s rtp://192.168.0.3:10001

       Send file to receiver with IPv4 source, repair, and control endpoints:

          $ roc-send -vv -i file:./input.wav -s rtp+rs8m://192.168.0.3:10001 \
              -r rs8m://192.168.0.3:10002 -c rtcp://192.168.0.3:10003

       Send file to receiver with IPv6 source, repair, and control endpoints:

          $ roc-send -vv -i file:./input.wav -s rtp+rs8m://[2001:db8::]:10001 \
              -r rs8m://[2001:db8::]:10002 -r rtcp://[2001:db8::]:10003

       Send file to two receivers, each with three endpoints:

          $ roc-send -vv \
              -i file:./input.wav \
              -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
       Capture sound from the default device (omit -i):

          $ roc-send -vv -s rtp://192.168.0.3:10001

       Capture sound from the default ALSA device:

          $ roc-send -vv -s rtp://192.168.0.3:10001 -i alsa://default

       Capture sound from a specific PulseAudio device:

          $ roc-send -vv -s rtp://192.168.0.3:10001 -i pulse://alsa_input.pci-0000_00_1f.3.analog-stereo

       Send WAV file (guess format by extension):

          $ roc-send -vv -s rtp://192.168.0.3:10001 -i file:./input.wav

       Send WAV file (specify format manually):

          $ roc-send -vv -s rtp://192.168.0.3:10001 -i file:./input.file --input-format wav

       Send WAV from stdin:

          $ roc-send -vv -s rtp://192.168.0.3:10001 -i file:- --input-format wav <./input.wav

       Send WAV file (specify absolute path):

          $ roc-send -vv -s rtp://192.168.0.3:10001 -i file:///home/user/input.wav

   Tuning examples
       Force a specific rate on the input device:

          $ roc-send -vv -s rtp://192.168.0.3:10001 --rate=44100

       Select the LDPC-Staircase FEC scheme and a larger block size:

          $ roc-send -vv -i file:./input.wav -s rtp+ldpc://192.168.0.3:10001 \
              -r ldpc://192.168.0.3:10002 -c ldpc://192.168.0.3:10003 \
              --nbsrc=1000 --nbrpr=500

       Select lower packet length:

          $ roc-send -vv -i file:./input.wav -s rtp+ldpc://192.168.0.3:10001 \
              --packet-length 2500us

       Select lower I/O latency and frame length:

          $ roc-send -vv -s rtp://192.168.0.3:10001 \
              --io-latency=20ms --frame-length 4ms

       Manually specify resampling parameters:

          $ roc-send -vv -s rtp://192.168.0.3:10001 \
              --resampler-backend=speex --resampler-profile=high

SEE ALSO

       roc-recv(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