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• Google’s YouTube Uses FFmpeg

  9 février 2011, par Multimedia Mike — General

  Controversy arose last week when Google accused Microsoft of stealing search engine results for their Bing search engine. It was a pretty novel sting operation and Google did a good job of visually illustrating their side of the story on their official blog.

  This reminds me of the fact that Google’s YouTube video hosting site uses FFmpeg for converting videos. Not that this is in the same league as the search engine shenanigans (it’s perfectly legit to use FFmpeg in this capacity, but to my knowledge, Google/YouTube has never confirmed FFmpeg usage), but I thought I would revisit this item and illustrate it with screenshots. This is not new information— I first empirically tested this fact 4 years ago. However, a lot of people wonder how exactly I can identify FFmpeg on the backend when I claim that I’ve written code that helps power YouTube.

  Short Answer
  How do I know YouTube uses FFmpeg to convert multimedia ? Because :

  1. FFmpeg can decode a number of impossibly obscure multimedia formats using code I wrote
  2. YouTube can transcode many of the same formats
  3. I screwed up when I wrote the code to support some of these weird formats
  4. My mistakes are still present when YouTube transcodes certain fringe formats

  Longer Answer (With Pictures !)
  Let’s take a video format named RoQ, developed by noted game designer Graeme Devine. Originated for use in the FMV-heavy game The 11th Hour, the format eventually found its way into the Quake 3 engine as well as many games derived from the same technology.

  Dr. Tim Ferguson reverse engineered the format (though it would later be open sourced along with the rest of the Q3 engine). I wrote a RoQ playback system for FFmpeg, and I messed up in doing so. I believe my coding error helps demonstrate the case I’m trying to make here.

  Observe what happened when I pushed the jk02.roq sample through YouTube in my original experiment 4 years ago :

  
  
  

  Do you see how the canyon walls bleed into the sky ? That’s not supposed to happen. FFmpeg doesn’t do that anymore but I was able to go back into the source code history to find when it did do that :

  
  
  

  Academic Answer
  FFmpeg fixed this bug in June of 2007 (thanks to Eric Lasota). The problem had to do with premature colorspace conversion in my original decoder.

  Leftovers
  I tried uploading the video again to see if the problem persists in YouTube’s transcoder. First bit of trivia : YouTube detects when you have uploaded the same video twice and rejects the subsequent attempts. So I created a double concatenation of the video and uploaded it. The problem is gone, illustrating that the backend is actually using a newer version of FFmpeg. This surprises me for somewhat esoteric reasons.

  Here’s another interesting bit of trivia for those who don’t do a lot of YouTube uploading— YouTube reports format details when you upload a video :

  
  
  

  So, yep, RoQ format. And you can wager that this will prompt me to go back through the litany of unusual formats that FFmpeg supports to see how YouTube responds.

• Computer crashing when using python tools in same script

  5 février 2023, par SL1997

  I am attempting to use the speech recognition toolkit VOSK and the speech diarization package Resemblyzer to transcibe audio and then identify the speakers in the audio.

  


  Tools :

  


  https://github.com/alphacep/vosk-api
  
https://github.com/resemble-ai/Resemblyzer

  


  I can do both things individually but run into issues when trying to do them when running the one python script.

  


  I used the following guide when setting up the diarization system :

  


  https://medium.com/saarthi-ai/who-spoke-when-build-your-own-speaker-diarization-module-from-scratch-e7d725ee279

  


  Computer specs are as follows :

  


  Intel(R) Core(TM) i3-7100 CPU @ 3.90GHz, 3912 Mhz, 2 Core(s), 4 Logical Processor(s)
  
32GB RAM

  


  The following is my code, I am not to sure if using threading is appropriate or if I even implemented it correctly, how can I best optimize this code as to achieve the results I am looking for and not crash.

