From 6cad635bd887634b1908c61ef6039420890b2ba2 Mon Sep 17 00:00:00 2001 From: Fog Date: Sat, 17 Jan 2015 13:47:07 -0500 Subject: [PATCH 1/2] Use ffmpeg for Windows Video Dumping instead of VFW --- Externals/ffmpeg/dev/README.txt | 111 + Externals/ffmpeg/dev/doc/developer.html | 777 + Externals/ffmpeg/dev/doc/examples/Makefile | 44 + Externals/ffmpeg/dev/doc/examples/README | 23 + .../ffmpeg/dev/doc/examples/avio_reading.c | 134 + .../dev/doc/examples/decoding_encoding.c | 665 + .../dev/doc/examples/demuxing_decoding.c | 386 + .../ffmpeg/dev/doc/examples/extract_mvs.c | 185 + .../ffmpeg/dev/doc/examples/filter_audio.c | 365 + .../ffmpeg/dev/doc/examples/filtering_audio.c | 280 + .../ffmpeg/dev/doc/examples/filtering_video.c | 262 + Externals/ffmpeg/dev/doc/examples/metadata.c | 56 + Externals/ffmpeg/dev/doc/examples/muxing.c | 670 + Externals/ffmpeg/dev/doc/examples/remuxing.c | 165 + .../dev/doc/examples/resampling_audio.c | 214 + .../ffmpeg/dev/doc/examples/scaling_video.c | 140 + 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445440 bytes Externals/ffmpeg/shared/bin/ffmpeg.exe | Bin 0 -> 332288 bytes Externals/ffmpeg/shared/bin/ffplay.exe | Bin 0 -> 546304 bytes Externals/ffmpeg/shared/bin/ffprobe.exe | Bin 0 -> 163328 bytes Externals/ffmpeg/shared/bin/postproc-53.dll | Bin 0 -> 129536 bytes Externals/ffmpeg/shared/bin/swresample-1.dll | Bin 0 -> 292864 bytes Externals/ffmpeg/shared/bin/swscale-3.dll | Bin 0 -> 468480 bytes Externals/ffmpeg/shared/doc/developer.html | 777 + Externals/ffmpeg/shared/doc/examples/Makefile | 44 + Externals/ffmpeg/shared/doc/examples/README | 23 + .../ffmpeg/shared/doc/examples/avio_reading.c | 134 + .../shared/doc/examples/decoding_encoding.c | 665 + .../shared/doc/examples/demuxing_decoding.c | 386 + .../ffmpeg/shared/doc/examples/extract_mvs.c | 185 + .../ffmpeg/shared/doc/examples/filter_audio.c | 365 + .../shared/doc/examples/filtering_audio.c | 280 + .../shared/doc/examples/filtering_video.c | 262 + .../ffmpeg/shared/doc/examples/metadata.c | 56 + Externals/ffmpeg/shared/doc/examples/muxing.c | 670 + .../ffmpeg/shared/doc/examples/remuxing.c | 165 + .../shared/doc/examples/resampling_audio.c | 214 + .../shared/doc/examples/scaling_video.c | 140 + .../shared/doc/examples/transcode_aac.c | 755 + .../ffmpeg/shared/doc/examples/transcoding.c | 583 + Externals/ffmpeg/shared/doc/faq.html | 719 + Externals/ffmpeg/shared/doc/fate.html | 286 + Externals/ffmpeg/shared/doc/ffmpeg-all.html | 27303 ++++++++++++++++ .../shared/doc/ffmpeg-bitstream-filters.html | 261 + .../ffmpeg/shared/doc/ffmpeg-codecs.html | 4474 +++ .../ffmpeg/shared/doc/ffmpeg-devices.html | 1810 + .../ffmpeg/shared/doc/ffmpeg-filters.html | 13382 ++++++++ .../ffmpeg/shared/doc/ffmpeg-formats.html | 2311 ++ .../ffmpeg/shared/doc/ffmpeg-protocols.html | 1545 + .../ffmpeg/shared/doc/ffmpeg-resampler.html | 357 + .../ffmpeg/shared/doc/ffmpeg-scaler.html | 231 + Externals/ffmpeg/shared/doc/ffmpeg-utils.html | 1468 + Externals/ffmpeg/shared/doc/ffmpeg.html | 2109 ++ Externals/ffmpeg/shared/doc/ffplay-all.html | 21308 ++++++++++++ Externals/ffmpeg/shared/doc/ffplay.html | 745 + Externals/ffmpeg/shared/doc/ffprobe-all.html | 21676 ++++++++++++ Externals/ffmpeg/shared/doc/ffprobe.html | 1113 + Externals/ffmpeg/shared/doc/general.html | 986 + Externals/ffmpeg/shared/doc/git-howto.html | 493 + Externals/ffmpeg/shared/doc/libavcodec.html | 76 + Externals/ffmpeg/shared/doc/libavdevice.html | 73 + Externals/ffmpeg/shared/doc/libavfilter.html | 72 + Externals/ffmpeg/shared/doc/libavformat.html | 76 + Externals/ffmpeg/shared/doc/libavutil.html | 95 + .../ffmpeg/shared/doc/libswresample.html | 95 + Externals/ffmpeg/shared/doc/libswscale.html | 89 + Externals/ffmpeg/shared/doc/nut.html | 211 + Externals/ffmpeg/shared/doc/platform.html | 447 + Externals/ffmpeg/shared/ff-prompt.bat | 35 + Externals/ffmpeg/shared/licenses/bzip2.txt | 42 + .../ffmpeg/shared/licenses/fontconfig.txt | 27 + Externals/ffmpeg/shared/licenses/freetype.txt | 169 + 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19 + .../shared/presets/libvpx-720p50_60.ffpreset | 19 + Source/Core/DolphinQt/DolphinQt.vcxproj | 2 +- Source/Core/DolphinWX/DolphinWX.vcxproj | 6 +- Source/Core/DolphinWX/VideoConfigDiag.cpp | 10 +- Source/Core/VideoBackends/D3D/Render.cpp | 15 +- Source/Core/VideoBackends/D3D/Render.h | 1 + Source/Core/VideoBackends/OGL/Render.cpp | 63 +- Source/Core/VideoCommon/AVIDump.cpp | 322 +- Source/Core/VideoCommon/AVIDump.h | 4 - Source/Core/VideoCommon/CMakeLists.txt | 6 +- Source/Core/VideoCommon/RenderBase.cpp | 5 - Source/Core/VideoCommon/RenderBase.h | 3 +- Source/Core/VideoCommon/VideoCommon.vcxproj | 20 + Source/UnitTests/UnitTests.vcxproj | 2 +- 313 files changed, 260584 insertions(+), 379 deletions(-) create mode 100644 Externals/ffmpeg/dev/README.txt create mode 100644 Externals/ffmpeg/dev/doc/developer.html create mode 100644 Externals/ffmpeg/dev/doc/examples/Makefile create mode 100644 Externals/ffmpeg/dev/doc/examples/README create mode 100644 Externals/ffmpeg/dev/doc/examples/avio_reading.c create mode 100644 Externals/ffmpeg/dev/doc/examples/decoding_encoding.c create mode 100644 Externals/ffmpeg/dev/doc/examples/demuxing_decoding.c create mode 100644 Externals/ffmpeg/dev/doc/examples/extract_mvs.c create mode 100644 Externals/ffmpeg/dev/doc/examples/filter_audio.c create mode 100644 Externals/ffmpeg/dev/doc/examples/filtering_audio.c create mode 100644 Externals/ffmpeg/dev/doc/examples/filtering_video.c create mode 100644 Externals/ffmpeg/dev/doc/examples/metadata.c create mode 100644 Externals/ffmpeg/dev/doc/examples/muxing.c create mode 100644 Externals/ffmpeg/dev/doc/examples/remuxing.c create mode 100644 Externals/ffmpeg/dev/doc/examples/resampling_audio.c create mode 100644 Externals/ffmpeg/dev/doc/examples/scaling_video.c create mode 100644 Externals/ffmpeg/dev/doc/examples/transcode_aac.c create mode 100644 Externals/ffmpeg/dev/doc/examples/transcoding.c create mode 100644 Externals/ffmpeg/dev/doc/faq.html create 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mode 100644 Externals/ffmpeg/shared/presets/ffprobe.xsd create mode 100644 Externals/ffmpeg/shared/presets/libvpx-1080p.ffpreset create mode 100644 Externals/ffmpeg/shared/presets/libvpx-1080p50_60.ffpreset create mode 100644 Externals/ffmpeg/shared/presets/libvpx-360p.ffpreset create mode 100644 Externals/ffmpeg/shared/presets/libvpx-720p.ffpreset create mode 100644 Externals/ffmpeg/shared/presets/libvpx-720p50_60.ffpreset diff --git a/Externals/ffmpeg/dev/README.txt b/Externals/ffmpeg/dev/README.txt new file mode 100644 index 0000000000..a82159f6d8 --- /dev/null +++ b/Externals/ffmpeg/dev/README.txt @@ -0,0 +1,111 @@ +This is a FFmpeg Win64 shared build by Kyle Schwarz. + +Zeranoe's FFmpeg Builds Home Page: + +This build was compiled on: , at: 22:19:40 + +FFmpeg version: 2015-01-15 git-cd960c8 + libavutil 54. 16.100 / 54. 16.100 + libavcodec 56. 20.100 / 56. 20.100 + libavformat 56. 18.101 / 56. 18.101 + libavdevice 56. 4.100 / 56. 4.100 + libavfilter 5. 7.101 / 5. 7.101 + libswscale 3. 1.101 / 3. 1.101 + libswresample 1. 1.100 / 1. 1.100 + libpostproc 53. 3.100 / 53. 3.100 + +This FFmpeg build was configured with: + --disable-static + --enable-shared + --enable-gpl + --enable-version3 + --disable-w32threads + --enable-avisynth + --enable-bzlib + --enable-fontconfig + --enable-frei0r + --enable-gnutls + --enable-iconv + --enable-libass + --enable-libbluray + --enable-libbs2b + --enable-libcaca + --enable-libfreetype + --enable-libgme + --enable-libgsm + --enable-libilbc + --enable-libmodplug + --enable-libmp3lame + --enable-libopencore-amrnb + --enable-libopencore-amrwb + --enable-libopenjpeg + --enable-libopus + --enable-librtmp + --enable-libschroedinger + --enable-libsoxr + --enable-libspeex + --enable-libtheora + --enable-libtwolame + --enable-libvidstab + --enable-libvo-aacenc + --enable-libvo-amrwbenc + --enable-libvorbis + --enable-libvpx + --enable-libwavpack + --enable-libwebp + --enable-libx264 + --enable-libx265 + --enable-libxavs + --enable-libxvid + --enable-lzma + --enable-decklink + --enable-zlib + +This build was compiled with the following external libraries: + bzip2 1.0.6 + Fontconfig 2.11.91 + Frei0r 20130909-git-10d8360 + GnuTLS 3.2.20 + libiconv 1.14 + libass 0.12.0 + libbluray 0.6.2 + libbs2b 3.1.0 + libcaca 0.99.beta18 + FreeType 2.5.4 + Game Music Emu 0.6.0 + GSM 1.0.13-4 + iLBC 20141214-git-ef04ebe + Modplug-XMMS 0.8.8.5 + LAME 3.99.5 + OpenCORE AMR 0.1.3 + OpenJPEG 1.5.2 + Opus 1.1 + RTMPDump 20140707-git-a1900c3 + Schroedinger 1.0.11 + libsoxr 0.1.1 + Speex 1.2rc2 + Theora 1.1.1 + TwoLAME 0.3.13 + vid.stab 0.98 + VisualOn AAC 0.1.3 + VisualOn AMR-WB 0.1.2 + Vorbis 1.3.4 + vpx 1.3.0 + WavPack 4.70.0 + WebP 0.4.2 + x264 20141220-git-40bb568 + x265 1.4 + XAVS svn-r55 + Xvid 1.3.3 + XZ Utils 5.2.0 + zlib 1.2.8 + +The source code for this FFmpeg build can be found at: + +This build was compiled on Debian 8.0 (64-bit): + +GCC 4.9.2 was used to compile this FFmpeg build: + +This build was compiled using the MinGW-w64 toolchain: + +Licenses for each library can be found in the 'licenses' folder. diff --git a/Externals/ffmpeg/dev/doc/developer.html b/Externals/ffmpeg/dev/doc/developer.html new file mode 100644 index 0000000000..ac67aae49a --- /dev/null +++ b/Externals/ffmpeg/dev/doc/developer.html @@ -0,0 +1,777 @@ + + + + + + + Developer Documentation + + + + + + +
+

