1. Introduction

FFmpeg is a very fast video and audio converter. It can also grab from a live audio/video source.

The command line interface is designed to be intuitive, in the sense that FFmpeg tries to figure out all parameters that can possibly be derived automatically. You usually only have to specify the target bitrate you want.

FFmpeg can also convert from any sample rate to any other, and resize video on the fly with a high quality polyphase filter.

2. Quick Start

2.1 Video and Audio grabbing

FFmpeg can use a video4linux compatible video source and any Open Sound System audio source:

ffmpeg /tmp/out.mpg

Note that you must activate the right video source and channel before launching FFmpeg with any TV viewer such as xawtv (http://bytesex.org/xawtv/) by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.

2.2 Video and Audio file format conversion

* FFmpeg can use any supported file format and protocol as input:

Examples:

* You can use YUV files as input:

ffmpeg -i /tmp/test%d.Y /tmp/out.mpg

It will use the files:

/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...

The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the `-s' option if FFmpeg cannot guess it.

* You can input from a raw YUV420P file:

ffmpeg -i /tmp/test.yuv /tmp/out.avi

test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.

* You can output to a raw YUV420P file:

ffmpeg -i mydivx.avi hugefile.yuv

* You can set several input files and output files:

ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg

Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.

* You can also do audio and video conversions at the same time:

ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2

Converts a.wav to MPEG audio at 22050Hz sample rate.

* You can encode to several formats at the same time and define a mapping from input stream to output streams:

ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0

Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map file:index' specifies which input stream is used for each output stream, in the order of the definition of output streams.

* You can transcode decrypted VOBs

ffmpeg -i snatch_1.vob -f avi -vcodec mpeg4 -b 800 -g 300 -bf 2 -acodec mp3 -ab 128 snatch.avi

This is a typical DVD ripping example; the input is a VOB file, the output an AVI file with MPEG-4 video and MP3 audio. Note that in this command we use B-frames so the MPEG-4 stream is DivX5 compatible, and GOP size is 300 which means one intra frame every 10 seconds for 29.97fps input video. Furthermore, the audio stream is MP3-encoded so you need to enable LAME support by passing --enable-mp3lame to configure. The mapping is particularly useful for DVD transcoding to get the desired audio language.

NOTE: To see the supported input formats, use ffmpeg -formats.

3. Invocation

3.1 Syntax

The generic syntax is:

ffmpeg [[infile options][`-i' infile]]... {[outfile options] outfile}...

If no input file is given, audio/video grabbing is done.

As a general rule, options are applied to the next specified file. For example, if you give the `-b 64' option, it sets the video bitrate of the next file. The format option may be needed for raw input files.

By default, FFmpeg tries to convert as losslessly as possible: It uses the same audio and video parameters for the outputs as the one specified for the inputs.

3.2 Main options

`-L'

Show license.

`-h'

Show help.

`-formats'

Show available formats, codecs, protocols, ...

`-f fmt'

Force format.

`-i filename'

input filename

`-y'

Overwrite output files.

`-t duration'

Set the recording time in seconds. hh:mm:ss[.xxx] syntax is also supported.

`-ss position'

Seek to given time position in seconds. hh:mm:ss[.xxx] syntax is also supported.

`-title string'

Set the title.

`-author string'

Set the author.

`-copyright string'

Set the copyright.

`-comment string'

Set the comment.

`-target type'

Specify target file type ("vcd", "svcd", "dvd", "dv", "pal-vcd", "ntsc-svcd", ... ). All the format options (bitrate, codecs, buffer sizes) are then set automatically. You can just type:

ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg

Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in:

ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg

`-hq'

Activate high quality settings.

`-itsoffset offset'

Set the input time offset in seconds. [-]hh:mm:ss[.xxx] syntax is also supported. This option affects all the input files that follow it. The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by 'offset' seconds.

3.3 Video Options

`-b bitrate'

Set the video bitrate in kbit/s (default = 200 kb/s).

`-r fps'

Set frame rate (default = 25).

`-s size'

Set frame size. The format is `wxh' (default = 160x128). The following abbreviations are recognized:

`sqcif'

128x96

`qcif'

176x144

`cif'

352x288

`4cif'

704x576

`-aspect aspect'

Set aspect ratio (4:3, 16:9 or 1.3333, 1.7777).