  


  from vosk import Model, KaldiRecognizer
from pydub import AudioSegment
import json
import sys
import os
import subprocess
import datetime
from resemblyzer import preprocess_wav, VoiceEncoder
from pathlib import Path
from resemblyzer.hparams import sampling_rate
from spectralcluster import SpectralClusterer
import threading
import queue
import gc



def recognition(queue, audio, FRAME_RATE):

    model = Model("Vosk_Models/vosk-model-small-en-us-0.15")

    rec = KaldiRecognizer(model, FRAME_RATE)
    rec.SetWords(True)

    rec.AcceptWaveform(audio.raw_data)
    result = rec.Result()

    transcript = json.loads(result)#["text"]

    #return transcript
    queue.put(transcript)



def diarization(queue, audio):

    wav = preprocess_wav(audio)
    encoder = VoiceEncoder("cpu")
    _, cont_embeds, wav_splits = encoder.embed_utterance(wav, return_partials=True, rate=16)
    print(cont_embeds.shape)

    clusterer = SpectralClusterer(
        min_clusters=2,
        max_clusters=100,
        p_percentile=0.90,
        gaussian_blur_sigma=1)

    labels = clusterer.predict(cont_embeds)

    def create_labelling(labels, wav_splits):

        times = [((s.start + s.stop) / 2) / sampling_rate for s in wav_splits]
        labelling = []
        start_time = 0

        for i, time in enumerate(times):
            if i > 0 and labels[i] != labels[i - 1]:
                temp = [str(labels[i - 1]), start_time, time]
                labelling.append(tuple(temp))
                start_time = time
            if i == len(times) - 1:
                temp = [str(labels[i]), start_time, time]
                labelling.append(tuple(temp))

        return labelling

    #return
    labelling = create_labelling(labels, wav_splits)
    queue.put(labelling)



def identify_speaker(queue1, queue2):

    transcript = queue1.get()
    labelling = queue2.get()

    for speaker in labelling:

        speakerID = speaker[0]
        speakerStart = speaker[1]
        speakerEnd = speaker[2]

        result = transcript['result']
        words = [r['word'] for r in result if speakerStart < r['start'] < speakerEnd]
        #return
        print("Speaker",speakerID,":",' '.join(words), "\n")





def main():

    queue1 = queue.Queue()
    queue2 = queue.Queue()

    FRAME_RATE = 16000
    CHANNELS = 1

    podcast = AudioSegment.from_mp3("Podcast_Audio/Film-Release-Clip.mp3")
    podcast = podcast.set_channels(CHANNELS)
    podcast = podcast.set_frame_rate(FRAME_RATE)

    first_thread = threading.Thread(target=recognition, args=(queue1, podcast, FRAME_RATE))
    second_thread = threading.Thread(target=diarization, args=(queue2, podcast))
    third_thread = threading.Thread(target=identify_speaker, args=(queue1, queue2))

    first_thread.start()
    first_thread.join()
    gc.collect()

    second_thread.start()
    second_thread.join()
    gc.collect()

    third_thread.start()
    third_thread.join()
    gc.collect()

    # transcript = recognition(podcast,FRAME_RATE)
    #
    # labelling = diarization(podcast)
    #
    # print(identify_speaker(transcript, labelling))


if __name__ == '__main__':
    main()


  


  When I say crash I mean everything freezes, I have to hold down the power button on the desktop and turn it back on again. No blue/blank screen, just frozen in my IDE looking at my code. Any help in resolving this issue would be greatly appreciated.

  


• Need help configuring FFMPEG to work with a webcams h264 stream

  9 août 2020, par The Welsh Dragon

  I have been trying to get a H264 stream from a H264 usb webcam working but I am not making much progress so I'm hoping someone knows FFMPEG better than me !

  


  There are dozens of questions/answers on SO but none solve my problem.

  


  In short, I get a very pixelated (or sometimes mostly green) screen. I am using VLC to test the stream which is coming via an RTSP server. I am using FFMPEG to copy the webcam stream to the local RTSP server.

