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• Ffmpeg gives out wrong audio C++

  22 août 2022, par Turgut

  I've made a program that takes a video as in input, edits it in opengl and gives an output by encoding it. So far the video section of the program worked just fine. Then I added audio, normally it finishes without any errors and the output video has an audio, but the video is wrong, just a distorted noise. I just want to decode an audio and use that decoded audio and put it on another video. I used the Muxing.c example of ffmpeg to create my encoder, normally there was a dummy audio before I removed it. Here is what it looks like now (I'm trimming some parts, mainly video encoding parts) :

  


  video_encoder.cpp :

  


  short video_encoder::encode_one_frame()
{
    if (enc_inf.encode_video || enc_inf.encode_audio) {
        /* select the stream to encode */
       if (enc_inf.encode_video &&
            (!enc_inf.encode_audio || av_compare_ts(enc_inf.video_st.next_pts, enc_inf.video_st.enc->time_base,
                                            enc_inf.audio_st.next_pts, enc_inf.audio_st.enc->time_base) <= 0)) {
            //std::cout << "Allocing video..." << std::endl;
            enc_inf.encode_video = !write_video_frame(enc_inf.oc, &enc_inf.video_st);
            return 1;
        } else {
            //std::cout << "Allocing audio..." << std::endl;
            enc_inf.encode_audio = !write_audio_frame(enc_inf.oc, &enc_inf.audio_st);
            return 2;
        }
    }
    return 0;
}

void video_encoder::set_audio_frame(AVFrame* audio, AVSampleFormat* format)
{
    audio_data = *audio;
    input_sample_fmt = *format;
}
int video_encoder::write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
    AVCodecContext *c;
    AVFrame *frame;
    int ret;
    int dst_nb_samples;

    c = ost->enc;

#if __AUDIO_ENABLED
    c->sample_fmt = input_sample_fmt;
#endif

    frame = get_audio_frame(ost);

    if (frame) {
        /* convert samples from native format to destination codec format, using the resampler */
        /* compute destination number of samples */
        dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
        //av_assert0(dst_nb_samples == frame->nb_samples);

        /* when we pass a frame to the encoder, it may keep a reference to it
         * internally;
         * make sure we do not overwrite it here
         */
        ret = av_frame_make_writable(ost->frame);
        if (ret < 0)
            exit(1);

        /* convert to destination format */
        ret = swr_convert(ost->swr_ctx,
                          ost->frame->data, dst_nb_samples,
                          (const uint8_t **)frame->data, frame->nb_samples);
        if (ret < 0) {
            fprintf(stderr, "Error while converting\n");
            exit(1);
        }
        frame = ost->frame;

        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
        ost->samples_count += dst_nb_samples;
    }


    return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
}

AVFrame* video_encoder::get_audio_frame(OutputStream *ost)
{
    AVFrame *frame = ost->tmp_frame;
    int j, i, v;
    int16_t *q = (int16_t*)frame->data[0];

#if __AUDIO_ENABLED
    *ost->tmp_frame = audio_data;
#endif
    //(int16_t)*audio_frame->data[0];
    /* check if we want to generate more frames */
    if (av_compare_ts(ost->next_pts, ost->enc->time_base,
                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
        return NULL;

    for (j = 0; j nb_samples; j++) {
        #if !__AUDIO_ENABLED
            v = (int)(sin(ost->t) * 10000);
        #endif
        for (i = 0; i < ost->enc->channels; i++)
        #if !__AUDIO_ENABLED
            *q++ = v;
        #endif
        ost->t     += ost->tincr;
        ost->tincr += ost->tincr2;
    }

    frame->pts = ost->next_pts;
    ost->next_pts  += frame->nb_samples;

#if __AUDIO_ENABLED        
    return &audio_data;
#else
    return frame;
#endif

}

void video_encoder::open_audio(AVFormatContext *oc, const AVCodec *codec,
                       OutputStream *ost, AVDictionary *opt_arg)
{
    AVCodecContext *c;
    int nb_samples;
    int ret;
    AVDictionary *opt = NULL;

    c = ost->enc;

    /* open it */
    av_dict_copy(&opt, opt_arg, 0);
    ret = avcodec_open2(c, codec, &opt);
    av_dict_free(&opt);
    if (ret < 0) {
        fprintf(stderr, "Could not open audio codec: %s\n", ret);
        exit(1);
    }

    /* init signal generator */
    ost->t     = 0;
    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
    /* increment frequency by 110 Hz per second */
    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
        nb_samples = 10000;
    else
        nb_samples = c->frame_size;