+ Developer Documentation +

+
+
+ + + + +

Table of Contents

+ +
+ + +
+ + + +

1 Developers Guide

+ + +

1.1 Notes for external developers

+ +

This document is mostly useful for internal FFmpeg developers. +External developers who need to use the API in their application should +refer to the API doxygen documentation in the public headers, and +check the examples in doc/examples and in the source code to +see how the public API is employed. +

+

You can use the FFmpeg libraries in your commercial program, but you +are encouraged to publish any patch you make. In this case the +best way to proceed is to send your patches to the ffmpeg-devel +mailing list following the guidelines illustrated in the remainder of +this document. +

+

For more detailed legal information about the use of FFmpeg in +external programs read the LICENSE file in the source tree and +consult http://ffmpeg.org/legal.html. +

+ +

1.2 Contributing

+ +

There are 3 ways by which code gets into ffmpeg. +

    +
  • Submitting Patches to the main developer mailing list + see Submitting patches for details. +
  • Directly committing changes to the main tree. +
  • Committing changes to a git clone, for example on github.com or + gitorious.org. And asking us to merge these changes. +
+ +

Whichever way, changes should be reviewed by the maintainer of the code +before they are committed. And they should follow the Coding Rules. +The developer making the commit and the author are responsible for their changes +and should try to fix issues their commit causes. +

+ +

1.3 Coding Rules

+ + +

1.3.1 Code formatting conventions

+ +

There are the following guidelines regarding the indentation in files: +

+
    +
  • Indent size is 4. + +
  • The TAB character is forbidden outside of Makefiles as is any +form of trailing whitespace. Commits containing either will be +rejected by the git repository. + +
  • You should try to limit your code lines to 80 characters; however, do so if +and only if this improves readability. +
+

The presentation is one inspired by ’indent -i4 -kr -nut’. +

+

The main priority in FFmpeg is simplicity and small code size in order to +minimize the bug count. +

+ +

1.3.2 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 is just one sentence. +All structures and their member variables should be documented, too. +

+

Avoid Qt-style and similar Doxygen syntax with ! in it, i.e. replace +//! with /// and similar. Also @ syntax should be employed +for markup commands, i.e. use @param and not \param. +

+
+
/**
+ * @file
+ * 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)
+...
+
+ + +

1.3.3 C language features

+ +

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 will not +accept patches to remove their use unless they absolutely do not impair +clarity and performance. +

+

All code must compile with recent versions of GCC and a number of other +currently supported compilers. To ensure compatibility, please do not use +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; })’). +
+ + +

1.3.4 Naming conventions

+

All names should be composed with underscores (_), not CamelCase. For example, +‘avfilter_get_video_buffer’ is an acceptable function name and +‘AVFilterGetVideo’ is not. The exception from this are type names, like +for example structs and enums; they should always be in the CamelCase +

+

There are the following conventions for naming variables and functions: +

+
    +
  • For local variables no prefix is required. + +
  • For file-scope variables and functions declared as static, no prefix +is required. + +
  • For variables and functions visible outside of file scope, but only used +internally by a library, an ff_ prefix should be used, +e.g. ‘ff_w64_demuxer’. + +
  • For variables and functions visible outside of file scope, used internally +across multiple libraries, use avpriv_ as prefix, for example, +‘avpriv_aac_parse_header’. + +
  • Each library has its own prefix for public symbols, in addition to the +commonly used av_ (avformat_ for libavformat, +avcodec_ for libavcodec, swr_ for libswresample, etc). +Check the existing code and choose names accordingly. +Note that some symbols without these prefixes are also exported for +retro-compatibility reasons. These exceptions are declared in the +lib<name>/lib<name>.v files. +
+ +

