Recherche avancée

Sur d’autres sites (7153)

• FFmpeg.swr_convert : audio to raw 16 bit pcm, to be used with xna SoundEffect. Audio cuts out when i convert

  21 mars 2019, par Robert Russell

  I want to resample mkv(vp8/ogg) and also raw 4 bit adpcm to raw 16bit pcm byte[] to be loaded into SoundEffect from xna library. So I can play it out while I’m using other code to display the frames (the video side is working).
  I can read a 16 bit wav file and play it. But when I goto resample something it doesn’t play 100%. One file is 3 mins and 15 secs. I only get 13 sec and 739 ms before it quits playing. I have been learning to do this by finding code samples in c++ and correcting it to work in c# using ffmpeg.autogen.

  the below is my best attempt at resampling.

              int nb_samples = Frame->nb_samples;
  
                      int output_nb_samples = nb_samples;
  
                      int nb_channels = ffmpeg.av_get_channel_layout_nb_channels(ffmpeg.AV_CH_LAYOUT_STEREO);
  
                      int bytes_per_sample = ffmpeg.av_get_bytes_per_sample(AVSampleFormat.AV_SAMPLE_FMT_S16) * nb_channels;
  
                      int bufsize = ffmpeg.av_samples_get_buffer_size(null, nb_channels, nb_samples,
  
                                                               AVSampleFormat.AV_SAMPLE_FMT_S16, 1);
  
  
  
                      byte*[] b = Frame->data;
  
                      fixed (byte** input = b)
  
                      {
  
                          byte* output = null;
  
                          ffmpeg.av_samples_alloc(&output, null,
  
                              nb_channels,
  
                              nb_samples,
  
                              (AVSampleFormat)Frame->format, 0);//
  
  
  
                          // Buffer input
  
  
  
                          Ret = ffmpeg.swr_convert(Swr, &output, output_nb_samples / 2, input, nb_samples);
  
                          CheckRet();
  
                          WritetoMs(output, 0, Ret * bytes_per_sample);
  
                          output_nb_samples -= Ret;
  
  
  
                          // Drain buffer
  
                          while ((Ret = ffmpeg.swr_convert(Swr, &output, output_nb_samples, null, 0)) > 0)
  
                          {
  
                              CheckRet();
  
                              WritetoMs(output, 0, Ret * bytes_per_sample);
  
                              output_nb_samples -= Ret;
  
                          }
  
                      }

  I changed that all to this but it cuts off sooner.

    Channels = ffmpeg.av_get_channel_layout_nb_channels(OutFrame->channel_layout);
  
                      int nb_channels = ffmpeg.av_get_channel_layout_nb_channels(ffmpeg.AV_CH_LAYOUT_STEREO);
  
                      int bytes_per_sample = ffmpeg.av_get_bytes_per_sample(AVSampleFormat.AV_SAMPLE_FMT_S16) * nb_channels;
  
  
  
                      if((Ret = ffmpeg.swr_convert_frame(Swr, OutFrame, Frame))>=0)
  
                          WritetoMs(*OutFrame->extended_data, 0, OutFrame->nb_samples * bytes_per_sample);
  
                      CheckRet();

  Both code use a function to set Swr it runs one time after the first frame is decoded.

          private void PrepareResampler()
  
      {
  
          ffmpeg.av_frame_copy_props(OutFrame, Frame);
  
          OutFrame->channel_layout = ffmpeg.AV_CH_LAYOUT_STEREO;
  
          OutFrame->format = (int)AVSampleFormat.AV_SAMPLE_FMT_S16;
  
          OutFrame->sample_rate = Frame->sample_rate;
  
          OutFrame->channels = 2;
  
          Swr = ffmpeg.swr_alloc();
  
          if (Swr == null)
  
              throw new Exception("SWR = Null");
  
          Ret = ffmpeg.swr_config_frame(Swr, OutFrame, Frame);
  
          CheckRet();
  
          Ret = ffmpeg.swr_init(Swr);
  
          CheckRet();
  
          Ret = ffmpeg.swr_is_initialized(Swr);
  
          CheckRet();
  
      }

  This is where I take the output and put it in the sound effect

  private void ReadAll()
  
      {
  
  
  
          using (Ms = new MemoryStream())
  
