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• aarch64 : vp9itxfm : Do separate functions for half/quarter idct16 and idct32

  22 novembre 2016, par Martin Storsjö

  aarch64 : vp9itxfm : Do separate functions for half/quarter idct16 and idct32
  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 14740 bytes to 24292 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 :
  
  vp9_inv_dct_dct_16x16_sub1_add_neon :     236.7
  
  vp9_inv_dct_dct_16x16_sub2_add_neon :    1051.0
  
  vp9_inv_dct_dct_16x16_sub4_add_neon :    1051.0
  
  vp9_inv_dct_dct_16x16_sub8_add_neon :    1051.0
  
  vp9_inv_dct_dct_16x16_sub12_add_neon :   1387.4
  
  vp9_inv_dct_dct_16x16_sub16_add_neon :   1387.6
  
  vp9_inv_dct_dct_32x32_sub1_add_neon :     554.1
  
  vp9_inv_dct_dct_32x32_sub2_add_neon :    5198.5
  
  vp9_inv_dct_dct_32x32_sub4_add_neon :    5198.6
  
  vp9_inv_dct_dct_32x32_sub8_add_neon :    5196.3
  
  vp9_inv_dct_dct_32x32_sub12_add_neon :   6183.4
  
  vp9_inv_dct_dct_32x32_sub16_add_neon :   6174.3
  
  vp9_inv_dct_dct_32x32_sub20_add_neon :   7151.4
  
  vp9_inv_dct_dct_32x32_sub24_add_neon :   7145.3
  
  vp9_inv_dct_dct_32x32_sub28_add_neon :   8119.3
  
  vp9_inv_dct_dct_32x32_sub32_add_neon :   8118.7
  After :
  
  vp9_inv_dct_dct_16x16_sub1_add_neon :     236.7
  
  vp9_inv_dct_dct_16x16_sub2_add_neon :     640.8
  
  vp9_inv_dct_dct_16x16_sub4_add_neon :     639.0
  
  vp9_inv_dct_dct_16x16_sub8_add_neon :     842.0
  
  vp9_inv_dct_dct_16x16_sub12_add_neon :   1388.3
  
  vp9_inv_dct_dct_16x16_sub16_add_neon :   1389.3
  
  vp9_inv_dct_dct_32x32_sub1_add_neon :     554.1
  
  vp9_inv_dct_dct_32x32_sub2_add_neon :    3685.5
  
  vp9_inv_dct_dct_32x32_sub4_add_neon :    3685.1
  
  vp9_inv_dct_dct_32x32_sub8_add_neon :    3684.4
  
  vp9_inv_dct_dct_32x32_sub12_add_neon :   5312.2
  
  vp9_inv_dct_dct_32x32_sub16_add_neon :   5315.4
  
  vp9_inv_dct_dct_32x32_sub20_add_neon :   7154.9
  
  vp9_inv_dct_dct_32x32_sub24_add_neon :   7154.5
  
  vp9_inv_dct_dct_32x32_sub28_add_neon :   8126.6
  
  vp9_inv_dct_dct_32x32_sub32_add_neon :   8127.2
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DBH] libavcodec/aarch64/vp9itxfm_neon.S

• ffmpeg php get frame / image after half time of the video or after 40 seconds

  6 mars 2016, par sonam Sharma

  hello all i am having this code in my php and i want to get an image after 50% of the duration of the video and also get the duration of the video in a variable i have installed ffmpeg in my php and computer

  the page has following code

  require 'vendor/autoload.php';
  
          //ececute ffmpeg generate mp4
  
          exec('ffmpeg -i '.$uploadfile.' -f mp4 -s 896x504 '.$new_flv.'');
  
          //execute ffmpeg and create thumb
  
          exec('ffmpeg  -i '.$uploadfile.' -f mjpeg -vframes 71 -s 768x432 -an '.$new_image_path.'');

  i want an image after 50% of the duration of the video and also store the video duration in a variable

  please give me some suggestions i am stuck over here

• FFmpeg transcoded sound (AAC) stops after half video time

  17 août 2015, par TheSHEEEP

  I have a strange problem in my C/C++ FFmpeg transcoder, which takes an input MP4 (varying input codecs) and produces and output MP4 (x264, baseline & AAC LC @44100 sample rate with libfdk_aac) :

  The resulting mp4 video has fine images (x264) and the audio (AAC LC) works fine as well, but is only played until exactly the half of the video.

