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  • Revert "v4l2 : setting device parameters early"

    8 novembre 2014, par Michael Niedermayer
    Revert "v4l2 : setting device parameters early"
    

    This reverts commit b1ad9312331759679a9c956233716a67ae681d89.

    Fixes Ticket #3517

    Conflicts :

    libavdevice/v4l2.c

    Requested-by : Giorgio Vazzana <mywing81@gmail.com>
    Merged-by : Michael Niedermayer <michaelni@gmx.at>

    • [DH] libavdevice/v4l2.c
  • 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.

    &#xA;

    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.

    &#xA;

    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.

    &#xA;

    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.

    &#xA;

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

    &#xA;

    The problematic python code is :

    &#xA;

    #!/usr/bin/env python3&#xA;# -*- coding: utf-8 -*-&#xA;"""&#xA;Recreating an error 22 when encoding sound with PyAV.&#xA;&#xA;Created on Sun Feb 19 08:10:29 2023&#xA;@author: andrewm&#xA;"""&#xA;import typing&#xA;import sys&#xA;import math&#xA;import fractions&#xA;&#xA;import av&#xA;from av import AudioFrame&#xA;&#xA;""" Ensure some PyAudio constants are still defined without changing &#xA;    the PyAudio recording callback function and without depending &#xA;    on PyAudio simply for reproducing the PyAV bug [Errno 22] thrown in &#xA;    File "av/filter/context.pyx", line 89, in av.filter.context.FilterContext.push&#xA;"""&#xA;class PA_Stub():&#xA;    paContinue = True&#xA;    paComplete= False&#xA;&#xA;pyaudio = PA_Stub()&#xA;&#xA;&#xA;"""Generate pure tone at given frequency with amplitude 0...1.0 at &#xA;   sampling frewuency fs and beginning at phase offset &#x27;phase&#x27;.&#xA;   Returns the new phase after the sinusoid has cycled over the &#xA;   sampling window length.&#xA;"""&#xA;def generate_tone(&#xA;        freq:int, phase:float, amp:float, fs, samp_fmt, buffer:bytearray&#xA;) -> float:&#xA;    assert samp_fmt == "s16", "Only s16 supported atm"&#xA;    samp_size_bytes = 2&#xA;    n_samples = int(len(buffer)/samp_size_bytes)&#xA;    window = [int(0) for i in range(n_samples)]&#xA;    theta = phase&#xA;    phase_inc = 2*math.pi * freq / fs&#xA;    for i in range(n_samples):&#xA;        v = amp * math.sin(theta)&#xA;        theta &#x2B;= phase_inc&#xA;        s = int((2**15-1)*v)&#xA;        window[i] = s&#xA;    for sample_i in range(len(window)):&#xA;        byte_i = sample_i * samp_size_bytes&#xA;        enc = window[sample_i].to_bytes(&#xA;                2, byteorder=sys.byteorder, signed=True&#xA;        )&#xA;        buffer[byte_i] = enc[0]&#xA;        buffer[byte_i&#x2B;1] = enc[1]&#xA;    return theta&#xA;&#xA;&#xA;channels = 1&#xA;fs = 44100  # Record at 44100 samples per second&#xA;fft_size_samps = 256&#xA;chunk_samps = fft_size_samps * 10  # Record in chunks that are multiples of fft windows.&#xA;&#xA;# print(f"fft_size_samps={fft_size_samps}\nchunk_samps={chunk_samps}")&#xA;&#xA;seconds = 3.0&#xA;out_filename = "testoutput.wav"&#xA;&#xA;# Store data in chunks for 3 seconds&#xA;sample_limit = int(fs * seconds)&#xA;sample_len = 0&#xA;frames = []  # Initialize array to store frames&#xA;&#xA;ffmpeg_codec_name = &#x27;mp2&#x27;  # flac, mp3, or libvorbis make same error.&#xA;&#xA;sample_size_bytes = 2&#xA;buffer = bytearray(int(chunk_samps*sample_size_bytes))&#xA;chunkperiod = chunk_samps / fs&#xA;total_chunks = int(math.ceil(seconds / chunkperiod))&#xA;phase = 0.0&#xA;&#xA;### uncomment if you want to see the synthetic data being used as a mic input.&#xA;# with open("test.raw","wb") as raw_out:&#xA;#     for ci in range(total_chunks):&#xA;#         phase = generate_tone(2600, phase, 0.8, fs, "s16", buffer)&#xA;#         raw_out.write(buffer)&#xA;# print("finished gen test")&#xA;# sys.exit(0)&#xA;# #---- &#xA;&#xA;# Using mp2 or mkv as the container format gets the same error.&#xA;with av.open(out_filename&#x2B;&#x27;.mp2&#x27;, "w", format="mp2") as output_con:&#xA;    output_con.metadata["title"] = "My title"&#xA;    output_con.metadata["key"] = "value"&#xA;    channel_layout = "mono"&#xA;    sample_fmt = "s16p"&#xA;&#xA;    ostream = output_con.add_stream(ffmpeg_codec_name, fs, layout=channel_layout)&#xA;    assert ostream is not None, "No stream!"&#xA;    cctx = ostream.codec_context&#xA;    cctx.sample_rate = fs&#xA;    cctx.time_base = fractions.Fraction(numerator=1,denominator=fs)&#xA;    cctx.format = sample_fmt&#xA;    cctx.channels = channels&#xA;    cctx.layout = channel_layout&#xA;    print(cctx, f"layout#{cctx.channel_layout}")&#xA;    &#xA;    # Define PyAudio-style callback for recording plus PyAV transcoding.&#xA;    def rec_callback(in_data, frame_count, time_info, status):&#xA;        global sample_len&#xA;        global ostream&#xA;        frames.append(in_data)&#xA;        nsamples = int(len(in_data) / (channels*sample_size_bytes))&#xA;        &#xA;        frame = AudioFrame(format=sample_fmt, layout=channel_layout, samples=nsamples)&#xA;        frame.sample_rate = fs&#xA;        frame.time_base = fractions.Fraction(numerator=1,denominator=fs)&#xA;        frame.pts = sample_len&#xA;        frame.planes[0].update(in_data)&#xA;        print(frame, len(in_data))&#xA;        &#xA;        for out_packet in ostream.encode(frame):&#xA;            output_con.mux(out_packet)&#xA;        for out_packet in ostream.encode(None):&#xA;            output_con.mux(out_packet)&#xA;        &#xA;        sample_len &#x2B;= nsamples&#xA;        retflag = pyaudio.paContinue if sample_lencode>

