Recherche avancée

Sur d’autres sites (13168)

• FFMPEG rtmp live stream with android fps is too low

  27 novembre 2015, par 蔡明原

  I’m interesting live stream from android to youtube.
  I have compile ffmpeg and write jni to live stream with android.
  but the fps is too low, it always has 3 5 fps, the user view is not good.
  how can i do to change the fps.
  this is my jni code with ffmpeg init.

  AVCodecContext *video_codec_ctx = video_stream->codec;
  
    video_codec_ctx->codec_id = video_codec->id;
  
    video_codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
  
    video_codec_ctx->level = 31;
  
  
  
    video_codec_ctx->width = width;
  
    video_codec_ctx->height = height;
  
    video_codec_ctx->pix_fmt = PIX_FMT_YUV420P;
  
  
  
    video_codec_ctx->rc_max_rate = 0;
  
    video_codec_ctx->rc_buffer_size = 0;
  
    video_codec_ctx->gop_size = 30;
  
    video_codec_ctx->max_b_frames = 0;
  
    video_codec_ctx->slices = 8;   
  
    video_codec_ctx->b_frame_strategy = 1;
  
    video_codec_ctx->coder_type = 0;
  
    video_codec_ctx->me_cmp = 1;
  
    video_codec_ctx->me_range = 16;
  
    video_codec_ctx->qmin = 10;
  
    video_codec_ctx->qmax = 51;
  
    video_codec_ctx->keyint_min = 25;
  
    video_codec_ctx->refs= 3;   
  
    video_codec_ctx->trellis = 0;
  
    video_codec_ctx->scenechange_threshold = 40;
  
    video_codec_ctx->flags |= CODEC_FLAG_LOOP_FILTER;
  
    video_codec_ctx->me_method = ME_HEX;
  
    video_codec_ctx->me_subpel_quality = 6;
  
    video_codec_ctx->i_quant_factor = 0.71;
  
    video_codec_ctx->qcompress = 0.6;
  
    video_codec_ctx->max_qdiff = 4;
  
    video_codec_ctx->time_base.den = 30;
  
    video_codec_ctx->time_base.num = 1;
  
    video_codec_ctx->bit_rate = 5200*1000;
  
    video_codec_ctx->bit_rate_tolerance = 0;
  
    video_codec_ctx->flags2 |= 0x00000100;
  
  
  
    fmt_context->bit_rate = 6000*1000;
  
  
  
    av_opt_set(video_codec_ctx, "partitions", "i8x8,i4x4,p8x8,b8x8", 0);
  
  
  
    av_opt_set_int(video_codec_ctx, "direct-pred", 1, 0);
  
    av_opt_set_int(video_codec_ctx, "rc-lookahead", 0, 0);
  
    av_opt_set_int(video_codec_ctx, "fast-pskip", 1, 0);
  
    av_opt_set_int(video_codec_ctx, "mixed-refs", 1, 0);
  
    av_opt_set_int(video_codec_ctx, "8x8dct", 0, 0);
  
    av_opt_set_int(video_codec_ctx, "weightb", 0, 0);
  
  
  
    if(fmt_context->oformat->flags & AVFMT_GLOBALHEADER)
  
        video_codec_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  
  
    LOGI("Opening video codec");
  
    AVDictionary *vopts = NULL;
  
    av_dict_set(&vopts, "profile", "main", 0);
  
    //av_dict_set(&vopts, "vprofile", "main", 0);
  
    av_dict_set(&vopts, "rc-lookahead", 0, 0);
  
    av_dict_set(&vopts, "tune", "film", 0);  
  
    av_dict_set(&vopts, "preset", "ultrafast", 0);  
  
    av_opt_set(video_codec_ctx->priv_data,"tune","film",0);
  
    av_opt_set(video_codec_ctx->priv_data,"preset","ultrafast",0);
  
    av_opt_set(video_codec_ctx->priv_data,"tune","film",0);
  
    int open_res = avcodec_open2(video_codec_ctx, video_codec, &vopts);

• ffmpeg trimming audio WAV files and setting timecode

  14 juillet 2022, par user19551045

  I am currently trying to cut an audio file to match the length of a video (without combining the two...just looking at timecodes) and produce a trimmed audio file that has a timecode that will match up with the video, the video is considered the absolute truth.

  


  Currently, the issue is that the timecodes from the original audio file do not get carried over into the new cropped audio file. So, the starting timecode is now 00:00:00:00 instead of say 07:20:02:14. Even using the -timecode commands and trying to hardcode the timecode that way doesn't seem to do the trick. I am wondering if there is any way around this ? I just want to do as minimal to the raw audio as possible...just change the audio file's length while setting the timecodes so the new audio will line up with the video. Any thoughts/suggestions welcome !

  


  Currently I have tried two options that don't seem to work :
using ffmpeg cmds :

  


  
        cmd2 = r'{} -ss "{}" -i "{}" -codec copy -timecode "{}" "{}"'.format(
            FFMPEG_PATH,
            abs(tc_diff_in_seconds),
            audio_path,
            "17074647",
            out_path
        )


  


  and also using pydub :

  


          current_audio = AudioSegment.from_wav("{}".format(audio_path))
        start_time_in_milli = abs(tc_diff_in_seconds*1000)
        end_time_in_milli = start_time_in_milli + video_dur_in_seconds * 1000
        trimmed_audio = current_audio[start_time_in_milli:end_time_in_milli]
        trimmed_audio.export('{}'.format(out_path), format='WAV', parameters=["-timecode", "17:07:46:47"])


  


  Any thoughts/suggestions welcome ! Thanks

  


• Hardware Accelerated H264 Decode using DirectX11 in Unity Plugin for UWP

  8 janvier 2019, par rohit n

  I’ve built an Unity plugin for my UWP app which converts raw h264 packets to RGB data and renders it to a texture. I’ve used FFMPEG to do this and it works fine.

  int framefinished = avcodec_send_packet(m_pCodecCtx, &packet);
  
  framefinished = avcodec_receive_frame(m_pCodecCtx, m_pFrame);
  
  // YUV to RGB conversion and render to texture after this

  Now, I’m trying to shift to hardware based decoding using DirectX11 DXVA2.0.

  Using this : https://docs.microsoft.com/en-us/windows/desktop/medfound/supporting-direct3d-11-video-decoding-in-media-foundation

  I was able to create a decoder(ID3D11VideoDecoder) but I don’t know how to supply it the raw H264 packets and get the YUV or NV12 data as output.
  (Or if its possible to render the output directly to the texture since I can get the ID3D11Texture2D pointer)

  so my question is, How do you send the raw h264 packets to this decoder and get the output from it ?

  Also, this is for real time operation so I’m trying to achieve minimal latency.

  Thanks in advance !