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• couldn't render an audio after extracting the audio from the video

  3 juin 2022, par Rama Ranneh

  hi … I am trying to use FFmpegFrameRecorder to extract an audio from a video using this code

  


          FFmpegFrameGrabber frameGrabber = new FFmpegFrameGrabber(mp4Path);
        frameGrabber.start();
        FFmpegFrameRecorder recorder = new FFmpegFrameRecorder(new 
        File("C:/Users/ASUS/Desktop/audio.aac"),audioChannels);
        recorder.setAudioMetadata(frameGrabber.getAudioMetadata());
        recorder.setFrameRate(frameGrabber.getFrameRate());
        recorder.setTimestamp(frameGrabber.getTimestamp());
        recorder.start(frameGrabber.getFormatContext());

        Frame f=null;
        //get audio sample and record it
        while ((f = frameGrabber.grabSamples()) != null) {
            recorder.record(f);
        }
        frameGrabber.stop();
        recorder.stop();
        recorder.close();
        LOGGER.info("C:/Users/ASUS/Desktop/audio.aac");


  


  but when I run it it gives me in terminal :

  


  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/Users/ASUS/Desktop/Lou.mp4':
Metadata:
major_brand     : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
encoder         : Lavf58.76.100
Duration: 00:00:39.10, start: 0.000000, bitrate: 116 kb/s
Stream #0:0[0x1](und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709, 
progressive), 256x144 [SAR 1:1 DAR 16:9], 63 kb/s, 23.98 fps, 23.98 tbr, 90k tbn (default)
Metadata:
  handler_name    : VideoHandler
  vendor_id       : [0][0][0][0]
  Stream #0:1[0x2](und): Audio: aac (HE-AAC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 46 kb/s (default)
Metadata:
  handler_name    : SoundHandler
  vendor_id       : [0][0][0][0]
 [aac @ 000000001afd7080] noise_facs_q 31 is invalid
 [aac @ 000000001afd7080] noise_facs_q 31 is invalid
 [aac @ 000000001afd7080] noise_facs_q 31 is invalid
 [aac @ 000000001afd7080] noise_facs_q 31 is invalid
 [aac @ 000000001afd7080] noise_facs_q 31 is invalid
 [aac @ 000000001afd7080] noise_facs_q 31 is invalid
 [aac @ 000000001afd7080] noise_facs_q 31 is invalid
 [h264 @ 000000001ad8bc40] mmco: unref short failure
 [h264 @ 000000001ad8bc40] mmco: unref short failure
 [h264 @ 000000001ad8bc40] mmco: unref short failure
 [h264 @ 000000001ad8bc40] mmco: unref short failure
 [h264 @ 000000001ad8bc40] mmco: unref short failure
 and so on ...
  Output #0, adts, to 'C:\Users\ASUS\Desktop\aud.aac':
 Metadata:
 encoder         : Lavf59.16.100
  Stream #0:0: Audio: aac (HE-AAC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 46 kb/s


  


  and when I try to run the audio it is not running
what's the wrong thing I did ?

  


• avcodec/asvenc : Avoid reversing output data twice

  13 octobre 2020, par Andreas Rheinhardt

  avcodec/asvenc : Avoid reversing output data twice
  The ASUS V2 format is designed for a little-endian bitstream reader, yet
  
  our encoder used an ordinary big-endian bitstream writer to write it ;
  
  the bits of every byte were swapped at the end and some data (namely the
  
  numbers not in static tables) had to be bitreversed before writing it at
  
  all, so that it would be reversed twice.
  This commit stops doing so ; instead, a little-endian bitstream writer is
  
  used. This also necessitated to switch certain static tables, which
  
  required trivial modifications to the decoder (that uses the same
  
  tables).
  Reviewed-by : Michael Niedermayer <michael@niedermayer.cc>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavcodec/asv.c
  • [DH] libavcodec/asv.h
  • [DH] libavcodec/asvdec.c
  • [DH] libavcodec/asvenc.c

• FFmpeg - how to set output sample_size

  23 décembre 2019, par DennisJ

  Trying to create a simple command line player for .dsf (DSD audio) files, and output to an alsa device that supports up to 24-bit 192 kHz sample rate. The following command almost works and it does play the track. Examining the bold text below, the dsf input file is converted to 24-bit/192 kHz, but the output is then truncated to 16-bit 192 kHz (pcm_s16le i.e, 16 bit little endian).

  ffmpeg -i ’01 - Sweet Georgia Brown.dsf’ -f alsa hw:0,0

  After displaying the ffmpeg banner and song metadata (tags), here is the result, bold is my emphasis :

  Duration : 00:05:14.83, start : 0.000000, bitrate : 9234 kb/s
  Stream #0:0 : Audio : flac, 192000 Hz, stereo, s32 (24 bit)
  Stream mapping :
  Stream #0:0 -> #0:0 (flac (native) -> pcm_s16le (native))
  Press [q] to stop, [?] for help
  Output #0, alsa, to ’hw:0,0’ :

  Since I can play this and many other tracks at full resolution using another player (foobar2000) it seems there might be an option in the encoder which is part of FFmpeg : Lavf57.83.100 I can find no information in any of the FFmpeg documentation that helps. Tried finding options in FFplay and even guessing using other FFmpeg options like this example.
  ffmpeg -sample_fmt s24 -i ’01 - Sweet Georgia Brown.dsf’ -f alsa hw:0,0 ***** same results.

  I’m stuck. Any suggestions ?

  Environment : Linux Mint 19.2, 64-bit, ASUS Xonar STXii sound card.