`-croptop size'

Set top crop band size (in pixels).

`-cropbottom size'

Set bottom crop band size (in pixels).

`-cropleft size'

Set left crop band size (in pixels).

`-cropright size'

Set right crop band size (in pixels).

`-padtop size'

Set top pad band size (in pixels).

`-padbottom size'

Set bottom pad band size (in pixels).

`-padleft size'

Set left pad band size (in pixels).

`-padright size'

Set right pad band size (in pixels).

`-padcolor (hex color)'

Set color of padded bands. The value for padcolor is expressed as a six digit hexadecimal number where the first two digits represent red, the middle two digits green and last two digits blue (default = 000000 (black)).

`-vn'

Disable video recording.

`-bt tolerance'

Set video bitrate tolerance (in kbit/s).

`-maxrate bitrate'

Set max video bitrate tolerance (in kbit/s).

`-minrate bitrate'

Set min video bitrate tolerance (in kbit/s).

`-bufsize size'

Set rate control buffer size (in kbit).

`-vcodec codec'

Force video codec to codec. Use the copy special value to tell that the raw codec data must be copied as is.

`-sameq'

Use same video quality as source (implies VBR).

`-pass n'

Select the pass number (1 or 2). It is useful to do two pass encoding. The statistics of the video are recorded in the first pass and the video is generated at the exact requested bitrate in the second pass.

`-passlogfile file'

Set two pass logfile name to file.

3.4 Advanced Video Options

`-g gop_size'

Set the group of pictures size.

`-intra'

Use only intra frames.

`-qscale q'

Use fixed video quantiser scale (VBR).

`-qmin q'

minimum video quantiser scale (VBR)

`-qmax q'

maximum video quantiser scale (VBR)

`-qdiff q'

maximum difference between the quantiser scales (VBR)

`-qblur blur'

video quantiser scale blur (VBR)

`-qcomp compression'

video quantiser scale compression (VBR)

`-rc_init_cplx complexity'

initial complexity for single pass encoding

`-b_qfactor factor'

qp factor between P- and B-frames

`-i_qfactor factor'

qp factor between P- and I-frames

`-b_qoffset offset'

qp offset between P- and B-frames

`-i_qoffset offset'

qp offset between P- and I-frames

`-rc_eq equation'

Set rate control equation (see section FFmpeg formula evaluator) (default = tex^qComp).

`-rc_override override'

rate control override for specific intervals

`-me method'

Set motion estimation method to method. Available methods are (from lowest to best quality):

`zero'

Try just the (0, 0) vector.

`phods'

`log'

`x1'

`epzs'

(default method)

`full'

exhaustive search (slow and marginally better than epzs)

`-dct_algo algo'

Set DCT algorithm to algo. Available values are:

`0'

FF_DCT_AUTO (default)

`1'

FF_DCT_FASTINT

`2'

FF_DCT_INT

`3'

FF_DCT_MMX

`4'

FF_DCT_MLIB

`5'

FF_DCT_ALTIVEC

`-idct_algo algo'

Set IDCT algorithm to algo. Available values are:

`0'

FF_IDCT_AUTO (default)

`1'

FF_IDCT_INT

`2'

FF_IDCT_SIMPLE

`3'

FF_IDCT_SIMPLEMMX

`4'

FF_IDCT_LIBMPEG2MMX

`5'

FF_IDCT_PS2

`6'

FF_IDCT_MLIB

`7'

FF_IDCT_ARM

`8'

FF_IDCT_ALTIVEC

`9'

FF_IDCT_SH4

`10'

FF_IDCT_SIMPLEARM

`-er n'

Set error resilience to n.

`1'

FF_ER_CAREFUL (default)

`2'

FF_ER_COMPLIANT

`3'

FF_ER_AGGRESSIVE

`4'

FF_ER_VERY_AGGRESSIVE

`-ec bit_mask'

Set error concealment to bit_mask. bit_mask is a bit mask of the following values:

`1'

FF_EC_GUESS_MVS (default = enabled)

`2'

FF_EC_DEBLOCK (default = enabled)

`-bf frames'

Use 'frames' B-frames (supported for MPEG-1, MPEG-2 and MPEG-4).

