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  • ffmpeg slower than downloading

    7 août 2017, par Krylic

    When trying to get a clip of a video from a remote source
    Input source :

    ffprobe version 3.3.2-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2007-2017 the FFmpeg developers
    Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'http://website.com/video.mp4':
     Metadata:
       major_brand     : mp42
       minor_version   : 0
       compatible_brands: mp42mp41
       creation_time   : 2017-07-13T15:44:58.000000Z
     Duration: 00:57:32.42, start: 0.000000, bitrate: 1939 kb/s
       Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv), 720x480 [SAR 10:11 DAR 15:11], 1745 kb/s, 29.97 fps, 29.97 tbr, 29970 tbn, 59.94 tbc (default)
       Metadata:
         creation_time   : 2017-07-13T15:44:58.000000Z
         handler_name    : Mainconcept MP4 Video Media Handler
         encoder         : AVC Coding
       Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 189 kb/s (default)
       Metadata:
         creation_time   : 2017-07-13T15:44:58.000000Z
         handler_name    : Mainconcept MP4 Sound Media Handler

    Current command :
    ffmpeg version 3.3.2-static
    ffmpeg.linux -threads 2 -y -ss 3273 -i "http://website.com/video.mp4" -an -movflags +faststart -preset veryfast -codec copy /outputfolder/trimmed_video.mp4
    This 5m35.102s to create a 45mb 2min file.

    If I download the file using wget it takes 28s and using ffmpeg only takes 0.243s

    If I add -vn OR -an to the output portion of the command it completes the trim in about 2.101s. Meaning it’s faster to download the two stream and merge them myself.

    Can anyone explain this behaviour and why my first command takes so long when on a lot of other video files it’s very fast ?

  • Trouble syncing libavformat/ffmpeg with x264 and RTP

    26 décembre 2012, par Jacob Peddicord

    I've been working on some streaming software that takes live feeds
    from various kinds of cameras and streams over the network using
    H.264. To accomplish this, I'm using the x264 encoder directly (with
    the "zerolatency" preset) and feeding NALs as they are available to
    libavformat to pack into RTP (ultimately RTSP). Ideally, this
    application should be as real-time as possible. For the most part,
    this has been working well.

    Unfortunately, however, there is some sort of synchronization issue :
    any video playback on clients seems to show a few smooth frames,
    followed by a short pause, then more frames ; repeat. Additionally,
    there appears to be approximately a 4-second delay. This happens with
    every video player I've tried : Totem, VLC, and basic gstreamer pipes.

    I've boiled it all down to a somewhat small test case :

    #include 
    
#include 
    
#include 
    
#include 
    
#include <libavformat></libavformat>avformat.h>
    
#include <libswscale></libswscale>swscale.h>
    

    
#define WIDTH       640
    
#define HEIGHT      480
    
#define FPS         30
    
#define BITRATE     400000
    
#define RTP_ADDRESS "127.0.0.1"
    
#define RTP_PORT    49990
    

    
struct AVFormatContext* avctx;
    
struct x264_t* encoder;
    
struct SwsContext* imgctx;
    

    
uint8_t test = 0x80;
    

    

    
void create_sample_picture(x264_picture_t* picture)
    
{
    
    // create a frame to store in
    
    x264_picture_alloc(picture, X264_CSP_I420, WIDTH, HEIGHT);
    

    
    // fake image generation
    
    // disregard how wrong this is; just writing a quick test
    
    int strides = WIDTH / 8;
    
    uint8_t* data = malloc(WIDTH * HEIGHT * 3);
    
    memset(data, test, WIDTH * HEIGHT * 3);
    
    test = (test &lt;&lt; 1) | (test >> (8 - 1));
    

    
    // scale the image
    
    sws_scale(imgctx, (const uint8_t* const*) &amp;data, &amp;strides, 0, HEIGHT,
    
              picture->img.plane, picture->img.i_stride);
    
}
    

    
int encode_frame(x264_picture_t* picture, x264_nal_t** nals)
    
{
    
    // encode a frame
    
    x264_picture_t pic_out;
    
    int num_nals;
    
    int frame_size = x264_encoder_encode(encoder, nals, &amp;num_nals, picture, &amp;pic_out);
    

    
    // ignore bad frames
    
    if (frame_size &lt; 0)
    
    {
    
        return frame_size;
    
    }
    

    
    return num_nals;
    
}
    

    
void stream_frame(uint8_t* payload, int size)
    
