The generic syntax is:
avserver [options] |
WARNING: avserver is unmaintained, largely broken and in need of a complete rewrite. It probably won’t work for you. Use at your own risk.
avserver is a streaming server for both audio and video. It supports several live feeds, streaming from files and time shifting on live feeds (you can seek to positions in the past on each live feed, provided you specify a big enough feed storage in avserver.conf).
This documentation covers only the streaming aspects of avserver / avconv. All questions about parameters for avconv, codec questions, etc. are not covered here. Read ‘avconv.html’ for more information.
avserver receives prerecorded files or FFM streams from some avconv instance as input, then streams them over RTP/RTSP/HTTP.
An avserver instance will listen on some port as specified in the configuration file. You can launch one or more instances of avconv and send one or more FFM streams to the port where avserver is expecting to receive them. Alternately, you can make avserver launch such avconv instances at startup.
Input streams are called feeds, and each one is specified by a <Feed> section in the configuration file.
For each feed you can have different output streams in various formats, each one specified by a <Stream> section in the configuration file.
avserver supports an HTTP interface which exposes the current status of the server.
Simply point your browser to the address of the special status stream specified in the configuration file.
For example if you have:
<Stream status.html> Format status # Only allow local people to get the status ACL allow localhost ACL allow 192.168.0.0 192.168.255.255 </Stream> |
then the server will post a page with the status information when the special stream ‘status.html’ is requested.
When properly configured and running, you can capture video and audio in real time from a suitable capture card, and stream it out over the Internet to either Windows Media Player or RealAudio player (with some restrictions).
It can also stream from files, though that is currently broken. Very often, a web server can be used to serve up the files just as well.
It can stream prerecorded video from .ffm files, though it is somewhat tricky to make it work correctly.
I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I’m using stock Linux 2.4.17 with the stock drivers. [Actually that isn’t true, I needed some special drivers for my motherboard-based sound card.]
I understand that FreeBSD systems work just fine as well.
First, build the kit. It *really* helps to have installed LAME first. Then when
you run the avserver ./configure, make sure that you have the
--enable-libmp3lame flag turned on.
LAME is important as it allows for streaming audio to Windows Media Player. Don’t ask why the other audio types do not work.
As a simple test, just run the following two command lines where INPUTFILE is some file which you can decode with avconv:
./avserver -f doc/avserver.conf & ./avconv -i INPUTFILE http://localhost:8090/feed1.ffm |
At this point you should be able to go to your Windows machine and fire up Windows Media Player (WMP). Go to Open URL and enter
http://<linuxbox>:8090/test.asf |
You should (after a short delay) see video and hear audio.
WARNING: trying to stream test1.mpg doesn’t work with WMP as it tries to transfer the entire file before starting to play. The same is true of AVI files.
You should edit the avserver.conf file to suit your needs (in terms of frame rates etc). Then install avserver and avconv, write a script to start them up, and off you go.
Maybe you didn’t install LAME, or got your ./configure statement wrong. Check the avconv output to see if a line referring to MP3 is present. If not, then your configuration was incorrect. If it is, then maybe your wiring is not set up correctly. Maybe the sound card is not getting data from the right input source. Maybe you have a really awful audio interface (like I do) that only captures in stereo and also requires that one channel be flipped. If you are one of these people, then export ’AUDIO_FLIP_LEFT=1’ before starting avconv.
Yes, they do.
Yes, it does. Who knows why?
Yes, it does. Any thoughts on this would be gratefully received. These differences extend to embedding WMP into a web page. [There are two object IDs that you can use: The old one, which does not play well, and the new one, which does (both tested on the same system). However, I suspect that the new one is not available unless you have installed WMP 7].
You can replay video from .ffm files that was recorded earlier. However, there are a number of caveats, including the fact that the avserver parameters must match the original parameters used to record the file. If they do not, then avserver deletes the file before recording into it. (Now that I write this, it seems broken).
You can fiddle with many of the codec choices and encoding parameters, and there are a bunch more parameters that you cannot control. Post a message to the mailing list if there are some ’must have’ parameters. Look in avserver.conf for a list of the currently available controls.
It will automatically generate the ASX or RAM files that are often used in browsers. These files are actually redirections to the underlying ASF or RM file. The reason for this is that the browser often fetches the entire file before starting up the external viewer. The redirection files are very small and can be transferred quickly. [The stream itself is often ’infinite’ and thus the browser tries to download it and never finishes.]