          {
  
              while (true)
  
              {
  
                  Ret = ffmpeg.av_read_frame(Format, Packet);
  
                  if (Ret == ffmpeg.AVERROR_EOF)
  
                      break;
  
                  CheckRet();
  
                  Decode();
  
              }
  
              if (Ms.Length > 0)
  
              {
  
                  se = new SoundEffect(Ms.ToArray(), 0, (int)Ms.Length, OutFrame->sample_rate, (AudioChannels)Channels, 0, 0);
  
                  //se.Duration; Stream->duration;
  
  
  
  
  
                  see = se.CreateInstance();
  
                  see.Play();
  
              } 
  
          }
  
      }

• html5 video wont play ANY mp4 encodes on ipad

  12 mars 2019, par nickg

  Im trying to embed a mp4 on a page with the HTML5 video tag. Everything works fine on Desktop but nothing will work on an iPad Version 9.3.4.
  I have the mime types in an .htaccess file. I’ve tried various encodes with handbrake, Miro and FFmpeg conversions.

  Even sample videos like at w3schools and videojs don’t play.
  The video will play if i actually sync it to the iPad, but nothing works over the web. An older iPad actually plays mp4s through the HTML5 video player.
  I’m ready to throw this POS iPad through a window.

   <video autoplay="false" width="320" height="240" controls="true">
  
    <source src="http://webnamehere.com/video/bunny.mp4" type="video/mp4; codecs=&quot;avc1.42E01E, mp4a.40.2&quot;">
  
  
  
  Your browser does not support the video tag.
  
  </source></video>

  Has anyone found a way to fix this ? Is there ANY encoding that this thing will actually play ? Thank you in advance for any help.

• Converting ffmpeg & ffprobe outputs to variables in an ffmpeg AWS Lambda Layer

  7 mars 2019, par Gracie

  I have two tasks I am trying to perform with the ffmpeg AWS Lambda layer.

  1) Convert an audio file from stereo to mono (with ffmpeg)

  2) Get the duration of the audio file and pass the result to a variable (with ffprobe)

  const { spawnSync } = require("child_process");
  
  const { readFileSync, writeFileSync, unlinkSync } = require("fs");
  
  const util = require('util');
  
  var fs = require('fs');
  
  
  
  exports.handler = (event, context, callback) => {
  
  
  
      // Windows 10 ffmpeg command to convert stereo to ffmpeg is
  
      // ffmpeg -i volando.flac -ac 1 volando-mono.flac
  
  
  
      // Convert from stereo to mono
  
      spawnSync(
  
          "/opt/bin/ffmpeg",
  
          [
  
              "-i",
  
              `volando.flac`,
  
              "-ac",
  
              "1",
  
              `/tmp/volando-mono.flac`
  
          ],
  
          { stdio: "inherit" }
  
      );
  
  
  
      //Pass result to a variable
  
      //var duration = stdio;
  
  
  
      //Read the content from the /tmp directory
  
      fs.readdir("/tmp/", function (err, data) {
  
          if (err) throw err;
  
          console.log('Contents of tmp file: ', data);
  
      });
  
  
  
      // Get duration of Flac file
  
      // Windows 10 ffmpeg command is 
  
      // ffprobe in.wav -show_entries stream=duration -select_streams a -of compact=p=0:nk=1 -v 0
  
      spawnSync(
  
          "/opt/bin/ffprobe",
  
          [
  
              `in.wav`,
  
              "-show_entries",
  
              "stream=duration",
  
              "-select_streams",
  
              "a",
  
              "-of",
  
              "compact=p=0:nk=1",
  
              "-v",
  
              "0"
  
          ],
  
          { stdio: "inherit" }
  
  
  
          //Pass result to a variable
  
          //var duration = stdio;
  
      );
  
  
  
  };

  Can anyone who has had success with this ffmpeg Lambda layer help get an output for these commands ?

  Here are some resources regarding the FFmpeg Lambda layer :

  https://serverless.com/blog/publish-aws-lambda-layers-serverless-framework/
  https://github.com/serverlesspub/ffmpeg-aws-lambda-layer
  https://devopstar.com/2019/01/28/serverless-watermark-using-aws-lambda-layers-ffmpeg/