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• FFmpeg generated video length doesn't match expected length [on hold]

  21 juin 2018, par BentCoder

  I am using command below to generate video out of images. The problem is the length out the video generated. It is always 5 seconds less than expected. If I have 4 images, I expect to see 20 seconds of video but I am getting 15 seconds instead. The very last image img-03.jpg just appears at the end of the video as if it is a cover image. Any idea why and solution ?

  Command

  ffmpeg -y -framerate 1/5 -f image2 \
  
  -i img-%2d.jpg \
  
  -c:v libvpx-vp9 \
  
  -r 25 \
  
  -crf 30 -b:v 0 \
  
  video.webm

  Images

  img-00.jpg
  
  img-01.jpg
  
  img-02.jpg
  
  img-03.jpg

  Output

  ffmpeg version 3.2.10-1~deb9u1~bpo8+1 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 4.9.2 (Debian 4.9.2-10)
  
    configuration: --prefix=/usr --extra-version='1~deb9u1~bpo8+1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --disable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    libavutil      55. 34.101 / 55. 34.101
  
    libavcodec     57. 64.101 / 57. 64.101
  
    libavformat    57. 56.101 / 57. 56.101
  
    libavdevice    57.  1.100 / 57.  1.100
  
    libavfilter     6. 65.100 /  6. 65.100
  
    libavresample   3.  1.  0 /  3.  1.  0
  
    libswscale      4.  2.100 /  4.  2.100
  
    libswresample   2.  3.100 /  2.  3.100
  
    libpostproc    54.  1.100 / 54.  1.100
  
  Input #0, image2, from 'img-%2d.jpg':
  
    Duration: 00:00:20.00, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 854x480 [SAR 1:1 DAR 427:240], 0.20 fps, 0.20 tbr, 0.20 tbn, 0.20 tbc
  
  [swscaler @ 0x55f2af60e800] deprecated pixel format used, make sure you did set range correctly
  
  [libvpx-vp9 @ 0x55f2af6276a0] v1.3.0
  
  Output #0, webm, to 'video.webm':
  
    Metadata:
  
      encoder         : Lavf57.56.101
  
      Stream #0:0: Video: vp9 (libvpx-vp9), yuv420p, 854x480 [SAR 1:1 DAR 427:240], q=-1--1, 25 fps, 1k tbn, 25 tbc
  
      Metadata:
  
        encoder         : Lavc57.64.101 libvpx-vp9
  
      Side data:
  
        cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mjpeg (native) -> vp9 (libvpx-vp9))
  
  Press [q] to stop, [?] for help
  
  Input stream #0:0 frame changed from size:854x480 fmt:yuvj444p to size:854x480 fmt:yuvj420p
  
  [swscaler @ 0x55f2af602b60] deprecated pixel format used, make sure you did set range correctly
  
  frame=    4 fps=0.0 q=0.0 Lsize=     214kB time=00:00:15.00 bitrate= 117.0kbits/s speed=  16x    
  
  video:214kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.234810%

• ffmpeg : Input link parameters do not match the corresponding output link parameters, when concatenating same size videos

  4 mars 2021, par mSourire

  I'm trying to combine a video and an audio, using the following command :

  


  ffmpeg -y -i 1.mkv -i 1.mka 
-max_muxing_queue_size 10000 -preset veryfast -r 30 -crf 20 -b:a 96000 -vbr on
-strict experimental
-filter_complex '
color=black:s=320x240:d=7ms[black0];
aevalsrc=0:d=15ms[silence1];
[black0][0]concat=n=2:v=1:a=0[video];
[1][silence1]concat=n=2:v=0:a=1[audio]'
-map [video] -map [audio] -c:v libvpx -c:a libopus output.webm


  


  But ffmpeg returns an error :

  


  [Parsed_concat_2 @ 0x7f8004506d00] Input link in0:v0 parameters (size 640x480, SAR 1:1) do not match the corresponding output link in0:v0 parameters (320x240, SAR 1:1)
[Parsed_concat_2 @ 0x7f8004506d00] Failed to configure output pad on Parsed_concat_2
Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #0:0
[libopus @ 0x7f800500c000] 1 frames left in the queue on closing
Conversion failed!