`-mbd mode'

macroblock decision

`0'

FF_MB_DECISION_SIMPLE: Use mb_cmp (cannot change it yet in FFmpeg).

`1'

FF_MB_DECISION_BITS: Choose the one which needs the fewest bits.

`2'

FF_MB_DECISION_RD: rate distortion

`-4mv'

Use four motion vector by macroblock (MPEG-4 only).

`-part'

Use data partitioning (MPEG-4 only).

`-bug param'

Work around encoder bugs that are not auto-detected.

`-strict strictness'

How strictly to follow the standards.

`-aic'

Enable Advanced intra coding (h263+).

`-umv'

Enable Unlimited Motion Vector (h263+)

`-deinterlace'

Deinterlace pictures.

`-interlace'

Force interlacing support in encoder (MPEG-2 and MPEG-4 only). Use this option if your input file is interlaced and you want to keep the interlaced format for minimum losses. The alternative is to deinterlace the input stream with `-deinterlace', but deinterlacing introduces losses.

`-psnr'

Calculate PSNR of compressed frames.

`-vstats'

Dump video coding statistics to `vstats_HHMMSS.log'.

`-vhook module'

Insert video processing module. module contains the module name and its parameters separated by spaces.

3.5 Audio Options

`-ar freq'

Set the audio sampling frequency (default = 44100 Hz).

`-ab bitrate'

Set the audio bitrate in kbit/s (default = 64).

`-ac channels'

Set the number of audio channels (default = 1).

`-an'

Disable audio recording.

`-acodec codec'

Force audio codec to codec. Use the copy special value to specify that the raw codec data must be copied as is.

3.6 Audio/Video grab options

`-vd device'

sEt video grab device (e.g. `/dev/video0').

`-vc channel'

Set video grab channel (DV1394 only).

`-tvstd standard'

Set television standard (NTSC, PAL (SECAM)).

`-dv1394'

Set DV1394 grab.

`-ad device'

Set audio device (e.g. `/dev/dsp').

3.7 Advanced options

`-map file:stream'

Set input stream mapping.

`-debug'

Print specific debug info.

`-benchmark'

Add timings for benchmarking.

`-hex'

Dump each input packet.

`-bitexact'

Only use bit exact algorithms (for codec testing).

`-ps size'

Set packet size in bits.

`-re'

Read input at native frame rate. Mainly used to simulate a grab device.

`-loop'

Loop over the input stream. Currently it works only for image streams. This option is used for automatic FFserver testing.

`-loop_output number_of_times'

Repeatedly loop output for formats that support looping such as animated GIF (0 will loop the output infinitely).

3.8 FFmpeg formula evaluator

When evaluating a rate control string, FFmpeg uses an internal formula evaluator.

The following binary operators are available: +, -, *, /, ^.

The following unary operators are available: +, -, (...).

The following functions are available:

sinh(x)

cosh(x)

tanh(x)

sin(x)

cos(x)

tan(x)

exp(x)

log(x)

squish(x)

gauss(x)

abs(x)

max(x, y)

min(x, y)

gt(x, y)

lt(x, y)

eq(x, y)

bits2qp(bits)

qp2bits(qp)

The following constants are available:

PI

E

iTex

pTex

tex

mv

fCode

iCount

mcVar

var

isI

isP

isB

avgQP

qComp

avgIITex

avgPITex

avgPPTex

avgBPTex

avgTex

3.9 Protocols

The filename can be `-' to read from standard input or to write to standard output.

FFmpeg also handles many protocols specified with an URL syntax.

Use 'ffmpeg -formats' to see a list of the supported protocols.

