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• How to contribute to open source, for companies

  18 octobre 2010, par Dark Shikari — development, open source, x264

  I have seen many nigh-incomprehensible attempts by companies to contribute to open source projects, including x264. Developers are often simply boggled, wondering why the companies seem incapable of proper communication. The companies assume the developers are being unreceptive, while the developers assume the companies are being incompetent, idiotic, or malicious. Most of this seems to boil down to a basic lack of understanding of how open source works, resulting in a wide variety of misunderstandings. Accordingly, this post will cover the dos and don’ts of corporate contribution to open source.

  Do : contact the project using their preferred medium of communication.

  Most open source projects use public methods of communication, such as mailing lists and IRC. It’s not the end of the world if you mistakenly make contact with the wrong people or via the wrong medium, but be prepared to switch to the correct one once informed ! You may not be experienced using whatever form of communication the project uses, but if you refuse to communicate through proper channels, they will likely not be as inclined to assist you. Larger open source projects are often much like companies in that they have different parts to their organization with different roles. Don’t assume that everyone is a major developer !

  If you don’t know what to do, a good bet is often to just ask someone.

  Don’t : contact only one person.

  Open source projects are a communal effort. Major contributions are looked over by multiple developers and are often discussed by the community as a whole. Yet many companies tend to contact only a single person in lieu of dealing with the project proper. This has many flaws : to begin with, it forces a single developer (who isn’t paid by you) to act as your liaison, adding yet another layer between what you want and the people you want to talk to. Contribution to open source projects should not be a game of telephone.

  Of course, there are exceptions to this : sometimes a single developer is in charge of the entirety of some particular aspect of a project that you intend to contribute to, in which case this might not be so bad.

  Do : make clear exactly what it is you are contributing.

  Are you contributing code ? Development resources ? Money ? API documentation ? Make it as clear as possible, from the start ! How developers react, which developers get involved, and their expectations will depend heavily on what they think you are providing. Make sure their expectations match reality. Great confusion can result when they do not.

  This also applies in the reverse — if there’s something you need from the project, such as support or assistance with development of your patch, make that explicitly clear.

  Don’t : code dump.

  Code does not have intrinsic value : it is only useful as part of a working, living project. Most projects react very negatively to large “dumps” of code without associated human resources. That is, they expect you to work with them to finalize the code until it is ready to be committed. Of course, it’s better to work with the project from the start : this avoids the situation of writing 50,000 lines of code independently and then finding that half of it needs to be rewritten. Or, worse, writing an enormous amount of code only to find it completely unnecessary.

  Of course, the reverse option — keeping such code to yourself — is often even more costly, as it forces you to maintain the code instead of the official developers.

  Do : ignore trolls.

  As mentioned above, many projects use public communication methods — which, of course, allow anyone to communicate, by nature of being public. Not everyone on a project’s IRC or mailing list is necessarily qualified to officially represent the project. It is not too uncommon for a prospective corporate contributor to be turned off by the uninviting words of someone who isn’t even involved in the project due to assuming that they were. Make sure you’re dealing with the right people before making conclusions.

  Don’t : disappear.

  If you are going to try to be involved in a project, you need to stay in contact. We’ve had all too many companies who simply disappear after the initial introduction. Some tell us that we’ll need an NDA, then never provide it or send status updates. You may know why you’re not in contact — political issues at the company, product launch crunches, a nice vacation to the Bahamas — but we don’t ! If you disappear, we will assume that you gave up.

  Above all, don’t assume that being at a large successful company makes you immune to these problems. If anything, these problems seem to be the most common at the largest companies. I didn’t name any names in this post, but practically every single one of these rules has been violated at some point by companies looking to contribute to x264. In the larger scale of open source, these problems happen constantly. Don’t fall into the same traps that many other companies have.

  If you’re an open source developer reading this post, remember it next time you see a company acting seemingly nonsensically in an attempt to contribute : it’s quite possible they just don’t know what to do. And just because they’re doing it wrong doesn’t mean that it isn’t your responsibility to try to help them do it right.