Furthermore, name space reserved for the system should not be invaded. +Identifiers ending in _t are reserved by +POSIX. +Also avoid names starting with __ or _ followed by an uppercase +letter as they are reserved by the C standard. Names starting with _ +are reserved at the file level and may not be used for externally visible +symbols. If in doubt, just avoid names starting with _ altogether. +

+ +

1.3.5 Miscellaneous conventions

+ +
    +
  • fprintf and printf are forbidden in libavformat and libavcodec, +please use av_log() instead. + +
  • Casts should be used only when necessary. Unneeded parentheses +should also be avoided if they don’t make the code easier to understand. +
+ + +

1.3.6 Editor configuration

+

In order to configure Vim to follow FFmpeg formatting conventions, paste +the following snippet into your .vimrc: +

+
" indentation rules for FFmpeg: 4 spaces, no tabs
+set expandtab
+set shiftwidth=4
+set softtabstop=4
+set cindent
+set cinoptions=(0
+" Allow tabs in Makefiles.
+autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
+" Trailing whitespace and tabs are forbidden, so highlight them.
+highlight ForbiddenWhitespace ctermbg=red guibg=red
+match ForbiddenWhitespace /\s\+$\|\t/
+" Do not highlight spaces at the end of line while typing on that line.
+autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@<!$/
+
+ +

For Emacs, add these roughly equivalent lines to your .emacs.d/init.el: +

+
(c-add-style "ffmpeg"
+             '("k&r"
+               (c-basic-offset . 4)
+               (indent-tabs-mode . nil)
+               (show-trailing-whitespace . t)
+               (c-offsets-alist
+                (statement-cont . (c-lineup-assignments +)))
+               )
+             )
+(setq c-default-style "ffmpeg")
+
+ + +

1.4 Development Policy

+ +
    +
  1. Contributions should be licensed under the +LGPL 2.1, +including an "or any later version" clause, or, if you prefer +a gift-style license, the +ISC or +MIT license. +GPL 2 including +an "or any later version" clause is also acceptable, but LGPL is +preferred. +If you add a new file, give it a proper license header. Do not copy and +paste it from a random place, use an existing file as template. + +
  2. 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. + +
  3. The commit message should have a short first line in the form of +a ‘topic: short description’ as a header, separated by a newline +from the body consisting of an explanation of why the change is necessary. +If the commit fixes a known bug on the bug tracker, the commit message +should include its bug ID. Referring to the issue on the bug tracker does +not exempt you from writing an excerpt of the bug in the commit message. + +
  4. You do not 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. + +
  5. Do not commit unrelated changes together, split them into self-contained +pieces. Also do not forget that if part B depends on part A, but A does not +depend on B, then A can and should be committed first and separate from B. +Keeping changes well split into self-contained parts makes reviewing and +understanding them on the commit log mailing list easier. This also helps +in case of debugging later on. +Also if you have doubts about splitting or not splitting, do not hesitate to +ask/discuss it on the developer mailing list. + +
  6. Do not change behavior of the programs (renaming options etc) or public +API or ABI 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. +

    +
  7. 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. + +
  8. 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 do not.). 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 (do not +move it to the right)! or do so in a separate commit +

    +
  9. 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. +Recommended format: + +
    +
    area changed: Short 1 line description
+
+details describing what and why and giving references.
+
    + +
  10. Make sure the author of the commit is set correctly. (see git commit –author) +If you apply a patch, send an +answer to ffmpeg-devel (or wherever you got the patch from) saying that +you applied the patch. + +
  11. When applying patches that have been discussed (at length) on the mailing +list, reference the thread in the log message. + +
  12. Do NOT commit to code actively maintained by others without permission. +Send a patch to ffmpeg-devel instead. If no one answers within a reasonable +timeframe (12h for build failures and security fixes, 3 days small changes, +1 week for big patches) then commit your patch if you think it is OK. +Also note, the maintainer can simply ask for more time to review! + +
  13. 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. + +
  14. 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. + +
  15. Try to keep important discussions and requests (also) on the public +developer mailing list, so that all developers can benefit from them. + +
  16. 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. + +
  17. 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. +Incrementing the first component means no backward compatibility to +previous versions (e.g. removal of a function from the public API). +Incrementing the second component means backward compatible change +(e.g. addition of a function to the public API or extension of an +existing data structure). +Incrementing the third component means a noteworthy binary compatible +change (e.g. encoder bug fix that matters for the decoder). The third +component always starts at 100 to distinguish FFmpeg from Libav. + +
  18. Compiler warnings indicate potential bugs or code with bad style. If a type of +warning always points to correct and clean code, that warning should +be disabled, not the code changed. +Thus the remaining warnings can either be bugs or correct code. +If it is a bug, the bug has to be fixed. If it is not, the code should +be changed to not generate a warning unless that causes a slowdown +or obfuscates the code. + +
  19. Make sure that no parts of the codebase that you maintain are missing from the +MAINTAINERS file. If something that you want to maintain is missing add it with +your name after it. +If at some point you no longer want to maintain some code, then please help +finding a new maintainer and also don’t forget updating the MAINTAINERS file. +
+ +

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

+ +

1.5 Submitting patches

+ +

First, read the Coding Rules above if you did not yet, in particular +the rules regarding patch submission. +

+

When you submit your patch, please use git format-patch or +git send-email. We cannot read other diffs :-) +

+

Also please do not submit a patch which contains several unrelated changes. +Split it into separate, self-contained pieces. This does not mean splitting +file by file. Instead, make the patch as small as possible while still +keeping it as a logical unit that contains an individual change, even +if it spans multiple files. This makes reviewing your patches much easier +for us and greatly increases your chances of getting your patch applied. +

+

Use the patcheck tool of FFmpeg to check your patch. +The tool is located in the tools directory. +

+

Run the Regression tests before submitting a patch in order to verify +it does not cause unexpected problems. +

+

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()’) +

+

Also please if you send several patches, send each patch as a separate mail, +do not attach several unrelated patches to the same mail. +

+

Patches should be posted to the +ffmpeg-devel +mailing list. Use git send-email when possible since it will properly +send patches without requiring extra care. If you cannot, then send patches +as base64-encoded attachments, so your patch is not trashed during +transmission. +

+

Your patch will be reviewed on the mailing list. You will likely be asked +to make some changes and are expected to send in an improved version that +incorporates the requests from the review. This process may go through +several iterations. Once your patch is deemed good enough, some developer +will pick it up and commit it to the official FFmpeg tree. +

+

Give us a few days to react. But if some time passes without reaction, +send a reminder by email. Your patch should eventually be dealt with. +

+ + +

1.6 New codecs or formats checklist

+ +
    +
  1. Did you use av_cold for codec initialization and close functions? + +
  2. Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or +AVInputFormat/AVOutputFormat struct? + +
  3. Did you bump the minor version number (and reset the micro version +number) in libavcodec/version.h or libavformat/version.h? + +
  4. Did you register it in allcodecs.c or allformats.c? + +
  5. Did you add the AVCodecID to avcodec.h? +When adding new codec IDs, also add an entry to the codec descriptor +list in libavcodec/codec_desc.c. + +
  6. If it has a FourCC, did you add it to libavformat/riff.c, +even if it is only a decoder? + +
  7. Did you add a rule to compile the appropriate files in the Makefile? +Remember to do this even if you’re just adding a format to a file that is +already being compiled by some other rule, like a raw demuxer. + +
  8. Did you add an entry to the table of supported formats or codecs in +doc/general.texi? + +
  9. Did you add an entry in the Changelog? + +
  10. If it depends on a parser or a library, did you add that dependency in +configure? + +
  11. Did you git add the appropriate files before committing? + +
  12. Did you make sure it compiles standalone, i.e. with +configure --disable-everything --enable-decoder=foo +(or --enable-demuxer or whatever your component is)? +
+ + + +