{
    
    // initalize a packet
    
    AVPacket p;
    
    av_init_packet(&amp;p);
    
    p.data = payload;
    
    p.size = size;
    
    p.stream_index = 0;
    
    p.flags = AV_PKT_FLAG_KEY;
    
    p.pts = AV_NOPTS_VALUE;
    
    p.dts = AV_NOPTS_VALUE;
    

    
    // send it out
    
    av_interleaved_write_frame(avctx, &amp;p);
    
}
    

    
int main(int argc, char* argv[])
    
{
    
    // initalize ffmpeg
    
    av_register_all();
    

    
    // set up image scaler
    
    // (in-width, in-height, in-format, out-width, out-height, out-format, scaling-method, 0, 0, 0)
    
    imgctx = sws_getContext(WIDTH, HEIGHT, PIX_FMT_MONOWHITE,
    
                            WIDTH, HEIGHT, PIX_FMT_YUV420P,
    
                            SWS_FAST_BILINEAR, NULL, NULL, NULL);
    

    
    // set up encoder presets
    
    x264_param_t param;
    
    x264_param_default_preset(&amp;param, "ultrafast", "zerolatency");
    

    
    param.i_threads = 3;
    
    param.i_width = WIDTH;
    
    param.i_height = HEIGHT;
    
    param.i_fps_num = FPS;
    
    param.i_fps_den = 1;
    
    param.i_keyint_max = FPS;
    
    param.b_intra_refresh = 0;
    
    param.rc.i_bitrate = BITRATE;
    
    param.b_repeat_headers = 1; // whether to repeat headers or write just once
    
    param.b_annexb = 1;         // place start codes (1) or sizes (0)
    

    
    // initalize
    
    x264_param_apply_profile(&amp;param, "high");
    
    encoder = x264_encoder_open(&amp;param);
    

    
    // at this point, x264_encoder_headers can be used, but it has had no effect
    

    
    // set up streaming context. a lot of error handling has been ommitted
    
    // for brevity, but this should be pretty standard.
    
    avctx = avformat_alloc_context();
    
    struct AVOutputFormat* fmt = av_guess_format("rtp", NULL, NULL);
    
    avctx->oformat = fmt;
    

    
    snprintf(avctx->filename, sizeof(avctx->filename), "rtp://%s:%d", RTP_ADDRESS, RTP_PORT);
    
    if (url_fopen(&amp;avctx->pb, avctx->filename, URL_WRONLY) &lt; 0)
    
    {
    
        perror("url_fopen failed");
    
        return 1;
    
    }
    
    struct AVStream* stream = av_new_stream(avctx, 1);
    

    
    // initalize codec
    
    AVCodecContext* c = stream->codec;
    
    c->codec_id = CODEC_ID_H264;
    
    c->codec_type = AVMEDIA_TYPE_VIDEO;
    
    c->flags = CODEC_FLAG_GLOBAL_HEADER;
    
    c->width = WIDTH;
    
    c->height = HEIGHT;
    
    c->time_base.den = FPS;
    
    c->time_base.num = 1;
    
    c->gop_size = FPS;
    
    c->bit_rate = BITRATE;
    
    avctx->flags = AVFMT_FLAG_RTP_HINT;
    

    
    // write the header
    
    av_write_header(avctx);
    

    
    // make some frames
    
    for (int frame = 0; frame &lt; 10000; frame++)
    
    {
    
        // create a sample moving frame
    
        x264_picture_t* pic = (x264_picture_t*) malloc(sizeof(x264_picture_t));
    
        create_sample_picture(pic);
    

    
        // encode the frame
    
        x264_nal_t* nals;
    
        int num_nals = encode_frame(pic, &amp;nals);
    

    
        if (num_nals &lt; 0)
    
            printf("invalid frame size: %d\n", num_nals);
    

    
        // send out NALs
    
        for (int i = 0; i &lt; num_nals; i++)
    
        {
    
            stream_frame(nals[i].p_payload, nals[i].i_payload);
    
        }
    

    
        // free up resources
    
        x264_picture_clean(pic);
    
        free(pic);
    

    
        // stream at approx 30 fps
    
        printf("frame %d\n", frame);
    
        usleep(33333);
    
    }
    

    
    return 0;
    
}

    This test shows black lines on a white background that
    should move smoothly to the left. It has been written for ffmpeg 0.6.5
    but the problem can be reproduced on 0.8 and 0.10 (from what I've tested so far). I've taken some shortcuts in error handling to make this example as short as
    possible while still showing the problem, so please excuse some of the
    nasty code. I should also note that while an SDP is not used here, I
    have tried using that already with similar results. The test can be
    compiled with :

    gcc -g -std=gnu99 streamtest.c -lswscale -lavformat -lx264 -lm -lpthread -o streamtest