• How to use audio frame after decode mp3 file using pyav, ffmpeg, python

  2 janvier 2021, par Long Tran Dai

  I am using using python with pyav, ffmpeg to decode mp3 in the memory. I know there are some other way to do it, like pipe ffmpeg command. However, I would like to explore pyav and ffmpeg API. So I have the following code. It works but the sound is very noisy, although hearable :

  


  import numpy as np&#xA;import av # to convert mp3 to wav using ffmpeg&#xA;import pyaudio # to play music&#xA;&#xA;mp3_path = &#x27;D:/MyProg/python/SauTimThiepHong.mp3&#x27;&#xA;&#xA;def decodeStream(mp3_path):&#xA;  # Run NOT OK&#xA;  &#xA;  container = av.open(mp3_path)&#xA;  stream = next(s for s in container.streams if s.type == &#x27;audio&#x27;)&#xA;  frame_count = 0&#xA;  data = bytearray()&#xA;  for packet in container.demux(stream):&#xA;    # <class>&#xA;    # We need to skip the "flushing" packets that `demux` generates.&#xA;    #if frame_count == 5000 : break         &#xA;    if packet.dts is None:&#xA;        continue&#xA;    for frame in packet.decode():   &#xA;        #&#xA;        # type(frame) : <class>&#xA;        #frame.samples = 1152 : 1152 diem du lieu : Number of audio samples (per channel)&#xA;        # moi frame co size = 1152 (diem) * 2 (channels) * 4 (bytes / diem) = 9216 bytes&#xA;        # 11021 frames&#xA;        #arr = frame.to_ndarray() # arr.nbytes = 9216&#xA;&#xA;        #channels = []  &#xA;        channels = frame.to_ndarray().astype("float16")&#xA;        #for plane in frame.planes:&#xA;            #channels.append(plane.to_bytes()) #plane has 4 bytes / sample, but audio has only 2 bytes&#xA;        #    channels.append(np.frombuffer(plane, dtype=np.single).astype("float16"))&#xA;            #channels.append(np.frombuffer(plane, dtype=np.single)) # kieu np.single co 4 bytes&#xA;        if not frame.is_corrupt:&#xA;            #data.extend(np.frombuffer(frame.planes[0], dtype=np.single).astype("float16")) # 1 channel: noisy&#xA;            # type(planes) : <class>&#xA;            frame_count &#x2B;= 1&#xA;            #print( &#x27;>>>> %04d&#x27; % frame_count, frame)   &#xA;            #if frame_count == 5000 : break     &#xA;            # mix channels:&#xA;            for i in range(frame.samples):                &#xA;                for ch in channels: # dec_ctx->channels&#xA;                    data.extend(ch[i]) #noisy&#xA;                    #fwrite(frame->data[ch] &#x2B; data_size*i, 1, data_size, outfile)&#xA;  return bytes(data)&#xA;</class></class></class>

  


  I use pipe ffmpeg to get decoded data to compare and find they are different :

  


  def RunFFMPEG(mp3_path, target_fs = "44100"):
    # Run OK
    import subprocess
    # init command
    ffmpeg_command = ["ffmpeg", "-i", mp3_path,
                   "-ab", "128k", "-acodec", "pcm_s16le", "-ac", "0", "-ar", target_fs, "-map",
                   "0:a", "-map_metadata", "-1", "-sn", "-vn", "-y",
                   "-f", "wav", "pipe:1"]
    # excute ffmpeg command
    pipe = subprocess.run(ffmpeg_command, stdout=subprocess.PIPE, stderr=subprocess.PIPE, bufsize= 10**8)
    # debug
    #print(pipe.stdout, pipe.stderr)
    # read signal as numpy array and assign sampling rate
    #audio_np = np.frombuffer(buffer=pipe.stdout, dtype=np.uint16, offset=44)
    #audio_np = np.frombuffer(buffer=pipe.stdout, dtype=np.uint16)
    #sig, fs  = audio_np, target_fs
    #return audio_np
    return pipe.stdout[78:]     


  