    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
                                       c->sample_rate, nb_samples);
    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
                                       c->sample_rate, nb_samples);

    /* copy the stream parameters to the muxer */
    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
    if (ret < 0) {
        fprintf(stderr, "Could not copy the stream parameters\n");
        exit(1);
    }

    /* create resampler context */
    ost->swr_ctx = swr_alloc();
    if (!ost->swr_ctx) {
        fprintf(stderr, "Could not allocate resampler context\n");
        exit(1);
    }

    /* set options */
    av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
    av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
    av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
    av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
    av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
    av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);

    /* initialize the resampling context */
    if ((ret = swr_init(ost->swr_ctx)) < 0) {
        fprintf(stderr, "Failed to initialize the resampling context\n");
        exit(1);
    }
}

    int video_encoder::write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
                       AVStream *st, AVFrame *frame, AVPacket *pkt)
    {
    int ret;
    // Conditional jump or move depends on uninitialised value
    // Use of uninitialised value of size 8
    // send the frame to the encoder

    // Error is about c.
    ret = avcodec_send_frame(c, frame);
    if (ret < 0) {
        fprintf(stderr, "Error sending a frame to the encoder: %s\n",
                ret);
        exit(1);
    }

    while (ret >= 0) {
        ret = avcodec_receive_packet(c, pkt);

        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            break;
        else if (ret < 0) {
            fprintf(stderr, "Error encoding a frame: %s\n", ret);
            exit(1);
        }

        /* rescale output packet timestamp values from codec to stream timebase */
        av_packet_rescale_ts(pkt, c->time_base, st->time_base);
        pkt->stream_index = st->index;

        /* Write the compressed frame to the media file. */
        //log_packet(fmt_ctx, pkt);

        //std::cout << "Packet: " << pkt << std::endl;

        ret = av_interleaved_write_frame(fmt_ctx, pkt);

        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
         * its contents and resets pkt), so that no unreferencing is necessary.
         * This would be different if one used av_write_frame(). */
        if (ret < 0) {
            fprintf(stderr, "Error while writing output packet: %s\n", ret);
            exit(1);
        }
    }

    return ret == AVERROR_EOF ? 1 : 0;
    }

video_encoder::video_encoder()
{
...
    enc_inf.video_st, enc_inf.audio_st = (struct OutputStream) { 0 };
    enc_inf.video_st.next_pts = 1; 
    enc_inf.audio_st.next_pts = 1;
    enc_inf.encode_audio, enc_inf.encode_video = 0;
...
    if (enc_inf.fmt->audio_codec != AV_CODEC_ID_NONE) {
        add_stream(&enc_inf.audio_st, enc_inf.oc, &audio_codec, enc_inf.fmt->audio_codec);
        enc_inf.have_audio = 1;
        enc_inf.encode_audio = 1;
    }

    /* Now that all the parameters are set, we can open the audio and
     * video codecs and allocate the necessary encode buffers. */
    if (enc_inf.have_video)
        open_video(enc_inf.oc, video_codec, &enc_inf.video_st, opt);

    if (enc_inf.have_audio)
        open_audio(enc_inf.oc, audio_codec, &enc_inf.audio_st, opt);

}



  


  And here is how I read the audio :

  


  video_decoder.cpp :

  


  int video_decode::video_reader_read_frame(){
      while (av_read_frame(av_format_ctx, av_packet) >= 0) {
            // Audio decode

            if (av_packet->stream_index == video_stream_index){

                response = avcodec_send_packet(av_codec_ctx, av_packet);
                if (response < 0) {
                    printf("Failed to decode packet: %s\n", av_make_error(response));
                    return false;
                }


                response = avcodec_receive_frame(av_codec_ctx, av_frame);
                if (response == AVERROR(EAGAIN) || response == AVERROR_EOF) {
                    av_packet_unref(av_packet);
                    continue;
                } else if (response < 0) {
                    printf("Failed to decode packet: %s\n", av_make_error(response));
                    return false;
                }


                *pts = av_frame->pts;
            ...
        #if __AUDIO_ENABLED
            else if (av_packet->stream_index == audio_stream_index){
                decode_audio(audio_codec_ctx, av_packet);
                av_packet_unref(av_packet);
                return 2;
            }
        #endif
       }
    }

uint8_t** video_decode::decode_audio(AVCodecContext *dec, const AVPacket *pkt)
    {
        auto& frame= state.av_frame;
        uint8_t ret = 0;
    