* When you connect to a live stream, most players (WMP, RA, etc) want to buffer a certain number of seconds of material so that they can display the signal continuously. However, avserver (by default) starts sending data in realtime. This means that there is a pause of a few seconds while the buffering is being done by the player. The good news is that this can be cured by adding a ’?buffer=5’ to the end of the URL. This means that the stream should start 5 seconds in the past – and so the first 5 seconds of the stream are sent as fast as the network will allow. It will then slow down to real time. This noticeably improves the startup experience.
You can also add a ’Preroll 15’ statement into the avserver.conf that will add the 15 second prebuffering on all requests that do not otherwise specify a time. In addition, avserver will skip frames until a key_frame is found. This further reduces the startup delay by not transferring data that will be discarded.
* You may want to adjust the MaxBandwidth in the avserver.conf to limit the amount of bandwidth consumed by live streams.
It turns out that (on my machine at least) the number of frames successfully grabbed is marginally less than the number that ought to be grabbed. This means that the timestamp in the encoded data stream gets behind realtime. This means that if you say ’Preroll 10’, then when the stream gets 10 or more seconds behind, there is no Preroll left.
Fixing this requires a change in the internals of how timestamps are handled.
?date= stuff work.Yes (subject to the limitation outlined above). Also note that whenever you start avserver, it deletes the ffm file (if any parameters have changed), thus wiping out what you had recorded before.
The format of the ?date=xxxxxx is fairly flexible. You should use one
of the following formats (the ’T’ is literal):
* YYYY-MM-DDTHH:MM:SS (localtime) * YYYY-MM-DDTHH:MM:SSZ (UTC) |
You can omit the YYYY-MM-DD, and then it refers to the current day. However note that ‘?date=16:00:00’ refers to 16:00 on the current day – this may be in the future and so is unlikely to be useful.
You use this by adding the ?date= to the end of the URL for the stream. For example: ‘http://localhost:8080/test.asf?date=2002-07-26T23:05:00’.
All the numerical options, if not specified otherwise, accept in input a string representing a number, which may contain one of the SI unit prefixes, for example ’K’, ’M’, ’G’. If ’i’ is appended after the prefix, binary prefixes are used, which are based on powers of 1024 instead of powers of 1000. The ’B’ postfix multiplies the value by 8, and can be appended after a unit prefix or used alone. This allows using for example ’KB’, ’MiB’, ’G’ and ’B’ as number postfix.
Options which do not take arguments are boolean options, and set the corresponding value to true. They can be set to false by prefixing with "no" the option name, for example using "-nofoo" in the command line will set to false the boolean option with name "foo".
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. -codec:a:1 ac3 option contains
a:1 stream specifer, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several stream, the option is then applied to all
of them. E.g. the stream specifier in -b:a 128k matches all audio
streams.
An empty stream specifier matches all streams, for example -codec copy
or -codec: copy would copy all the streams without reencoding.
Possible forms of stream specifiers are:
Matches the stream with this index. E.g. -threads:1 4 would set the
thread count for the second stream to 4.
stream_type is one of: ’v’ for video, ’a’ for audio, ’s’ for subtitle, ’d’ for data and ’t’ for attachments. If stream_index is given, then matches stream number stream_index of this type. Otherwise matches all streams of this type.
If stream_index is given, then matches stream number stream_index in program with id program_id. Otherwise matches all streams in this program.
These options are shared amongst the av* tools.
Show license.
Show help. An optional parameter may be specified to print help about a specific item.
Possible values of arg are:
Print detailed information about the decoder named decoder_name. Use the ‘-decoders’ option to get a list of all decoders.
Print detailed information about the encoder named encoder_name. Use the ‘-encoders’ option to get a list of all encoders.
Print detailed information about the demuxer named demuxer_name. Use the ‘-formats’ option to get a list of all demuxers and muxers.
Print detailed information about the muxer named muxer_name. Use the ‘-formats’ option to get a list of all muxers and demuxers.
Print detailed information about the filter name filter_name. Use the ‘-filters’ option to get a list of all filters.
Show version.
Show available formats.
The fields preceding the format names have the following meanings:
Decoding available
Encoding available
Show all codecs known to libavcodec.
