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    Jolie sélection multiple
    Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
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    PHP5 est obligatoire, vous pouvez l’installer en suivant ce tutoriel spécifique.
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Sur d’autres sites (9489)

  • Revision 5215b83aea : Simplifying and inlining k_cvtlo_epi16 and k_cvthi_epi16 Simplify the k_cvtlo_e

    3 octobre 2013, par A.Mahfoodh

    Changed Paths :
     Modify /vp9/encoder/x86/vp9_dct32x32_sse2.c



    Simplifying and inlining k_cvtlo_epi16 and k_cvthi_epi16

    Simplify the k_cvtlo_epi16 and k_cvthi_epi16 to only two
    instructions. Then inlined them.

    quoting from intel MMX_App_Compute_16bit_Vector.pdf‎
    "The PMADDWD instruction multiplies four
    pairs of 16-bit numbers and produces partial sums of the results
    and can do so once per clock (with a three-clock latency)."
    so I am assuming that there will be three clock overhead after the
    last _mm_madd_pi16 command.
    Even with the overhead the number of clocks in general should be
    smaller. I am not sure though becasue I could not find information
    about number of clocks required for instructions in k_cvtlo_epi16
    and k_cvthi_epi16. I will run a test and compare the execution time.

    Change-Id : Ieda4aa338f69ad3dd196ac6e7892da3cf1b47ea7

  • Stream ffmpeg transcoding result to S3

    7 juin 2019, par mabead

    I want to transcode a large file using FFMPEG and store the result directly on AWS S3. This will be done inside of an AWS Lambda that has limited tmp space so I can’t store the transcoding result locally and then upload it to S3 in a second step. I won’t have enough tmp space. I therefore want to store the FFMPEG output directly on S3.

    I therefore created a S3 pre-signed url that allows ’PUT’ :

    var outputPath = s3Client.GetPreSignedURL(new Amazon.S3.Model.GetPreSignedUrlRequest
    {
       BucketName = "my-bucket",
       Expires = DateTime.UtcNow.AddMinutes(5),
       Key = "output.mp3",
       Verb = HttpVerb.PUT,
    });

    I then called ffmpeg with the resulting pre-signed url :

    ffmpeg -i C:\input.wav -y -vn -ar 44100 -ac 2 -ab 192k -f mp3 https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550427237&Signature=%2BE8Wc%2F%2FQYrvGxzc%2FgXnsvauKnac%3D

    FFMPEG returns an exit code of 1 with the following output :

    ffmpeg version N-93120-ga84af760b8 Copyright (c) 2000-2019 the FFmpeg developers
     built with gcc 8.2.1 (GCC) 20190212
     configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
     libavutil      56. 26.100 / 56. 26.100
     libavcodec     58. 47.100 / 58. 47.100
     libavformat    58. 26.101 / 58. 26.101
     libavdevice    58.  6.101 / 58.  6.101
     libavfilter     7. 48.100 /  7. 48.100
     libswscale      5.  4.100 /  5.  4.100
     libswresample   3.  4.100 /  3.  4.100
     libpostproc    55.  4.100 / 55.  4.100
    Guessed Channel Layout for Input Stream #0.0 : stereo
    Input #0, wav, from 'C:\input.wav':
     Duration: 00:04:16.72, bitrate: 3072 kb/s
       Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s
    Stream mapping:
     Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help
    Output #0, mp3, to 'https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550427237&Signature=%2BE8Wc%2F%2FQYrvGxzc%2FgXnsvauKnac%3D':
     Metadata:
       TSSE            : Lavf58.26.101
       Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s32p, 192 kb/s
       Metadata:
         encoder         : Lavc58.47.100 libmp3lame
    size=     577kB time=00:00:24.58 bitrate= 192.2kbits/s speed=49.1x    
    size=    1109kB time=00:00:47.28 bitrate= 192.1kbits/s speed=47.2x    
    [tls @ 000001d73d786b00] Error in the push function.
    av_interleaved_write_frame(): I/O error
    Error writing trailer of https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550427237&Signature=%2BE8Wc%2F%2FQYrvGxzc%2FgXnsvauKnac%3D: I/O error
    size=    1143kB time=00:00:48.77 bitrate= 192.0kbits/s speed=  47x    
    video:0kB audio:1144kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
    [tls @ 000001d73d786b00] The specified session has been invalidated for some reason.
    [tls @ 000001d73d786b00] Error in the pull function.
    [https @ 000001d73d784fc0] URL read error:  -5
    Conversion failed!