    It can be played with gtreamer directly :

    gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink

    You should immediately notice the stuttering. One common "fix" I've
    seen all over the Internet is to add sync=false to the pipeline :

    gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink sync=false

    This causes playback to be smooth (and near-realtime), but is a
    non-solution and only works with gstreamer. I'd like to fix the
    problem at the source. I've been able to stream with near-identical
    parameters using raw ffmpeg and haven't had any issues :

    ffmpeg -re -i sample.mp4 -vcodec libx264 -vpre ultrafast -vpre baseline -b 400000 -an -f rtp rtp://127.0.0.1:49990 -an

    So clearly I'm doing something wrong. But what is it ?

  • Trouble syncing libavformat/ffmpeg with x264 and RTP

    26 décembre 2012, par Jacob Peddicord

    I've been working on some streaming software that takes live feeds
    from various kinds of cameras and streams over the network using
    H.264. To accomplish this, I'm using the x264 encoder directly (with
    the "zerolatency" preset) and feeding NALs as they are available to
    libavformat to pack into RTP (ultimately RTSP). Ideally, this
    application should be as real-time as possible. For the most part,
    this has been working well.

    Unfortunately, however, there is some sort of synchronization issue :
    any video playback on clients seems to show a few smooth frames,
    followed by a short pause, then more frames ; repeat. Additionally,
    there appears to be approximately a 4-second delay. This happens with
    every video player I've tried : Totem, VLC, and basic gstreamer pipes.

    I've boiled it all down to a somewhat small test case :

    #include 
    
#include 
    
#include 
    
#include 
    
#include <libavformat></libavformat>avformat.h>
    
#include <libswscale></libswscale>swscale.h>
    

    
#define WIDTH       640
    
#define HEIGHT      480
    
#define FPS         30
    
#define BITRATE     400000
    
#define RTP_ADDRESS "127.0.0.1"
    
#define RTP_PORT    49990
    

    
struct AVFormatContext* avctx;
    
struct x264_t* encoder;
    
struct SwsContext* imgctx;
    

    
uint8_t test = 0x80;
    

    

    
void create_sample_picture(x264_picture_t* picture)
    
{
    
    // create a frame to store in
    
    x264_picture_alloc(picture, X264_CSP_I420, WIDTH, HEIGHT);
    

    
    // fake image generation
    
    // disregard how wrong this is; just writing a quick test
    
    int strides = WIDTH / 8;
    
    uint8_t* data = malloc(WIDTH * HEIGHT * 3);
    
    memset(data, test, WIDTH * HEIGHT * 3);
    
    test = (test &lt;&lt; 1) | (test >> (8 - 1));
    

    
    // scale the image
    
    sws_scale(imgctx, (const uint8_t* const*) &amp;data, &amp;strides, 0, HEIGHT,
    
              picture->img.plane, picture->img.i_stride);
    
}
    

    
int encode_frame(x264_picture_t* picture, x264_nal_t** nals)
    
{
    
    // encode a frame
    
    x264_picture_t pic_out;
    
    int num_nals;
    
    int frame_size = x264_encoder_encode(encoder, nals, &amp;num_nals, picture, &amp;pic_out);
    

    
    // ignore bad frames
    
    if (frame_size &lt; 0)
    
    {
    
        return frame_size;
    
    }
    

    
    return num_nals;
    
}
    

    
void stream_frame(uint8_t* payload, int size)
    
{
    
    // initalize a packet
    
    AVPacket p;
    
    av_init_packet(&amp;p);
    
    p.data = payload;
    
    p.size = size;
    
    p.stream_index = 0;
    
    p.flags = AV_PKT_FLAG_KEY;
    
    p.pts = AV_NOPTS_VALUE;
    
    p.dts = AV_NOPTS_VALUE;
    

    
    // send it out
    
    av_interleaved_write_frame(avctx, &amp;p);
    
}
    

    
int main(int argc, char* argv[])
    
{
    
    // initalize ffmpeg
    
    av_register_all();
    

    
    // set up image scaler
    
    // (in-width, in-height, in-format, out-width, out-height, out-format, scaling-method, 0, 0, 0)
    
    imgctx = sws_getContext(WIDTH, HEIGHT, PIX_FMT_MONOWHITE,
    
                            WIDTH, HEIGHT, PIX_FMT_YUV420P,
    
                            SWS_FAST_BILINEAR, NULL, NULL, NULL);
    

