Recherche avancée

Sur d’autres sites (3241)

• ffmpeg library pcm to ac3 encoding

  16 juillet 2014, par Dave Camp

  I’m new to the ffmpeg library and Im working on a custom directshow filter. I decided to use the ffmpeg library for the encoding of what I need to achieve. I’m a little confused with some of the parameters and the correct values with what ffmpeg is expecting.

  I’m currently working on the ac3 part of the custom filter.
  I’ve looked through the example of the encoding audio ( for MP2 encoding ) in the ffmpeg doc and I understand it, but I don’t understand how I should adapt it to my specific needs.

  The incoming samples are at 48K samples per second 16bit per sample and are stereo interleaved. The upstream filter is giving them to me at 25fps so I get an incoming ’audio sample packet’ of 1920 bytes for each audio frame. I want to encode that data into an ac3 data packet that I pass on to the next process that I’ll be doing myself.

  But I’m unsure of the correct parameters for each component in the following code...

  The code I have so far. There are several questions in the comments at key places.

  AVCodec*         g_pCodec = nullptr;
  
  AVCodecContext*  g_pContext = nullptr;
  
  AVFrame*         g_pFrame = nullptr;
  
  AVPacket         g_pPacket;
  
  LPVOID           g_pInSampleBuffer;
  
  
  
  avcodec_register_all();
  
  g_pCodec = avcodec_find_encoder(CODEC_ID_AC3);
  
  
  
  // What am I'm describing here? the incoming sample params or the outgoing sample params?
  
  // An educated guess is the outgoing sample params
  
  g_pContext = avcodec_alloc_context3(g_pCodec);
  
  g_pContext->bit_rate = 448000;
  
  g_pContext->sample_rate = 48000;
  
  g_pContext->channels = 2;
  
  g_pContext->sample_fmt = AV_SAMPLE_FMT_FLTP;
  
  g_pContext->channel_layout = AV_CH_LAYOUT_STEREO;
  
  
  
  // And this is the incoming sample params?
  
  g_pFrame = av_frame_alloc();
  
  g_pFrame->nb_samples = 1920; ?? What figure is the codec expecting me to give it here? 1920 / bytes_per_sample? 
  
  g_pFrame->format = AV_SAMPLE_FMT_S16;
  
  g_pFrame->channel_layout = AV_CH_LAYOUT_STEREO;
  
  
  
  // I assume this going to give me the size of a buffer that I use to fill with my incoming samples? I get a dwSize of 15360 but my samples are only coming in at 1920, does this matter?
  
  dwSize = av_samples_get_buffer_size(nullptr,2,1920,AV_SAMPLE_FMT_S16,0);
  
  
  
  // do I need to use av_malloc and copy my samples into g_pInSampleBuffer or can I supply the address of my own buffer ( created outside of the libav framework ) ?
  
  g_pInSampleBuffer = (LPVOID)av_malloc(dwSize)
  
  avcodec_fill_audio_frame(g_pFrame,2,AV_SAMPLE_FMT_S16,(const uint8_t*)g_pInSampleBuffer,*dwSize,0);
  
  
  
  // Encoding loop - samples are given to me through a directshow interface - DSInbuffer is the buffer containing the incoming samples
  
  av_init_packet(&g_pPacket);
  
  g_pPacket.data = nullptr;
  
  g_pPacket.size = 0;
  
  
  
  int gotpacket = 0;
  
  int ok = avcodec_encode_audio2(g_pContext,&g_pPacket,g_pFrame,&gotpacket);
  
  if((ok == 0) && gotpacket){
  
     // I then copy from g_pPacket.data an amount of g_pPacket.size bytes into another directshow interface buffer that sends the encoded sample downstream.
  
  
  
      av_free_packet(&g_pPacket);
  
  }

  Currently it will crash at the avcodec_encode_audio2 call. If I change the format parameter to AV_SAMPLE_FMT_FLTP in the avcodec_fill_audio_frame call then it doesnt crash but it only encodes 1 frame of data and i get error -22 on the next frame. The pPacket.size parameter is 1792 ( 7 * 256 ) after the first avcode_encode_audio2 call.

  As I’m new to ffmpeg I’m sure it will probably be something quite straight forward that I’ve missed or I’m misunderstanding, and I’m confused at where the parameters for the incoming samples are and the outgoing samples.