1.7 patch submission checklist

+ +
    +
  1. Does make fate pass with the patch applied? + +
  2. Was the patch generated with git format-patch or send-email? + +
  3. Did you sign off your patch? (git commit -s) +See http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches for the meaning +of sign off. + +
  4. Did you provide a clear git commit log message? + +
  5. Is the patch against latest FFmpeg git master branch? + +
  6. Are you subscribed to ffmpeg-devel? +(the list is subscribers only due to spam) + +
  7. Have you checked that the changes are minimal, so that the same cannot be +achieved with a smaller patch and/or simpler final code? + +
  8. If the change is to speed critical code, did you benchmark it? + +
  9. If you did any benchmarks, did you provide them in the mail? + +
  10. Have you checked that the patch does not introduce buffer overflows or +other security issues? + +
  11. Did you test your decoder or demuxer against damaged data? If no, see +tools/trasher, the noise bitstream filter, and +zzuf. Your decoder or demuxer +should not crash, end in a (near) infinite loop, or allocate ridiculous +amounts of memory when fed damaged data. + +
  12. Does the patch not mix functional and cosmetic changes? + +
  13. Did you add tabs or trailing whitespace to the code? Both are forbidden. + +
  14. Is the patch attached to the email you send? + +
  15. Is the mime type of the patch correct? It should be text/x-diff or +text/x-patch or at least text/plain and not application/octet-stream. + +
  16. If the patch fixes a bug, did you provide a verbose analysis of the bug? + +
  17. If the patch fixes a bug, did you provide enough information, including +a sample, so the bug can be reproduced and the fix can be verified? +Note please do not attach samples >100k to mails but rather provide a +URL, you can upload to ftp://upload.ffmpeg.org + +
  18. Did you provide a verbose summary about what the patch does change? + +
  19. Did you provide a verbose explanation why it changes things like it does? + +
  20. Did you provide a verbose summary of the user visible advantages and +disadvantages if the patch is applied? + +
  21. Did you provide an example so we can verify the new feature added by the +patch easily? + +
  22. If you added a new file, did you insert a license header? It should be +taken from FFmpeg, not randomly copied and pasted from somewhere else. + +
  23. You should maintain alphabetical order in alphabetically ordered lists as +long as doing so does not break API/ABI compatibility. + +
  24. Lines with similar content should be aligned vertically when doing so +improves readability. + +
  25. Consider to add a regression test for your code. + +
  26. If you added YASM code please check that things still work with –disable-yasm + +
  27. Make sure you check the return values of function and return appropriate +error codes. Especially memory allocation functions like av_malloc() +are notoriously left unchecked, which is a serious problem. + +
  28. Test your code with valgrind and or Address Sanitizer to ensure it’s free +of leaks, out of array accesses, etc. +
+ + +

1.8 Patch review process

+ +

All patches posted to ffmpeg-devel will be reviewed, unless they contain a +clear note that the patch is not for the git master branch. +Reviews and comments will be posted as replies to the patch on the +mailing list. The patch submitter then has to take care of every comment, +that can be by resubmitting a changed patch or by discussion. Resubmitted +patches will themselves be reviewed like any other patch. If at some point +a patch passes review with no comments then it is approved, that can for +simple and small patches happen immediately while large patches will generally +have to be changed and reviewed many times before they are approved. +After a patch is approved it will be committed to the repository. +

+

We will review all submitted patches, but sometimes we are quite busy so +especially for large patches this can take several weeks. +

+

If you feel that the review process is too slow and you are willing to try to +take over maintainership of the area of code you change then just clone +git master and maintain the area of code there. We will merge each area from +where its best maintained. +

+

When resubmitting patches, please do not make any significant changes +not related to the comments received during review. Such patches will +be rejected. Instead, submit significant changes or new features as +separate patches. +

+ +

1.9 Regression tests

+ +

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

+

Running ’make fate’ accomplishes this, please see fate.html for details. +

+

[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]. +

+ +

1.9.1 Adding files to the fate-suite dataset

+ +

When there is no muxer or encoder available to generate test media for a +specific test then the media has to be included in the fate-suite. +First please make sure that the sample file is as small as possible to test the +respective decoder or demuxer sufficiently. Large files increase network +bandwidth and disk space requirements. +Once you have a working fate test and fate sample, provide in the commit +message or introductory message for the patch series that you post to +the ffmpeg-devel mailing list, a direct link to download the sample media. +

+ + +

1.9.2 Visualizing Test Coverage

+ +

The FFmpeg build system allows visualizing the test coverage in an easy +manner with the coverage tools gcov/lcov. This involves +the following steps: +

+
    +
  1. Configure to compile with instrumentation enabled: + configure --toolchain=gcov. + +
  2. Run your test case, either manually or via FATE. This can be either + the full FATE regression suite, or any arbitrary invocation of any + front-end tool provided by FFmpeg, in any combination. + +
  3. Run make lcov to generate coverage data in HTML format. + +
  4. View lcov/index.html in your preferred HTML viewer. +
+ +

You can use the command make lcov-reset to reset the coverage +measurements. You will need to rerun make lcov after running a +new test. +

+ +

1.9.3 Using Valgrind

+ +

The configure script provides a shortcut for using valgrind to spot bugs +related to memory handling. Just add the option +--toolchain=valgrind-memcheck or --toolchain=valgrind-massif +to your configure line, and reasonable defaults will be set for running +FATE under the supervision of either the memcheck or the +massif tool of the valgrind suite. +

+

In case you need finer control over how valgrind is invoked, use the +--target-exec='valgrind <your_custom_valgrind_options> option in +your configure line instead. +

+ +

1.10 Release process

+ +

FFmpeg maintains a set of release branches, which are the +recommended deliverable for system integrators and distributors (such as +Linux distributions, etc.). At regular times, a release +manager prepares, tests and publishes tarballs on the +http://ffmpeg.org website. +

+

There are two kinds of releases: +

+
    +
  1. Major releases always include the latest and greatest +features and functionality. + +
  2. Point releases are cut from release branches, +which are named release/X, with X being the release +version number. +
+ +

Note that we promise to our users that shared libraries from any FFmpeg +release never break programs that have been compiled against +previous versions of the same release series in any case! +

+

However, from time to time, we do make API changes that require adaptations +in applications. Such changes are only allowed in (new) major releases and +require further steps such as bumping library version numbers and/or +adjustments to the symbol versioning file. Please discuss such changes +on the ffmpeg-devel mailing list in time to allow forward planning. +

+ +

1.10.1 Criteria for Point Releases

+ +

Changes that match the following criteria are valid candidates for +inclusion into a point release: +

+
    +
  1. Fixes a security issue, preferably identified by a CVE +number issued by http://cve.mitre.org/. + +
  2. Fixes a documented bug in https://trac.ffmpeg.org. + +
  3. Improves the included documentation. + +
  4. Retains both source code and binary compatibility with previous +point releases of the same release branch. +
+ +

The order for checking the rules is (1 OR 2 OR 3) AND 4. +

+ + +

1.10.2 Release Checklist

+ +

The release process involves the following steps: +

+
    +
  1. Ensure that the RELEASE file contains the version number for +the upcoming release. + +
  2. Add the release at https://trac.ffmpeg.org/admin/ticket/versions. + +
  3. Announce the intent to do a release to the mailing list. + +
  4. Make sure all relevant security fixes have been backported. See +https://ffmpeg.org/security.html. + +
  5. Ensure that the FATE regression suite still passes in the release +branch on at least i386 and amd64 +(cf. Regression tests). + +
  6. Prepare the release tarballs in bz2 and gz formats, and +supplementing files that contain gpg signatures + +
  7. Publish the tarballs at http://ffmpeg.org/releases. Create and +push an annotated tag in the form nX, with X +containing the version number. + +
  8. Propose and send a patch to the ffmpeg-devel mailing list +with a news entry for the website. + +
  9. Publish the news entry. + +
  10. Send announcement to the mailing list. +
+ + +

