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  • ffmpeg Command in Docker with Rust Tokio Closes Warp Server Connection (curl 52 Error)

    3 juin 2024, par user762345

    I’m encountering an issue where executing an ffmpeg concatenation command through Rust’s Tokio process in a Docker container causes subsequent HTTP requests to fail. The error occurs exclusively after running the ffmpeg command and making immediate requests, resulting in a “curl 52 empty response from server” error with the connection being closed. Notably, this issue does not occur when running the same setup outside of Docker. Additionally, if no HTTP requests are made after the ffmpeg command, the curl 52 error does not occur.

    


    Here is the verbose curl output of my minimum reproducible example (see below).

    


    curl -v "http://localhost:3030"
*   Trying 127.0.0.1:3030...
* Connected to localhost (127.0.0.1) port 3030 (#0)
> GET / HTTP/1.1
> Host: localhost:3030
> User-Agent: curl/8.1.2
> Accept: */*
> 
* Empty reply from server
* Closing connection 0
curl: (52) Empty reply from server


    


    Here are Docker logs from my minimum reproducible example (see below). The wav files are concatenated successfully, then the container appears to rebuild.

    


    [2024-06-03T05:26:58Z INFO  minimal_docker_webserver_post_error] Starting server on 0.0.0.0:3030
[2024-06-03T05:26:58Z INFO  warp::server] Server::run; addr=0.0.0.0:3030
[2024-06-03T05:26:58Z INFO  warp::server] listening on http://0.0.0.0:3030
[2024-06-03T05:27:07Z INFO  minimal_docker_webserver_post_error] WAV files concatenated successfully
[Running 'cargo run']
    Finished `dev` profile [unoptimized + debuginfo] target(s) in 0.06s
     Running `target/debug/minimal_docker_webserver_post_error`
[2024-06-03T05:27:08Z INFO  minimal_docker_webserver_post_error] Starting server on 0.0.0.0:3030
[2024-06-03T05:27:08Z INFO  warp::server] Server::run; addr=0.0.0.0:3030
[2024-06-03T05:27:08Z INFO  warp::server] listening on http://0.0.0.0:3030


    


    What I have tried :
I tried using different web frameworks (Warp, Actix-web) and request crates (reqwest, ureq). I also tried running the setup outside of Docker, which worked as expected without any issues. Additionally, I tried running the setup in Docker without making any HTTP requests after the ffmpeg command, and the connection closed successfully without errors. I also tried posting to httpbin with a minimal request, but the issue persisted.

    


    Minimum reproducible example :

    


    main.rs

    


    use warp::Filter;&#xA;use reqwest::Client;&#xA;use std::convert::Infallible;&#xA;use log::{info, error};&#xA;use env_logger;&#xA;use tokio::process::Command;&#xA;&#xA;#[tokio::main]&#xA;async fn main() {&#xA;    std::env::set_var("RUST_LOG", "debug");&#xA;    env_logger::init();&#xA;&#xA;    let route = warp::path::end()&#xA;        .and_then(handle_request);&#xA;&#xA;    info!("Starting server on 0.0.0.0:3030");&#xA;    warp::serve(route)&#xA;        .run(([0, 0, 0, 0], 3030))&#xA;        .await;&#xA;}&#xA;&#xA;async fn handle_request() -> Result<impl infallible="infallible"> {&#xA;    let client = Client::new();&#xA;&#xA;    let output = Command::new("ffmpeg")&#xA;        .args(&amp;[&#xA;            "y",&#xA;            "-i", "concat:/usr/src/minimal_docker_webserver_post_error/file1.wav|/usr/src/minimal_docker_webserver_post_error/file2.wav",&#xA;            "-c", "copy",&#xA;            "/usr/src/minimal_docker_webserver_post_error/combined.wav"&#xA;        ])&#xA;        .output()&#xA;        .await;&#xA;&#xA;    match output {&#xA;        Ok(output) => {&#xA;            if output.status.success() {&#xA;                info!("WAV files concatenated successfully");&#xA;            } else {&#xA;                error!("Failed to concatenate WAV files: {:?}", output);&#xA;                return Ok(warp::reply::with_status("Failed to concatenate WAV files", warp::http::StatusCode::INTERNAL_SERVER_ERROR));&#xA;            }&#xA;        },&#xA;        Err(e) => {&#xA;            error!("Failed to execute ffmpeg: {:?}", e);&#xA;            return Ok(warp::reply::with_status("Failed to execute ffmpeg", warp::http::StatusCode::INTERNAL_SERVER_ERROR));&#xA;        }&#xA;    }&#xA;&#xA;    // ISSUE: Connection closes with curl: (52) Empty reply from server&#xA;    match client.get("https://httpbin.org/get").send().await {&#xA;        Ok(response) => info!("GET request successful: {:?}", response),&#xA;        Err(e) => error!("GET request failed: {:?}", e),&#xA;    }&#xA;&#xA;    match client.post("https://httpbin.org/post")&#xA;        .body("field1=value1&amp;field2=value2")&#xA;        .send().await {&#xA;        Ok(response) => info!("POST request successful: {:?}", response),&#xA;        Err(e) => error!("POST request failed: {:?}", e),&#xA;    }&#xA;&#xA;    Ok(warp::reply::with_status("Request handled", warp::http::StatusCode::OK))&#xA;}&#xA;</impl>

