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Le plugin : Podcasts.
14 juillet 2010, parLe problème du podcasting est à nouveau un problème révélateur de la normalisation des transports de données sur Internet.
Deux formats intéressants existent : Celui développé par Apple, très axé sur l’utilisation d’iTunes dont la SPEC est ici ; Le format "Media RSS Module" qui est plus "libre" notamment soutenu par Yahoo et le logiciel Miro ;
Types de fichiers supportés dans les flux
Le format d’Apple n’autorise que les formats suivants dans ses flux : .mp3 audio/mpeg .m4a audio/x-m4a .mp4 (...) -
Sélection de projets utilisant MediaSPIP
29 avril 2011, parLes exemples cités ci-dessous sont des éléments représentatifs d’usages spécifiques de MediaSPIP pour certains projets.
Vous pensez avoir un site "remarquable" réalisé avec MediaSPIP ? Faites le nous savoir ici.
Ferme MediaSPIP @ Infini
L’Association Infini développe des activités d’accueil, de point d’accès internet, de formation, de conduite de projets innovants dans le domaine des Technologies de l’Information et de la Communication, et l’hébergement de sites. Elle joue en la matière un rôle unique (...) -
Selection of projects using MediaSPIP
2 mai 2011, parThe examples below are representative elements of MediaSPIP specific uses for specific projects.
MediaSPIP farm @ Infini
The non profit organizationInfini develops hospitality activities, internet access point, training, realizing innovative projects in the field of information and communication technologies and Communication, and hosting of websites. It plays a unique and prominent role in the Brest (France) area, at the national level, among the half-dozen such association. Its members (...)
Sur d’autres sites (2532)
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ffmpeg : Input link parameters do not match the corresponding output link parameters, when concatenating same size videos
4 mars 2021, par mSourireI'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!



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Using PyAV to encode mono audio to file, params match docs, but still causes Errno 22
20 février 2023, par andrew8088While 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):

 File "Dev/multichan_recording/av_encode.py", line 147, in <module>
 ret_data, ret_flag = rec_callback(buffer, ci, {}, 1)

 File "Dev/multichan_recording/av_encode.py", line 121, in rec_callback
 for out_packet in ostream.encode(frame):

 File "av/stream.pyx", line 153, in av.stream.Stream.encode

 File "av/codec/context.pyx", line 484, in av.codec.context.CodecContext.encode

 File "av/audio/codeccontext.pyx", line 42, in av.audio.codeccontext.AudioCodecContext._prepare_frames_for_encode

 File "av/audio/resampler.pyx", line 101, in av.audio.resampler.AudioResampler.resample

 File "av/filter/graph.pyx", line 211, in av.filter.graph.Graph.push

 File "av/filter/context.pyx", line 89, in av.filter.context.FilterContext.push

 File "av/error.pyx", line 336, in av.error.err_check

ValueError: [Errno 22] Invalid argument

</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 fromres = lib.av_buffersrc_write_frame(self.ptr, frame.ptr)

Trying to dig intoav_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 ?


-
Adding FFmpeg library as dependency in Visual studio project to get rid of LNK2019 error
13 mars 2019, par MSD PaulWhile compiling a project on Microsoft visual studio (to generate dll), I am getting the following linking errors (LNK2019)
Severity Code Description Project File Line Suppression State
Error LNK2019 unresolved external symbol av_strerror referenced in function "private : void __cdecl DecoderFFmpeg::printErrorMsg(int)" (?printErrorMsg@DecoderFFmpeg@@AEAAXH@Z)
Severity Code Description Project File Line Suppression State
Error LNK2019 unresolved external symbol av_get_channel_layout_nb_channels referenced in function "private : int __cdecl DecoderFFmpeg::initSwrContext(void)"several other functions like av_dict_set, av_dict_free, avcodec_decode2 which is related to the ffmpeg libraries. Can anyone please tell me how to properly add "ffmpeg" as dependenciesto this project and link it properly ?