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  • ffmpeg generates file with only audio from ts

    8 novembre 2016, par Seoman

    I’m trying to get some parts of a ts video and then merge them into a new video. The problem is that when doing the split, the video is lost. The splits only contains audio. Here is the video source information :

    ffmpeg -i source_file.ts

    ffmpeg version 3.0.1 Copyright (c) 2000-2016 the FFmpeg developers
    
built with Apple LLVM version 7.3.0 (clang-703.0.29)
    
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.0.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-vda
    
libavutil      55. 17.103 / 55. 17.103
    
libavcodec     57. 24.102 / 57. 24.102
    
libavformat    57. 25.100 / 57. 25.100
    
libavdevice    57.  0.101 / 57.  0.101
    
libavfilter     6. 31.100 /  6. 31.100
    
libavresample   3.  0.  0 /  3.  0.  0
    
libswscale      4.  0.100 /  4.  0.100
    
libswresample   2.  0.101 /  2.  0.101
    
libpostproc    54.  0.100 / 54.  0.100
    
Input #0, mpegts, from '97274181.ts':
    
Duration: 00:02:30.70, start: 60.000000, bitrate: 3582 kb/s
    
Program 1 
    
Stream #0:0[0x100]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 139 kb/s
    
Stream #0:1[0x101]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(tv, unknown/bt709/unknown), 1920x1080, 60 fps, 60 tbr, 90k tbn, 2k tbc
    
Stream #0:2[0x102]: Data: timed_id3 (ID3  / 0x20334449)
    
At least one output file must be specified

    This is the command used for doing the split :

    ffmpeg -ss 10 -i source_file.ts -t 5 -c copy cut1.ts

    And this is the video information after the cut :

    ffmpeg version 3.0.1 Copyright (c) 2000-2016 the FFmpeg developers
    
built with Apple LLVM version 7.3.0 (clang-703.0.29)
    
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.0.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-vda
    
libavutil      55. 17.103 / 55. 17.103
    
libavcodec     57. 24.102 / 57. 24.102
    
libavformat    57. 25.100 / 57. 25.100
    
libavdevice    57.  0.101 / 57.  0.101
    
libavfilter     6. 31.100 /  6. 31.100
    
libavresample   3.  0.  0 /  3.  0.  0
    
libswscale      4.  0.100 /  4.  0.100
    
libswresample   2.  0.101 /  2.  0.101
    
libpostproc    54.  0.100 / 54.  0.100
    
[NULL @ 0x7fc9ba008000] start time for stream 0 is not set in estimate_timings_from_pts
    
[mpegts @ 0x7fc9b9800000] Could not find codec parameters for stream 0 (Video: h264 ([27][0][0][0] / 0x001B), none): unspecified size
    
Consider increasing the value for the 'analyzeduration' and 'probesize' options
    
Input #0, mpegts, from 'siri1.ts':
    
Duration: 00:00:05.08, start: 1.400000, bitrate: 148 kb/s
    
Program 1 
    
Metadata:
    
  service_name    : Service01
    
  service_provider: FFmpeg
    
Stream #0:0[0x100]: Video: h264 ([27][0][0][0] / 0x001B), none, 90k tbr, 90k tbn, 180k tbc
    
Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 120 kb/s
    
At least one output file must be specified

    If I reproduce cut1.ts I can only hear the audio. Video is gone.

  • FFmpeg(C/libav) VPX to mpeg2video stream cannot be reproduce in VLC

    21 septembre 2017, par caiomcg

    I am currently trying to transcode a VPX(VP8/VP9) video to a mpeg2video and stream it over UDP with mpegts.

    I have initialized all of the contexts and the streams and as long as I stream it to ffplay it works, if I send the stream to VLC or another player, the receiver only display the first frame and do nothing else. If I do the same thing through the command line it works flawlessly - ffmpeg -re -i video.webm -an -f mpegts udp://127.0.0.1:8080