+ This document was generated on January 14, 2015 using makeinfo. +

+
+ + diff --git a/Externals/ffmpeg/dev/doc/examples/Makefile b/Externals/ffmpeg/dev/doc/examples/Makefile new file mode 100644 index 0000000000..9f03f04b57 --- /dev/null +++ b/Externals/ffmpeg/dev/doc/examples/Makefile @@ -0,0 +1,44 @@ +# use pkg-config for getting CFLAGS and LDLIBS +FFMPEG_LIBS= libavdevice \ + libavformat \ + libavfilter \ + libavcodec \ + libswresample \ + libswscale \ + libavutil \ + +CFLAGS += -Wall -g +CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS) +LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS) + +EXAMPLES= avio_reading \ + decoding_encoding \ + demuxing_decoding \ + extract_mvs \ + filtering_video \ + filtering_audio \ + metadata \ + muxing \ + remuxing \ + resampling_audio \ + scaling_video \ + transcode_aac \ + transcoding \ + +OBJS=$(addsuffix .o,$(EXAMPLES)) + +# the following examples make explicit use of the math library +avcodec: LDLIBS += -lm +decoding_encoding: LDLIBS += -lm +muxing: LDLIBS += -lm +resampling_audio: LDLIBS += -lm + +.phony: all clean-test clean + +all: $(OBJS) $(EXAMPLES) + +clean-test: + $(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg + +clean: clean-test + $(RM) $(EXAMPLES) $(OBJS) diff --git a/Externals/ffmpeg/dev/doc/examples/README b/Externals/ffmpeg/dev/doc/examples/README new file mode 100644 index 0000000000..c1ce619d35 --- /dev/null +++ b/Externals/ffmpeg/dev/doc/examples/README @@ -0,0 +1,23 @@ +FFmpeg examples README +---------------------- + +Both following use cases rely on pkg-config and make, thus make sure +that you have them installed and working on your system. + + +Method 1: build the installed examples in a generic read/write user directory + +Copy to a read/write user directory and just use "make", it will link +to the libraries on your system, assuming the PKG_CONFIG_PATH is +correctly configured. + +Method 2: build the examples in-tree + +Assuming you are in the source FFmpeg checkout directory, you need to build +FFmpeg (no need to make install in any prefix). Then just run "make examples". +This will build the examples using the FFmpeg build system. You can clean those +examples using "make examplesclean" + +If you want to try the dedicated Makefile examples (to emulate the first +method), go into doc/examples and run a command such as +PKG_CONFIG_PATH=pc-uninstalled make. diff --git a/Externals/ffmpeg/dev/doc/examples/avio_reading.c b/Externals/ffmpeg/dev/doc/examples/avio_reading.c new file mode 100644 index 0000000000..02474e907a --- /dev/null +++ b/Externals/ffmpeg/dev/doc/examples/avio_reading.c @@ -0,0 +1,134 @@ +/* + * Copyright (c) 2014 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavformat AVIOContext API example. + * + * Make libavformat demuxer access media content through a custom + * AVIOContext read callback. + * @example avio_reading.c + */ + +#include +#include +#include +#include + +struct buffer_data { + uint8_t *ptr; + size_t size; ///< size left in the buffer +}; + +static int read_packet(void *opaque, uint8_t *buf, int buf_size) +{ + struct buffer_data *bd = (struct buffer_data *)opaque; + buf_size = FFMIN(buf_size, bd->size); + + printf("ptr:%p size:%zu\n", bd->ptr, bd->size); + + /* copy internal buffer data to buf */ + memcpy(buf, bd->ptr, buf_size); + bd->ptr += buf_size; + bd->size -= buf_size; + + return buf_size; +} + +int main(int argc, char *argv[]) +{ + AVFormatContext *fmt_ctx = NULL; + AVIOContext *avio_ctx = NULL; + uint8_t *buffer = NULL, *avio_ctx_buffer = NULL; + size_t buffer_size, avio_ctx_buffer_size = 4096; + char *input_filename = NULL; + int ret = 0; + struct buffer_data bd = { 0 }; + + if (argc != 2) { + fprintf(stderr, "usage: %s input_file\n" + "API example program to show how to read from a custom buffer " + "accessed through AVIOContext.\n", argv[0]); + return 1; + } + input_filename = argv[1]; + + /* register codecs and formats and other lavf/lavc components*/ + av_register_all(); + + /* slurp file content into buffer */ + ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL); + if (ret < 0) + goto end; + + /* fill opaque structure used by the AVIOContext read callback */ + bd.ptr = buffer; + bd.size = buffer_size; + + if (!(fmt_ctx = avformat_alloc_context())) { + ret = AVERROR(ENOMEM); + goto end; + } + + avio_ctx_buffer = av_malloc(avio_ctx_buffer_size); + if (!avio_ctx_buffer) { + ret = AVERROR(ENOMEM); + goto end; + } + avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size, + 0, &bd, &read_packet, NULL, NULL); + if (!avio_ctx) { + ret = AVERROR(ENOMEM); + goto end; + } + fmt_ctx->pb = avio_ctx; + + ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL); + if (ret < 0) { + fprintf(stderr, "Could not open input\n"); + goto end; + } + + ret = avformat_find_stream_info(fmt_ctx, NULL); + if (ret < 0) { + fprintf(stderr, "Could not find stream information\n"); + goto end; + } + + av_dump_format(fmt_ctx, 0, input_filename, 0); + +end: + avformat_close_input(&fmt_ctx); + /* note: the internal buffer could have changed, and be != avio_ctx_buffer */ + if (avio_ctx) { + av_freep(&avio_ctx->buffer); + av_freep(&avio_ctx); + } + av_file_unmap(buffer, buffer_size); + + if (ret < 0) { + fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); + return 1; + } + + return 0; +} diff --git a/Externals/ffmpeg/dev/doc/examples/decoding_encoding.c b/Externals/ffmpeg/dev/doc/examples/decoding_encoding.c new file mode 100644 index 0000000000..80da66431b --- /dev/null +++ b/Externals/ffmpeg/dev/doc/examples/decoding_encoding.c @@ -0,0 +1,665 @@ +/* + * Copyright (c) 2001 Fabrice Bellard + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavcodec API use example. + * + * @example decoding_encoding.c + * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...), + * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the + * format handling + */ + +#include + +#include +#include +#include +#include +#include +#include +#include + +#define INBUF_SIZE 4096 +#define AUDIO_INBUF_SIZE 20480 +#define AUDIO_REFILL_THRESH 4096 + +/* check that a given sample format is supported by the encoder */ +static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt) +{ + const enum AVSampleFormat *p = codec->sample_fmts; + + while (*p != AV_SAMPLE_FMT_NONE) { + if (*p == sample_fmt) + return 1; + p++; + } + return 0; +} + +/* just pick the highest supported samplerate */ +static int select_sample_rate(AVCodec *codec) +{ + const int *p; + int best_samplerate = 0; + + if (!codec->supported_samplerates) + return 44100; + + p = codec->supported_samplerates; + while (*p) { + best_samplerate = FFMAX(*p, best_samplerate); + p++; + } + return best_samplerate; +} + +/* select layout with the highest channel count */ +static int select_channel_layout(AVCodec *codec) +{ + const uint64_t *p; + uint64_t best_ch_layout = 0; + int best_nb_channels = 0; + + if (!codec->channel_layouts) + return AV_CH_LAYOUT_STEREO; + + p = codec->channel_layouts; + while (*p) { + int nb_channels = av_get_channel_layout_nb_channels(*p); + + if (nb_channels > best_nb_channels) { + best_ch_layout = *p; + best_nb_channels = nb_channels; + } + p++; + } + return best_ch_layout; +} + +/* + * Audio encoding example + */ +static void audio_encode_example(const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + AVFrame *frame; + AVPacket pkt; + int i, j, k, ret, got_output; + int buffer_size; + FILE *f; + uint16_t *samples; + float t, tincr; + + printf("Encode audio file %s\n", filename); + + /* find the MP2 encoder */ + codec = avcodec_find_encoder(AV_CODEC_ID_MP2); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } + + /* put sample parameters */ + c->bit_rate = 64000; + + /* check that the encoder supports s16 pcm input */ + c->sample_fmt = AV_SAMPLE_FMT_S16; + if (!check_sample_fmt(codec, c->sample_fmt)) { + fprintf(stderr, "Encoder does not support sample format %s", + av_get_sample_fmt_name(c->sample_fmt)); + exit(1); + } + + /* select other audio parameters supported by the encoder */ + c->sample_rate = select_sample_rate(codec); + c->channel_layout = select_channel_layout(codec); + c->channels = av_get_channel_layout_nb_channels(c->channel_layout); + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "wb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + /* frame containing input raw audio */ + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate