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• Make sound settings created by ffmpeg apply to entire Windows applications [closed]

  7 septembre 2023, par rabeh koo

  The recording sound of my microphone is weak, so I resorted to using ffmpeg to add a volume increase effect and some other effects. I created a simple batch file and recorded the sound with it, and it works very well.
This is the batch file code :

  


  ffmpeg -f dshow -i audio="Microphone (USBAudio2.0)" -channels 1 -af " stereotools=level_in=2 , afftdn=nf=-25 , volume=10 " Output.wav


  


  

  • But the problem occurs when I want to make calls or use the microphone in Windows applications, the recording is very weak. Is there a way to make the ffmpeg effects that are in the batch file apply to all system applications, and how can that be done ?

  


  


  I searched and found that I have to create a virtual microphone device with some programs and make it the main microphone for the system. I tried these programs, but there was no option to add ffmpeg codes or pass the audio through the batch file that I created earlier.

  


• Decoding an MKA audio file into raw data of the pcm_alaw type (MKA Audio to pcm_alaw)

  2 octobre 2020, par bbdd

  My task is to open an existing audio file with the mka extension (Matroska container) and extract the raw audio data. If the audio data is different from pcm_alaw, then convert it to pcm_alaw before saving it to another file ( or buffer). This example shows only an example of extracting raw data from an mp2 file. I do not know how to do this with the mka container. I would like to have something like this :

  


  UPD

  


  I found an option to save audio data in the format in which it was recorded in the audio file. An example is shown below.

  


  PS. This is only a test version and most likely there are memory leaks and other problems.

  


  &#xA;#include <qfile>&#xA;#include <qdebug>&#xA;#include "audiodecoder.h"&#xA;&#xA;int main(int argc, char* argv[])&#xA;{&#xA;    AudioDecoder decoder("/home/test/test.mka");&#xA;    bool started = decoder.start();&#xA;    if (!started) {&#xA;        return EXIT_FAILURE;&#xA;    }&#xA;&#xA;    QFile file("/home/test/rawData.bin");&#xA;    file.open(QIODevice::WriteOnly);&#xA;&#xA;    while (true) {&#xA;        auto data = decoder.getData(255);&#xA;        if (data.isEmpty()) {&#xA;            break;&#xA;        }&#xA;        file.write(data.data(), data.size());&#xA;    }&#xA;    file.close();&#xA;    return EXIT_SUCCESS;&#xA;}&#xA;&#xA;</qdebug></qfile>

  


  audiodecoder.h

  


  class AudioDecoder {
public:
    AudioDecoder(const QString& fileName);
    AudioDecoder& operator=(const AudioDecoder& rhs) = delete;
    AudioDecoder& operator=(AudioDecoder&& rhs) = delete;
    AudioDecoder(const AudioDecoder& rhs) = delete;
    AudioDecoder(AudioDecoder&& rhs) = delete;
    virtual ~AudioDecoder(void);

    virtual bool start(void) noexcept;
    virtual QByteArray getData(const quint16& size) noexcept;
    virtual bool stop(void) noexcept;

protected:
    bool m_initialized;
    QString m_fileName;

    AVFrame* p_frame = nullptr;
    AVPacket* p_packet = nullptr;
    AVCodecContext* p_cdcCtx = nullptr;
    AVFormatContext* p_frmCtx = nullptr;
};


  


  audiodecoder.cpp

  


  
static void logging(const char* message)
{
    qDebug() << message;
}

AudioDecoder::AudioDecoder(const QString& fileName)
    : m_initialized(false)
    , m_fileName(fileName)
    , p_cdcCtx(nullptr)
    , p_frmCtx(nullptr)
{
    av_register_all();
}

QByteArray AudioDecoder::getData(const quint16& dataSize) noexcept
{
    QByteArray data;
    qint32 response = 0;
    if (av_read_frame(p_frmCtx, p_packet) >= 0) {
        //logging(QString("AVPacket->pts %1").arg(p_packet->pts).toStdString().c_str());
        //response = decode_packet(p_packet, p_cdcCtx, p_frame);
        response = avcodec_send_packet(p_cdcCtx, p_packet);
        if (response < 0) {
            logging("Error while sending a packet to the decoder");
            return {};
        }
        while (response >= 0) {
            response = avcodec_receive_frame(p_cdcCtx, p_frame);
            if (response == AVERROR(EAGAIN) || response == AVERROR_EOF) {
                break;
            }
            else if (response < 0) {
                logging("Error while receiving a frame from the decoder");
                return {};
            }
            if (response >= 0) {
                logging(QString("Frame %1 (type=%2, size=%3 bytes) pts %4 key_frame %5 [DTS %6], duration[%7]")
                            .arg(p_cdcCtx->frame_number)
                            .arg(av_get_picture_type_char(p_frame->pict_type))
                            .arg(p_frame->pkt_size)
                            .arg(p_frame->pts)
                            .arg(p_frame->key_frame)
                            .arg(p_frame->coded_picture_number)
                            .arg(p_frame->pkt_duration)
                            .toStdString()
                            .c_str());