  Then I use pyaudio to play data and find it very noisy

  


  p = pyaudio.PyAudio()
streamOut = p.open(format=pyaudio.paInt16, channels=2, rate= 44100, output=True)
#streamOut = p.open(format=pyaudio.paInt16, channels=1, rate= 44100, output=True)

mydata = decodeStream(mp3_path)
print("bytes of mydata = ", len(mydata))
#print("bytes of mydata = ", mydata.nbytes)

ffMpegdata = RunFFMPEG(mp3_path)
print("bytes of ffMpegdata = ", len(ffMpegdata)) 
#print("bytes of ffMpegdata = ", ffMpegdata.nbytes)

minlen = min(len(mydata), len(ffMpegdata))
print("mydata == ffMpegdata", mydata[:minlen] == ffMpegdata[:minlen]) # ffMpegdata.tobytes()[:minlen] )

#bytes of mydata =  50784768
#bytes of ffMpegdata =  50784768
#mydata == ffMpegdata False

streamOut.write(mydata)
streamOut.write(ffMpegdata)
streamOut.stop_stream()
streamOut.close()
p.terminate()


  


  Please help me to understand decoded frame of pyav api (after for frame in packet.decode() :). Should it be processed more ? or I have some error ?

  


  It makes me crazy for 3 days. I could not guess where to go.

  


  Thank you very much.

  


• ffmpeg : Extracted wav from mp4 video does not have equal duration as the original video

  26 juillet 2021, par John Smith

  I have a mp4 video that is 0.92 seconds, and I am trying to extract the audio of the video to a wav format. I have tried several commands (I have provided a list of some of the commands that I have tried), however, the resulting wav does not have the same duration as the original video (the resulting wav often has a duration of 0.96 seconds instead of 0.92 seconds). Ensuring that the video and audio are synchronous is crucial for what I am doing (the videos are typically videos of a person speaking, and it is important that the speech (audio) is in-sync with the mouth movements of the speaker).

  


  I find it odd that, by extracting audio from a video, the duration changes, even despite what is happening under the hood for the conversion (in terms of codecs used, etc).

  


  Some of the commands that I've tried include :

  


  ffmpeg -i <input /> -c copy -map 0:a -sample_rate 16000 <output>&#xA;&#xA;ffmpeg -i <input /> -async  1 -f wav <output>&#xA;&#xA;ffmpeg -i <input /> -vn -acodec copy <output>&#xA;&#xA;ffmpeg -i <input /> -ac 2 -f wav <output>&#xA;</output></output></output></output>

  


  Any insight would be highly appreciated.
Thanks !

  


  Edit Output of the command ffmpeg -ignore_editlist true -i 00026.mp4 output.wav

  


  ffmpeg version 2021-02-28-git-85ab9deb98-full_build-www.gyan.dev Copyright (c) 2000-2021 the FFmpeg developers
  built with gcc 10.2.0 (Rev6, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma --enable-lib
snappy --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca --enable-sdl2 --enable-libdav1d --enable-libzvbi --enable-librav1e --enable-libsvtav1 --en
able-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi --enable-libvidstab --e
nable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-libglslang --enable-vulkan --
enable-opencl --enable-libcdio --enable-libgme --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame --enable-libvo-amrwben
c --enable-libilbc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite --enable-libmysofa --enable-librubberband --en
able-libsoxr --enable-chromaprint
  libavutil      56. 66.100 / 56. 66.100
  libavcodec     58.126.100 / 58.126.100
  libavformat    58. 68.100 / 58. 68.100
  libavdevice    58. 12.100 / 58. 12.100
  libavfilter     7.107.100 /  7.107.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '00026.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    encoder         : Lavf57.37.101
  Duration: 00:00:00.98, start: 0.000000, bitrate: 599 kb/s
  Stream #0:0(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 160x160 [SAR 1:1 DAR 1:1], 556 kb/s, 25 fps, 25 tbr, 12800 tbn, 25 tbc (default)
    Metadata:
      handler_name    : VideoHandler
      vendor_id       : [0][0][0][0]
  Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 16000 Hz, mono, fltp, 65 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      vendor_id       : [0][0][0][0]
Stream mapping:
  Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'output.wav':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2mp41
    ISFT            : Lavf58.68.100
  Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
      vendor_id       : [0][0][0][0]
      encoder         : Lavc58.126.100 pcm_s16le
size=      32kB time=00:00:00.96 bitrate= 273.7kbits/s speed= 212x
video:0kB audio:32kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.238037%