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• Java.lang.UnsatisfiedLinkError : No implementation found for int [duplicate]

  28 mars 2017, par Muthukumar Subramaniam

  This question already has an answer here :

  • Android NDK C++ JNI (no implementation found for native…)
    
    6 answers
    

  I executed youtube watch me android application project. I just add some classes in my project and build with ndk. I got the error like

  java.lang.UnsatisfiedLinkError : No implementation found for int com.ephronsystem.mobilizerapp.Ffmpeg.encodeVideoFrame(byte[]) (tried Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame and Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame___3B).

  My code :

  package com.ephronsystem.mobilizerapp;
  
  
  
  public class Ffmpeg {
  
  
  
       static {
  
          System.loadLibrary("ffmpeg");
  
      }
  
  
  
      public static native boolean init(int width, int height, int audio_sample_rate, String rtmpUrl);
  
  
  
      public static native void shutdown();
  
  
  
      // Returns the size of the encoded frame.
  
      public static native int encodeVideoFrame(byte[] yuv_image);
  
  
  
      public static native int encodeAudioFrame(short[] audio_data, int length);
  
  }

  This is ffmpeg-jni.c

   #include <android></android>log.h>
  
  #include 
  
  #include 
  
  #include "libavcodec/avcodec.h"
  
  #include "libavformat/avformat.h"
  
  #include "libavutil/opt.h"
  
  
  
  #ifdef __cplusplus
  
  extern "C" {
  
  #endif
  
  
  
  JNIEXPORT jboolean JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_init(JNIEnv *env, jobject thiz,
  
                                                                   jint width, jint height,
  
                                                                   jint audio_sample_rate,
  
                                                                   jstring rtmp_url);
  
  JNIEXPORT void JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_shutdown(JNIEnv *env,
  
  jobject thiz
  
  );
  
  JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame(JNIEnv
  
  *env,
  
  jobject thiz,
  
          jbyteArray
  
  yuv_image);
  
  JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeAudioFrame(JNIEnv *env,
  
                                                                               jobject thiz,
  
                                                                               jshortArray audio_data,
  
                                                                               jint length);
  
  
  
  #ifdef __cplusplus
  
  }
  
  #endif
  
  
  
  #define LOGI(...) __android_log_print(ANDROID_LOG_INFO, "ffmpeg-jni", __VA_ARGS__)
  
  #define URL_WRONLY 2
  
          static AVFormatContext *fmt_context;
  
          static AVStream *video_stream;
  
          static AVStream *audio_stream;
  
  
  
          static int pts
  
  = 0;
  
  static int last_audio_pts = 0;
  
  
  
  // Buffers for UV format conversion
  
  static unsigned char *u_buf;
  
  static unsigned char *v_buf;
  
  
  
  static int enable_audio = 1;
  
  static int64_t audio_samples_written = 0;
  
  static int audio_sample_rate = 0;
  
  
  
  // Stupid buffer for audio samples. Not even a proper ring buffer
  
  #define AUDIO_MAX_BUF_SIZE 16384  // 2x what we get from Java
  
  static short audio_buf[AUDIO_MAX_BUF_SIZE];
  
  static int audio_buf_size = 0;
  
  
  
  void AudioBuffer_Push(const short *audio, int num_samples) {
  
      if (audio_buf_size >= AUDIO_MAX_BUF_SIZE - num_samples) {
  
          LOGI("AUDIO BUFFER OVERFLOW: %i + %i > %i", audio_buf_size, num_samples,
  
               AUDIO_MAX_BUF_SIZE);
  
          return;
  
      }
  
      for (int i = 0; i &lt; num_samples; i++) {
  
          audio_buf[audio_buf_size++] = audio[i];
  
      }
  
  }
  
  
  
  int AudioBuffer_Size() { return audio_buf_size; }
  
  
  
  short *AudioBuffer_Get() { return audio_buf; }
  
  
  
  void AudioBuffer_Pop(int num_samples) {
  
      if (num_samples > audio_buf_size) {
  
          LOGI("Audio buffer Pop WTF: %i vs %i", num_samples, audio_buf_size);
  
          return;
  
      }
  
      memmove(audio_buf, audio_buf + num_samples, num_samples * sizeof(short));
  
      audio_buf_size -= num_samples;
  