The protocol http: is currently used only to communicate with FFserver (see the FFserver documentation). When FFmpeg will be a video player it will also be used for streaming :-)

4. Tips

• For streaming at very low bitrate application, use a low frame rate and a small GOP size. This is especially true for RealVideo where the Linux player does not seem to be very fast, so it can miss frames. An example is:

  ffmpeg -g 3 -r 3 -t 10 -b 50 -s qcif -f rv10 /tmp/b.rm

• The parameter 'q' which is displayed while encoding is the current quantizer. The value 1 indicates that a very good quality could be achieved. The value 31 indicates the worst quality. If q=31 appears too often, it means that the encoder cannot compress enough to meet your bitrate. You must either increase the bitrate, decrease the frame rate or decrease the frame size.
• If your computer is not fast enough, you can speed up the compression at the expense of the compression ratio. You can use '-me zero' to speed up motion estimation, and '-intra' to disable motion estimation completely (you have only I-frames, which means it is about as good as JPEG compression).
• To have very low audio bitrates, reduce the sampling frequency (down to 22050 kHz for MPEG audio, 22050 or 11025 for AC3).
• To have a constant quality (but a variable bitrate), use the option '-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst quality).
• When converting video files, you can use the '-sameq' option which uses the same quality factor in the encoder as in the decoder. It allows almost lossless encoding.

5. Supported File Formats and Codecs

You can use the -formats option to have an exhaustive list.

5.1 File Formats

FFmpeg supports the following file formats through the libavformat library:

X means that encoding (resp. decoding) is supported.

5.2 Image Formats

FFmpeg can read and write images for each frame of a video sequence. The following image formats are supported:

X means that encoding (resp. decoding) is supported.

5.3 Video Codecs

X means that encoding (resp. decoding) is supported.

See http://mplayerhq.hu/~michael/codec-features.html to get a precise comparison of the FFmpeg MPEG-4 codec compared to other implementations.

5.4 Audio Codecs

X means that encoding (resp. decoding) is supported.

I means that an integer-only version is available, too (ensures high performance on systems without hardware floating point support).

6. Platform Specific information

6.1 Linux

FFmpeg should be compiled with at least GCC 2.95.3. GCC 3.2 is the preferred compiler now for FFmpeg. All future optimizations will depend on features only found in GCC 3.2.

6.2 BSD

BSD make will not build FFmpeg, you need to install and use GNU Make (`gmake').

6.3 Windows

6.3.1 Native Windows compilation

• Install the current versions of MSYS and MinGW from http://www.mingw.org/. You can find detailed installation instructions in the download section and the FAQ.
• If you want to test the FFplay, also download the MinGW development library of SDL 1.2.x (`SDL-devel-1.2.x-mingw32.tar.gz') from http://www.libsdl.org. Unpack it in a temporary directory, and unpack the archive `i386-mingw32msvc.tar.gz' in the MinGW tool directory. Edit the `sdl-config' script so that it gives the correct SDL directory when invoked.
• Extract the current version of FFmpeg.
• Start the MSYS shell (file `msys.bat').
• Change to the FFmpeg directory and follow the instructions of how to compile FFmpeg (file `INSTALL'). Usually, launching `./configure' and `make' suffices. If you have problems using SDL, verify that `sdl-config' can be launched from the MSYS command line.
• You can install FFmpeg in `Program Files/FFmpeg' by typing `make install'. Don't forget to copy `SDL.dll' to the place you launch `ffplay' from.

Notes:

• The target `make wininstaller' can be used to create a Nullsoft based Windows installer for FFmpeg and FFplay. `SDL.dll' must be copied to the FFmpeg directory in order to build the installer.
• By using ./configure --enable-shared when configuring FFmpeg, you can build `avcodec.dll' and `avformat.dll'. With make install you install the FFmpeg DLLs and the associated headers in `Program Files/FFmpeg'.
• Visual C++ compatibility: If you used ./configure --enable-shared when configuring FFmpeg, FFmpeg tries to use the Microsoft Visual C++ lib tool to build avcodec.lib and avformat.lib. With these libraries you can link your Visual C++ code directly with the FFmpeg DLLs (see below).

6.3.2 Visual C++ compatibility

FFmpeg will not compile under Visual C++ - and it has too many dependencies on the GCC compiler to make a port viable. However, if you want to use the FFmpeg libraries in your own applications, you can still compile those applications using Visual C++. An important restriction to this is that you have to use the dynamically linked versions of the FFmpeg libraries (i.e. the DLLs), and you have to make sure that Visual-C++-compatible import libraries are created during the FFmpeg build process.

This description of how to use the FFmpeg libraries with Visual C++ is based on Visual C++ 2005 Express Edition Beta 2. If you have a different version, you might have to modify the procedures slightly.