        // submit the packet to the decoder
        ret = avcodec_send_packet(dec, pkt);
        if (ret < 0) {
            std::cout << "Error submitting a packet for decoding" << std::endl;
        }
    
        // get all the available frames from the decoder
        while (ret >= 0) {
            ret = avcodec_receive_frame(dec, frame);
            if (ret < 0) {
                // those two return values are special and mean there is no output
                // frame available, but there were no errors during decoding
                if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)){ 
                    return nullptr;
                }
                std::cout << "Decode err" << std::endl;
            }
    
            return frame->data;
        }
        return nullptr;
    }

 bool video_decode::video_reader_open(const char* filename) 
    {
...
if (open_codec_context(&audio_stream_index, &audio_dec_ctx, av_format_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
            audio_stream = av_format_ctx->streams[audio_stream_index];
        }
...
}



  


  And here is how everything comes together inside main :

  


  int main()
{
   ...
   video_encoder encoder = new video_encoder();
   video_decode decoder = new video_decode("file_name");
   decoder.video_reader_open();
   int frame_Type = 0;
   ...
   while(true)
   {
      ...
      decoder->video_reader_read_frame();
      ...
      if(encoder->enc_inf.encode_video || encoder->enc_inf.encode_audio)
            {
                if(!time_line.audio_only)
                {   
                    set_video_frame(fbo);
                    encoder->set_encode_framebuffer(fbo);
                }else{
                    uint8_t* fbo = nullptr;
                    encoder->set_encode_framebuffer(fbo);
                }

        #if __AUDIO_ENABLED
                if(frame_type == 2)
                    frame_type = encoder->set_audio_frame(audio_test->get_state().av_frame, &audio_test->get_state().audio_codec_ctx->sample_fmt);
        #endif  
            }else{
                break;
            }
   }
}



  


  I know I'm doing something wrong when connecting the decoder and the encoder but not quite sure where I went wrong.

  


  Using ubuntu.

  


• Ffmpeg gives out distorted audio C++/C

  23 août 2022, par Turgut

  I've made a program that takes a video as in input, edits it in opengl and gives an output by encoding it. So far the video section of the program worked just fine. Then I added audio, normally it finishes without any errors and the output video has an audio, but the audio is wrong, I can hear the intended audio in the background but its mostly a distorted noise. I just want to decode an audio and use that decoded audio and put it on another video. I used the Muxing.c example of ffmpeg to create my encoder, normally there was a dummy audio before I removed it. Here is what it looks like now (I'm trimming some parts, mainly video encoding parts) :

  


  video_encoder.cpp :

  


  short video_encoder::encode_one_frame()
{
    if (enc_inf.encode_video || enc_inf.encode_audio) {
        /* select the stream to encode */
       if (enc_inf.encode_video &&
            (!enc_inf.encode_audio || av_compare_ts(enc_inf.video_st.next_pts, enc_inf.video_st.enc->time_base,
                                            enc_inf.audio_st.next_pts, enc_inf.audio_st.enc->time_base) <= 0)) {
            //std::cout << "Allocing video..." << std::endl;
            enc_inf.encode_video = !write_video_frame(enc_inf.oc, &enc_inf.video_st);
            return 1;
        } else {
            //std::cout << "Allocing audio..." << std::endl;
            enc_inf.encode_audio = !write_audio_frame(enc_inf.oc, &enc_inf.audio_st);
            return 2;
        }
    }
    return 0;
}

void video_encoder::set_audio_frame(AVFrame* audio, AVSampleFormat* format)
{
    audio_data = *audio;
    input_sample_fmt = *format;
}
int video_encoder::write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
    AVCodecContext *c;
    AVFrame *frame;
    int ret;
    int dst_nb_samples;

    c = ost->enc;

#if __AUDIO_ENABLED
    c->sample_fmt = input_sample_fmt;
#endif

    frame = get_audio_frame(ost);

    if (frame) {
        /* convert samples from native format to destination codec format, using the resampler */
        /* compute destination number of samples */
        dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
        //av_assert0(dst_nb_samples == frame->nb_samples);

        /* when we pass a frame to the encoder, it may keep a reference to it
         * internally;
         * make sure we do not overwrite it here
         */
        ret = av_frame_make_writable(ost->frame);
        if (ret < 0)
            exit(1);

        /* convert to destination format */
        ret = swr_convert(ost->swr_ctx,
                          ost->frame->data, dst_nb_samples,
                          (const uint8_t **)frame->data, frame->nb_samples);
        if (ret < 0) {
            fprintf(stderr, "Error while converting\n");
            exit(1);
        }
        frame = ost->frame;