  }
  
  
  
  void AudioBuffer_Clear() {
  
      memset(audio_buf, 0, sizeof(audio_buf));
  
      audio_buf_size = 0;
  
  }
  
  
  
  static void log_callback(void *ptr, int level, const char *fmt, va_list vl) {
  
      char x[2048];
  
      vsnprintf(x, 2048, fmt, vl);
  
      LOGI(x);
  
  }
  
  
  
  JNIEXPORT jboolean JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_init(JNIEnv *env, jobject thiz,
  
                                                                   jint width, jint height,
  
                                                                   jint audio_sample_rate_param,
  
                                                                   jstring rtmp_url) {
  
      avcodec_register_all();
  
      av_register_all();
  
      av_log_set_callback(log_callback);
  
  
  
      fmt_context = avformat_alloc_context();
  
      AVOutputFormat *ofmt = av_guess_format("flv", NULL, NULL);
  
      if (ofmt) {
  
          LOGI("av_guess_format returned %s", ofmt->long_name);
  
      } else {
  
          LOGI("av_guess_format fail");
  
          return JNI_FALSE;
  
      }
  
  
  
      fmt_context->oformat = ofmt;
  
      LOGI("creating video stream");
  
      video_stream = av_new_stream(fmt_context, 0);
  
  
  
      if (enable_audio) {
  
          LOGI("creating audio stream");
  
          audio_stream = av_new_stream(fmt_context, 1);
  
      }
  
  
  
      // Open Video Codec.
  
      // ======================
  
      AVCodec *video_codec = avcodec_find_encoder(AV_CODEC_ID_H264);
  
      if (!video_codec) {
  
          LOGI("Did not find the video codec");
  
          return JNI_FALSE;  // leak!
  
      } else {
  
          LOGI("Video codec found!");
  
      }
  
      AVCodecContext *video_codec_ctx = video_stream->codec;
  
      video_codec_ctx->codec_id = video_codec->id;
  
      video_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
  
      video_codec_ctx->level = 31;
  
  
  
      video_codec_ctx->width = width;
  
      video_codec_ctx->height = height;
  
      video_codec_ctx->pix_fmt = PIX_FMT_YUV420P;
  
      video_codec_ctx->rc_max_rate = 0;
  
      video_codec_ctx->rc_buffer_size = 0;
  
      video_codec_ctx->gop_size = 12;
  
      video_codec_ctx->max_b_frames = 0;
  
      video_codec_ctx->slices = 8;
  
      video_codec_ctx->b_frame_strategy = 1;
  
      video_codec_ctx->coder_type = 0;
  
      video_codec_ctx->me_cmp = 1;
  
      video_codec_ctx->me_range = 16;
  
      video_codec_ctx->qmin = 10;
  
      video_codec_ctx->qmax = 51;
  
      video_codec_ctx->keyint_min = 25;
  
      video_codec_ctx->refs = 3;
  
      video_codec_ctx->trellis = 0;
  
      video_codec_ctx->scenechange_threshold = 40;
  
      video_codec_ctx->flags |= CODEC_FLAG_LOOP_FILTER;
  
      video_codec_ctx->me_method = ME_HEX;
  
      video_codec_ctx->me_subpel_quality = 6;
  
      video_codec_ctx->i_quant_factor = 0.71;
  
      video_codec_ctx->qcompress = 0.6;
  
      video_codec_ctx->max_qdiff = 4;
  
      video_codec_ctx->time_base.den = 10;
  
      video_codec_ctx->time_base.num = 1;
  
      video_codec_ctx->bit_rate = 3200 * 1000;
  
      video_codec_ctx->bit_rate_tolerance = 0;
  
      video_codec_ctx->flags2 |= 0x00000100;
  
  
  
      fmt_context->bit_rate = 4000 * 1000;
  
  
  
      av_opt_set(video_codec_ctx, "partitions", "i8x8,i4x4,p8x8,b8x8", 0);
  
      av_opt_set_int(video_codec_ctx, "direct-pred", 1, 0);
  
      av_opt_set_int(video_codec_ctx, "rc-lookahead", 0, 0);
  
      av_opt_set_int(video_codec_ctx, "fast-pskip", 1, 0);
  
      av_opt_set_int(video_codec_ctx, "mixed-refs", 1, 0);
  
      av_opt_set_int(video_codec_ctx, "8x8dct", 0, 0);
  
      av_opt_set_int(video_codec_ctx, "weightb", 0, 0);
  