Here are the step-by-step instructions for building the FFmpeg libraries so they can be used with Visual C++:

1. Install Visual C++ (if you haven't done so already).
2. Install MinGW and MSYS as described above.
3. Add a call to `vcvars32.bat' (which sets up the environment variables for the Visual C++ tools) as the first line of `msys.bat'. The standard location for `vcvars32.bat' is `C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat', and the standard location for `msys.bat' is `C:\msys\1.0\msys.bat'. If this corresponds to your setup, add the following line as the first line of `msys.bat':

   call "C:\Program Files\Microsoft Visual Studio 8\VC\bin\vcvars32.bat"

4. Start the MSYS shell (file `msys.bat') and type link.exe. If you get a help message with the command line options of link.exe, this means your environment variables are set up correctly, the Microsoft linker is on the path and will be used by FFmpeg to create Visual-C++-compatible import libraries.
5. Extract the current version of FFmpeg and change to the FFmpeg directory.
6. Type the command ./configure --enable-shared --disable-static --enable-memalign-hack to configure and, if that didn't produce any errors, type make to build FFmpeg.
7. The subdirectories `libavformat', `libavcodec', and `libavutil' should now contain the files `avformat.dll', `avformat.lib', `avcodec.dll', `avcodec.lib', `avutil.dll', and `avutil.lib', respectively. Copy the three DLLs to your System32 directory (typically `C:\Windows\System32').

And here is how to use these libraries with Visual C++:

1. Create a new console application ("File / New / Project") and then select "Win32 Console Application". On the appropriate page of the Application Wizard, uncheck the "Precompiled headers" option.
2. Write the source code for your application, or, for testing, just copy the code from an existing sample application into the source file that Visual C++ has already created for you. (Note that your source filehas to have a .cpp extension; otherwise, Visual C++ won't compile the FFmpeg headers correctly because in C mode, it doesn't recognize the inline keyword.) For example, you can copy `output_example.c' from the FFmpeg distribution (but you will have to make minor modifications so the code will compile under C++, see below).
3. Open the "Project / Properties" dialog box. In the "Configuration" combo box, select "All Configurations" so that the changes you make will affect both debug and release builds. In the tree view on the left hand side, select "C/C++ / General", then edit the "Additional Include Directories" setting to contain the complete paths to the `libavformat', `libavcodec', and `libavutil' subdirectories of your FFmpeg directory. Note that the directories have to be separated using semicolons. Now select "Linker / General" from the tree view and edit the "Additional Library Directories" setting to contain the same three directories.
4. Still in the "Project / Properties" dialog box, select "Linker / Input" from the tree view, then add the files `avformat.lib', `avcodec.lib', and `avutil.lib' to the end of the "Additional Dependencies". Note that the names of the libraries have to be separated using spaces.
5. Now, select "C/C++ / Code Generation" from the tree view. Select "Debug" in the "Configuration" combo box. Make sure that "Runtime Library" is set to "Multi-threaded Debug DLL". Then, select "Release" in the "Configuration" combo box and make sure that "Runtime Library" is set to "Multi-threaded DLL".
6. Click "OK" to close the "Project / Properties" dialog box and build the application. Hopefully, it should compile and run cleanly. If you used `output_example.c' as your sample application, you will get a few compiler errors, but they are easy to fix. The first type of error occurs because Visual C++ doesn't allow an int to be converted to an enum without a cast. To solve the problem, insert the required casts (this error occurs once for a CodecID and once for a CodecType). The second type of error occurs because C++ requires the return value of malloc to be cast to the exact type of the pointer it is being assigned to. Visual C++ will complain that, for example, (void *) is being assigned to (uint8_t *) without an explicit cast. So insert an explicit cast in these places to silence the compiler. The third type of error occurs because the snprintf library function is called _snprintf under Visual C++. So just add an underscore to fix the problem. With these changes, `output_example.c' should compile under Visual C++, and the resulting executable should produce valid video files.

6.3.3 Cross compilation for Windows with Linux

You must use the MinGW cross compilation tools available at http://www.mingw.org/.