    
    // set up encoder presets
    
    x264_param_t param;
    
    x264_param_default_preset(&amp;param, "ultrafast", "zerolatency");
    

    
    param.i_threads = 3;
    
    param.i_width = WIDTH;
    
    param.i_height = HEIGHT;
    
    param.i_fps_num = FPS;
    
    param.i_fps_den = 1;
    
    param.i_keyint_max = FPS;
    
    param.b_intra_refresh = 0;
    
    param.rc.i_bitrate = BITRATE;
    
    param.b_repeat_headers = 1; // whether to repeat headers or write just once
    
    param.b_annexb = 1;         // place start codes (1) or sizes (0)
    

    
    // initalize
    
    x264_param_apply_profile(&amp;param, "high");
    
    encoder = x264_encoder_open(&amp;param);
    

    
    // at this point, x264_encoder_headers can be used, but it has had no effect
    

    
    // set up streaming context. a lot of error handling has been ommitted
    
    // for brevity, but this should be pretty standard.
    
    avctx = avformat_alloc_context();
    
    struct AVOutputFormat* fmt = av_guess_format("rtp", NULL, NULL);
    
    avctx->oformat = fmt;
    

    
    snprintf(avctx->filename, sizeof(avctx->filename), "rtp://%s:%d", RTP_ADDRESS, RTP_PORT);
    
    if (url_fopen(&amp;avctx->pb, avctx->filename, URL_WRONLY) &lt; 0)
    
    {
    
        perror("url_fopen failed");
    
        return 1;
    
    }
    
    struct AVStream* stream = av_new_stream(avctx, 1);
    

    
    // initalize codec
    
    AVCodecContext* c = stream->codec;
    
    c->codec_id = CODEC_ID_H264;
    
    c->codec_type = AVMEDIA_TYPE_VIDEO;
    
    c->flags = CODEC_FLAG_GLOBAL_HEADER;
    
    c->width = WIDTH;
    
    c->height = HEIGHT;
    
    c->time_base.den = FPS;
    
    c->time_base.num = 1;
    
    c->gop_size = FPS;
    
    c->bit_rate = BITRATE;
    
    avctx->flags = AVFMT_FLAG_RTP_HINT;
    

    
    // write the header
    
    av_write_header(avctx);
    

    
    // make some frames
    
    for (int frame = 0; frame &lt; 10000; frame++)
    
    {
    
        // create a sample moving frame
    
        x264_picture_t* pic = (x264_picture_t*) malloc(sizeof(x264_picture_t));
    
        create_sample_picture(pic);
    

    
        // encode the frame
    
        x264_nal_t* nals;
    
        int num_nals = encode_frame(pic, &amp;nals);
    

    
        if (num_nals &lt; 0)
    
            printf("invalid frame size: %d\n", num_nals);
    

    
        // send out NALs
    
        for (int i = 0; i &lt; num_nals; i++)
    
        {
    
            stream_frame(nals[i].p_payload, nals[i].i_payload);
    
        }
    

    
        // free up resources
    
        x264_picture_clean(pic);
    
        free(pic);
    

    
        // stream at approx 30 fps
    
        printf("frame %d\n", frame);
    
        usleep(33333);
    
    }
    

    
    return 0;
    
}

    This test shows black lines on a white background that
    should move smoothly to the left. It has been written for ffmpeg 0.6.5
    but the problem can be reproduced on 0.8 and 0.10 (from what I've tested so far). I've taken some shortcuts in error handling to make this example as short as
    possible while still showing the problem, so please excuse some of the
    nasty code. I should also note that while an SDP is not used here, I
    have tried using that already with similar results. The test can be
    compiled with :

    gcc -g -std=gnu99 streamtest.c -lswscale -lavformat -lx264 -lm -lpthread -o streamtest

    It can be played with gtreamer directly :

    gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink

    You should immediately notice the stuttering. One common "fix" I've
    seen all over the Internet is to add sync=false to the pipeline :

    gst-launch udpsrc port=49990 ! application/x-rtp,payload=96,clock-rate=90000 ! rtph264depay ! decodebin ! xvimagesink sync=false

    This causes playback to be smooth (and near-realtime), but is a
    non-solution and only works with gstreamer. I'd like to fix the
    problem at the source. I've been able to stream with near-identical
    parameters using raw ffmpeg and haven't had any issues :

    ffmpeg -re -i sample.mp4 -vcodec libx264 -vpre ultrafast -vpre baseline -b 400000 -an -f rtp rtp://127.0.0.1:49990 -an

    So clearly I'm doing something wrong. But what is it ?

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