    &#xA;

    FFMPEG command to generate the two wav files for concatenation

    &#xA;

    ffmpeg -f lavfi -i "sine=frequency=1000:duration=5" file1.wav &amp;&amp; ffmpeg -f lavfi -i "sine=frequency=500:duration=5" file2.wav&#xA;

    &#xA;

    Dockerfile

    &#xA;

    # Use the official Rust image as the base image&#xA;FROM rust:latest&#xA;&#xA;# Install cargo-watch&#xA;RUN cargo install cargo-watch&#xA;&#xA;# Install ffmpeg&#xA;RUN apt-get update &amp;&amp; apt-get install -y ffmpeg&#xA;&#xA;# Set the working directory inside the container&#xA;WORKDIR /usr/src/minimal_docker_webserver_post_error&#xA;&#xA;# Copy the Cargo.toml and Cargo.lock files&#xA;COPY Cargo.toml Cargo.lock ./&#xA;&#xA;# Copy the source code&#xA;COPY src ./src&#xA;&#xA;# Copy wav files&#xA;COPY file1.wav /usr/src/minimal_docker_webserver_post_error/file1.wav&#xA;COPY file2.wav /usr/src/minimal_docker_webserver_post_error/file2.wav&#xA;&#xA;# Install dependencies&#xA;RUN cargo build --release&#xA;&#xA;# Expose the port that the application will run on&#xA;EXPOSE 3030&#xA;&#xA;# Set the entry point to use cargo-watch&#xA;CMD ["cargo", "watch", "-x", "run"]&#xA;

    &#xA;

    Cargo.toml

    &#xA;

    [package]&#xA;name = "minimal_docker_webserver_post_error"&#xA;version = "0.1.0"&#xA;edition = "2021"&#xA;&#xA;# See more keys and their definitions at https://doc.rust-lang.org/cargo/reference/manifest.html&#xA;&#xA;[dependencies]&#xA;warp = "0.3"&#xA;reqwest = { version = "0.12.4", features = ["json"] }&#xA;tokio = { version = "1", features = ["full"] }&#xA;log = "0.4"&#xA;env_logger = "0.11.3"&#xA;

    &#xA;

    Making the request to the warp server

    &#xA;

    curl -v "http://localhost:3030"&#xA;

    &#xA;

  • ffmpeg 4 : Using the stream_loop parameter to loop the audio during a video ends up with an infinite loop

    17 juin 2020, par JarsOfJam-Scheduler

    Summary

    &#xA;&#xA;

      &#xA;
    1. Context
    2. &#xA;

    3. The software I use
    4. &#xA;

    5. The problem
    6. &#xA;

    7. Results
      &#xA;4.1. Actual Results

      &#xA;&#xA;

      4.2. Expected Results

    8. &#xA;

    9. What did I try to fix the bug ?

    10. &#xA;

    11. How to reproduce this bug : minimal and testable example with the provided required data

    12. &#xA;

    13. The question

    14. &#xA;

    15. Sources

    16. &#xA;

    &#xA;&#xA;


    &#xA;&#xA;

    Context

    &#xA;&#xA;

    I would want to set an audio WAV as the background sound of a video WEBM. The video can be shorter or longer than the audio. At the moment I add the audio over the video, I don't know the length of both streams. The audio must repeat until the video ends (the audio can be truncated if the video ends before the end of the last repetition of the audio).

    &#xA;&#xA;

    The software I use

    &#xA;&#xA;

    I use ffmpeg version 4.2.2-1ubuntu1 18.04.sav0.

    &#xA;&#xA;

    The problem

    &#xA;&#xA;

    ffmpeg seems to enter in an infinite loop when it proccesses in order to mix the audio and the video. Also, the length of the currently-generating-output-file (which contains both video and audio) is equal to the length of the audio, instead of the length of the video.