  


  The webcam also supports YUYV which I can get working - it is just the h264 stream causing me problems.

  


  So this is how the device is presented :

  


  H264 USB Camera: USB Camera (usb-20980000.usb-1):
        /dev/video0
        /dev/video1
        /dev/video2
        /dev/video3


  


  /dev/video0 is the YUYV and MPEG stream
/dev/video2 is the h264 stream that has the following capabilities :

  


  ioctl: VIDIOC_ENUM_FMT
        Type: Video Capture

        [0]: 'H264' (H.264, compressed)
                Size: Discrete 1920x1080
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                Size: Discrete 1280x720
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                Size: Discrete 800x600
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                Size: Discrete 640x480
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                Size: Discrete 640x360
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                Size: Discrete 352x288
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                Size: Discrete 320x240
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                Size: Discrete 1920x1080
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)
                        Interval: Discrete 0.033s (30.000 fps)
                        Interval: Discrete 0.040s (25.000 fps)
                        Interval: Discrete 0.067s (15.000 fps)


  


  I have tried various resolutions, the smaller giving slightly less pixelated images but none are usable and definitely dont compare to the YUYV high resolution results.

  


  This (YUYV) command works :

  


  ffmpeg -input_format yuyv422 -f video4linux2 -s 1280x720 -r 10 -i /dev/video0 -c:v h264_omx -r 10 -b:v 2M -an -f rtsp rtsp://localhost:80/live/stream


  


  These two h264 options dont work :

  


  ffmpeg -input_format h264 -f video4linux2 -video_size 1920x1080 -framerate 30 -i /dev/video0 -c:v copy -an -f rtsp rtsp://localhost:80/live/stream


  


  ffmpeg -re -i /dev/video2 -video_size 800x600 -framerate 15 -pix_fmt yuv420p -tune zerolatency -c:v copy -an -f rtsp rtsp://localhost:80/live/stream


  


  For that last command the FFMPEG output looks like this :

  


  ffmpeg version git-2020-08-07-6fdf3cc Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 8 (Raspbian 8.3.0-6+rpi1)
  configuration: --extra-ldflags=-latomic --arch=armel --target-os=linux --enable-gpl --enable-omx --enable-omx-rpi --enable-nonfree --enable-libfreetype --enable-libx264 --enable-libmp3lame --enable-mmal --enable-indev=alsa --enable-outdev=alsa
  libavutil      56. 58.100 / 56. 58.100
  libavcodec     58.100.100 / 58.100.100
  libavformat    58. 50.100 / 58. 50.100
  libavdevice    58. 11.101 / 58. 11.101
  libavfilter     7. 87.100 /  7. 87.100
  libswscale      5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc    55.  8.100 / 55.  8.100
Input #0, video4linux2,v4l2, from '/dev/video2':
  Duration: N/A, start: 1353.265049, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 30 fps, 30 tbr, 1000k tbn, 2000k tbc
[udp @ 0x38c29f0] attempted to set receive buffer to size 393216 but it only ended up set as 360448
[udp @ 0x38d7b50] attempted to set receive buffer to size 393216 but it only ended up set as 360448
Output #0, rtsp, to 'rtsp://localhost:80/live/stream':
  Metadata:
    encoder         : Lavf58.50.100
    Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, q=2-31, 30 fps, 30 tbr, 90k tbn, 1000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[rtsp @ 0x38fd890] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
[rtsp @ 0x38fd890] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file.
frame=  348 fps= 18 q=-1.0 size=N/A time=00:00:21.03 bitrate=N/A speed=1.09x


  


  The issue looks like it is bandwidth related or the lack of processing power in the device being used BUT the YUYV works at a high resolution and (taking a completely different approach i.e. not using FFMPEG) I can get a very decent MPEG stream working on the same device.

  


  So any FFMPEG experts out there who can help me with getting the correct parameters for a h264 stream ?