  This is obviously and extract from the main function that I’ve created, and I’ve manually typed into the forum. If there are spelling mistakes that are by mistake here, then original code compiles and runs.

  Dave.

• Cutting and rejoining videos causes audio de-synchronization using ffmpeg [closed]

  10 décembre 2023, par Donotalo

  This is the original post. I'm not getting any answer there so I thought may be the programming community knows the answer.

  


  Main Post

  


  I'm trying to cut a video into pieces and rejoin them using ffmpeg on Windows 11 x64. Here's the details of ffmpeg :

  


  ffmpeg version 2023-11-22-git-0008e1c5d5-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg developers
  built with gcc 12.2.0 (Rev10, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --pkg-config=pkgconf --disable-w32threads
--disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp
--enable-bzlib --enable-lzma --enable-libsnappy --enable-zlib --enable-librist --enable-libsrt
--enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2
--enable-libaribb24 --enable-libaribcaption --enable-libdav1d --enable-libdavs2 --enable-libuavs3d
--enable-libzvbi --enable-librav1e --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl --enable-libopenjpeg --enable-libvpx
--enable-mediafoundation --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi
--enable-libharfbuzz --enable-liblensfun --enable-libvidstab --enable-libvmaf --enable-libzimg
--enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc
--enable-dxva2 --enable-d3d11va --enable-libvpl --enable-libshaderc --enable-vulkan --enable-libplacebo
--enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt
--enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame
--enable-libvo-amrwbenc --enable-libcodec2 --enable-libilbc --enable-libgsm --enable-libopencore-amrnb
--enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite
--enable-libmysofa --enable-librubberband --enable-libsoxr --enable-chromaprint
  libavutil      58. 32.100 / 58. 32.100
  libavcodec     60. 34.100 / 60. 34.100
  libavformat    60. 17.100 / 60. 17.100
  libavdevice    60.  4.100 / 60.  4.100
  libavfilter     9. 13.100 /  9. 13.100
  libswscale      7.  6.100 /  7.  6.100
  libswresample   4. 13.100 /  4. 13.100
  libpostproc    57.  4.100 / 57.  4.100


  


  I've a video file in MKV format. I've cut it into 3 pieces using the following command :

  


  ffmpeg.exe -ss 0:0:0 -i input.mkv -t 0:54:15 -c:v hevc_amf -b:v 3M -c:s mov_text seg-01.mp4
ffmpeg.exe -ss 0:54:29 -i input.mkv -t 0:35:35 -c:v hevc_amf -b:v 3M -c:s mov_text seg-02.mp4
ffmpeg.exe -ss 1:30:12 -i input.mkv -t 0:4:10 -c:v hevc_amf -b:v 3M -c:s mov_text seg-03.mp4


  


  The audio is copied in all 3 pieces like original. Now I'm joining them using the following command :

  


  ffmpeg.exe -y -f concat -safe 0 -i .\join.txt -c:v hevc_amf -c:a copy -c:s copy -fflags +genpts out.mp4


  


  where, join.txt is :

  


  file seg-01.mp4
file seg-02.mp4
file seg-03.mp4


  


  ffmpeg throws the following warning :

  


  [mp4 @ 000002a5cf9cc040] Non-monotonic DTS in output stream 0:1; previous: 156240896, current: 156240084; changing to 156240897.
This may result in incorrect timestamps in the output file.
[mp4 @ 000002a5cf9cc040] Non-monotonic DTS in output stream 0:1; previous: 258720980, current: 258720462; changing to 258720981.
This may result in incorrect timestamps in the output file.


  


  I've observed that audio is de-synchronized after the places where I cut.

  


  How to keep audio, video and subtitle synchronized after cutting and rejoining the video using ffmpeg ?