audio frame\n"); + exit(1); + } + + frame->nb_samples = c->frame_size; + frame->format = c->sample_fmt; + frame->channel_layout = c->channel_layout; + + /* the codec gives us the frame size, in samples, + * we calculate the size of the samples buffer in bytes */ + buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, + c->sample_fmt, 0); + if (buffer_size < 0) { + fprintf(stderr, "Could not get sample buffer size\n"); + exit(1); + } + samples = av_malloc(buffer_size); + if (!samples) { + fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", + buffer_size); + exit(1); + } + /* setup the data pointers in the AVFrame */ + ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (const uint8_t*)samples, buffer_size, 0); + if (ret < 0) { + fprintf(stderr, "Could not setup audio frame\n"); + exit(1); + } + + /* encode a single tone sound */ + t = 0; + tincr = 2 * M_PI * 440.0 / c->sample_rate; + for (i = 0; i < 200; i++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + for (j = 0; j < c->frame_size; j++) { + samples[2*j] = (int)(sin(t) * 10000); + + for (k = 1; k < c->channels; k++) + samples[2*j + k] = samples[2*j]; + t += tincr; + } + /* encode the samples */ + ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding audio frame\n"); + exit(1); + } + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + + /* get the delayed frames */ + for (got_output = 1; got_output; i++) { + ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + fclose(f); + + av_freep(&samples); + av_frame_free(&frame); + avcodec_close(c); + av_free(c); +} + +/* + * Audio decoding. + */ +static void audio_decode_example(const char *outfilename, const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int len; + FILE *f, *outfile; + uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; + AVPacket avpkt; + AVFrame *decoded_frame = NULL; + + av_init_packet(&avpkt); + + printf("Decode audio file %s to %s\n", filename, outfilename); + + /* find the mpeg audio decoder */ + codec = avcodec_find_decoder(AV_CODEC_ID_MP2); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "rb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + outfile = fopen(outfilename, "wb"); + if (!outfile) { + av_free(c); + exit(1); + } + + /* decode until eof */ + avpkt.data = inbuf; + avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); + + while (avpkt.size > 0) { + int i, ch; + int got_frame = 0; + + if (!decoded_frame) { + if (!(decoded_frame = av_frame_alloc())) { + fprintf(stderr, "Could not allocate audio frame\n"); + exit(1); + } + } + + len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt); + if (len < 0) { + fprintf(stderr, "Error while decoding\n"); + exit(1); + } + if (got_frame) { + /* if a frame has been decoded, output it */ + int data_size = av_get_bytes_per_sample(c->sample_fmt); + if (data_size < 0) { + /* This should not occur, checking just for paranoia */ + fprintf(stderr, "Failed to calculate data size\n"); + exit(1); + } + for (i=0; inb_samples; i++) + for (ch=0; chchannels; ch++) + fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile); + } + avpkt.size -= len; + avpkt.data += len; + avpkt.dts = + avpkt.pts = AV_NOPTS_VALUE; + if (avpkt.size < AUDIO_REFILL_THRESH) { + /* Refill the input buffer, to avoid trying to decode + * incomplete frames. Instead of this, one could also use + * a parser, or use a proper container format through + * libavformat. */ + memmove(inbuf, avpkt.data, avpkt.size); + avpkt.data = inbuf; + len = fread(avpkt.data + avpkt.size, 1, + AUDIO_INBUF_SIZE - avpkt.size, f); + if (len > 0) + avpkt.size += len; + } + } + + fclose(outfile); + fclose(f); + + avcodec_close(c); + av_free(c); + av_frame_free(&decoded_frame); +} + +/* + * Video encoding example + */ +static void video_encode_example(const char *filename, int codec_id) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int i, ret, x, y, got_output; + FILE *f; + AVFrame *frame; + AVPacket pkt; + uint8_t endcode[] = { 0, 0, 1, 0xb7 }; + + printf("Encode video file %s\n", filename); + + /* find the mpeg1 video encoder */ + codec = avcodec_find_encoder(codec_id); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } + + /* put sample parameters */ + c->bit_rate = 400000; + /* resolution must be a multiple of two */ + c->width = 352; + c->height = 288; + /* frames per second */ + c->time_base = (AVRational){1,25}; + /* emit one intra frame every ten frames + * check frame pict_type before passing frame + * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I + * then gop_size is ignored and the output of encoder + * will always be I frame irrespective to gop_size + */ + c->gop_size = 10; + c->max_b_frames = 1; + c->pix_fmt = AV_PIX_FMT_YUV420P; + + if (codec_id == AV_CODEC_ID_H264) + av_opt_set(c->priv_data, "preset", "slow", 0); + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "wb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + frame->format = c->pix_fmt; + frame->width = c->width; + frame->height = c->height; + + /* the image can be allocated by any means and av_image_alloc() is + * just the most convenient way if av_malloc() is to be used */ + ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height, + c->pix_fmt, 32); + if (ret < 0) { + fprintf(stderr, "Could not allocate raw picture buffer\n"); + exit(1); + } + + /* encode 1 second of video */ + for (i = 0; i < 25; i++) { + av_init_packet(&pkt); + pkt.data = NULL; // packet data will be allocated by the encoder + pkt.size = 0; + + fflush(stdout); + /* prepare a dummy image */ + /* Y */ + for (y = 0; y < c->height; y++) { + for (x = 0; x < c->width; x++) { + frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3; + } + } + + /* Cb and Cr */ + for (y = 0; y < c->height/2; y++) { + for (x = 0; x < c->width/2; x++) { + frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2; + frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5; + } + } + + frame->pts = i; + + /* encode the image */ + ret = avcodec_encode_video2(c, &pkt, frame, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + printf("Write frame %3d (size=%5d)\n", i, pkt.size); + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + + /* get the delayed frames */ + for (got_output = 1; got_output; i++) { + fflush(stdout); + + ret = avcodec_encode_video2(c, &pkt, NULL, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); + exit(1); + } + + if (got_output) { + printf("Write frame %3d (size=%5d)\n", i, pkt.size); + fwrite(pkt.data, 1, pkt.size, f); + av_free_packet(&pkt); + } + } + + /* add sequence end code to have a real mpeg file */ + fwrite(endcode, 1, sizeof(endcode), f); + fclose(f); + + avcodec_close(c); + av_free(c); + av_freep(&frame->data[0]); + av_frame_free(&frame); + printf("\n"); +} + +/* + * Video decoding example + */ + +static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize, + char *filename) +{ + FILE *f; + int i; + + f = fopen(filename,"w"); + fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255); + for (i = 0; i < ysize; i++) + fwrite(buf + i * wrap, 1, xsize, f); + fclose(f); +} + +static int decode_write_frame(const char *outfilename, AVCodecContext *avctx, + AVFrame *frame, int *frame_count, AVPacket *pkt, int last) +{ + int len, got_frame; + char buf[1024]; + + len = avcodec_decode_video2(avctx, frame, &got_frame, pkt); + if (len < 0) { + fprintf(stderr, "Error while decoding frame %d\n", *frame_count); + return len; + } + if (got_frame) { + printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count); + fflush(stdout); + + /* the picture is allocated by the decoder, no need to free it */ + snprintf(buf, sizeof(buf), outfilename, *frame_count); + pgm_save(frame->data[0], frame->linesize[0], + avctx->width, avctx->height, buf); + (*frame_count)++; + } + if (pkt->data) { + pkt->size -= len; + pkt->data += len; + } + return 0; +} + +static void video_decode_example(const char *outfilename, const char *filename) +{ + AVCodec *codec; + AVCodecContext *c= NULL; + int frame_count; + FILE *f; + AVFrame *frame; + uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; + AVPacket avpkt; + + av_init_packet(&avpkt); + + /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */ + memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE); + + printf("Decode video file %s to %s\n", filename, outfilename); + + /* find the mpeg1 video decoder */ + codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO); + if (!codec) { + fprintf(stderr, "Codec not found\n"); + exit(1); + } + + c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } + + if(codec->capabilities&CODEC_CAP_TRUNCATED) + c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */ + + /* For some codecs, such as msmpeg4 and mpeg4, width and height + MUST be initialized there because this information is not + available in the bitstream. */ + + /* open it */ + if (avcodec_open2(c, codec, NULL) < 0) { + fprintf(stderr, "Could not open codec\n"); + exit(1); + } + + f = fopen(filename, "rb"); + if (!f) { + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + + frame_count = 0; + for (;;) { + avpkt.size = fread(inbuf, 1, INBUF_SIZE, f); + if (avpkt.size == 0) + break; + + /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio) + and this is the only method to use them because you cannot + know the compressed data size before analysing it. + + BUT some other codecs (msmpeg4, mpeg4) are inherently frame + based, so you must call them with all the data for one + frame exactly. You must also initialize 'width' and + 'height' before initializing them. */ + + /* NOTE2: some codecs allow the raw parameters (frame size, + sample rate) to be changed at any frame. We handle this, so + you should also take care of it */ + + /* here, we use a stream based decoder (mpeg1video), so we + feed decoder and see if it could decode a frame */ + avpkt.data = inbuf; + while (avpkt.size > 0) + if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0) + exit(1); + } + + /* some codecs, such as MPEG, transmit the I and P frame with a + latency of one frame. You must do the following to have a + chance to get the last frame of the video */ + avpkt.data = NULL; + avpkt.size = 0; + decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1); + + fclose(f); + + avcodec_close(c); + av_free(c); + av_frame_free(&frame); + printf("\n"); +} + +int main(int argc, char **argv) +{ + const char *output_type; + + /* register all the codecs */ + avcodec_register_all(); + + if (argc < 2) { + printf("usage: %s output_type\n" + "API example program to decode/encode a media stream with libavcodec.\n" + "This program generates a synthetic stream and encodes it to a file\n" + "named test.h264, test.mp2 or test.mpg depending on output_type.\n" + "The encoded stream is then decoded and written to a raw data output.\n" + "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n", + argv[0]); + return 1; + } + output_type = argv[1]; + + if (!strcmp(output_type, "h264")) { + video_encode_example("test.h264", AV_CODEC_ID_H264); + } else if (!strcmp(output_type, "mp2")) { + audio_encode_example("test.mp2"); + audio_decode_example("test.pcm", "test.mp2"); + } else if (!strcmp(output_type, "mpg")) { + video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO); + video_decode_example("test%02d.pgm", "test.mpg"); + } else { + fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n", + output_type); + return 1; + } + + return 0; +} diff --git a/Externals/ffmpeg/dev/doc/examples/demuxing_decoding.c b/Externals/ffmpeg/dev/doc/examples/demuxing_decoding.c new file mode 100644 index 0000000000..2ce4018c79 --- /dev/null +++ b/Externals/ffmpeg/dev/doc/examples/demuxing_decoding.c @@ -0,0 +1,386 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * Demuxing and decoding example. + * + * Show how to use the libavformat and libavcodec API to demux and + * decode audio and video data. + * @example demuxing_decoding.c + */ + +#include +#include +#include +#include + +static AVFormatContext *fmt_ctx = NULL; +static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx; +static AVStream *video_stream = NULL, *audio_stream = NULL; +static const char *src_filename = NULL; +static const char *video_dst_filename = NULL; +static const char *audio_dst_filename = NULL; +static FILE *video_dst_file = NULL; +static FILE *audio_dst_file = NULL; + +static uint8_t *video_dst_data[4] = {NULL}; +static int video_dst_linesize[4]; +static int video_dst_bufsize; + +static int video_stream_idx = -1, audio_stream_idx = -1; +static AVFrame *frame = NULL; +static AVPacket pkt; +static int video_frame_count = 0; +static int audio_frame_count = 0; + +/* The different ways of decoding and managing data memory. You are not + * supposed to support all the modes in your application but pick the one most + * appropriate to your needs. Look for the use of api_mode in this example to + * see what are the differences of API usage between them */ +enum { + API_MODE_OLD = 0, /* old method, deprecated */ + API_MODE_NEW_API_REF_COUNT = 1, /* new method, using the frame reference counting */ + API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */ +}; + +static int api_mode = API_MODE_OLD; + +static int decode_packet(int *got_frame, int cached) +{ + int ret = 0; + int decoded = pkt.size; + + *got_frame = 0; + + if (pkt.stream_index == video_stream_idx) { + /* decode video frame */ + ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); + return ret; + } + + if (*got_frame) { + printf("video_frame%s n:%d coded_n:%d pts:%s\n", + cached ? "(cached)" : "", + video_frame_count++, frame->coded_picture_number, + av_ts2timestr(frame->pts, &video_dec_ctx->time_base)); + + /* copy decoded frame to destination buffer: + * this is required since rawvideo expects non aligned data */ + av_image_copy(video_dst_data, video_dst_linesize, + (const uint8_t **)(frame->data), frame->linesize, + video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height); + + /* write to rawvideo file */ + fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file); + } + } else if (pkt.stream_index == audio_stream_idx) { + /* decode audio frame */ + ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret)); + return ret; + } + /* Some audio decoders decode only part of the packet, and have to be + * called again with the remainder of the packet data. + * Sample: fate-suite/lossless-audio/luckynight-partial.shn + * Also, some decoders might over-read the packet. */ + decoded = FFMIN(ret, pkt.size); + + if (*got_frame) { + size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format); + printf("audio_frame%s n:%d nb_samples:%d pts:%s\n", + cached ? "(cached)" : "", + audio_frame_count++, frame->nb_samples, + av_ts2timestr(frame->pts, &audio_dec_ctx->time_base)); + + /* Write the raw audio data samples of the first plane. This works + * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However, + * most audio decoders output planar audio, which uses a separate + * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P). + * In other words, this code will write only the first audio channel + * in these cases. + * You should use libswresample or libavfilter to convert the frame + * to packed data. */ + fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file); + } + } + + /* If we use the new API with reference counting, we own the data and need + * to de-reference it when we don't use it anymore */ + if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT) + av_frame_unref(frame); + + return decoded; +} + +static int open_codec_context(int *stream_idx, + AVFormatContext *fmt_ctx, enum AVMediaType type) +{ + int ret; + AVStream *st; + AVCodecContext *dec_ctx = NULL; + AVCodec *dec = NULL; + AVDictionary *opts = NULL; + + ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); + if (ret < 0) { + fprintf(stderr, "Could not find %s stream in input file '%s'\n", + av_get_media_type_string(type), src_filename); + return ret; + } else { + *stream_idx = ret; + st = fmt_ctx->streams[*stream_idx]; + + /* find decoder for the stream */ + dec_ctx = st->codec; + dec = avcodec_find_decoder(dec_ctx->codec_id); + if (!dec) { + fprintf(stderr, "Failed to find %s codec\n", + av_get_media_type_string(type)); + return AVERROR(EINVAL); + } + + /* Init the decoders, with or without reference counting */ + if (api_mode == API_MODE_NEW_API_REF_COUNT) + av_dict_set(&opts, "refcounted_frames", "1", 0); + if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) { + fprintf(stderr, "Failed to open %s codec\n", + av_get_media_type_string(type)); + return ret; + } + } + + return 0; +} + +static int get_format_from_sample_fmt(const char **fmt, + enum AVSampleFormat sample_fmt) +{ + int i; + struct sample_fmt_entry { + enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; + } sample_fmt_entries[] = { + { AV_SAMPLE_FMT_U8, "u8", "u8" }, + { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, + { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, + { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, + { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, + }; + *fmt = NULL; + + for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { + struct sample_fmt_entry *entry = &sample_fmt_entries[i]; + if (sample_fmt == entry->sample_fmt) { + *fmt = AV_NE(entry->fmt_be, entry->fmt_le); + return 0; + } + } + + fprintf(stderr, + "sample format %s is not supported as output format\n", + av_get_sample_fmt_name(sample_fmt)); + return -1; +} + +int main (int argc, char **argv) +{ + int ret = 0, got_frame; + + if (argc != 4 && argc != 5) { + fprintf(stderr, "usage: %s [-refcount=] " + "input_file video_output_file audio_output_file\n" + "API example program to show how to read frames from an input file.