  


  It looks like ffmpeg is unable to concatenate video and black frames, complaining, that video has a distinct resolution, but that's not true :

  


  #> ffprobe 1.mkv
Input #0, matroska,webm, from '1.mkv':
  Metadata:
    encoder         : GStreamer matroskamux version 1.8.1.1
    creation_time   : 2021-03-02T13:44:03.000000Z
  Duration: 00:01:48.41, start: 0.710000, bitrate: 757 kb/s
    Stream #0:0(eng): Video: vp8, yuv420p(progressive), 320x240, SAR 1:1 DAR 4:3, 120 tbr, 1k tbn, 1k tbc (default)
    Metadata:
      title           : Video


  


  So, both the input source and the filter has the same resolution.

  


  When I'm trying to change the color filter to "color=s=640x480", ffmpeg says an opposite thing :

  


  [Parsed_concat_2 @ 0x7fca3ca185c0] Input link in0:v0 parameters (size 320x240, SAR 1:1) do not match the corresponding output link in0:v0 parameters (640x480, SAR 1:1)
[Parsed_concat_2 @ 0x7fca3ca185c0] Failed to configure output pad on Parsed_concat_2
Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument


  


  Help please to solve this !

  


  Full listing :

  


  ffmpeg -y -i 1.mkv -i 1.mka
-max_muxing_queue_size 10000
-preset veryfast -r 30 -crf 20 -b:a 96000 -vbr on
-strict experimental
-filter_complex '
color=black:s=320x240:d=7ms[black0];
aevalsrc=0:d=15ms[silence1];
[black0][0]concat=n=2:v=1:a=0[video];
[1][silence1]concat=n=2:v=0:a=1[audio]'
-map [video] -map [audio] -c:v libvpx -c:a libopus output.webm

ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_9 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
Input #0, matroska,webm, from '1.mkv':
  Metadata:
    encoder         : GStreamer matroskamux version 1.8.1.1
    creation_time   : 2021-03-02T13:44:03.000000Z
  Duration: 00:01:48.41, start: 0.710000, bitrate: 757 kb/s
    Stream #0:0(eng): Video: vp8, yuv420p(progressive), 320x240, SAR 1:1 DAR 4:3, 120 tbr, 1k tbn, 1k tbc (default)
    Metadata:
      title           : Video
Input #1, matroska,webm, from '1.mka':
  Metadata:
    encoder         : GStreamer matroskamux version 1.8.1.1
    creation_time   : 2021-03-02T13:44:03.000000Z
  Duration: 00:01:48.40, start: 0.703000, bitrate: 38 kb/s
    Stream #1:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
    Metadata:
      title           : Audio
Codec AVOption preset (Configuration preset) specified for output file #0 (output.webm) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
Stream mapping:
  Stream #0:0 (vp8) -> concat:in1:v0
  Stream #1:0 (opus) -> concat:in0:a0
  concat -> Stream #0:0 (libvpx)
  concat -> Stream #0:1 (libopus)
Press [q] to stop, [?] for help
[libvpx @ 0x7fe08a80bc00] v1.9.0
[libvpx @ 0x7fe08a80bc00] Bitrate not specified for constrained quality mode, using default of 256kbit/sec
Output #0, webm, to 'output.webm':
  Metadata:
    encoder         : Lavf58.45.100
    Stream #0:0: Video: vp8 (libvpx), yuv420p, 320x240 [SAR 1:1 DAR 4:3], q=-1--1, 256 kb/s, 30 fps, 1k tbn, 30 tbc (default)
    Metadata:
      encoder         : Lavc58.91.100 libvpx
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
    Stream #0:1: Audio: opus (libopus), 48000 Hz, stereo, flt, 96 kb/s (default)
    Metadata:
      encoder         : Lavc58.91.100 libopus
[Parsed_color_0 @ 0x7fe089815940] EOF timestamp not reliable
[Parsed_concat_2 @ 0x7fe088501980] Input link in0:v0 parameters (size 640x480, SAR 1:1) do not match the corresponding output link in0:v0 parameters (320x240, SAR 1:1)
[Parsed_concat_2 @ 0x7fe088501980] Failed to configure output pad on Parsed_concat_2
Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #0:0
[libopus @ 0x7fe08a810c00] 1 frames left in the queue on closing
Conversion failed!