  
  
      if (fmt_context->oformat->flags &amp; AVFMT_GLOBALHEADER)
  
          video_codec_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
      LOGI("Opening video codec");
  
      AVDictionary *vopts = NULL;
  
      av_dict_set(&amp;vopts, "profile", "main", 0);
  
      //av_dict_set(&amp;vopts, "vprofile", "main", 0);
  
      av_dict_set(&amp;vopts, "rc-lookahead", 0, 0);
  
      av_dict_set(&amp;vopts, "tune", "film", 0);
  
      av_dict_set(&amp;vopts, "preset", "ultrafast", 0);
  
      av_opt_set(video_codec_ctx->priv_data, "tune", "film", 0);
  
      av_opt_set(video_codec_ctx->priv_data, "preset", "ultrafast", 0);
  
      av_opt_set(video_codec_ctx->priv_data, "tune", "film", 0);
  
      int open_res = avcodec_open2(video_codec_ctx, video_codec, &amp;vopts);
  
      if (open_res &lt; 0) {
  
          LOGI("Error opening video codec: %i", open_res);
  
          return JNI_FALSE;   // leak!
  
      }
  
  
  
      // Open Audio Codec.
  
      // ======================
  
  
  
      if (enable_audio) {
  
          AudioBuffer_Clear();
  
          audio_sample_rate = audio_sample_rate_param;
  
          AVCodec *audio_codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
  
          if (!audio_codec) {
  
              LOGI("Did not find the audio codec");
  
              return JNI_FALSE;  // leak!
  
          } else {
  
              LOGI("Audio codec found!");
  
          }
  
          AVCodecContext *audio_codec_ctx = audio_stream->codec;
  
          audio_codec_ctx->codec_id = audio_codec->id;
  
          audio_codec_ctx->codec_type = AVMEDIA_TYPE_AUDIO;
  
          audio_codec_ctx->bit_rate = 128000;
  
          audio_codec_ctx->bit_rate_tolerance = 16000;
  
          audio_codec_ctx->channels = 1;
  
          audio_codec_ctx->profile = FF_PROFILE_AAC_LOW;
  
          audio_codec_ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
  
          audio_codec_ctx->sample_rate = 44100;
  
  
  
          LOGI("Opening audio codec");
  
          AVDictionary *opts = NULL;
  
          av_dict_set(&amp;opts, "strict", "experimental", 0);
  
          open_res = avcodec_open2(audio_codec_ctx, audio_codec, &amp;opts);
  
          LOGI("audio frame size: %i", audio_codec_ctx->frame_size);
  
  
  
          if (open_res &lt; 0) {
  
              LOGI("Error opening audio codec: %i", open_res);
  
              return JNI_FALSE;   // leak!
  
          }
  
      }
  
  
  
      const jbyte *url = (*env)->GetStringUTFChars(env, rtmp_url, NULL);
  
  
  
      // Point to an output file
  
      if (!(ofmt->flags &amp; AVFMT_NOFILE)) {
  
          if (avio_open(&amp;fmt_context->pb, url, URL_WRONLY) &lt; 0) {
  
              LOGI("ERROR: Could not open file %s", url);
  
              return JNI_FALSE;  // leak!
  
          }
  
      }
  
      (*env)->ReleaseStringUTFChars(env, rtmp_url, url);
  
  
  
      LOGI("Writing output header.");
  
      // Write file header
  
      if (avformat_write_header(fmt_context, NULL) != 0) {
  
          LOGI("ERROR: av_write_header failed");
  
          return JNI_FALSE;
  
      }
  
  
  
      pts = 0;
  
      last_audio_pts = 0;
  
      audio_samples_written = 0;
  
  
  
      // Initialize buffers for UV format conversion
  
      int frame_size = video_codec_ctx->width * video_codec_ctx->height;
  
      u_buf = (unsigned char *) av_malloc(frame_size / 4);
  
      v_buf = (unsigned char *) av_malloc(frame_size / 4);
  
  
  
      LOGI("ffmpeg encoding init done");
  
      return JNI_TRUE;
  
  }
  
  
  
  JNIEXPORT void JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_shutdown(JNIEnv
  