Note that the term ’codec’ is used throughout this documentation as a shortcut for what is more correctly called a media bitstream format.
Show available decoders.
Show all available encoders.
Show available bitstream filters.
Show available protocols.
Show available libavfilter filters.
Show available pixel formats.
Show available sample formats.
Set the logging level used by the library. loglevel is a number or a string containing one of the following values:
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting
the environment variable AV_LOG_FORCE_COLOR.
The use of the environment variable NO_COLOR is deprecated and
will be dropped in a following Libav version.
Set a mask that’s applied to autodetected CPU flags. This option is intended for testing. Do not use it unless you know what you’re doing.
These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the list of available AVOptions, use the ‘-help’ option. They are separated into two categories:
These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs.
These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs.
For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the ‘id3v2_version’ private option of the MP3 muxer:
avconv -i input.flac -id3v2_version 3 out.mp3 |
All codec AVOptions are obviously per-stream, so the chapter on stream specifiers applies to them
Note ‘-nooption’ syntax cannot be used for boolean AVOptions, use ‘-option 0’/‘-option 1’.
Note2 old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name is now obsolete and will be removed soon.
set bitrate (in bits/s)
Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far ratecontrol is willing to deviate from the target average bitrate value. This is not related to minimum/maximum bitrate. Lowering tolerance too much has an adverse effect on quality.
Possible values:
allow decoders to produce unaligned output
use four motion vectors per macroblock (MPEG-4)
use 1/4-pel motion compensation
use loop filter
use fixed qscale
use gmc
always try a mb with mv=<0,0>
use internal 2-pass ratecontrol in first pass mode
use internal 2-pass ratecontrol in second pass mode
only decode/encode grayscale
do not draw edges
error[?] variables will be set during encoding
normalize adaptive quantization
use interlaced DCT
force low delay
place global headers in extradata instead of every keyframe
use only bitexact functions (except (I)DCT)
H.263 advanced intra coding / MPEG-4 AC prediction
interlaced motion estimation
closed GOP
Output even potentially corrupted frames
set motion estimation method
Possible values:
zero motion estimation (fastest)
full motion estimation (slowest)
EPZS motion estimation (default)
esa motion estimation (alias for full)
tesa motion estimation
diamond motion estimation (alias for EPZS)
log motion estimation
phods motion estimation
X1 motion estimation
hex motion estimation
umh motion estimation
set the group of picture (GOP) size
set audio sampling rate (in Hz)
set number of audio channels
set cutoff bandwidth
video quantizer scale compression (VBR). Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
video quantizer scale blur (VBR)
minimum video quantizer scale (VBR)
maximum video quantizer scale (VBR)
maximum difference between the quantizer scales (VBR)
use ’frames’ B frames
QP factor between P- and B-frames
ratecontrol method
strategy to choose between I/P/B-frames
RTP payload size in bytes
work around not autodetected encoder bugs
Possible values:
some old lavc-generated MSMPEG4v3 files (no autodetection)
Xvid interlacing bug (autodetected if FOURCC == XVIX)
(autodetected if FOURCC == UMP4)
padding bug (autodetected)
illegal VLC bug (autodetected per FOURCC)
old standard qpel (autodetected per FOURCC/version)
direct-qpel-blocksize bug (autodetected per FOURCC/version)
edge padding bug (autodetected per FOURCC/version)
work around various bugs in Microsoft’s broken decoders
truncated frames
how strictly to follow the standards
Possible values:
strictly conform to a older more strict version of the spec or reference software
strictly conform to all the things in the spec no matter what the consequences
allow unofficial extensions
allow non-standardized experimental things
QP offset between P- and B-frames
set error detection flags
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
use MPEG quantizers instead of H.263
how to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function)
experimental quantizer modulation
experimental quantizer modulation
Set rate control equation. When computing the expression, besides the standard functions defined in the section ’Expression Evaluation’, the following functions are available: bits2qp(bits), qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.
Set maximum bitrate tolerance (in bits/s). Requires bufsize to be set.