    My output context :

    this->output_codec_ctx_->codec_type    = AVMEDIA_TYPE_VIDEO; // Set media type
    
this->output_codec_ctx_->pix_fmt       = AV_PIX_FMT_YUV420P; // Set stream pixel format
    
this->output_codec_ctx_->time_base.den = ceil(av_q2d(input_stream->r_frame_rate));     // Add the real video framerate. Eg.: 29.9
    
this->output_codec_ctx_->time_base.num = 1;                  // Numerator of the framerate. Eg.: num/29.9
    
this->output_codec_ctx_->width         = input_stream->codecpar->width;  // Video width
    
this->output_codec_ctx_->height        = input_stream->codecpar->height; // Video height
    
this->output_codec_ctx_->bit_rate      = 400000; // Video quality
    
this->output_codec_ctx_->gop_size      = 12;
    
this->output_codec_ctx_->max_b_frames  = 2;
    
this->output_codec_ctx_->framerate     = this->input_codec_ctx_->framerate;
    
this->output_codec_ctx_->sample_aspect_ratio     = this->input_codec_ctx_->sample_aspect_ratio;

    My av_dump :

    Output #0, mpegts, to 'udp://127.0.0.1:20010':
    
  Metadata:
    
    encoder         : Lavf57.72.101
    
    Stream #0:0: Video: mpeg2video (Main), 1 reference frame, yuv420p, 480x640 (0x0), q=2-31, 400 kb/s, SAR 1:1 DAR 3:4, 24 fps, 24 tbr, 90k tbn

    FFMPEG av_dump :

    Output #0, mpegts, to 'udp://127.0.0.1:20010':
    
  Metadata:
    
    title           : Tears of Steel
    
    encoder         : Lavf57.72.101
    
    Stream #0:0: Video: mpeg2video (Main), yuv420p, 480x640 [SAR 1:1 DAR 3:4], q=2-31, 200 kb/s, 24 fps, 90k tbn, 24 tbc (default)
    
    Metadata:
    
      encoder         : Lavc57.96.101 mpeg2video
    
    Side data:
    
      cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1

    Any ideia on what I may be doing wrong ?

  • ffmpeg create RTP stream

    27 novembre 2016, par DankMemes

    I’m trying to encode and stream using ffmpeg (libavcodec/libavformat - MSVC x64 with Zeranoe builds)

    Here is my code, largely adapted from the encoding example, error handling removed

    #include "stdafx.h"
    
extern "C" {
    
#include <libavformat></libavformat>avformat.h>
    
#include <libavcodec></libavcodec>avcodec.h>
    

    
#include <libavutil></libavutil>opt.h>
    
#include <libavutil></libavutil>channel_layout.h>
    
#include <libavutil></libavutil>common.h>
    
#include <libavutil></libavutil>imgutils.h>
    
#include <libavutil></libavutil>mathematics.h>
    
#include <libavutil></libavutil>samplefmt.h>
    
}
    
#pragma comment(lib, "avformat.lib")
    
#pragma comment(lib, "avutil.lib")
    
#pragma comment(lib, "avcodec.lib")
    

    
int main() {
    
    avcodec_register_all();
    
    av_register_all();
    
    avformat_network_init();
    

    
    AVCodecID codec_id = AV_CODEC_ID_H264;
    
    AVCodec *codec;
    
    AVCodecContext *c = NULL;
    
    int i, ret, x, y, got_output;
    
    AVFrame *frame;
    
    AVPacket pkt;
    

    
    codec = avcodec_find_encoder(codec_id);
    
    c = avcodec_alloc_context3(codec);
    

    
    c->bit_rate = 400000;
    
    c->width = 352;
    
    c->height = 288;
    
    c->time_base.num = 1;
    
    c->time_base.den = 25;
    
    c->gop_size = 25;
    
    c->max_b_frames = 1;
    
    c->pix_fmt = AV_PIX_FMT_YUV420P;
    
    c->codec_type = AVMEDIA_TYPE_VIDEO;
    
    c->flags = CODEC_FLAG_GLOBAL_HEADER;
    

    
    if (codec_id == AV_CODEC_ID_H264) {
    
        ret = av_opt_set(c->priv_data, "preset", "ultrafast", 0);
    
        ret = av_opt_set(c->priv_data, "tune", "zerolatency", 0);
    
    }
    

    
    avcodec_open2(c, codec, NULL)
    

    
    frame = av_frame_alloc();
    
    frame->format = c->pix_fmt;
    
    frame->width = c->width;
    
    frame->height = c->height;
    
    ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
    
        c->pix_fmt, 32);
    

    
    AVFormatContext* avfctx;
    