  *env,
  
  jobject thiz
  
  ) {
  
  av_write_trailer(fmt_context);
  
  avio_close(fmt_context
  
  ->pb);
  
  avcodec_close(video_stream
  
  ->codec);
  
  if (enable_audio) {
  
  avcodec_close(audio_stream
  
  ->codec);
  
  }
  
  av_free(fmt_context);
  
  av_free(u_buf);
  
  av_free(v_buf);
  
  
  
  fmt_context = NULL;
  
  u_buf = NULL;
  
  v_buf = NULL;
  
  }
  
  
  
  JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeVideoFrame(JNIEnv
  
  *env,
  
  jobject thiz,
  
          jbyteArray
  
  yuv_image) {
  
  int yuv_length = (*env)->GetArrayLength(env, yuv_image);
  
  unsigned char *yuv_data = (*env)->GetByteArrayElements(env, yuv_image, 0);
  
  
  
  AVCodecContext *video_codec_ctx = video_stream->codec;
  
  //LOGI("Yuv size: %i w: %i h: %i", yuv_length, video_codec_ctx->width, video_codec_ctx->height);
  
  
  
  int frame_size = video_codec_ctx->width * video_codec_ctx->height;
  
  
  
  const unsigned char *uv = yuv_data + frame_size;
  
  
  
  // Convert YUV from NV12 to I420. Y channel is the same so we don't touch it,
  
  // we just have to deinterleave UV.
  
  for (
  
  int i = 0;
  
  i &lt; frame_size / 4; i++) {
  
  v_buf[i] = uv[i * 2];
  
  u_buf[i] = uv[i * 2 + 1];
  
  }
  
  
  
  AVFrame source;
  
  memset(&amp;source, 0, sizeof(AVFrame));
  
  source.data[0] =
  
  yuv_data;
  
  source.data[1] =
  
  u_buf;
  
  source.data[2] =
  
  v_buf;
  
  source.linesize[0] = video_codec_ctx->
  
  width;
  
  source.linesize[1] = video_codec_ctx->width / 2;
  
  source.linesize[2] = video_codec_ctx->width / 2;
  
  
  
  // only for bitrate regulation. irrelevant for sync.
  
  source.
  
  pts = pts;
  
  pts++;
  
  
  
  int out_length = frame_size + (frame_size / 2);
  
  unsigned char *out = (unsigned char *) av_malloc(out_length);
  
  int compressed_length = avcodec_encode_video(video_codec_ctx, out, out_length, &amp;source);
  
  
  
  (*env)->
  
  ReleaseByteArrayElements(env, yuv_image, yuv_data,
  
  0);
  
  
  
  // Write to file too
  
  if (compressed_length > 0) {
  
  AVPacket pkt;
  
  av_init_packet(&amp;pkt);
  
  pkt.
  
  pts = last_audio_pts;
  
  if (video_codec_ctx->coded_frame &amp;&amp; video_codec_ctx->coded_frame->key_frame) {
  
  pkt.flags |= 0x0001;
  
  }
  
  pkt.
  
  stream_index = video_stream->index;
  
  pkt.
  
  data = out;
  
  pkt.
  
  size = compressed_length;
  
  if (
  
  av_interleaved_write_frame(fmt_context,
  
  &amp;pkt) != 0) {
  
  LOGI("Error writing video frame");
  
  }
  
  } else {
  
  LOGI("??? compressed_length &lt;= 0");
  
  }
  
  
  
  last_audio_pts++;
  
  
  
  av_free(out);
  
  return
  
  compressed_length;
  
  }
  
  
  
  JNIEXPORT jint JNICALL Java_com_ephronsystem_mobilizerapp_Ffmpeg_encodeAudioFrame(JNIEnv
  
  *env,
  
  jobject thiz,
  
          jshortArray
  
  audio_data,
  
  jint length
  
  ) {
  
  if (!enable_audio) {
  
  return 0;
  
  }
  
  
  
  short *audio = (*env)->GetShortArrayElements(env, audio_data, 0);
  
  //LOGI("java audio buffer size: %i", length);
  
  
  
  AVCodecContext *audio_codec_ctx = audio_stream->codec;
  
  
  
  unsigned char *out = av_malloc(128000);
  
  
  
  AudioBuffer_Push(audio, length
  
  );
  
  
  
  int total_compressed = 0;
  
  while (
  
  
  
  AudioBuffer_Size()
  
  
  
  >= audio_codec_ctx->frame_size) {
  
  AVPacket pkt;
  
  av_init_packet(&amp;pkt);
  
