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• Adding a image resource over video file from sd card using ffmpeg or MediaMuxer for android

  18 septembre 2014, par Alin

  I am stuck in this area which I am not comfortable at all to work in.

  Here is what I did so far :

  • Made an Ubuntu VirtualBox machine
  • Downloaded latest ffmpeg version which is 2.3.3
  • Compiled ffmpeg to be compatible with armv7-a so in the end I get two folders : include and lib. In include I have the headers and in libs the *.so files (just as in http://www.roman10.net/how-to-build-ffmpeg-with-ndk-r9/)

  

  I have created a new android project and made a jni folder and this is how far I went... Even this, with all the struggle being new to linux and compiling took me almost a week to reach.

  Adding a watermark in ffmpeg I believe it is done on libavfilter ? I have to dig on this matter, however the original ffmpeg I need to translate into my project is :

  ffmpeg -i input.avi -i logo.png -filter_complex 'overlay=10:main_h-overlay_h-10' output.avi

  As far as I am studying now I need to do inside jni :

  • create a add_watermark.c file in which I need to somehow call the function that does the filter overlay call

  • create Android.mk to load this and the ffmpeg needed libraries

    LOCAL_PATH := $(call my-dir)

    include $(CLEAR_VARS)

    LOCAL_MODULE := add-watermark

    LOCAL_SRC_FILES := add-watermark.c

    LOCAL_LDLIBS := -llog -ljnigraphics -lz

    LOCAL_SHARED_LIBRARIES := libavformat libavcodec libswscale libavutil

    include $(BUILD_SHARED_LIBRARY)

    $(call import-module,ffmpeg-2.3.3/android/armv7-a)

  • create Application.mk

    APP_ABI := armeabi-v7a

    APP_PLATFORM := android-8

  • run ndk-build and use the generated libraries in my android project.

  I really need help on continuing, so every answer is received with great attention and pleasure.

  Later Edit :
  Would it be possible to somehow build ffmpeg.exe as a library and call its main with the exact same parameters as the original exe ? I do not want to run ffmpeg as a standalone executable, but have it integrated within the project. Something like http://www.roman10.net/how-to-port-ffmpeg-the-program-to-androidideas-and-thoughts/ What downsides would this approach have ?

  Later edit 2 : if this is possible by using MediaMuxer or other APIs added in android 4.3 I am open to it you sample codes are provided. I did look over the MediaCodec and MediaMuxer samples also Grafik and haven’t found a proper way to do what I wanted. I prefer ffmpeg approach better if it works

• FFMPEG GIF with transparency from png image sequence

  11 février 2020, par Nick S

  I’ve been trying to use ffmpeg to create a gif with a transparent background, but whenever the movement goes on top of the background, the pixels stay there. It’s a tree with a wind animation, this is how it ends up : https://i.imgur.com/pq4ArBG.png

  I first try to create the palette, and then the gif :

  ffmpeg -i Tree_%04d.png -vf palettegen=reserve_transparent=1 palette.png
  
  
  
  ffmpeg -framerate 30 -i Tree_%04d.png -i palette.png -lavfi paletteuse=alpha_threshold=128 treegif.gif

  It seems the previous frames simply stay there, but I can’t figure out how to dispose of them.

• Adjust Opus Ogg page size using Go

  15 mars 2024, par matthew

  Using the code from Go's Pion WebRTC library play-from-disk example I'm able to create a WebRTC connection and send audio from a local Ogg file to peers.

  


  The play-from-disk example README.md details how to first convert the page size of the Ogg file to 20,000 using ffmpeg, like so :

  


  ffmpeg -i $INPUT_FILE -c:a libopus -page_duration 20000 -vn output.ogg


  


  I'd like to make this same adjustment to Ogg data natively in Go, without using ffmpeg. How can this be done ?

  


  I've tried using Pion's oggreader and oggwriter, but using these requires deep Opus file format and RTP protocol knowledge that neither I nor ChatGPT seem to have.

  


  For additional context, I'm using a Text-to-Speech (TTS) API to generate my Ogg data as follows :

  


  req, err := http.NewRequest("POST", "https://api.openai.com/v1/audio/speech", bytes.NewBuffer([]byte(fmt.Sprintf(`{
    "model": "tts-1",
    "voice": "alloy",
    "input": %q,
    "response_format": "opus",
}`, text))))

req.Header.Add("Authorization", "Bearer "+token)
req.Header.Add("Content-Type", "application/json; charset=UTF-8")

client := &http.Client{}
resp, err := client.Do(req)



  


  As I'm trying to create a real-time audio app, ideally, I'd like to pipe the response to WebRTC performing the conversion on chunks as these are received so that peers can start to listen to the audio before it has been fully received on the server.