    AVOutputFormat* fmt = av_guess_format("rtp", NULL, NULL);
    

    
    ret = avformat_alloc_output_context2(&amp;avfctx, fmt, fmt->name,
    
        "rtp://127.0.0.1:49990");
    

    
    printf("Writing to %s\n", avfctx->filename);
    

    
    avio_open(&amp;avfctx->pb, avfctx->filename, AVIO_FLAG_WRITE)
    

    
    struct AVStream* stream = avformat_new_stream(avfctx, codec);
    
    stream->codecpar->bit_rate = 400000;
    
    stream->codecpar->width = 352;
    
    stream->codecpar->height = 288;
    
    stream->codecpar->codec_id = AV_CODEC_ID_H264;
    
    stream->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
    
    stream->time_base.num = 1;
    
    stream->time_base.den = 25;
    

    
    avformat_write_header(avfctx, NULL);
    
    char buf[200000];
    
    AVFormatContext *ac[] = { avfctx };
    
    av_sdp_create(ac, 1, buf, 20000);
    
    printf("sdp:\n%s\n", buf);
    
    FILE* fsdp;
    
    fopen_s(&amp;fsdp, "test.sdp", "w");
    
    fprintf(fsdp, "%s", buf);
    
    fclose(fsdp);
    
    system("PAUSE");
    
    system("start "" \"C:\\Program Files (x86)\\VideoLAN\\VLC\\vlc.exe\" test.sdp");
    

    
    int j = 0;
    
    for (i = 0; i &lt; 10000; i++) {
    
        av_init_packet(&amp;pkt);
    
        pkt.data = NULL;    // packet data will be allocated by the encoder
    
        pkt.size = 0;
    
        fflush(stdout);
    
        /* prepare a dummy image */
    
        /* Y */
    
        for (y = 0; y &lt; c->height; y++) {
    
            for (x = 0; x &lt; c->width; x++) {
    
                frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
    
            }
    
        }
    
        /* Cb and Cr */
    
        for (y = 0; y &lt; c->height / 2; y++) {
    
            for (x = 0; x &lt; c->width / 2; x++) {
    
                frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
    
                frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
    
            }
    
        }
    
        frame->pts = i;
    
        /* encode the image */
    
        ret = avcodec_send_frame(c, frame);
    
        ret = avcodec_receive_packet(c, &amp;pkt);
    

    
        if (ret == AVERROR_EOF) {
    
            got_output = false;
    
            printf("Stream EOF\n");
    
        } else if(ret == AVERROR(EAGAIN)) {
    
            got_output = false;
    
            printf("Stream EAGAIN\n");
    
        } else {
    
            got_output = true;
    
        }
    

    
        if (got_output) {
    
            printf("Write frame %3d (size=%5d)\n", j++, pkt.size);
    
            av_interleaved_write_frame(avfctx, &amp;pkt);
    
            av_packet_unref(&amp;pkt);
    
        }
    

    
        Sleep(40);
    
    }
    

    
    // end
    
    ret = avcodec_send_frame(c, NULL);
    

    
    /* get the delayed frames */
    
    for (; ; i++) {
    
        fflush(stdout);
    
        ret = avcodec_receive_packet(c, &amp;pkt);
    
        if (ret == AVERROR_EOF) {
    
            printf("Stream EOF\n");
    
            break;
    
        } else if (ret == AVERROR(EAGAIN)) {
    
            printf("Stream EAGAIN\n");
    
            got_output = false;
    
        } else {
    
            got_output = true;
    
        }
    

    
        if (got_output) {
    
            printf("Write frame %3d (size=%5d)\n", j++, pkt.size);
    
            av_interleaved_write_frame(avfctx, &amp;pkt);
    
            av_packet_unref(&amp;pkt);
    
        }
    
    }
    

    
    avcodec_close(c);
    
    av_free(c);
    
    av_freep(&amp;frame->data[0]);
    
    av_frame_free(&amp;frame);
    
    printf("\n");
    
    system("pause");
    
    return 0;
    
}

    However VLC (opened with the generated SDP file) isn’t able to play the stream. Messages has this

    core error: ES_OUT_RESET_PCR called

    followed by repeated

    packetizer_h264 warning: waiting for SPS/PPS
    
core debug: Buffering <some percent="percent">%
    
</some>

    What am I doing wrong ?

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