  
  
  int compressed_length = avcodec_encode_audio(audio_codec_ctx, out, 128000,
  
                                               AudioBuffer_Get());
  
  
  
  total_compressed +=
  
  compressed_length;
  
  audio_samples_written += audio_codec_ctx->
  
  frame_size;
  
  
  
  int new_pts = (audio_samples_written * 1000) / audio_sample_rate;
  
  if (compressed_length > 0) {
  
  pkt.
  
  size = compressed_length;
  
  pkt.
  
  pts = new_pts;
  
  last_audio_pts = new_pts;
  
  //LOGI("audio_samples_written: %i  comp_length: %i   pts: %i", (int)audio_samples_written, (int)compressed_length, (int)new_pts);
  
  pkt.flags |= 0x0001;
  
  pkt.
  
  stream_index = audio_stream->index;
  
  pkt.
  
  data = out;
  
  if (
  
  av_interleaved_write_frame(fmt_context,
  
  &amp;pkt) != 0) {
  
  LOGI("Error writing audio frame");
  
  }
  
  }
  
  AudioBuffer_Pop(audio_codec_ctx
  
  ->frame_size);
  
  }
  
  
  
  (*env)->
  
  ReleaseShortArrayElements(env, audio_data, audio,
  
  0);
  
  
  
  av_free(out);
  
  return
  
  total_compressed;
  
  }

• SOLVED ffmpeg : consider increasing probesize error, but it is never satisfied

  27 octobre 2020, par Jaz

  I was trying to use an Arch solution for streaming to twitch today through FFMPEG, but all of my attempts were in vain because of one simple thing on FFMPEG. it says that the probesize is not large enough, so I instinctively increased the probesize value more and more... and now it is -probesize "500M" yet it is still saying it is not enough. here is the code snippet

  


  [x11grab @ 0x5631f846cd00] Stream #0: not enough frames to estimate rate; consider increasing probesize
Input #0, x11grab, from ':0.0':
  Duration: N/A, start: 1603397505.341400, bitrate: 1007124 kb/s
    Stream #0:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768, 1007124 kb/s, 30 fps, 1000k tbr, 1000k tbn, 1000k tbc
0: Input/output error


  


  and the code

  


  #!/bin/bash
     INRES="1366x768" # input resolution
     OUTRES="1366x768" # output resolution
     FPS="30" # target FPS
     GOP="60" # i-frame interval, should be double of FPS,
     GOPMIN="30" # min i-frame interval, should be equal to fps,
     THREADS="2" # max 6
     CBR="1000k" # constant bitrate (should be between 1000k - 3000k)
     QUALITY="ultrafast"  # one of the many FFMPEG preset
     AUDIO_RATE="44100"
     PROBESZ="500M" # specify a size for the ffmpeg tool to assess frames
     STREAM_KEY="$1" # paste the stream key after calling stream_now
     SERVER="live-mia" # twitch server in miami Florida, see https://stream.twitch.tv/ingests/ for list

     ffmpeg -f x11grab -s "$INRES" -r "$FPS" -i :0.0 -f pulse -i 0 -f flv -ac 2 -ar $AUDIO_RATE \
       -vcodec libx264 -g $GOP -keyint_min $GOPMIN -b:v $CBR -minrate $CBR -maxrate $CBR -pix_fmt yuv420p\
       -s $OUTRES -preset $QUALITY -tune film -acodec aac -threads $THREADS -strict normal \
       -bufsize $CBR -probesize $PROBESZ "rtmp://$SERVER.twitch.tv/app/$STREAM_KEY"


  


  even though it was a solution to store in .bashrc, I stored it in a script to call manually.

  


  and if this is helpful, here is the fancy banner ffmpeg shows before the error

  


  ffmpeg version n4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 10.1.0 (GCC)
  configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-avisynth --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librav1e --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100


  


• Encoding a growing video file in realtime fails prematurely

  17 janvier 2023, par Macster

  This batch script is repeatedly concatenating video clips from an textfile. The output file is then beeing encoded in realtime into dash format. Unfortunately the realtime encoding will always end prematurely and I can't figure out why. From what I observed, it shouldn't be possible that the realtime encoding would catch up to the concating - which is happening each time after the duration of the clip that was just added - because I'm setting an offset, to when the encoding has to start, via timeout.