    &#xA;

    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.

    &#xA;

    The normal program console output up until the exception is :

    &#xA;

    mp2 at 0x7f8e38202cf0> layout#4&#xA;Beginning&#xA; 5120&#xA;. 5120&#xA;

    &#xA;

    The stack trace is :

    &#xA;

    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>

    &#xA;

    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.

    &#xA;

    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)
    &#xA;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.

    &#xA;

    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.

    &#xA;

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

    &#xA;

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

    &#xA;

  • FFmpeg - Concat videos with different time base

    21 janvier 2017, par Xys

    I’m trying to concat videos with the concat demuxer, but it does not work when using one video ("video2.mp4" below). By does not work, I mean playing the concatenated video on a player will work until the second video part starts (it just cannot read the video anymore). It works with the concat filter though. They are both mp4 videos, so I think it’s because of the time base ? I can concat other videos with the concat demuxer and it works fine (even with different resolutions/bitrate). It only happens when trying to concat "video2.mp4".

    Also, I have a lot of warning/errors like this, probably when ffmpeg starts concatenating the 2nd video :

    [mp4 @ 0x7f847a814800] Non-monotonous DTS in output stream 0:0; previous: 906906, current: 302359; changing to 906907. This may result in incorrect timestamps in the output file.

    What would be the best way to have a minimum concat time ? Do I really need to use the concat filter or can I change the time base of "video1.mp4" if it’s really the problem ?

    Any help would be appreciated, thanks !

    Video 1 :

    ffprobe version 3.2.2 Copyright (c) 2007-2016 the FFmpeg developers
     built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
     configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
     libavutil      55. 34.100 / 55. 34.100
     libavcodec     57. 64.101 / 57. 64.101
     libavformat    57. 56.100 / 57. 56.100
     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, mov,mp4,m4a,3gp,3g2,mj2, from 'video1.mp4':
     Metadata:
       major_brand     : mp42
       minor_version   : 0
       compatible_brands: isommp42
       creation_time   : 2016-08-17T22:50:35.000000Z
     Duration: 00:00:10.11, start: 0.000000, bitrate: 38018 kb/s
       Stream #0:0(eng): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 1696x848, 37832 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc (default)
       Metadata:
         creation_time   : 2016-08-18T00:02:24.000000Z
         handler_name    : VideoHandle
       Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 189 kb/s (default)
       Metadata:
         creation_time   : 2016-08-18T00:02:24.000000Z
         handler_name    : SoundHandle

