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• libav C++ sw_scale returns variable byte size of BGR24 buffer

  6 mars 2021, par igor

  I'm trying to convert my local /dev/video0 cam to BGR24 format. It works if I resize the image (although not 100% of the time, more like 85% of the time), but I'd like to keep the same size as input video.

  


  I initialize BGR image like so including the sws context :

  


  AVPixelFormat outputPixFormat = AV_PIX_FMT_BGR24;

    AVFrame* pFrameBGR = av_frame_alloc();
    pFrameBGR->width = decoder->video_codec_context->width;
    pFrameBGR->height = decoder->video_codec_context->height;
    pFrameBGR->format = outputPixFormat;

    int alloRet = av_image_alloc(pFrameBGR->data, pFrameBGR->linesize, decoder->video_codec_context->width, decoder->video_codec_context->height, outputPixFormat, 1);
    if (alloRet < 0) {
        logging("failed to allocate image");
        return -1;
    }

    struct SwsContext *sws_ctx = NULL;

    sws_ctx = sws_getContext(decoder->video_codec_context->width,
    decoder->video_codec_context->height,
    decoder->video_codec_context->pix_fmt,
    decoder->video_codec_context->width,
    decoder->video_codec_context->height,
    outputPixFormat,
    SWS_DIRECT_BGR,
    0,
    0,
    0
    );


  


  This is the portion of my decoding loop :

  


  int response = avcodec_send_packet(pCodecContext, pPacket);
    if (response < 0) {
        logging("Error while sending a packet to decoder: %s", av_err2str(response));
        return response;
    }

    while (response >= 0) {
        response = avcodec_receive_frame(pCodecContext, pFrame);
         if (response == AVERROR(EAGAIN) || response == AVERROR_EOF) {
            break;
        } else if (response < 0) {
            logging("Error while receiving a frame from the decoder: %s", av_err2str(response));
            return response;
        }
        if (response >= 0) {

            sws_scale(sws_ctx,  (uint8_t const * const *)pFrame->data, pFrame->linesize, 0, pCodecContext->height, pFrameBGR->data, pFrameBGR->linesize);



  


  THe question is how to copy the plane of AVFrame into a buffer :

  


    size_t rgb_size = av_image_get_buffer_size(AV_PIX_FMT_BGR24, bgrFrame->width, bgrFrame->height, 1);

    uint8_t *dst_data;
    dst_data = (uint8_t *)(av_malloc(rgb_size));

    av_image_copy_to_buffer(dst_data, rgb_size, (const uint8_t* const *)bgrFrame->data, bgrFrame->linesize, AV_PIX_FMT_BGR24, bgrFrame->width, bgrFrame->height, 1);



  


  If I try to save to file the BGR image is correctly copied :

  


  char filebuf[256];
snprintf(filebuf, sizeof filebuf, "%s%d%s", "out_", pPacket->dts, ".rgb");
std::FILE *output=fopen(filebuf,"wb+");  

fwrite(bgrFrame->data[0],(pFrame->width)*(pFrame->height)*3,1,output);  
std::fclose(output);


  


  So it looks like my copy to buffer function is faulty, but I can figure out what's wrong with it :

  


    uint8_t *dst_data;
    dst_data = (uint8_t *)(av_malloc(rgb_size));

    av_image_copy_to_buffer(dst_data, rgb_size, (const uint8_t* const *)bgrFrame->data, bgrFrame->linesize, AV_PIX_FMT_BGR24, bgrFrame->width, bgrFrame->height, 1);


  


• Creating a Discord Bot to play .mp3 files

  31 décembre 2020, par TheGr3atJosh

  My goal at the moment is to create a Discord Bot that can play .mp3 files for my private Discord Server. This is my code. The file test.mp3 is in the same folder as the .py file. The error message is : "NameError : name 'FFmpegPCMAudio' is not defined". I added the ffmpeg executable in my path environment variable. Can anyone help ?

  


  import discord
import asyncio
import time

client = discord.Client()

@client.event
async def on_message(message):
    if "test" in message.content:
        user = message.author
        voice_channel = user.voice.channel
        channel = None
        if voice_channel != None:
            channel = voice_channel.name
            vc = await voice_channel.connect()
            player = vc.play(FFmpegPCMAudio("test.mp3"), after=lambda: print('done'))
            player.start()
            while not player.is_done():
                await asyncio.sleep(1)
            player.stop()
            await vc.disconnect()

client.run(TOKEN)


  


• Cannot play H264/AAC, mpeg-ts, HTTP stream in browser [on hold]

  19 mars 2019, par Ishaan Shringi

  I have a H264/AAC based MPEG-2 transport stream from a server.

  The browser is unable to play the stream natively, probably because of the transport stream container format.

  I could not find a javascript player that could support this stream.

  Is this possible using any js library, plugin or an extension implementation ?