    &#xA;&#xA;

    The problem seems to be triggered by this command line :

    &#xA;&#xA;

    ffmpeg -i directory_1/video.webm -stream_loop -1 -fflags &#x2B;shortest -max_interleave_delta 50000 -i directory_2/audio.wav directory_3/video_and_audio.webm&#xA;

    &#xA;&#xA;

    Results

    &#xA;&#xA;

    Actual Results

    &#xA;&#xA;

    Three things :

    &#xA;&#xA;

      &#xA;
    1. The infinite loop of the ffmpeg process : I must manually stop the ffmpeg process

    2. &#xA;

    3. The output video file with music (which is currently generating but output anyway) : it contains both audio and video. But the length of the output file is equal to the length of the audio, instead of the length of the video.

    4. &#xA;

    5. The following output logs :

    6. &#xA;

    &#xA;&#xA;

    &#xA;

    ffmpeg version 4.2.2-1ubuntu1 18.04.sav0 Copyright (c) 2000-2019 the&#xA; FFmpeg developers built with gcc 7 (Ubuntu 7.5.0-3ubuntu1 18.04)
    &#xA; configuration : —prefix=/usr —extra-version='1ubuntu1 18.04.sav0'&#xA; —toolchain=hardened —libdir=/usr/lib/x86_64-linux-gnu —incdir=/usr/include/x86_64-linux-gnu —arch=amd64 —enable-gpl —disable-stripping —enable-avresample —disable-filter=resample —enable-avisynth —enable-gnutls —enable-ladspa —enable-libaom —enable-libass —enable-libbluray —enable-libbs2b —enable-libcaca —enable-libcdio —enable-libcodec2 —enable-libflite —enable-libfontconfig —enable-libfreetype —enable-libfribidi —enable-libgme —enable-libgsm —enable-libjack —enable-libmp3lame —enable-libmysofa —enable-libopenjpeg —enable-libopenmpt —enable-libopus —enable-libpulse —enable-librsvg —enable-librubberband —enable-libshine —enable-libsnappy —enable-libsoxr —enable-libspeex —enable-libssh —enable-libtheora —enable-libtwolame —enable-libvidstab —enable-libvorbis —enable-libvpx —enable-libwavpack —enable-libwebp —enable-libx265 —enable-libxml2 —enable-libxvid —enable-libzmq —enable-libzvbi —enable-lv2 —enable-omx —enable-openal —enable-opencl —enable-opengl —enable-sdl2 —enable-libdc1394 —enable-libdrm —enable-libiec61883 —enable-nvenc —enable-chromaprint —enable-frei0r —enable-libx264 —enable-shared libavutil 56. 31.100 / 56. 31.100 libavcodec 58. 54.100 / 58. 54.100 libavformat 58. 29.100 / 58. 29.100 libavdevice 58. 8.100 /&#xA; 58. 8.100 libavfilter 7. 57.100 / 7. 57.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 5.100 / 5. 5.100 libswresample 3. 5.100 / 3. 5.100 libpostproc 55. 5.100 /&#xA; 55. 5.100 Input #0, matroska,webm, from 'youtubed/my_youtube_video.webm' : Metadata :&#xA; encoder : Chrome Duration : N/A, start : 0.000000, bitrate : N/A&#xA; Stream #0:0(eng) : Video : vp8, yuv420p(progressive), 3200x1608, SAR 1:1 DAR 400:201, 1k tbr, 1k tbn, 1k tbc (default)&#xA; Metadata :&#xA; alpha_mode : 1 Guessed Channel Layout for Input Stream #1.0 : stereo Input #1, wav, from 'tmp_music/original_music.wav' :
    &#xA; Duration : 00:00:11.78, bitrate : 1411 kb/s&#xA; Stream #1:0 : Audio : pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Stream mapping : Stream #0:0 -> #0:0 (vp8&#xA; (native) -> vp9 (libvpx-vp9)) Stream #1:0 -> #0:1 (pcm_s16le&#xA; (native) -> opus (libopus)) Press [q] to stop, [?] for help&#xA; [libvpx-vp9 @ 0x5645268aed80] v1.8.2 [libopus @ 0x5645268b09c0] No bit&#xA; rate set. Defaulting to 96000 bps. Output #0, webm, to&#xA; 'youtubed/my_youtube_video_with_music.webm' : Metadata :&#xA; encoder : Lavf58.29.100&#xA; Stream #0:0(eng) : Video : vp9 (libvpx-vp9), yuv420p(progressive), 3200x1608 [SAR 1:1 DAR 400:201], q=-1—1, 200 kb/s, 1k fps, 1k tbn, 1k&#xA; tbc (default)&#xA; Metadata :&#xA; alpha_mode : 1&#xA; encoder : Lavc58.54.100 libvpx-vp9&#xA; Side data :&#xA; cpb : bitrate max/min/avg : 0/0/0 buffer size : 0 vbv_delay : -1&#xA; Stream #0:1 : Audio : opus (libopus), 48000 Hz, stereo, s16, 96 kb/s&#xA; Metadata :&#xA; encoder : Lavc58.54.100 libopus