Then configure FFmpeg with the following options:

./configure --enable-mingw32 --cross-prefix=i386-mingw32msvc-

(you can change the cross-prefix according to the prefix chosen for the MinGW tools).

Then you can easily test FFmpeg with Wine (http://www.winehq.com/).

6.4 Mac OS X

6.5 BeOS

The configure script should guess the configuration itself. Networking support is currently not finished. errno issues fixed by Andrew Bachmann.

Old stuff:

François Revol - revol at free dot fr - April 2002

The configure script should guess the configuration itself, however I still didn't test building on the net_server version of BeOS.

FFserver is broken (needs poll() implementation).

There are still issues with errno codes, which are negative in BeOS, and that FFmpeg negates when returning. This ends up turning errors into valid results, then crashes. (To be fixed)

7. Developers Guide

7.1 API

• libavcodec is the library containing the codecs (both encoding and decoding). Look at `libavcodec/apiexample.c' to see how to use it.
• libavformat is the library containing the file format handling (mux and demux code for several formats). Look at `ffplay.c' to use it in a player. See `output_example.c' to use it to generate audio or video streams.

7.2 Integrating libavcodec or libavformat in your program

You can integrate all the source code of the libraries to link them statically to avoid any version problem. All you need is to provide a 'config.mak' and a 'config.h' in the parent directory. See the defines generated by ./configure to understand what is needed.

You can use libavcodec or libavformat in your commercial program, but any patch you make must be published. The best way to proceed is to send your patches to the FFmpeg mailing list.

7.3 Coding Rules

FFmpeg is programmed in the ISO C90 language with a few additional features from ISO C99, namely:

• the `inline' keyword;
• `//' comments;
• designated struct initializers (`struct s x = { .i = 17 };')
• compound literals (`x = (struct s) { 17, 23 };')

These features are supported by all compilers we care about, so we won't accept patches to remove their use unless they absolutely don't impair clarity and performance.

All code must compile with GCC 2.95 and GCC 3.3. Currently, FFmpeg also compiles with several other compilers, such as the Compaq ccc compiler or Sun Studio 9, and we would like to keep it that way unless it would be exceedingly involved. To ensure compatibility, please don't use any additional C99 features or GCC extensions. Especially watch out for:

• mixing statements and declarations;
• `long long' (use `int64_t' instead);
• `__attribute__' not protected by `#ifdef __GNUC__' or similar;
• GCC statement expressions (`(x = ({ int y = 4; y; })').

Indent size is 4. The presentation is the one specified by 'indent -i4 -kr -nut'. The TAB character is forbidden outside of Makefiles as is any form of trailing whitespace. Commits containing either will be rejected by the Subversion repository.

Main priority in FFmpeg is simplicity and small code size (=less bugs).

Comments: Use the JavaDoc/Doxygen format (see examples below) so that code documentation can be generated automatically. All nontrivial functions should have a comment above them explaining what the function does, even if it's just one sentence. All structures and their member variables should be documented, too.

/**
 * @file mpeg.c
 * MPEG codec.
 * @author ...
 */

/**
 * Summary sentence.
 * more text ...
 * ...
 */
typedef struct Foobar{
    int var1; /**< var1 description */
    int var2; ///< var2 description
    /** var3 description */
    int var3;
} Foobar;

/**
 * Summary sentence.
 * more text ...
 * ...
 * @param my_parameter description of my_parameter
 * @return return value description
 */
int myfunc(int my_parameter)
...

fprintf and printf are forbidden in libavformat and libavcodec, please use av_log() instead.

7.4 Development Policy

1. You must not commit code which breaks FFmpeg! (Meaning unfinished but enabled code which breaks compilation or compiles but does not work or breaks the regression tests) You can commit unfinished stuff (for testing etc), but it must be disabled (#ifdef etc) by default so it does not interfere with other developers' work.
2. You don't have to over-test things. If it works for you, and you think it should work for others, then commit. If your code has problems (portability, triggers compiler bugs, unusual environment etc) they will be reported and eventually fixed.
3. Do not commit unrelated changes together, split them into self-contained pieces.
4. Do not change behavior of the program (renaming options etc) without first discussing it on the ffmpeg-devel mailing list. Do not remove functionality from the code. Just improve!

   Note: Redundant code can be removed.