        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
        ost->samples_count += dst_nb_samples;
    }


    return write_frame(oc, c, ost->st, frame, ost->tmp_pkt);
}

AVFrame* video_encoder::get_audio_frame(OutputStream *ost)
{
    AVFrame *frame = ost->tmp_frame;
    int j, i, v;
    int16_t *q = (int16_t*)frame->data[0];

#if __AUDIO_ENABLED
    *ost->tmp_frame = audio_data;
#endif
    //(int16_t)*audio_frame->data[0];
    /* check if we want to generate more frames */
    if (av_compare_ts(ost->next_pts, ost->enc->time_base,
                      STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
        return NULL;

    for (j = 0; j nb_samples; j++) {
        #if !__AUDIO_ENABLED
            v = (int)(sin(ost->t) * 10000);
        #endif
        for (i = 0; i < ost->enc->channels; i++)
        #if !__AUDIO_ENABLED
            *q++ = v;
        #endif
        ost->t     += ost->tincr;
        ost->tincr += ost->tincr2;
    }

    frame->pts = ost->next_pts;
    ost->next_pts  += frame->nb_samples;

#if __AUDIO_ENABLED        
    return &audio_data;
#else
    return frame;
#endif

}

void video_encoder::open_audio(AVFormatContext *oc, const AVCodec *codec,
                       OutputStream *ost, AVDictionary *opt_arg)
{
    AVCodecContext *c;
    int nb_samples;
    int ret;
    AVDictionary *opt = NULL;

    c = ost->enc;

    /* open it */
    av_dict_copy(&opt, opt_arg, 0);
    ret = avcodec_open2(c, codec, &opt);
    av_dict_free(&opt);
    if (ret < 0) {
        fprintf(stderr, "Could not open audio codec: %s\n", ret);
        exit(1);
    }

    /* init signal generator */
    ost->t     = 0;
    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
    /* increment frequency by 110 Hz per second */
    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
        nb_samples = 10000;
    else
        nb_samples = c->frame_size;

    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
                                       c->sample_rate, nb_samples);
    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
                                       c->sample_rate, nb_samples);

    /* copy the stream parameters to the muxer */
    ret = avcodec_parameters_from_context(ost->st->codecpar, c);
    if (ret < 0) {
        fprintf(stderr, "Could not copy the stream parameters\n");
        exit(1);
    }

    /* create resampler context */
    ost->swr_ctx = swr_alloc();
    if (!ost->swr_ctx) {
        fprintf(stderr, "Could not allocate resampler context\n");
        exit(1);
    }

    /* set options */
    av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
    av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
    av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
    av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
    av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
    av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);

    /* initialize the resampling context */
    if ((ret = swr_init(ost->swr_ctx)) < 0) {
        fprintf(stderr, "Failed to initialize the resampling context\n");
        exit(1);
    }
}

    int video_encoder::write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
                       AVStream *st, AVFrame *frame, AVPacket *pkt)
    {
    int ret;
    // Conditional jump or move depends on uninitialised value
    // Use of uninitialised value of size 8
    // send the frame to the encoder

    // Error is about c.
    ret = avcodec_send_frame(c, frame);
    if (ret < 0) {
        fprintf(stderr, "Error sending a frame to the encoder: %s\n",
                ret);
        exit(1);
    }

    while (ret >= 0) {
        ret = avcodec_receive_packet(c, pkt);

        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
            break;
        else if (ret < 0) {
            fprintf(stderr, "Error encoding a frame: %s\n", ret);
            exit(1);
        }

        /* rescale output packet timestamp values from codec to stream timebase */
        av_packet_rescale_ts(pkt, c->time_base, st->time_base);
        pkt->stream_index = st->index;

        /* Write the compressed frame to the media file. */
        //log_packet(fmt_ctx, pkt);

        //std::cout << "Packet: " << pkt << std::endl;

        ret = av_interleaved_write_frame(fmt_ctx, pkt);

        /* pkt is now blank (av_interleaved_write_frame() takes ownership of
         * its contents and resets pkt), so that no unreferencing is necessary.
         * This would be different if one used av_write_frame(). */
        if (ret < 0) {
            fprintf(stderr, "Error while writing output packet: %s\n", ret);
            exit(1);
        }
    }

    return ret == AVERROR_EOF ? 1 : 0;
    }

video_encoder::video_encoder()
{
...
    enc_inf.video_st, enc_inf.audio_st = (struct OutputStream) { 0 };
    enc_inf.video_st.next_pts = 1; 
    enc_inf.audio_st.next_pts = 1;
    enc_inf.encode_audio, enc_inf.encode_video = 0;
...
    if (enc_inf.fmt->audio_codec != AV_CODEC_ID_NONE) {
        add_stream(&enc_inf.audio_st, enc_inf.oc, &audio_codec, enc_inf.fmt->audio_codec);
        enc_inf.have_audio = 1;
        enc_inf.encode_audio = 1;
    }