\n" + "This program reads frames from a file, decodes them, and writes decoded\n" + "video frames to a rawvideo file named video_output_file, and decoded\n" + "audio frames to a rawaudio file named audio_output_file.\n\n" + "If the -refcount option is specified, the program use the\n" + "reference counting frame system which allows keeping a copy of\n" + "the data for longer than one decode call. If unset, it's using\n" + "the classic old method.\n" + "\n", argv[0]); + exit(1); + } + if (argc == 5) { + const char *mode = argv[1] + strlen("-refcount="); + if (!strcmp(mode, "old")) api_mode = API_MODE_OLD; + else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT; + else if (!strcmp(mode, "new_refcount")) api_mode = API_MODE_NEW_API_REF_COUNT; + else { + fprintf(stderr, "unknow mode '%s'\n", mode); + exit(1); + } + argv++; + } + src_filename = argv[1]; + video_dst_filename = argv[2]; + audio_dst_filename = argv[3]; + + /* register all formats and codecs */ + av_register_all(); + + /* open input file, and allocate format context */ + if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) { + fprintf(stderr, "Could not open source file %s\n", src_filename); + exit(1); + } + + /* retrieve stream information */ + if (avformat_find_stream_info(fmt_ctx, NULL) < 0) { + fprintf(stderr, "Could not find stream information\n"); + exit(1); + } + + if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) { + video_stream = fmt_ctx->streams[video_stream_idx]; + video_dec_ctx = video_stream->codec; + + video_dst_file = fopen(video_dst_filename, "wb"); + if (!video_dst_file) { + fprintf(stderr, "Could not open destination file %s\n", video_dst_filename); + ret = 1; + goto end; + } + + /* allocate image where the decoded image will be put */ + ret = av_image_alloc(video_dst_data, video_dst_linesize, + video_dec_ctx->width, video_dec_ctx->height, + video_dec_ctx->pix_fmt, 1); + if (ret < 0) { + fprintf(stderr, "Could not allocate raw video buffer\n"); + goto end; + } + video_dst_bufsize = ret; + } + + if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) { + audio_stream = fmt_ctx->streams[audio_stream_idx]; + audio_dec_ctx = audio_stream->codec; + audio_dst_file = fopen(audio_dst_filename, "wb"); + if (!audio_dst_file) { + fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename); + ret = 1; + goto end; + } + } + + /* dump input information to stderr */ + av_dump_format(fmt_ctx, 0, src_filename, 0); + + if (!audio_stream && !video_stream) { + fprintf(stderr, "Could not find audio or video stream in the input, aborting\n"); + ret = 1; + goto end; + } + + /* When using the new API, you need to use the libavutil/frame.h API, while + * the classic frame management is available in libavcodec */ + if (api_mode == API_MODE_OLD) + frame = avcodec_alloc_frame(); + else + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate frame\n"); + ret = AVERROR(ENOMEM); + goto end; + } + + /* initialize packet, set data to NULL, let the demuxer fill it */ + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + if (video_stream) + printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename); + if (audio_stream) + printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename); + + /* read frames from the file */ + while (av_read_frame(fmt_ctx, &pkt) >= 0) { + AVPacket orig_pkt = pkt; + do { + ret = decode_packet(&got_frame, 0); + if (ret < 0) + break; + pkt.data += ret; + pkt.size -= ret; + } while (pkt.size > 0); + av_free_packet(&orig_pkt); + } + + /* flush cached frames */ + pkt.data = NULL; + pkt.size = 0; + do { + decode_packet(&got_frame, 1); + } while (got_frame); + + printf("Demuxing succeeded.\n"); + + if (video_stream) { + printf("Play the output video file with the command:\n" + "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", + av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height, + video_dst_filename); + } + + if (audio_stream) { + enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt; + int n_channels = audio_dec_ctx->channels; + const char *fmt; + + if (av_sample_fmt_is_planar(sfmt)) { + const char *packed = av_get_sample_fmt_name(sfmt); + printf("Warning: the sample format the decoder produced is planar " + "(%s). This example will output the first channel only.\n", + packed ? packed : "?"); + sfmt = av_get_packed_sample_fmt(sfmt); + n_channels = 1; + } + + if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0) + goto end; + + printf("Play the output audio file with the command:\n" + "ffplay -f %s -ac %d -ar %d %s\n", + fmt, n_channels, audio_dec_ctx->sample_rate, + audio_dst_filename); + } + +end: + avcodec_close(video_dec_ctx); + avcodec_close(audio_dec_ctx); + avformat_close_input(&fmt_ctx); + if (video_dst_file) + fclose(video_dst_file); + if (audio_dst_file) + fclose(audio_dst_file); + if (api_mode == API_MODE_OLD) + avcodec_free_frame(&frame); + else + av_frame_free(&frame); + av_free(video_dst_data[0]); + + return ret < 0; +} diff --git a/Externals/ffmpeg/dev/doc/examples/extract_mvs.c b/Externals/ffmpeg/dev/doc/examples/extract_mvs.c new file mode 100644 index 0000000000..d6fd61335e --- /dev/null +++ b/Externals/ffmpeg/dev/doc/examples/extract_mvs.c @@ -0,0 +1,185 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * Copyright (c) 2014 Clément Bœsch + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +#include +#include + +static AVFormatContext *fmt_ctx = NULL; +static AVCodecContext *video_dec_ctx = NULL; +static AVStream *video_stream = NULL; +static const char *src_filename = NULL; + +static int video_stream_idx = -1; +static AVFrame *frame = NULL; +static AVPacket pkt; +static int video_frame_count = 0; + +static int decode_packet(int *got_frame, int cached) +{ + int decoded = pkt.size; + + *got_frame = 0; + + if (pkt.stream_index == video_stream_idx) { + int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); + return ret; + } + + if (*got_frame) { + int i; + AVFrameSideData *sd; + + video_frame_count++; + sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS); + if (sd) { + const AVMotionVector *mvs = (const AVMotionVector *)sd->data; + for (i = 0; i < sd->size / sizeof(*mvs); i++) { + const AVMotionVector *mv = &mvs[i]; + printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n", + video_frame_count, mv->source, + mv->w, mv->h, mv->src_x, mv->src_y, + mv->dst_x, mv->dst_y, mv->flags); + } + } + } + } + + return decoded; +} + +static int open_codec_context(int *stream_idx, + AVFormatContext *fmt_ctx, enum AVMediaType type) +{ + int ret; + AVStream *st; + AVCodecContext *dec_ctx = NULL; + AVCodec *dec = NULL; + AVDictionary *opts = NULL; + + ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); + if (ret < 0) { + fprintf(stderr, "Could not find %s stream in input file '%s'\n", + av_get_media_type_string(type), src_filename); + return ret; + } else { + *stream_idx = ret; + st = fmt_ctx->streams[*stream_idx]; + + /* find decoder for the stream */ + dec_ctx = st->codec; + dec = avcodec_find_decoder(dec_ctx->codec_id); + if (!dec) { + fprintf(stderr, "Failed to find %s codec\n", + av_get_media_type_string(type)); + return AVERROR(EINVAL); + } + + /* Init the video decoder */ + av_dict_set(&opts, "flags2", "+export_mvs", 0); + if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) { + fprintf(stderr, "Failed to open %s codec\n", + av_get_media_type_string(type)); + return ret; + } + } + + return 0; +} + +int main(int argc, char **argv) +{ + int ret = 0, got_frame; + + if (argc != 2) { + fprintf(stderr, "Usage: %s