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• imdct15 : remove the AArch64 assembly

  4 janvier 2017, par Rostislav Pehlivanov

  imdct15 : remove the AArch64 assembly
  Prep work for the next commit, which will add a new FFT algorithm
  
  which makes the iMDCT over 3x faster than it is currently (standalone,
  
  the FFT is with some framesizes over 10x faster).
  The new FFT algorithm uses the already thouroughly SIMD’d power of two
  
  FFT which already has SIMD for AArch64, so users of that platform will
  
  still see an improvement.
  The previous FFT+SIMD was barely 2.5x faster than the C versions on these
  
  platforms.
  Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
  

  • [DH] libavcodec/aarch64/Makefile
  • [DH] libavcodec/aarch64/imdct15_init.c
  • [DH] libavcodec/aarch64/imdct15_neon.S
  • [DH] libavcodec/imdct15.c
  • [DH] libavcodec/imdct15.h

• How to 'convert' MP3 file to numpy array or list

  30 mai 2021, par Ajayi Olamide

  I'm working on an audio-related project that connects with Django backend via rest api. Part of the front-end requires to display waveforms of associated mp3 files and for this, it in turn requires optimized data of each mp3 file in form of an array, which the front-end (javascript) then processes and converts to a waveform. I can pick the associated mp3 file from backend storage, the problem is converting it into an array which I can serve to the front-end api. I have tried several methods but none seem to be working. I tried this How to read a MP3 audio file into a numpy array / save a numpy array to MP3 ? which leaves my computer hanging until I forced it to restart by holding the power button down. I have a working ffmpeg and so, I have also tried this Trying to convert an mp3 file to a Numpy Array, and ffmpeg just hangs which continues to raise TypeError on np.fromstring(data[data.find("data")+4:], np.int16). I can't actually say what the problem is and I really hope someone can help. Thank you in advance !

  


  EDIT
This is the django view for retrieving the waveform data :

  


  NB : I've only included useful codes as I'm typing with my mobile phone.

  


  def waveform(self, request, ptype, id):
    project = Project.objects.get(pk=id)
    audio = project.audio

    mp3_path = os.path.join(cdn_dir, audio) 
    cmd = ['ffmpeg', '-i', mp3_path, '-f', 'wav', '-']
    p = Popen(cmd, stdin=PIPE, stdout=PIPE, stderr=PIPE, creationflags=0x8000000)
    data = p.communicate()[0]
    array = np.fromstring(data[data.find("data")+4:], np.int16)

    return Response(array)


  


  The TypeError I get is this :
TypeError: argument should be integer or bytes-like object, not "str"

  


• Compile FFmpeg project for ARM in PC Linux 64-bits

  4 avril 2017, par Dang_Ho

  I want to compile a simple FFmpeg project for my Arrow Sockit Board with an arm-linux-gnueabihf architecture from my Linux-64bit PC. I don’t want to compile the project in the board directly because of my low CPU and that is not convenient for me.

  I’m using FFmpeg version 2.8.11 and this is my Makefile and my "main.c". I know, my Makefile has something wrong in it. If I command "make", it will compile depending on my PC’s architecture, I can’t use that binary file on my board. So, can someone please tell me how to do it.

  I Cross-Compiled the FFmpeg package and installed into the board. I tested all functions such as ffmpeg, ffplay. All them work. The source code folder is located to /home/hohaidang/ffmpeg-2.8.11

  #include 
  
  #include <libavcodec></libavcodec>avcodec.h>
  
  #include <libavformat></libavformat>avformat.h>
  
  #include <libswscale></libswscale>swscale.h>
  
  
  
  int main(int argc, char *argv[]){
  
          av_register_all();
  
          return 0;
  
  }

  # use pkg-config for getting CFLAGS and LDLIBS
  
  FFMPEG_LIBS=    libavdevice                        \
  
                  libavformat                        \
  
                  libavfilter                        \
  
                  libavcodec                         \
  
                  libswresample                      \
  
                  libswscale                         \
  
                  libavutil                          \
  
  
  
  CFLAGS += -Wall -g
  
  CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
  
  LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
  
  
  
  EXAMPLES=       main
  
  
  
  OBJS=$(addsuffix .o,$(EXAMPLES))
  
  
  
  # the following examples make explicit use of the math library
  
  avcodec:           LDLIBS += -lm
  
  decoding_encoding: LDLIBS += -lm
  
  muxing:            LDLIBS += -lm
  
  resampling_audio:  LDLIBS += -lm
  
  
  
  .phony: all clean-test clean
  
  
  
  all: $(OBJS) $(EXAMPLES)
  
  
  
  clean-test:
  
          $(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
  
  
  
  clean: clean-test
  
          $(RM) $(EXAMPLES) $(OBJS)