    As you can see, I have a URL read error. This is a little surprising to me since I want to output to this url and not read it.

    Anybody know how I can store directly my FFMPEG output directly to S3 without having to store it locally first ?

    Edit 1
    I then tried to use the -method PUT parameter and use http instead of https to remove TLS from the equation. Here’s the output that I got when running ffmpeg with the -v trace option.

    ffmpeg version N-93120-ga84af760b8 Copyright (c) 2000-2019 the FFmpeg developers
     built with gcc 8.2.1 (GCC) 20190212
     configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
     libavutil      56. 26.100 / 56. 26.100
     libavcodec     58. 47.100 / 58. 47.100
     libavformat    58. 26.101 / 58. 26.101
     libavdevice    58.  6.101 / 58.  6.101
     libavfilter     7. 48.100 /  7. 48.100
     libswscale      5.  4.100 /  5.  4.100
     libswresample   3.  4.100 /  3.  4.100
     libpostproc    55.  4.100 / 55.  4.100
    Splitting the commandline.
    Reading option '-i' ... matched as input url with argument 'C:\input.wav'.
    Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
    Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'.
    Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'.
    Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'.
    Reading option '-ab' ... matched as option 'ab' (audio bitrate (please use -b:a)) with argument '192k'.
    Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'.
    Reading option '-method' ... matched as AVOption 'method' with argument 'PUT'.
    Reading option '-v' ... matched as option 'v' (set logging level) with argument 'trace'.
    Reading option 'https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D' ... matched as output url.
    Finished splitting the commandline.
    Parsing a group of options: global .
    Applying option y (overwrite output files) with argument 1.
    Applying option v (set logging level) with argument trace.
    Successfully parsed a group of options.
    Parsing a group of options: input url C:\input.wav.
    Successfully parsed a group of options.
    Opening an input file: C:\input.wav.
    [NULL @ 000001fb37abb180] Opening 'C:\input.wav' for reading
    [file @ 000001fb37abc180] Setting default whitelist 'file,crypto'
    Probing wav score:99 size:2048
    [wav @ 000001fb37abb180] Format wav probed with size=2048 and score=99
    [wav @ 000001fb37abb180] Before avformat_find_stream_info() pos: 54 bytes read:65590 seeks:1 nb_streams:1
    [wav @ 000001fb37abb180] parser not found for codec pcm_s32le, packets or times may be invalid.
       Last message repeated 1 times
    [wav @ 000001fb37abb180] All info found
    [wav @ 000001fb37abb180] stream 0: start_time: -192153584101141.156 duration: 256.716
    [wav @ 000001fb37abb180] format: start_time: -9223372036854.775 duration: 256.716 bitrate=3072 kb/s
    [wav @ 000001fb37abb180] After avformat_find_stream_info() pos: 204854 bytes read:294966 seeks:1 frames:50
    Guessed Channel Layout for Input Stream #0.0 : stereo
    Input #0, wav, from 'C:\input.wav':
     Duration: 00:04:16.72, bitrate: 3072 kb/s
       Stream #0:0, 50, 1/48000: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s
    Successfully opened the file.
    Parsing a group of options: output url https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D.
    Applying option vn (disable video) with argument 1.
    Applying option ar (set audio sampling rate (in Hz)) with argument 44100.
    Applying option ac (set number of audio channels) with argument 2.
    Applying option ab (audio bitrate (please use -b:a)) with argument 192k.
    Applying option f (force format) with argument mp3.
    Successfully parsed a group of options.
    Opening an output file: https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D.
    [http @ 000001fb37b15140] Setting default whitelist 'http,https,tls,rtp,tcp,udp,crypto,httpproxy'