5. Do not commit changes to the build system (Makefiles, configure script) which change behavior, defaults etc, without asking first. The same applies to compiler warning fixes, trivial looking fixes and to code maintained by other developers. We usually have a reason for doing things the way we do. Send your changes as patches to the ffmpeg-devel mailing list, and if the code maintainers say OK, you may commit. This does not apply to files you wrote and/or maintain.
6. We refuse source indentation and other cosmetic changes if they are mixed with functional changes, such commits will be rejected and removed. Every developer has his own indentation style, you should not change it. Of course if you (re)write something, you can use your own style, even though we would prefer if the indentation throughout FFmpeg was consistent (Many projects force a given indentation style - we don't.). If you really need to make indentation changes (try to avoid this), separate them strictly from real changes.

   NOTE: If you had to put if(){ .. } over a large (> 5 lines) chunk of code, then either do NOT change the indentation of the inner part within (don't move it to the right)! or do so in a separate commit

7. Always fill out the commit log message. Describe in a few lines what you changed and why. You can refer to mailing list postings if you fix a particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
8. If you apply a patch by someone else, include the name and email address in the log message. Since the ffmpeg-cvslog mailing list is publicly archived you should add some SPAM protection to the email address. Send an answer to ffmpeg-devel (or wherever you got the patch from) saying that you applied the patch.
9. Do NOT commit to code actively maintained by others without permission. Send a patch to ffmpeg-devel instead.
10. Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits are sent there and reviewed by all the other developers. Bugs and possible improvements or general questions regarding commits are discussed there. We expect you to react if problems with your code are uncovered.
11. Update the documentation if you change behavior or add features. If you are unsure how best to do this, send a patch to ffmpeg-devel, the documentation maintainer(s) will review and commit your stuff.
12. Never write to unallocated memory, never write over the end of arrays, always check values read from some untrusted source before using them as array index or other risky things.
13. Remember to check if you need to bump versions for the specific libav parts (libavutil, libavcodec, libavformat) you are changing. You need to change the version integer and the version string. Incrementing the first component means no backward compatibility to previous versions (e.g. removal of a function). Incrementing the second component means backward compatible change (e.g. addition of a function). Incrementing the third component means a noteworthy binary compatible change (e.g. encoder bug fix that matters for the decoder).
14. If you add a new codec, remember to update the changelog, add it to the supported codecs table in the documentation and bump the second component of the `libavcodec' version number appropriately. If it has a fourcc, add it to `libavformat/avienc.c', even if it is only a decoder.

We think our rules are not too hard. If you have comments, contact us.

Note, these rules are mostly borrowed from the MPlayer project.

7.4.1 Renaming/moving files or content of files

You CANNOT do that. Post a request for such a change to the mailing list Do NOT remove & readd a file - it will kill the changelog!!!!

7.5 Submitting patches

First, (see section Coding Rules) above if you didn't yet.

When you submit your patch, try to send a unified diff (diff '-up' option). I cannot read other diffs :-)

Also please do not submit patches which contain several unrelated changes. Split them into individual self-contained patches; this makes reviewing them much easier.

Run the regression tests before submitting a patch so that you can verify that there are no big problems.

Patches should be posted as base64 encoded attachments (or any other encoding which ensures that the patch won't be trashed during transmission) to the ffmpeg-devel mailing list, see http://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel

It also helps quite a bit if you tell us what the patch does (for example 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant and has no lrint()')

We reply to all submitted patches and either apply or reject with some explanation why, but sometimes we are quite busy so it can take a week or two.

7.6 Regression tests

Before submitting a patch (or committing to the repository), you should at least test that you did not break anything.

The regression tests build a synthetic video stream and a synthetic audio stream. These are then encoded and decoded with all codecs or formats. The CRC (or MD5) of each generated file is recorded in a result file. A 'diff' is launched to compare the reference results and the result file.

The regression tests then go on to test the FFserver code with a limited set of streams. It is important that this step runs correctly as well.

Run 'make test' to test all the codecs and formats.

Run 'make fulltest' to test all the codecs, formats and FFserver.

[Of course, some patches may change the results of the regression tests. In this case, the reference results of the regression tests shall be modified accordingly].

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