    &#xA;

    &#xA;&#xA;

    Expected Results

    &#xA;&#xA;

      &#xA;
    1. No infinite loop during the ffmpeg process

    2. &#xA;

    3. Concerning the output logs, I don't know what it should look.

    4. &#xA;

    5. The output file with the audio and the video should :

      &#xA;&#xA;

      3.1. If the video is longer than the audio, then the audio is repeated until it exactly fits the video. The audio can be truncated.

      &#xA;&#xA;

      3.2. If the video is shorter than the audio, then the audio is truncated and exactly fits the video.

      &#xA;&#xA;

      3.3. If both video and audio are of the same length, then the audio exactly fits the video.

    6. &#xA;

    &#xA;&#xA;

    How to reproduce this bug ? (+ required data)

    &#xA;&#xA;

      &#xA;
    1. Download the following files (resp. audio and video) (I must refresh these download links every 24 hours) :

      &#xA;&#xA;

      1.1. https://a.uguu.se/dmgsmItjJMDq_audio.wav

      &#xA;&#xA;

      1.2. https://a.uguu.se/w3qHDlGq6mOW_video.webm

    2. &#xA;

    3. Move them into the directory/directories of your choice.

    4. &#xA;

    5. Open your CLI, move to the adequat directory and copy/paste/execute the instruction given in Part. The Problem (don't forget to eventually modify this instruction by indicating the adequat directories, according to step 2.).

    6. &#xA;

    7. You'll face my problem.

    8. &#xA;

    &#xA;&#xA;

    What did I try to fix the bug ?

    &#xA;&#xA;

    Nothing, since I don't even understand why the bug occures.

    &#xA;&#xA;

    The question

    &#xA;&#xA;

    How to correct my command in order to mix these audio and video streams without any infinite loop during the ffmpeg process, keeping in mind that I don't know their length, and that audio must be repeated in order to fit the video, even if audio must be truncated (in the case of the last repetition of the audio file must be truncated because the video stream has just ended) ?

    &#xA;&#xA;

    Sources

    &#xA;&#xA;

    The source is the command line you can find in Part. The problem.

    &#xA;

  • FFmpeg with Nvidia GPU - full HW transcode with 50i to 50p deinterlacing

    5 janvier 2018, par Jernej Stopinšek

    I’m trying to do a full hardware transcode of an udp stream to hls
    with 50i to 50p deinterlacing.

    I’m using ffmpeg and Nvidia GPU.

    Since HLS requires deinterlacing

    https://developer.apple.com/library/content/documentation/General/Reference/HLSAuthoringSpec/Requirements.html

    I would like to deinterlace an interlaced source stream and preserve
    as much smooth motion and picture quality as possible.

    My hardware, software and driver info :

    GPU : Tesla P100-PCIE-12GB
    Nvidia Driver Version : 387.26
    Cuda compilation tools, release 9.1, V9.1.85
    FFmpeg from git on 20171218

    ffmpeg version N-89520-g3f88744067 Copyright (c) 2000-2017 the FFmpeg
    developers built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
    configuration : —enable-gpl
    —enable-cuda-sdk —enable-libx264 —enable-libx265 —enable-nonfree —enable-libnpp —enable-opengl —enable-opencl —enable-libfreetype —enable-openssl —enable-libzvbi —enable-libfontconfig —enable-libfreetype —enable-libfribidi —extra-cflags=-I/usr/local/cuda/include —extra-ldflags=-L/usr/local/cuda/lib64 —arch=x86_64

    libavutil 56. 6.100 / 56. 6.100
    libavcodec 58. 8.100 / 58.
    8.100
    libavformat 58. 3.100 / 58. 3.100
    libavdevice 58. 0.100 / 58. 0.100
    libavfilter 7. 7.100 / 7. 7.100
    libswscale 5.
    0.101 / 5. 0.101
    libswresample 3. 0.101 / 3. 0.101
    libpostproc 55. 0.100 / 55. 0.100