    /* Now that all the parameters are set, we can open the audio and
     * video codecs and allocate the necessary encode buffers. */
    if (enc_inf.have_video)
        open_video(enc_inf.oc, video_codec, &enc_inf.video_st, opt);

    if (enc_inf.have_audio)
        open_audio(enc_inf.oc, audio_codec, &enc_inf.audio_st, opt);

}



  


  And here is how I read the audio :

  


  video_decoder.cpp :

  


  int video_decode::video_reader_read_frame(){
      while (av_read_frame(av_format_ctx, av_packet) >= 0) {
            // Audio decode

            if (av_packet->stream_index == video_stream_index){

                response = avcodec_send_packet(av_codec_ctx, av_packet);
                if (response < 0) {
                    printf("Failed to decode packet: %s\n", av_make_error(response));
                    return false;
                }


                response = avcodec_receive_frame(av_codec_ctx, av_frame);
                if (response == AVERROR(EAGAIN) || response == AVERROR_EOF) {
                    av_packet_unref(av_packet);
                    continue;
                } else if (response < 0) {
                    printf("Failed to decode packet: %s\n", av_make_error(response));
                    return false;
                }


                *pts = av_frame->pts;
            ...
        #if __AUDIO_ENABLED
            else if (av_packet->stream_index == audio_stream_index){
                decode_audio(audio_codec_ctx, av_packet);
                av_packet_unref(av_packet);
                return 2;
            }
        #endif
       }
    }

uint8_t** video_decode::decode_audio(AVCodecContext *dec, const AVPacket *pkt)
    {
        auto& frame= state.av_frame;
        uint8_t ret = 0;
    
        // submit the packet to the decoder
        ret = avcodec_send_packet(dec, pkt);
        if (ret < 0) {
            std::cout << "Error submitting a packet for decoding" << std::endl;
        }
    
        // get all the available frames from the decoder
        while (ret >= 0) {
            ret = avcodec_receive_frame(dec, frame);
            if (ret < 0) {
                // those two return values are special and mean there is no output
                // frame available, but there were no errors during decoding
                if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)){ 
                    return nullptr;
                }
                std::cout << "Decode err" << std::endl;
            }
    
            return frame->data;
        }
        return nullptr;
    }

 bool video_decode::video_reader_open(const char* filename) 
    {
...
if (open_codec_context(&audio_stream_index, &audio_dec_ctx, av_format_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
            audio_stream = av_format_ctx->streams[audio_stream_index];
        }
...
}



  


  And here is how everything comes together inside main :

  


  int main()
{
   ...
   video_encoder encoder = new video_encoder();
   video_decode decoder = new video_decode("file_name");
   decoder.video_reader_open();
   int frame_Type = 0;
   ...
   while(true)
   {
      ...
      decoder->video_reader_read_frame();
      ...
      if(encoder->enc_inf.encode_video || encoder->enc_inf.encode_audio)
            {
                if(!time_line.audio_only)
                {   
                    set_video_frame(fbo);
                    encoder->set_encode_framebuffer(fbo);
                }else{
                    uint8_t* fbo = nullptr;
                    encoder->set_encode_framebuffer(fbo);
                }

        #if __AUDIO_ENABLED
                if(frame_type == 2)
                    frame_type = encoder->set_audio_frame(audio_test->get_state().av_frame, &audio_test->get_state().audio_codec_ctx->sample_fmt);
        #endif  
            }else{
                break;
            }
   }
}



  


  I know I'm doing something wrong when connecting the decoder and the encoder but not quite sure where I went wrong.

  


  Using ubuntu.

  


• drawertext color with conditon ffmpeg

  31 octobre 2022, par abderrahim khadri

  I have a coin price banner where I use ffmpeg to stream it and use reload:1 to render it in real time. I want to color the negative with red and the positive with green , haw to do it and is it posible with fflmpeg

  


  -vf 'drawtext=enable='between(t,18.93,20.28)':fontfile=fonts/cousine-bold.ttf:fontsize=144:fontcolor_expr=%{eif\\: if(between(0,1000)\, green\, red) \\: x}:x=82:y=288:text=coinsprice.txt:reload:1'