Advanced search

On other websites (4352)

• Transfer audio with ffmpeg, unexpected silent at the beginning [duplicate]

  4 May 2018, by LouisCJT

  This question already has an answer here:

  • ffmpeg wrong audio file after conversion in AAC
    
    1 answer
    
  • Why does libmp3lame add zeros to the start of the MP3?
    
    1 answer
    

  I was using ffmpeg to transfer the format of audio files, like from .wav to mp3. The transfer seemed fine and the audio sounds as expected. However when I check its waveform, I found that after the transfer,

  1. A small portion of silent (and with some very tiny noise) is at the beginning of the output audio. The length is about 1 ms 0.001 sec.

  2. As the silent part is appended, the end of the audio is altered - the length of the output audio is longer than the input file. (my input is exactly 10 sec, now the output is around 10.1 sec.

  Need the transfer precise because need to do further analysis frame by frame, and this situation could not go well. I used the following command to transfer a file.

  ffmpeg -I ..\wav_1K_32bit_24576kbps_384000Hz_stereo.wav -vn -ar 12000 -ac 2 -ab 320000 -f MP3 MP3_12000Hz_32kbps_stereo_VBROff.MP3

  ---

  Please refer to the screen crop of waveform view.

  Beginning of audio, input (U) & output (L)

  End of audio, input (U) & output (L)

  Thank you!

• Why am I getting FPE when using swresample 1.1?

  24 January 2013, by mystafer

  I am building a project to view a video feed from an IP camera in Android using FFmpeg 1.1.

  I'm attempting to use swresample in an Android project and getting a floating point exception when calling swr_convert. I stepped through the swresample code and found one line in libswresample/swresample.c function swri_realloc_audio where the variables a->bps and a->ch_count are zero causing the FPE.

  int swri_realloc_audio(AudioData *a, int count){
  
      int i, countb;
  
      AudioData old;
  
  
  
      LOGD("in swri_realloc_audio - bps[%d], ch_count[%d]", a->bps, a->ch_count);
  
      if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
  
      return AVERROR(EINVAL);

  01-21 17:29:09.612: D/swresample.c(18789): in swri_realloc_audio - bps[0], ch_count[0]

  I found bug ticket #1834 in the FFmpeg project that sounds like the exact same issue, but it was resolved by calling swr_init. However, my code does call this function and still crashes. Here is my JNI code:

  SwrContext* resampleCtx = swr_alloc_set_opts(NULL,
  
          AV_CH_LAYOUT_MONO, AV_SAMPLE_FMT_S16, pAudioCodecCtx->sample_rate,
  
          pAudioCodecCtx->channel_layout, pAudioCodecCtx->sample_fmt,
  
          pAudioCodecCtx->sample_rate, 0, 0);
  
  
  
  swr_init(resampleCtx);
  
  LOGD("Resample context initialized");
  
  
  
  int dataSize = swr_convert(resampleCtx,
  
          &pAudioOutBuffer, AVCODEC_MAX_AUDIO_FRAME_SIZE / 2,
  
          (const uint8_t**) &(pFrame->data[0]), pFrame->nb_samples);
  
  LOGD("Resample conversion complete");
  
  
  
  swr_free(&resampleCtx);
  
  LOGD("Obtained data size - dataSize[%d]", dataSize);

  I'm confused because I don't seem to have any control over the variable a in the swri_realloc_audio function. I stepped through the code and noticed that it is from the variable resampleCtx->postin. This variable is copied from resampleCtx->in in the swr_init function, but I don't see where in is ever set to anything.

  What am I doing wrong? Is it in my code or is there a problem in swresample?

• How can I stream raw video frames AND audio to FFMPEG with Python 2.7?

  18 November 2017, by Just Askin

  I am streaming raw video frames from Pygame to FFMPEG, then sending to a rtmp stream, but for the life of me, I can’t figure out how to send live audio using the same Python module. It does not need to be the Pygame mixer, but I am not opposed to using it if that is where the best answer lies. I’m pretty sure it’s not though.

  My question is this: What is the best strategy to send live audio output from a program to FFMPEG along with raw video frames simultaneously from the same Python module?

  My program is large, and eventually I would like to build options to switch audio inputs from a queue of music, a microphone, or any other random sounds from any program I want to use. But for the time being, I just want something to work. I am starting off with a simple Espeak command.

  Here is my Python commands:

  command = ['ffmpeg', '-re', '-framerate', '22', '-s', '1280x720', '-pix_fmt', 'rgba', '-f', 'rawvideo', '-i', '-', '-f', 's16le', '-ar', '22500', '-i', '/tmp/audio', '-preset', ultrafast', '-pix_fmt', 'rgba', '-b:v', '2500', '-s', 'hd720', '-r', '25', '-g', '50', '-crf', '20', '-f', 'flv', 'rtmp://xxx' ]
  
  
  
  pipe = sp.Popen(command, stdin=sp.PIPE) 

  Then I send my frames to stdin from within my main while True: loop.

  The problem I run into with this strategy is I can’t figure out how to shove audio into FFMPEG from within Python without blocking the pipe. After hours of research, I am pretty confident I can’t use the pipe to send the audio along with the frames. I thought the named pipe was my solution (which works running Espeak outside of Python), but it blocks Python until the Espeak is done... so no good.

  I assume I need threading for multiprocessing, but I cannot figure out from the official documentation or any other resources as to how I can solve my problem with it.

  The ['-f', 's16le', '-ar', '22500', '-i', '/tmp/audio'] are settings that work if I run espeak from a separate terminal with espeak 'some text' --stdout > /tmp/audio.

  I am using Centos 7, Python 2.7, pygame, the latest build of FFMPEG,