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• ffmpeg bit_rate error - Android [duplicate]

  6 juin 2016, par Adroid Freak

  This question already has an answer here :

  • ffmpeg Too many bits per frame requested
    
    2 answers
    

  I’m getting the following error "maybe incorrect parameters such as bit_rate, rate, width or height" in some android phones, what am I doing wrong ?

  /data/data/com.exampleapp.android/app_bin/ffmpeg -y -i /storage/emulated/0/example/test/recording.mp4 -af asetrate=9600*4/3 -strict -2 /storage/emulated/0/example/media/example/test.m4a 
  
     libavutil      54.  7.100 / 54.  7.100
  
     libavcodec     56.  1.100 / 56.  1.100
  
     libavformat    56.  4.101 / 56.  4.101
  
     libavdevice    56.  0.100 / 56.  0.100
  
     libavfilter     5.  1.100 /  5.  1.100
  
     libswscale      3.  0.100 /  3.  0.100
  
     libswresample   1.  1.100 /  1.  1.100
  
     libpostproc    53.  0.100 / 53.  0.100
  
   Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/example/test/recording.mp4':
  
     Metadata:
  
       major_brand     : isom
  
       minor_version   : 0
  
       compatible_brands: isom3gp4
  
       creation_time   : 2016-06-06 00:51:05
  
     Duration: 00:00:03.33, start: 0.000000, bitrate: 14 kb/s
  
       Stream #0:0(eng): Audio: aac (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 12 kb/s (default)
  
       Metadata:
  
         creation_time   : 2016-06-06 00:51:05
  
         handler_name    : SoundHandle
  
   [aac @ 0x42136170] Too many bits per frame requested
  
   Output #0, ipod, to '/storage/emulated/0/example/media/example/test.m4a':
  
     Metadata:
  
       major_brand     : isom
  
       minor_version   : 0
  
       compatible_brands: isom3gp4
  
       Stream #0:0(eng): Audio: aac, 0 channels, 128 kb/s (default)
  
       Metadata:
  
         creation_time   : 2016-06-06 00:51:05
  
         handler_name    : SoundHandle
  
         encoder         : Lavc56.1.100 aac
  
   Stream mapping:
  
     Stream #0:0 -> #0:0 (aac (native) -> aac (native))
  
   Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
  
  exitValue 1

• Error decoding a simple audio file using FFmpeg library

  29 mars 2017, par satyres

  After successfuly compiling the latest version of FFmpeg library and generated .a library in Ubuntu I’ve been struggling now for more than a week to decode and play a simple mp3 file in Android without a success !
  I’ve followed this tutorial given by FFmpeg team in Github i’ve tried to use it in Android but no luck !
  here is the Native code.

  void Java_com_example_home_hellondk_MainActivity_audio_1decode_1example(JNIEnv * env, jobject obj, jstring file, jbyteArray array) {
  
      jboolean isfilenameCopy;
  
      const char * filename = ( * env) - > GetStringUTFChars(env, file, &
  
          isfilenameCopy);
  
      jclass cls = ( * env) - > GetObjectClass(env, obj);
  
      jmethodID play = ( * env) - > GetMethodID(env, cls, "playSound", "([BI)V");
  
      AVCodec * codec;
  
      AVCodecContext * c = NULL;
  
      int len;
  
      FILE * f, * outfile;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFrame * decoded_frame = NULL;
  
  
  
      av_init_packet( & avpkt);
  
  
  
      printf("Decode audio file %s \n", filename);
  
      LOGE("Decode audio file %s\n", filename);
  
      /* find the MPEG audio decoder */
  
      codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found\n");
  
          LOGE("Codec not found\n");
  
          exit(1);
  
      }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio codec context\n");
  
          LOGE("Could not allocate audio codec context\n");
  
          exit(1);
  
      }
  
  
  
      /* open it */
  
      if (avcodec_open2(c, codec, NULL) < 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
          LOGE("Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
          fprintf(stderr, "Could not open %s\n", filename);
  
          LOGE("Could not open %s\n", filename);
  
          exit(1);
  
      }
  
  
  
  
  
      /* decode until eof */
  
      avpkt.data = inbuf;
  
      avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
  
  
      while (avpkt.size > 0) {
  
          int i, ch;
  
          int got_frame = 0;
  
  
  
          if (!decoded_frame) {
  
              if (!(decoded_frame = av_frame_alloc())) {
  
                  fprintf(stderr, "Could not allocate audio frame\n");
  
                  LOGE("Could not allocate audio frame\n");
  
                  exit(1);
  
              }
  
          }
  
  
  
          len = avcodec_decode_audio4(c, decoded_frame, & got_frame, & avpkt);
  
          if (len < 0) {
  
              fprintf(stderr, "Error while decoding\n");
  
              LOGE("Error while decoding\n");
  
              exit(1);
  