    Input stream info :

    ffmpeg -t 00:05:00 -i udp://xxx.xxx.xxx.xxx:xxxx -map 0:0 -vf idet -c rawvideo -y -f rawvideo /dev/null

    Input #0, mpegts, from ’udp ://xxx.xxx.xxx.xxx:xxxx’ :
    Duration :
    N/A, start : 49634.159411, bitrate : N/A
    Program xxxxx
    Metadata : service_name :
    service_provider : Stream
    #0:0[0x44d] : Video : h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k
    tbn, 50 tbc
    Stream #0:10x19de : Audio : mp2 ([3][0][0][0] /
    0x0003), 48000 Hz, stereo, s16p, 192 kb/s
    Stream
    #0:20x19e1 : Subtitle : dvb_subtitle ([6][0][0][0] / 0x0006)

    Output #0, rawvideo, to ’/dev/null’ :
    Metadata :
    encoder :
    Lavf58.3.100
    Stream #0:0 : Video : rawvideo (I420 / 0x30323449),
    yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 622080 kb/s, 25 fps, 25
    tbn, 25 tbc
    Metadata :
    encoder : Lavc58.8.100 rawvideo
    frame= 7538 fps= 25 q=-0.0 Lsize=22896675kB time=00:05:01.52
    bitrate=622080.0kbits/s dup=38 drop=0 speed=1.02x
    video:22896675kB audio:0kB subtitle:0kB other streams:0kB global
    headers:0kB muxing overhead : 0.000000%
    [Parsed_idet_0 @
    0x56370b3c5080] Repeated Fields : Neither : 7458 Top : 24 Bottom : 18
    [Parsed_idet_0 @ 0x56370b3c5080] Single frame detection : TFF : 281 BFF :
    13 Progressive : 5639 Undetermined : 1567
    [Parsed_idet_0 @
    0x56370b3c5080] Multi frame detection : TFF : 380 BFF : 0 Progressive :
    7120 Undetermined : 0


    This is my command for adaptive hardware deinterlacing. It gives great results with picture, but sound is out of sync.

    ffmpeg -y -err_detect ignore_err -loglevel debug -vsync -1 -hwaccel cuvid -hwaccel_device 1 -c:v h264_cuvid -deint adaptive -r:v 50 -gpu:v 1 -i "udp://xxx.xxx.xxx.xxx:xxxx=?overrun_nonfatal=1&amp;fifo_size=84450&amp;buffer_size=33554432" -map 0:0 -map 0:1 -c:a aac -b:a 196k -c:v h264_nvenc -flags -global_header+cgop -gpu:v 1 -g:v 50 -bf:v 4 -coder:v cabac -b_adapt:v false -b:v 5184000 -minrate:v 5184000 -maxrate:v 5184000 -bufsize:v 2488320 -rc:v cbr_hq -2pass:v true -rc-lookahead:v 25 -no-scenecut:v 1 -profile:v high -preset:v slow -color_range:v 1 -color_trc:v 1 -color_primaries:v 1 -colorspace:v 1 -f hls -hls_time 5 -hls_list_size 3 -start_number 0 -hls_flags delete_segments /srv/hls/program_01/1080p/index.m3u8

    If I add option "-drop_second_field 1" to h264_cuvid and remove -r:v 50 from input and put it to h264_nvenc - then transcoded stream has synced audio, but I think I’m losing quality due to drop_second_field option.

    ffmpeg -y -err_detect ignore_err -loglevel debug -vsync -1 -hwaccel cuvid -hwaccel_device 1 -c:v h264_cuvid -deint adaptive -drop_second_field 1 -gpu:v 1 -i "udp://xxx.xxx.xxx.xxx:xxxx=?overrun_nonfatal=1&amp;fifo_size=84450&amp;buffer_size=33554432" -map 0:0 -map 0:1 -c:a aac -b:a 196k -c:v h264_nvenc -flags -global_header+cgop -gpu:v 1 -g:v 50 -r:v 50 -bf:v 4 -coder:v cabac -b_adapt:v false -b:v 5184000 -minrate:v 5184000 -maxrate:v 5184000 -bufsize:v 2488320 -rc:v cbr_hq -2pass:v true -rc-lookahead:v 25 -no-scenecut:v 1 -profile:v high -preset:v slow -color_range:v 1 -color_trc:v 1 -color_primaries:v 1 -colorspace:v 1 -f hls -hls_time 5 -hls_list_size 3 -start_number 0 -hls_flags delete_segments /srv/hls/program_01/1080p/index.m3u8

    Could someone please point me in the right direction how to properly deinterlace with cuvid and minimal possible loss of quality ?