Set minimum bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use otherwise.
set ratecontrol buffer size (in bits)
currently useless
QP factor between P- and I-frames
QP offset between P- and I-frames
initial complexity for 1-pass encoding
DCT algorithm
Possible values:
autoselect a good one (default)
fast integer
accurate integer
floating point AAN DCT
compresses bright areas stronger than medium ones
temporal complexity masking
spatial complexity masking
inter masking
compresses dark areas stronger than medium ones
select IDCT implementation
Possible values:
floating point AAN IDCT
set error concealment strategy
Possible values:
iterative motion vector (MV) search (slow)
use strong deblock filter for damaged MBs
prediction method
Possible values:
sample aspect ratio
print specific debug info
Possible values:
picture info
rate control
macroblock (MB) type
per-block quantization parameter (QP)
motion vector
error recognition
memory management control operations (H.264)
visualize quantization parameter (QP), lower QP are tinted greener
visualize block types
picture buffer allocations
threading operations
visualize motion vectors (MVs)
Possible values:
forward predicted MVs of P-frames
forward predicted MVs of B-frames
backward predicted MVs of B-frames
full-pel ME compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
sub-pel ME compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
macroblock compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
interlaced DCT compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
diamond type & size for motion estimation
amount of motion predictors from the previous frame
pre motion estimation
pre motion estimation compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
diamond type & size for motion estimation pre-pass
sub-pel motion estimation quality
limit motion vectors range (1023 for DivX player)
intra quant bias
inter quant bias
Possible values:
variable length coder / Huffman coder
arithmetic coder
raw (no encoding)
run-length coder
deflate-based coder
context model
macroblock decision algorithm (high quality mode)
Possible values:
use mbcmp (default)
use fewest bits
use best rate distortion
scene change threshold
minimum Lagrange factor (VBR)
maximum Lagrange factor (VBR)
noise reduction
number of bits which should be loaded into the rc buffer before decoding starts
Possible values:
allow non-spec-compliant speedup tricks
skip bitstream encoding
ignore cropping information from sps
place global headers at every keyframe instead of in extradata
Possible values:
autodetect a suitable number of threads to use
motion estimation threshold
macroblock threshold
intra_dc_precision
nsse weight
number of macroblock rows at the top which are skipped
number of macroblock rows at the bottom which are skipped
Possible values:
Possible values:
frame skip threshold
frame skip factor
frame skip exponent
frame skip compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
increase the quantizer for macroblocks close to borders
minimum macroblock Lagrange factor (VBR)
maximum macroblock Lagrange factor (VBR)
motion estimation bitrate penalty compensation (1.0 = 256)
Possible values:
Possible values:
Possible values:
refine the two motion vectors used in bidirectional macroblocks
downscale frames for dynamic B-frame decision
minimum interval between IDR-frames (x264)
reference frames to consider for motion compensation
chroma QP offset from luma
rate-distortion optimal quantization
multiplied by qscale for each frame and added to scene_change_score
adjust sensitivity of b_frame_strategy 1
GOP timecode frame start number, in non-drop-frame format
set desired number of audio channels
Possible values:
Possible values:
number of slices, used in parallelized encoding
select multithreading type
Possible values:
audio service type
Possible values:
Main Audio Service
Effects
Visually Impaired
Hearing Impaired
Dialogue
Commentary
Emergency
Voice Over
Karaoke
Possible values:
8-bit unsigned integer
16-bit signed integer
32-bit signed integer
32-bit float
64-bit double
8-bit unsigned integer planar
16-bit signed integer planar
32-bit signed integer planar
32-bit float planar
64-bit double planar
set probing size
set packet size
Possible values:
reduce the latency by flushing out packets immediately
ignore index
generate pts
do not fill in missing values that can be exactly calculated
disable AVParsers, this needs nofillin too
ignore dts
discard corrupted frames
reduce the latency introduced by optional buffering
how many microseconds are analyzed to estimate duration
decryption key
max memory used for timestamp index (per stream)
max memory used for buffering real-time frames
print specific debug info
Possible values:
maximum muxing or demuxing delay in microseconds
number of frames used to probe fps
set error detection flags (deprecated; use err_detect, save via avconv)
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
set error detection flags
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
maximum buffering duration for interleaving
Use ‘configfile’ instead of ‘/etc/avserver.conf’.
Enable no-launch mode. This option disables all the Launch directives within the various <Stream> sections. Since avserver will not launch any avconv instances, you will have to launch them manually.
Enable debug mode. This option increases log verbosity, directs log messages to stdout.