    Video 2 :

    ffprobe version 3.2.2 Copyright (c) 2007-2016 the FFmpeg developers
     built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
     configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
     libavutil      55. 34.100 / 55. 34.100
     libavcodec     57. 64.101 / 57. 64.101
     libavformat    57. 56.100 / 57. 56.100
     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, mov,mp4,m4a,3gp,3g2,mj2, from 'video2.mp4':
     Metadata:
       major_brand     : mp42
       minor_version   : 0
       compatible_brands: mp42mp41
       creation_time   : 2017-01-06T22:30:23.000000Z
     Duration: 00:00:08.19, start: 0.000000, bitrate: 101474 kb/s
       Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 4096x2048 [SAR 1:1 DAR 2:1], 101549 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
       Metadata:
         creation_time   : 2017-01-06T22:30:23.000000Z
         handler_name    : ?Mainconcept Video Media Handler
         encoder         : AVC Coding
       Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 125 kb/s (default)
       Metadata:
         creation_time   : 2017-01-06T22:30:23.000000Z
         handler_name    : #Mainconcept MP4 Sound Media Handler

    FFMpeg Command :

    ffmpeg -f concat -safe 0 -i concat.txt -c copy result.mp4

    concat.txt :

    file '/path/to/video1.mp4'
    file '/path/to/video2.mp4'

    Command result :

    ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
     built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
     configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
     libavutil      55. 34.100 / 55. 34.100
     libavcodec     57. 64.101 / 57. 64.101
     libavformat    57. 56.100 / 57. 56.100
     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
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbd8b808c00] Auto-inserting h264_mp4toannexb bitstream filter
    Input #0, concat, from 'concat.txt':
     Duration: N/A, start: 0.000000, bitrate: 38021 kb/s
       Stream #0:0(eng): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 1696x848, 37832 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc
       Metadata:
         creation_time   : 2016-08-18T00:02:24.000000Z
         handler_name    : VideoHandle
       Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 189 kb/s
       Metadata:
         creation_time   : 2016-08-18T00:02:24.000000Z
         handler_name    : SoundHandle
    Output #0, mp4, to 'result.mp4':
     Metadata:
       encoder         : Lavf57.56.100
       Stream #0:0(eng): Video: h264 (Baseline) ([33][0][0][0] / 0x0021), yuv420p, 1696x848, q=2-31, 37832 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
       Metadata:
         creation_time   : 2016-08-18T00:02:24.000000Z
         handler_name    : VideoHandle
       Stream #0:1(eng): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, 189 kb/s
       Metadata:
         creation_time   : 2016-08-18T00:02:24.000000Z
         handler_name    : SoundHandle
    Stream mapping:
     Stream #0:0 -> #0:0 (copy)
     Stream #0:1 -> #0:1 (copy)
    Press [q] to stop, [?] for help
    frame=  112 fps=0.0 q=-1.0 size=   12701kB time=00:00:03.70 bitrate=28092.2kbits/s speed= 7.4x
    frame=  151 fps=151 q=-1.0 size=   18853kB time=00:00:05.00 bitrate=30857.5kbits/s speed=   5x
    frame=  224 fps=149 q=-1.0 size=   30042kB time=00:00:07.44 bitrate=33074.8kbits/s speed=4.95x
    frame=  268 fps=134 q=-1.0 size=   36596kB time=00:00:08.90 bitrate=33650.8kbits/s speed=4.44x
    [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbd8a808000] Auto-inserting h264_mp4toannexb bitstream filter.9x
    [mp4 @ 0xb545d000] Non-monotonous DTS in output stream 0:1; previous: 484352, current: 445939; changing to 484353. This may result in incorrect timestamps in the output file.
    [concat @ 0xb545c400] DTS 304057 &lt; 906906 out of order
    [mp4 @ 0xb545d000] Non-monotonous DTS in output stream 0:0; previous: 906906, current: 304057; changing to 906907. This may result in incorrect timestamps in the output file.
    [mp4 @ 0xb545d000] Non-monotonous DTS in output stream 0:1; previous: 484353, current: 446963; changing to 484354. This may result in incorrect timestamps in the output file.
    ...
    ... like 100 DTS errors ...
    ...
    [mp4 @ 0xb545d000] Non-monotonous DTS in output stream 0:0; previous: 907150, current: 548301; changing to 907151. This may result in incorrect timestamps in the output file.
    frame=  548 fps=169 q=-1.0 Lsize=  148399kB time=00:00:18.28 bitrate=66493.7kbits/s speed=5.64x
    video:148027kB audio:359kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.008622%

    Thank you in advance !