    [tcp @ 000001fb37b16c80] Original list of addresses:
    [tcp @ 000001fb37b16c80] Address 52.216.8.203 port 80
    [tcp @ 000001fb37b16c80] Interleaved list of addresses:
    [tcp @ 000001fb37b16c80] Address 52.216.8.203 port 80
    [tcp @ 000001fb37b16c80] Starting connection attempt to 52.216.8.203 port 80
    [tcp @ 000001fb37b16c80] Successfully connected to 52.216.8.203 port 80
    [http @ 000001fb37b15140] request: PUT /output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D HTTP/1.1
    Transfer-Encoding: chunked
    User-Agent: Lavf/58.26.101
    Accept: */*
    Connection: close
    Host: landr-distribution-reportsdev-mb.s3.amazonaws.com
    Icy-MetaData: 1
    Successfully opened the file.
    Stream mapping:
     Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (libmp3lame))
    Press [q] to stop, [?] for help
    cur_dts is invalid (this is harmless if it occurs once at the start per stream)
    detected 8 logical cores
    [graph_0_in_0_0 @ 000001fb37b21080] Setting 'time_base' to value '1/48000'
    [graph_0_in_0_0 @ 000001fb37b21080] Setting 'sample_rate' to value '48000'
    [graph_0_in_0_0 @ 000001fb37b21080] Setting 'sample_fmt' to value 's32'
    [graph_0_in_0_0 @ 000001fb37b21080] Setting 'channel_layout' to value '0x3'
    [graph_0_in_0_0 @ 000001fb37b21080] tb:1/48000 samplefmt:s32 samplerate:48000 chlayout:0x3
    [format_out_0_0 @ 000001fb37b22cc0] Setting 'sample_fmts' to value 's32p|fltp|s16p'
    [format_out_0_0 @ 000001fb37b22cc0] Setting 'sample_rates' to value '44100'
    [format_out_0_0 @ 000001fb37b22cc0] Setting 'channel_layouts' to value '0x3'
    [format_out_0_0 @ 000001fb37b22cc0] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
    [AVFilterGraph @ 000001fb37b0d940] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
    [auto_resampler_0 @ 000001fb37b251c0] picking s32p out of 3 ref:s32
    [auto_resampler_0 @ 000001fb37b251c0] [SWR @ 000001fb37b252c0] Using fltp internally between filters
    [auto_resampler_0 @ 000001fb37b251c0] ch:2 chl:stereo fmt:s32 r:48000Hz -> ch:2 chl:stereo fmt:s32p r:44100Hz
    Output #0, mp3, to 'https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D':
     Metadata:
       TSSE            : Lavf58.26.101
       Stream #0:0, 0, 1/44100: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s32p, delay 1105, 192 kb/s
       Metadata:
         encoder         : Lavc58.47.100 libmp3lame
    cur_dts is invalid (this is harmless if it occurs once at the start per stream)
       Last message repeated 6 times
    size=     649kB time=00:00:27.66 bitrate= 192.2kbits/s speed=55.3x    
    size=    1207kB time=00:00:51.48 bitrate= 192.1kbits/s speed=51.5x    
    av_interleaved_write_frame(): Unknown error
    No more output streams to write to, finishing.
    [libmp3lame @ 000001fb37b147c0] Trying to remove 47 more samples than there are in the queue
    Error writing trailer of https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D: Error number -10054 occurred
    size=    1251kB time=00:00:53.39 bitrate= 192.0kbits/s speed=51.5x    
    video:0kB audio:1252kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
    Input file #0 (C:\input.wav):
     Input stream #0:0 (audio): 5014 packets read (20537344 bytes); 5014 frames decoded (2567168 samples);
     Total: 5014 packets (20537344 bytes) demuxed
    Output file #0 (https://my-bucket.s3.amazonaws.com/output.mp3?AWSAccessKeyId=AKIAJDSGJWM63VQEXHIQ&Expires=1550695990&Signature=dy3RVqDlX%2BlJ0INlDkl0Lm1Rqb4%3D):
     Output stream #0:0 (audio): 2047 frames encoded (2358144 samples); 2045 packets muxed (1282089 bytes);
     Total: 2045 packets (1282089 bytes) muxed
    5014 frames successfully decoded, 0 decoding errors
    [AVIOContext @ 000001fb37b1f440] Statistics: 0 seeks, 2046 writeouts
    [http @ 000001fb37b15140] URL read error:  -10054
    [AVIOContext @ 000001fb37ac4400] Statistics: 20611126 bytes read, 1 seeks
    Conversion failed!