  


• arm : vp9itxfm : Do a simpler half/quarter idct16/idct32 when possible

  22 novembre 2016, par Martin Storsjö

  arm : vp9itxfm : Do a simpler half/quarter idct16/idct32 when possible
  This work is sponsored by, and copyright, Google.
  This avoids loading and calculating coefficients that we know will
  
  be zero, and avoids filling the temp buffer with zeros in places
  
  where we know the second pass won’t read.
  This gives a pretty substantial speedup for the smaller subpartitions.
  The code size increases from 12388 bytes to 19784 bytes.
  The idct16/32_end macros are moved above the individual functions ; the
  
  instructions themselves are unchanged, but since new functions are added
  
  at the same place where the code is moved from, the diff looks rather
  
  messy.
  Before :                              Cortex A7       A8       A9      A53
  
  vp9_inv_dct_dct_16x16_sub1_add_neon :     273.0    189.5    212.0    235.8
  
  vp9_inv_dct_dct_16x16_sub2_add_neon :    2102.1   1521.7   1736.2   1265.8
  
  vp9_inv_dct_dct_16x16_sub4_add_neon :    2104.5   1533.0   1736.6   1265.5
  
  vp9_inv_dct_dct_16x16_sub8_add_neon :    2484.8   1828.7   2014.4   1506.5
  
  vp9_inv_dct_dct_16x16_sub12_add_neon :   2851.2   2117.8   2294.8   1753.2
  
  vp9_inv_dct_dct_16x16_sub16_add_neon :   3239.4   2408.3   2543.5   1994.9
  
  vp9_inv_dct_dct_32x32_sub1_add_neon :     758.3    456.7    864.5    553.9
  
  vp9_inv_dct_dct_32x32_sub2_add_neon :   10776.7   7949.8   8567.7   6819.7
  
  vp9_inv_dct_dct_32x32_sub4_add_neon :   10865.6   8131.5   8589.6   6816.3
  
  vp9_inv_dct_dct_32x32_sub8_add_neon :   12053.9   9271.3   9387.7   7564.0
  
  vp9_inv_dct_dct_32x32_sub12_add_neon :  13328.3  10463.2  10217.0   8321.3
  
  vp9_inv_dct_dct_32x32_sub16_add_neon :  14176.4  11509.5  11018.7   9062.3
  
  vp9_inv_dct_dct_32x32_sub20_add_neon :  15301.5  12999.9  11855.1   9828.2
  
  vp9_inv_dct_dct_32x32_sub24_add_neon :  16482.7  14931.5  12650.1  10575.0
  
  vp9_inv_dct_dct_32x32_sub28_add_neon :  17589.5  15811.9  13482.8  11333.4
  
  vp9_inv_dct_dct_32x32_sub32_add_neon :  18696.2  17049.2  14355.6  12089.7
  After :
  
  vp9_inv_dct_dct_16x16_sub1_add_neon :     273.0    189.5    211.7    235.8
  
  vp9_inv_dct_dct_16x16_sub2_add_neon :    1203.5    998.2   1035.3    763.0
  
  vp9_inv_dct_dct_16x16_sub4_add_neon :    1203.5    998.1   1035.5    760.8
  
  vp9_inv_dct_dct_16x16_sub8_add_neon :    1926.1   1610.6   1722.1   1271.7
  
  vp9_inv_dct_dct_16x16_sub12_add_neon :   2873.2   2129.7   2285.1   1757.3
  
  vp9_inv_dct_dct_16x16_sub16_add_neon :   3221.4   2520.3   2557.6   2002.1
  
  vp9_inv_dct_dct_32x32_sub1_add_neon :     753.0    457.5    866.6    554.6
  
  vp9_inv_dct_dct_32x32_sub2_add_neon :    7554.6   5652.4   6048.4   4920.2
  
  vp9_inv_dct_dct_32x32_sub4_add_neon :    7549.9   5685.0   6046.9   4925.7
  
  vp9_inv_dct_dct_32x32_sub8_add_neon :    8336.9   6704.5   6604.0   5478.0
  
  vp9_inv_dct_dct_32x32_sub12_add_neon :  10914.0   9777.2   9240.4   7416.9
  
  vp9_inv_dct_dct_32x32_sub16_add_neon :  11859.2  11223.3   9966.3   8095.1
  
  vp9_inv_dct_dct_32x32_sub20_add_neon :  15237.1  13029.4  11838.3   9829.4
  
  vp9_inv_dct_dct_32x32_sub24_add_neon :  16293.2  14379.8  12644.9  10572.0
  
  vp9_inv_dct_dct_32x32_sub28_add_neon :  17424.3  15734.7  13473.0  11326.9
  
  vp9_inv_dct_dct_32x32_sub32_add_neon :  18531.3  17457.0  14298.6  12080.0
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] libavcodec/arm/vp9itxfm_neon.S