                for (int i = 0; i < p_frame->linesize[0]; ++i) {
                    data.push_back(p_frame->data[0][i]);
                }
            }
        }
        av_packet_unref(p_packet);
        return data;
    }
    return {};
}

bool AudioDecoder::start(void) noexcept
{
    if (m_initialized) {
        return true;
    }

    int error;
    // Open the input file to read from it.
    if ((error = avformat_open_input(&p_frmCtx,
             m_fileName.toStdString().c_str(), nullptr, nullptr))
        < 0) {
        qDebug() << "Could not open input file: " << m_fileName;
        p_frmCtx = nullptr;
        return false;
    }
    // Get information on the input file (number of streams etc.).
    if ((error = avformat_find_stream_info(p_frmCtx, nullptr)) < 0) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    // Make sure that there is only one stream in the input file.
    if ((p_frmCtx)->nb_streams != 1) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }

    if (p_frmCtx->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }

    // Find a decoder for the audio stream.
    AVCodec* input_codec = nullptr;
    if (!(input_codec = avcodec_find_decoder((p_frmCtx)->streams[0]->codecpar->codec_id))) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    // Allocate a new decoding context.
    AVCodecContext* avctx = avcodec_alloc_context3(input_codec);
    if (!avctx) {
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    // Initialize the stream parameters with demuxer information.
    error = avcodec_parameters_to_context(avctx, (p_frmCtx)->streams[0]->codecpar);
    if (error < 0) {
        avformat_close_input(&p_frmCtx);
        avcodec_free_context(&avctx);
        qDebug() << __LINE__;
        return false;
    }
    /* Open the decoder for the audio stream to use it later. */
    if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
        avcodec_free_context(&avctx);
        avformat_close_input(&p_frmCtx);
        qDebug() << __LINE__;
        return false;
    }
    /* Save the decoder context for easier access later. */
    p_cdcCtx = avctx;
    av_dump_format(p_frmCtx, 0, m_fileName.toStdString().c_str(), 0);

    p_frame = av_frame_alloc();
    if (!p_frame) {
        logging("failed to allocated memory for AVFrame");
        return false;
    }
    p_packet = av_packet_alloc();
    if (!p_packet) {
        logging("failed to allocated memory for AVPacket");
        return false;
    }
    return m_initialized = true;
}

bool AudioDecoder::stop(void) noexcept
{
    if (p_cdcCtx != nullptr) {
        avcodec_free_context(&p_cdcCtx);
    }
    if (p_frmCtx != nullptr) {
        avformat_close_input(&p_frmCtx);
    }
    return true;
}

AudioDecoder::~AudioDecoder(void)
{
    stop();
}


  


  But the problem in this example is that I didn't implement the ability to get exactly the requested size of audio data. In my case, it's just ignored. And also, in this case, I extract and save audio data in the format in which it was originally recorded. How do I convert any audio data format to the one I need. For example, I get PCM_S16LE, but I want to convert it to PCM_ALAW.

  


• ffmpeg SAP announcements with UDP multicast stream

  5 août 2015, par user4535092

  I have been trying to get this working and crawling the forums but I just can’t find the solution.

  What I want, is to capture my windows desktop using dshow, encode it using x264 into a multicast stream, then announce that stream using SAP so that others on the network can identify and decode that stream easily.

  I have everything working rather nicely, EXCEPT for the SAP announcement.

  Here is what I’m using :

  ffmpeg -y -f dshow -framerate 24 -i video="screen-capture-recorder":audio="virtual-audio-capturer" -r 10 -vcodec libx264 -preset ultrafast -crf 18 -async 1 -acodec libmp3lame -ab 24k -ar 22050 -bsf:v h264_mp4toannexb -bufsize 8000k -f mpegts udp://224.100.0.1:5001

  I’ve investigated using the sap muxer (-f sap sap://224.100.0.1:5001?same_port=1) but that doesn’t seem to format a usable mpegts stream and it has limited options.

  If someone knows how I might generate the SDP information and attach it as a sap announcement for this stream, it would be much appreciated !

  PS I’ve posted this on ffmpeg forum and mailing list also but no responses.