    So it looks like it is able to connect to my S3 pre-signed url but I still have the Error writing trailer error coupled with a URL read error.

  • Unable to integrate ffmpeg as a native dependency for an Android NDK project

    28 février 2019, par mmark

    I have an Android NDK project which consists entirely of C/C++ code, and it basically processes images without using any external libraries.

    I’m using Android Studio + Gradle NDK Experimental plugin (0.7.0-alpha1).

    Now, I need to integrate ffmpeg as a native library to use it from the C/C++ code in order to decode a H.264 video frame.

    These are the questions I’ve found here regarding this issue :

    Android - Integrating ffmpeg and android-ndk-r9c

    Android NDK w/ ffmpeg library - error running project

    Using FFmpeg native libraries with Android-NDK

    Can not build with prebuilt static libraries using gradle-experimental

    Here is my build.gradle file :

    apply plugin: 'com.android.model.application'

    def ffmpeg_path = file(project(':ffmpeg').projectDir).absolutePath + "/ffmpeg-android"

    model {

       repositories {
           libs(PrebuiltLibraries) {
               libavcodec {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libavcodec.a")
                   }
               }
               libavutil {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libavutil.a")
                   }
               }
               libswresample {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libswresample.a")
                   }
               }
               libswscale {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libswscale.a")
                   }
               }
               libavformat {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libavformat.a")
                   }
               }
               libavdevice {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libavdevice.a")
                   }
               }
               libavfilter {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libavfilter.a")
                   }
               }
               libpostproc {
                   headers.srcDir "${ffmpeg_path}/include"
                   binaries.withType(StaticLibraryBinary) {
                       staticLibraryFile = file("${ffmpeg_path}/${targetPlatform.getName()}/lib/libpostproc.a")
                   }
               }
           }
       }

       android {
           compileSdkVersion = 23
           buildToolsVersion = "23.0.2"

           defaultConfig.with {
               applicationId = "com.example.hellojni"
               minSdkVersion.apiLevel = 4
               targetSdkVersion.apiLevel = 23
           }
       }

       /*
        * native build settings
        */
       android.ndk {
           moduleName = "hello-jni"
           platformVersion = 9 //same as minSdkVersion.apiLevel for better compatibility
           stl    = "c++_static"
           abiFilters.addAll(["armeabi", "armeabi-v7a", "x86"]) //filtering ABIs on the ones Google Play Games library is compatible with.
       }
       android.buildTypes {
           release {
               minifyEnabled = false
               proguardFiles.add(file('proguard-rules.txt'))
           }
       }

       android.sources {
           main {
               jni {
                   dependencies {
                       library "libavcodec" linkage "static"
                       library "libavutil" linkage "static"
                       library "libswresample" linkage "static"
                       library "libswscale" linkage "static"
                       library "libavformat" linkage "static"
                       library "libavdevice" linkage "static"
                       library "libavfilter" linkage "static"
                       library "libpostproc" linkage "static"
                   }
               }
           }
       }

    }

    I based on this sample from Google which integrates an external native library.