  


  I've tried other formats like .mp4 and .h264 and other options, but nothing seems to help. So my assumption is, that there is a conflict when read/write operation is made and these operations overlap at a certain point. But how do I find out when and how to avoid it ? I haven't had the feeling that something was happening at the exact same time, when observing the command promt.

  


  
The screenshot was taken right at failing. As you can see, the concat file queue1.webm is already more than 10 seconds longer than the realtime encoding at its failing position. That's why I don't think it has to do with catching up too fast. It will fail randomly, so one time it fails at 25 seconds and next time it might fail at 2 minutes and 20 seconds.

  


  To avoid the possibility of different video settings causing troubble, I'm using only one video file. I will link it here : BigBuckBunny (Mega NZ) It's a 10 sec snippet from BigBuckBunny. I hope this is legal !? But you can use what ever clip you want.

  


  IMPORTANT : If you try to reproduce the behaviour, please make sure you make at least one entry,
  like file 'bigbuckbunny_webm.webm' in mylist.txt, because adding something if the file is empty is kinda broken :)

  


  So here is the code :

  


  Just the FFMPEG commands :

  


  ffmpeg -f concat -i "mylist.txt" -safe 0  -c copy -f webm -reset_timestamps 1 -streaming 1 -live 1 -y queue1.webm
[..]
ffmpeg -re -i queue1.webm -c copy -map 0:v -use_timeline 1 -use_template 1 -remove_at_exit 0 -window_size 10 -adaptation_sets "id=0,streams=v" -streaming 1 -live 1 -f dash -y queue.mpd


  


  makedir.bat

  


  @ECHO on

:: Create new queue
IF NOT EXIST "queue1.webm" mkfifo "queue1.webm"

setlocal EnableDelayedExpansion

set string=file 'bigbuckbunny_webm.webm'
set video_path=""
SET /a c=0
set file=-1
set file_before=""

:loop
::Get last entry from "mylist.txt"
for /f "delims=" %%a in ('type mylist.txt ^| findstr /b /c:"file"') do (
  set video_path=%%a
)
echo %video_path%

::Insert file 'bigbuckbunny_webm.webm' if mylist.txt is empty.
if "%video_path%" EQU """" (echo %string% >> mylist.txt && set file=%string:~6,-1%) else (set file=%video_path:~6,-1%)

::Insert file 'bigbuckbunny_webm.webm' into mylit.txt if actual entry(%file%) is the same than before(file 'bigbuckbunny_webm.webm').
if "%file%" EQU "%file_before%" (echo. >> mylist.txt && echo %string%>>mylist.txt) 

echo %file%

::Get the video duration
for /f "tokens=1* delims=:" %%a in ('ffmpeg -i %file% 2^>^&1 ^| findstr "Duration"') do (set duration=%%b)
echo %duration%

::Crop format to HH:MM:SS
set duration=%duration:~1,11%
echo %duration%

::Check if seconds are double digits, less than 10, like 09. Then use only 9.
if %duration:~6,1% EQU 0 (
  set /a sec=%duration:~7,1% 
    ) else ( 
        set /a sec=%duration:~6,2%

)
echo %sec%

::Convert duration into seconds
set /a duration=%duration:~0,2%*3600+%duration:~3,2%*60+%sec%
echo %duration%

::echo %duration%

::Increase iteration count.
set /a c=c+1

::Add new clip to queue.
ffmpeg -f concat -i "mylist.txt" -safe 0  -c copy -f webm -reset_timestamps 1 -streaming 1 -live 1 -y queue1.webm

::Start realtime encoding queue1, if a first clip was added.
if !c! EQU 1 (
  start cmd /k "startRealtimeEncoding.bat"
)

::Wait the duration of the inserted video 
timeout /t %duration%

::Set the actual filename as the previous file for the next iteration.
set file_before=%file%

::Stop after c loops.
if !c! NEQ 20 goto loop

echo %c%

endlocal

:end  


  


  startRealtimeEncoding.bat

  


  @ECHO off

timeout /t 5
ffmpeg -re -i queue1.webm -c copy -map 0:v -seg_duration 2 -keyint_min 48 -use_timeline 1 -use_template 1 -remove_at_exit 0 -window_size 10 -adaptation_sets "id=0,streams=v" -streaming 1 -live 1 -f dash -y queue.mpd

:end