  


• Using PyAV to encode mono audio to file, params match docs, but still causes Errno 22

  20 février 2023, par andrew8088

  While trying to use PyAV to encode live mono audio from a microphone to a compressed audio stream (using mp2 or flac as encoder), the program kept raising an exception ValueError: [Errno 22] Invalid argument.

  


  To remove the live microphone source as a cause of the problem, and to make the problematic code easier for others to run/test, I have removed the mic source and now just generate a pure tone as a sequence of input buffers.

  


  All attempts to figure out the missing or mismatched or incorrect argument have just resulted in seeing documentation and examples that are the same as my code.

  


  I would like to know from someone who has used PyAV successfully for mono audio what the correct method and parameters are for encoding mono frames into the mono stream.

  


  The package used is av 10.0.0 installed with
pip3 install av --no-binary av
so it uses my package-manager provided ffmpeg library, which is version 4.2.7.

  


  The problematic python code is :

  


  #!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
Recreating an error 22 when encoding sound with PyAV.

Created on Sun Feb 19 08:10:29 2023
@author: andrewm
"""
import typing
import sys
import math
import fractions

import av
from av import AudioFrame

""" Ensure some PyAudio constants are still defined without changing 
    the PyAudio recording callback function and without depending 
    on PyAudio simply for reproducing the PyAV bug [Errno 22] thrown in 
    File "av/filter/context.pyx", line 89, in av.filter.context.FilterContext.push
"""
class PA_Stub():
    paContinue = True
    paComplete= False

pyaudio = PA_Stub()


"""Generate pure tone at given frequency with amplitude 0...1.0 at 
   sampling frewuency fs and beginning at phase offset 'phase'.
   Returns the new phase after the sinusoid has cycled over the 
   sampling window length.
"""
def generate_tone(
        freq:int, phase:float, amp:float, fs, samp_fmt, buffer:bytearray
) -> float:
    assert samp_fmt == "s16", "Only s16 supported atm"
    samp_size_bytes = 2
    n_samples = int(len(buffer)/samp_size_bytes)
    window = [int(0) for i in range(n_samples)]
    theta = phase
    phase_inc = 2*math.pi * freq / fs
    for i in range(n_samples):
        v = amp * math.sin(theta)
        theta += phase_inc
        s = int((2**15-1)*v)
        window[i] = s
    for sample_i in range(len(window)):
        byte_i = sample_i * samp_size_bytes
        enc = window[sample_i].to_bytes(
                2, byteorder=sys.byteorder, signed=True
        )
        buffer[byte_i] = enc[0]
        buffer[byte_i+1] = enc[1]
    return theta


channels = 1
fs = 44100  # Record at 44100 samples per second
fft_size_samps = 256
chunk_samps = fft_size_samps * 10  # Record in chunks that are multiples of fft windows.

# print(f"fft_size_samps={fft_size_samps}\nchunk_samps={chunk_samps}")

seconds = 3.0
out_filename = "testoutput.wav"

# Store data in chunks for 3 seconds
sample_limit = int(fs * seconds)
sample_len = 0
frames = []  # Initialize array to store frames

ffmpeg_codec_name = 'mp2'  # flac, mp3, or libvorbis make same error.

sample_size_bytes = 2
buffer = bytearray(int(chunk_samps*sample_size_bytes))
chunkperiod = chunk_samps / fs
total_chunks = int(math.ceil(seconds / chunkperiod))
phase = 0.0

### uncomment if you want to see the synthetic data being used as a mic input.
# with open("test.raw","wb") as raw_out:
#     for ci in range(total_chunks):
#         phase = generate_tone(2600, phase, 0.8, fs, "s16", buffer)
#         raw_out.write(buffer)
# print("finished gen test")
# sys.exit(0)
# #---- 