    This is the ffmpeg project I compiled before importing it into mine :
    https://github.com/WritingMinds/ffmpeg-android

    I’ve added all the .a and .h files and I’m able to import the headers in my C/C++ code, but when I try to compile it I get the following error :

    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function try_decode_frame: error: undefined reference to 'avpriv_h264_has_num_reorder_frames'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function compute_pkt_fields: error: undefined reference to 'avpriv_h264_has_num_reorder_frames'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function parse_packet: error: undefined reference to 'av_parser_parse2'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function parse_packet: error: undefined reference to 'av_parser_close'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function read_frame_internal: error: undefined reference to 'av_parser_init'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function avformat_find_stream_info: error: undefined reference to 'av_parser_init'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function avformat_find_stream_info: error: undefined reference to 'avcodec_pix_fmt_to_codec_tag'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function avformat_find_stream_info: error: undefined reference to 'avpriv_get_raw_pix_fmt_tags'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function ff_stream_add_bitstream_filter: error: undefined reference to 'av_bitstream_filter_init'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function ff_stream_add_bitstream_filter: error: undefined reference to 'av_bitstream_filter_close'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(utils.o):utils.c:function av_apply_bitstream_filters: error: undefined reference to 'av_bitstream_filter_filter'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(aadec.o):aadec.c:function aa_read_packet: error: undefined reference to 'av_tea_init'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(aadec.o):aadec.c:function aa_read_packet: error: undefined reference to 'av_tea_crypt'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(aadec.o):aadec.c:function aa_read_header: error: undefined reference to 'av_tea_alloc'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(aadec.o):aadec.c:function aa_read_header: error: undefined reference to 'av_tea_init'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(aadec.o):aadec.c:function aa_read_header: error: undefined reference to 'av_tea_crypt'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(ac3dec.o):ac3dec.c:function ac3_eac3_probe.isra.0: error: undefined reference to 'avpriv_ac3_parse_header'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(adtsenc.o):adtsenc.c:function adts_write_header: error: undefined reference to 'avpriv_mpeg4audio_get_config'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(adtsenc.o):adtsenc.c:function adts_write_header: error: undefined reference to 'avpriv_copy_pce_data'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(asfcrypt.o):asfcrypt.c:function ff_asfcrypt_dec: error: undefined reference to 'av_rc4_alloc'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(asfcrypt.o):asfcrypt.c:function ff_asfcrypt_dec: error: undefined reference to 'av_rc4_init'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(asfcrypt.o):asfcrypt.c:function ff_asfcrypt_dec: error: undefined reference to 'av_rc4_crypt'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(asfcrypt.o):asfcrypt.c:function ff_asfcrypt_dec: error: undefined reference to 'av_rc4_init'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(asfcrypt.o):asfcrypt.c:function ff_asfcrypt_dec: error: undefined reference to 'av_rc4_crypt'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(matroska.o):matroska.c:function ff_mkv_stereo3d_conv: error: undefined reference to 'av_stereo3d_alloc'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(nut.o):nut.c:function ff_nut_add_sp: error: undefined reference to 'av_tree_node_alloc'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(nut.o):nut.c:function ff_nut_add_sp: error: undefined reference to 'av_tree_insert'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(nut.o):nut.c:function ff_nut_free_sp: error: undefined reference to 'av_tree_enumerate'
    /Users/marcos/Documents/Android/fdecoder/ffmpeg/ffmpeg-android/armeabi-v7a/lib/libavformat.a(nut.o):nut.c:function ff_nut_free_sp: error: undefined reference to 'av_tree_destroy'
    Error:error: ld returned 1 exit status
    Error:Execution failed for task ':app:linkHello-jniArmeabi-v7aDebugSharedLibrary'.
    > A build operation failed.
         Linker failed while linking libhello-jni.so.
     See the complete log at: file:///Users/marcos/Documents/Android/fdecoder/app/build/tmp/linkHello-jniArmeabi-v7aDebugSharedLibrary/output.txt
    Information:BUILD FAILED
    Information:Total time: 14.993 secs
    Information:2 errors
    Information:0 warnings
    Information:See complete output in console

    There’s obviously something I’m not importing properly, but can’t figure exactly what’s missing.