          }
  
          if (got_frame) {
  
              /* if a frame has been decoded, output it */
  
              int data_size = av_get_bytes_per_sample(c - > sample_fmt);
  
              if (data_size < 0) {
  
                  /* This should not occur, checking just for paranoia */
  
                  fprintf(stderr, "Failed to calculate data size\n");
  
                  LOGE("Failed to calculate data size\n");
  
                  exit(1);
  
              }
  
              if (data_size > 0) {
  
  
  
                  jbyte * bytes = ( * env) - > GetByteArrayElements(env, array, NULL);
  
                  memcpy(bytes, decoded_frame, got_frame); //
  
                  ( * env) - > ReleaseByteArrayElements(env, array, bytes, 0);
  
                  ( * env) - > CallVoidMethod(env, obj, play, array, got_frame);
  
                  LOGE("DECODING ERROR5");
  
              }
  
          }
  
          avpkt.size -= len;
  
          avpkt.data += len;
  
          avpkt.dts =
  
              avpkt.pts = AV_NOPTS_VALUE;
  
          if (avpkt.size < AUDIO_REFILL_THRESH) {
  
              /* Refill the input buffer, to avoid trying to decode
  
               * incomplete frames. Instead of this, one could also use
  
               * a parser, or use a proper container format through
  
               * libavformat. */
  
              memmove(inbuf, avpkt.data, avpkt.size);
  
              avpkt.data = inbuf;
  
              len = fread(avpkt.data + avpkt.size, 1,
  
                  AUDIO_INBUF_SIZE - avpkt.size, f);
  
              if (len > 0)
  
                  avpkt.size += len;
  
          }
  
      }
  
  
  
  
  
      fclose(f);
  
  
  
      avcodec_free_context( & c);
  
      av_frame_free( & decoded_frame);
  
  
  
  }

  The Java code :

  package com.example.home.hellondk;
  
  
  
  import android.media.AudioFormat;
  
  import android.media.AudioManager;
  
  import android.media.AudioTrack;
  
  import android.media.MediaPlayer;
  
  import android.support.v7.app.AppCompatActivity;
  
  import android.os.Bundle;
  
  import android.util.Log;
  
  
  
  import java.io.File;
  
  import java.io.FileNotFoundException;
  
  import java.io.FileOutputStream;
  
  import java.io.IOException;
  
  
  
  public class MainActivity extends AppCompatActivity {
  
      static {
  
          System.loadLibrary("MyLibraryPlayer");
  
      }
  
      public native void createEngine();
  
  
  
      public native void audio_decode_example(String outfilename, byte[] array);
  
  
  
  
  
  
  
      private AudioTrack track;
  
      private FileOutputStream os;
  
  
  
      @Override
  
      protected void onCreate(Bundle savedInstanceState) {
  
          super.onCreate(savedInstanceState);
  
          setContentView(R.layout.activity_main);
  
          createEngine();
  
  
  
   /*        MediaPlayer mp = new MediaPlayer();
  
          mp.start();*/
  
  
  
          int bufSize = AudioTrack.getMinBufferSize(32000,
  
                  AudioFormat.CHANNEL_CONFIGURATION_STEREO,
  
                  AudioFormat.ENCODING_PCM_16BIT);
  
  
  
          track = new AudioTrack(AudioManager.STREAM_MUSIC,
  
                  32000,
  
                  AudioFormat.CHANNEL_CONFIGURATION_STEREO,
  
                  AudioFormat.ENCODING_PCM_16BIT,
  
                  bufSize,
  
                  AudioTrack.MODE_STREAM);
  
  
  
          byte[] bytes = new byte[bufSize];
  
  
  
          try {
  
              os = new FileOutputStream("/storage/emulated/0/Cloud Radio/a.out", false);
  
          } catch (FileNotFoundException e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }
  
     audio_decode_example("/storage/emulated/0/Cloud Radio/test.mp3", bytes);
  
  
  
      }
  
  
  
      void playSound(byte[] buf, int size) {
  
          //android.util.Log.v("ROHAUPT", "RAH Playing");
  
          if (track.getPlayState() != AudioTrack.PLAYSTATE_PLAYING)
  
              track.play();
  
          track.write(buf, 0, size);
  
  
  
          try {
  
              os.write(buf, 0, size);
  
          } catch (IOException e) {
  
              // TODO Auto-generated catch block
  
              e.printStackTrace();
  
          }
  
      }
  
  }

  I always got this error : Error while decoding .
  i’ve tried to change the decoder "AV_CODEC_ID_MP3" no sucess !
  Thank you so much for your help.
  Kind regards

• avformat/mxfdec : Fix Sign error in mxf_read_primer_pack()

  29 août 2017, par 孙浩(晓黑)

  avformat/mxfdec : Fix Sign error in mxf_read_primer_pack()
  Fixes : 20170829B.mxf
  Co-Author : 张洪亮(望初)" <wangchu.zhl@alibaba-inc.com>
  
  Found-by : Xiaohei and Wangchu from Alibaba Security Team
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/mxfdec.c