  The audio is not slowed down, not stretched and doesn’t stutter. It just stops right in the middle of the video.

  One hint may be that the input file has a sample rate of 22050 and 22050/44100 is 0.5, but I really don’t get why this would make the sound just stop after half the time. I’d expect such an error leading to sound being at the wrong speed. Everything works just fine if I don’t try to enforce 44100 and instead just use the incoming sample_rate.

  Another guess would be that the pts calculation doesn’t work. But the audio sounds just fine (until it stops) and I do exactly the same for the video part, where it works flawlessly. "Exactly", as in the same code, but "audio"-variables replaced with "video"-variables.

  FFmpeg reports no errors during the whole process. I also flush the decoders/encoders/interleaved_writing after all the package reading from the input is done. It works well for the video so I doubt there is much wrong with my general approach.

  Here are the functions of my code (stripped off the error handling & other class stuff) :

  AudioCodecContext Setup

  outContext->_audioCodec = avcodec_find_encoder(outContext->_audioTargetCodecID);
  
  outContext->_audioStream = 
  
          avformat_new_stream(outContext->_formatContext, outContext->_audioCodec);
  
  outContext->_audioCodecContext = outContext->_audioStream->codec;
  
  outContext->_audioCodecContext->channels = 2;
  
  outContext->_audioCodecContext->channel_layout = av_get_default_channel_layout(2);
  
  outContext->_audioCodecContext->sample_rate = 44100;
  
  outContext->_audioCodecContext->sample_fmt = outContext->_audioCodec->sample_fmts[0];
  
  outContext->_audioCodecContext->bit_rate = 128000;
  
  outContext->_audioCodecContext->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
  
  outContext->_audioCodecContext->time_base = 
  
          (AVRational){1, outContext->_audioCodecContext->sample_rate};
  
  outContext->_audioStream->time_base = (AVRational){1, outContext->_audioCodecContext->sample_rate};
  
  int retVal = avcodec_open2(outContext->_audioCodecContext, outContext->_audioCodec, NULL);

  Resampler Setup

  outContext->_audioResamplerContext = 
  
          swr_alloc_set_opts( NULL, outContext->_audioCodecContext->channel_layout,
  
                              outContext->_audioCodecContext->sample_fmt,
  
                              outContext->_audioCodecContext->sample_rate,
  
                              _inputContext._audioCodecContext->channel_layout,
  
                              _inputContext._audioCodecContext->sample_fmt,
  
                              _inputContext._audioCodecContext->sample_rate,
  
                              0, NULL);
  
  int retVal = swr_init(outContext->_audioResamplerContext);

  Decoding

  decodedBytes = avcodec_decode_audio4(   _inputContext._audioCodecContext, 
  
                                          _inputContext._audioTempFrame, 
  
                                          &p_gotAudioFrame, &_inputContext._currentPacket);

  Converting (only if decoding produced a frame, of course)

  int retVal = swr_convert(   outContext->_audioResamplerContext, 
  
                              outContext->_audioConvertedFrame->data, 
  
                              outContext->_audioConvertedFrame->nb_samples, 
  
                              (const uint8_t**)_inputContext._audioTempFrame->data, 
  
                              _inputContext._audioTempFrame->nb_samples);

  Encoding (only if decoding produced a frame, of course)

  outContext->_audioConvertedFrame->pts = 
  
          av_frame_get_best_effort_timestamp(_inputContext._audioTempFrame);
  
  
  
  // Init the new packet
  
  av_init_packet(&outContext->_audioPacket);
  
  outContext->_audioPacket.data = NULL;
  
  outContext->_audioPacket.size = 0;
  
  
  
  // Encode
  
  int retVal = avcodec_encode_audio2( outContext->_audioCodecContext, 
  
                                      &outContext->_audioPacket, 
  
                                      outContext->_audioConvertedFrame,
  
                                      &p_gotPacket);
  
  
  
  
  
  // Set pts/dts time stamps for writing interleaved
  
  av_packet_rescale_ts(   &outContext->_audioPacket, 
  
                          outContext->_audioCodecContext->time_base,
  
                          outContext->_audioStream->time_base);
  
  outContext->_audioPacket.stream_index = outContext->_audioStream->index;

  Writing (only if encoding produced a packet, of course)

  int retVal = av_interleaved_write_frame(outContext->_formatContext, &outContext->_audioPacket);

  I am quite out of ideas about what would cause such a behaviour.