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• Ffmpeg Android - Minimum binary size to convert WAV to MP3

  22 novembre 2020, par timson

  The only thing I want to do is convert wav files to mp3 inside my Android application.

  


  I am currently using https://github.com/tanersener/mobile-ffmpeg and with audio-release everything is working fine. As the lib size is about 40 MB and I only need a single command, I'd like to build my own .aar file as described in the Wiki to reduce the application size.

  


  I edited the android-ffmpeh.sh ./configure:

  


  --disable-everything \    
--enable-pthreads \
--enable-avcodec \
--enable-avformat \
--enable-swresample \
--enable-avfilter \
--enable-libmp3lame \
--enable-parser=mpegaudio \
--enable-demuxer=mp3,wav,pcm_s16le \
--enable-muxer=mp3,wav,pcm_s16le \
--enable-decoder=pcm*,mp3*,wav,pcm_s16le \
--enable-encoder=pcm*,pcm_s16le,wav,mp3,libmp3lame \
--enable-filter=aresample \
--enable-protocol=file \


  


  and then ran ./android.sh -l --enable-lame --enable-libiconv

  


  In my Android app FFmpeg loads but the conversion doesn't succed with following error :

  


  E/mobile-ffmpeg: [AVFilterGraph @ 0x7209dfec40] No such filter: 'anull'
E/mobile-ffmpeg: Error reinitializing filters!
E/mobile-ffmpeg: Failed to inject frame into filter network: Invalid argument
E/mobile-ffmpeg: Error while processing the decoded data for stream #0:0
I/mobile-ffmpeg: Conversion failed!


  


  Does anyone know what I'm missing or another config to build a minimal size binary for this.
Any help is highly appreciated !

  


• Is it possible to re-translate RTMP stream without losing speed ? [closed]

  3 août 2024, par Lunavod

  I've been working on a stream proxy - the idea is that instead of streaming directly to Twitch, OBS streams to a local RTMP server running on the same machine. The server decodes flv from the rtmp stream into rawvideo using ffmpeg, modifies pixels, and encodes back into flv, streaming the result to twitch. Again, using ffmpeg.

  


  However, I was not able to make this setup work reliably - I always run into buffering issues on Twitch. Even if ffmpeg shows a stable bitrate and 60fps, twitch slowly loses buffer size, then pauses to buffer, and then slowly loses buffer again... This results in endlessly growing delays and frequent pauses.

  


  I simplified this setup, removing the rawvideo part together with frame modification. A simplified setup accepts the rtmp stream, and dumps it into FFmpeg, which sends it to Twitch with minimal overhead (I hope).
But even with this setup, Twitch still increases latency, although considerably slower.

  


  The connection between rtmp server and ffmpeg is done with TCP sockets.
I tried using stdin, but it works even worse.
I also tried using windows named pipes but ran into a bottleneck - writing rawvideo from ffmpeg and reading it from script worked fine, as well as writing from a script and reading from ffmpeg. However, running both simultaneously in two different pipes slowed down.

  


  Initially, all of this was written in python, but I also tried using go, hoping that rtmp server realisation in python was the problem.

  


  Am I missing something fundamental here ? Is this idea possible at all ?

  


• "Error : more samples than frame size" while encoding audio to opus codec using FFMPEG

  28 avril 2023, par lokit khemka

  I am converting audio from codec AAC to Opus using libavcodec library of FFMPEG. The input codec details are as follows : Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 6 channels, fltp, 391 kb/s (default)

  


  The codec options that I have used for the output encoding are as follows :

  


      int OUTPUT_CHANNELS = 2;
    int OUTPUT_BIT_RATE = 32000;
int sample_rate = 48000;
    encoder_sc->audio_avcc->channels = OUTPUT_CHANNELS;
    encoder_sc->audio_avcc->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
    encoder_sc->audio_avcc->sample_rate = sample_rate;
    encoder_sc->audio_avcc->sample_fmt = encoder_sc->audio_avc->sample_fmts[0];
    encoder_sc->audio_avcc->bit_rate = OUTPUT_BIT_RATE;
    encoder_sc->audio_avcc->time_base = (AVRational){1, sample_rate};


  


  I am using the code in the file as it is, with minimal changes : https://github.com/leandromoreira/ffmpeg-libav-tutorial/blob/master/3_transcoding.c for reference. Look for the function prepare_audio_encoder in the file.

  


  When the run the program, I keep getting the error : " more samples than frame size". I don't know much about Audio Processing, so I cannot debug this error. Any help is greatly appreciated.