# Using mp2 or mkv as the container format gets the same error.
with av.open(out_filename+'.mp2', "w", format="mp2") as output_con:
    output_con.metadata["title"] = "My title"
    output_con.metadata["key"] = "value"
    channel_layout = "mono"
    sample_fmt = "s16p"

    ostream = output_con.add_stream(ffmpeg_codec_name, fs, layout=channel_layout)
    assert ostream is not None, "No stream!"
    cctx = ostream.codec_context
    cctx.sample_rate = fs
    cctx.time_base = fractions.Fraction(numerator=1,denominator=fs)
    cctx.format = sample_fmt
    cctx.channels = channels
    cctx.layout = channel_layout
    print(cctx, f"layout#{cctx.channel_layout}")
    
    # Define PyAudio-style callback for recording plus PyAV transcoding.
    def rec_callback(in_data, frame_count, time_info, status):
        global sample_len
        global ostream
        frames.append(in_data)
        nsamples = int(len(in_data) / (channels*sample_size_bytes))
        
        frame = AudioFrame(format=sample_fmt, layout=channel_layout, samples=nsamples)
        frame.sample_rate = fs
        frame.time_base = fractions.Fraction(numerator=1,denominator=fs)
        frame.pts = sample_len
        frame.planes[0].update(in_data)
        print(frame, len(in_data))
        
        for out_packet in ostream.encode(frame):
            output_con.mux(out_packet)
        for out_packet in ostream.encode(None):
            output_con.mux(out_packet)
        
        sample_len += nsamples
        retflag = pyaudio.paContinue if sample_lencode>

  


  If you uncomment the RAW output part you will find the generated data can be imported as PCM s16 Mono 44100Hz into Audacity and plays the expected tone, so the generated audio data does not seem to be the problem.

  


  The normal program console output up until the exception is :

  


  mp2 at 0x7f8e38202cf0> layout#4
Beginning
 5120
. 5120


  


  The stack trace is :

  


  Traceback (most recent call last):&#xA;&#xA;  File "Dev/multichan_recording/av_encode.py", line 147, in <module>&#xA;    ret_data, ret_flag = rec_callback(buffer, ci, {}, 1)&#xA;&#xA;  File "Dev/multichan_recording/av_encode.py", line 121, in rec_callback&#xA;    for out_packet in ostream.encode(frame):&#xA;&#xA;  File "av/stream.pyx", line 153, in av.stream.Stream.encode&#xA;&#xA;  File "av/codec/context.pyx", line 484, in av.codec.context.CodecContext.encode&#xA;&#xA;  File "av/audio/codeccontext.pyx", line 42, in av.audio.codeccontext.AudioCodecContext._prepare_frames_for_encode&#xA;&#xA;  File "av/audio/resampler.pyx", line 101, in av.audio.resampler.AudioResampler.resample&#xA;&#xA;  File "av/filter/graph.pyx", line 211, in av.filter.graph.Graph.push&#xA;&#xA;  File "av/filter/context.pyx", line 89, in av.filter.context.FilterContext.push&#xA;&#xA;  File "av/error.pyx", line 336, in av.error.err_check&#xA;&#xA;ValueError: [Errno 22] Invalid argument&#xA;&#xA;</module>

  


  edit : It's interesting that the error happens on the 2nd AudioFrame, as apparently the first one was encoded okay, because they are given the same attribute values aside from the Presentation Time Stamp (pts), but leaving this out and letting PyAV/ffmpeg generate the PTS by itself does not fix the error, so an incorrect PTS does not seem the cause.

  


  After a brief glance in av/filter/context.pyx the exception must come from a bad return value from res = lib.av_buffersrc_write_frame(self.ptr, frame.ptr)
  
Trying to dig into av_buffersrc_write_frame from the ffmpeg source it is not clear what could be causing this error. The only obvious one is a mismatch between channel layouts, but my code is setting the layout the same in the Stream and the Frame. That problem had been found by an old question pyav - cannot save stream as mono and their answer (that one parameter required is undocumented) is the only reason the code now has the layout='mono' argument when making the stream.

  


  The program output shows layout #4 is being used, and from https://github.com/FFmpeg/FFmpeg/blob/release/4.2/libavutil/channel_layout.h you can see this is the value for symbol AV_CH_FRONT_CENTER which is the only channel in the MONO layout.

  


  The mismatch is surely some other object property or an undocumented parameter requirement.

  


  How do you encode mono audio to a compressed stream with PyAV ?