Joining Portaudio and Opus - c++

Hi im trying to take sound from an open PortAudio Stream, encode it with opus, decode it and reproduce it again with portaudio.
Im doing this as a prototype just to try and understand the mechanics of this systems so, no real interest on following this concrete flow.
Thing is, portaudio gives buffers where OPUS needs Frames. Mi thought lead me to this in the portaudio side:
err = (Pa_ReadStream(stream, readBuffer, FRAMES_PER_BUFFER));
if (err = paNoError){
qDebug()<<"Fail read";
qDebug()<<Pa_GetErrorText(err);
// blockingRecord = false;
}
while (pos<FRAMES_PER_BUFFER){
memcpy(frameBuffer,readBuffer+(pos*FRAME_SIZE*NUM_CHANNELS),FRAME_SIZE*CHANNELS);
compressedSound = om.encodeOpus(frameBuffer);
unCompressedSound = om.decodeOpus(compressedSound);
memcpy(readBuffer+(pos*FRAME_SIZE*NUM_CHANNELS),unCompressedSound,FRAME_SIZE*CHANNELS);
pos++;
}
pos = 0;
err = (Pa_WriteStream(stream, readBuffer, FRAMES_PER_BUFFER));
if (err != paNoError)
{
qDebug() << "FAIL WRITE";
qDebug()<<Pa_GetErrorText(err);
//blockingRecord = false;
}
And this on the OPUS side:
unsigned char * OpusManager::encodeOpus(unsigned char *frame){
memcpy(encoded, frame, FRAME_SIZE*CHANNELS);
int ret = opus_encode(enc, encoded, FRAME_SIZE, compressed_buffer, encoded_data_size);
if (ret<0){
qDebug()<<"Failure while compressing sound";
return NULL;
}
return (compressed_buffer);
}
unsigned char * OpusManager::decodeOpus(unsigned char *frame){
int ret= opus_decode(dec, frame, encoded_data_size, decoded, FRAME_SIZE, 0);
if (ret<0){
qDebug()<<"Failure while decompressing sound";
return NULL;
}
memcpy(uncompressed_buffer, decoded, FRAME_SIZE*CHANNELS);
return (uncompressed_buffer);
}
No errors without encocing and perfect soud. With encode i get no errors till the PA_Writestream call, where i get a "Output underflowed" PaError. I suppose the way of taking the frames ive implemmented must be waaay wrong, but cant find info to help me with this.

It seems your interpretation of Opus' frame_size parameters to opus_encode and opus_decode is incorrect. If I understand your code correctly you're recording a packet of size FRAMES_PER_BUFFER frames and then try to turn it into N packets of size FRAME_SIZE. Instead, it seems to me that Opus wants to turn your packet of FRAMES_PER_BUFFER into another packet of equal frame count, and in doing so, only uses it's FRAME_SIZE parameter as some sort of quality control parameter for the encoding process. Below you'll find a complete sample that I believe does what you want. Play around with the '480' magic number in encode()/decode() and hear audio quality change.
int opusErr;
PaError paErr;
std::string s;
int const channels = 2;
int const bufferSize = 480;
int const sampleRate = 48000;
int const durationSeconds = 5;
opus_int32 enc_bytes;
opus_int32 dec_bytes;
int framesProcessed = 0;
std::vector<unsigned short> captured(bufferSize * channels);
std::vector<unsigned short> decoded(bufferSize * channels);
// * 2: byte count, 16 bit samples
std::vector<unsigned char> encoded(bufferSize * channels * 2);
// initialize opus
OpusEncoder* enc = opus_encoder_create(
sampleRate, channels, OPUS_APPLICATION_AUDIO, &opusErr);
if (opusErr != OPUS_OK)
{
std::cout << "opus_encoder_create failed: " << opusErr << "\n";
std::getline(std::cin, s);
return 1;
}
OpusDecoder* dec = opus_decoder_create(
sampleRate, channels, &opusErr);
if (opusErr != OPUS_OK)
{
std::cout << "opus_decoder_create failed: " << opusErr << "\n";
std::getline(std::cin, s);
return 1;
}
// initialize portaudio
if ((paErr = Pa_Initialize()) != paNoError)
{
std::cout << "Pa_Initialize failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
PaStream* stream = nullptr;
if ((paErr = Pa_OpenDefaultStream(&stream,
channels, channels, paInt16, sampleRate,
bufferSize, nullptr, nullptr)) != paNoError)
{
std::cout << "Pa_OpenDefaultStream failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
// start stream
if ((paErr = Pa_StartStream(stream)) != paNoError)
{
std::cout << "Pa_StartStream failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
// capture, encode, decode & render durationSeconds of audio
while (framesProcessed < sampleRate * durationSeconds)
{
if ((paErr = Pa_ReadStream(stream,
captured.data(), bufferSize)) != paNoError)
{
std::cout << "Pa_ReadStream failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
if ((enc_bytes = opus_encode(enc, reinterpret_cast<opus_int16 const*>(
captured.data()), 480, encoded.data(), encoded.size())) < 0)
{
std::cout << "opus_encode failed: " << enc_bytes << "\n";
std::getline(std::cin, s);
return 1;
}
if ((dec_bytes = opus_decode(dec, encoded.data(), enc_bytes,
reinterpret_cast<opus_int16*>(decoded.data()), 480, 0)) < 0)
{
std::cout << "opus_decode failed: " << dec_bytes << "\n";
std::getline(std::cin, s);
return 1;
}
if ((paErr = Pa_WriteStream(stream, decoded.data(), bufferSize)) != paNoError)
{
std::cout << "Pa_WriteStream failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
framesProcessed += bufferSize;
}
// stop stream
if ((paErr = Pa_StopStream(stream)) != paNoError)
{
std::cout << "Pa_StopStream failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
// cleanup portaudio
if ((paErr = Pa_CloseStream(stream)) != paNoError)
{
std::cout << "Pa_CloseStream failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
if ((paErr = Pa_Terminate()) != paNoError)
{
std::cout << "Pa_Terminate failed: " << Pa_GetErrorText(paErr) << "\n";
std::getline(std::cin, s);
return 1;
}
// cleanup opus
opus_decoder_destroy(dec);
opus_encoder_destroy(enc);

Related

UDP Packets coming in with wrong formatting

I'm currently working to get some software that is currently running on windows ported over to ubuntu. One aspect of this software handles udp packets with the location of some of our assets using SFML. For some reason after moving OS's the packets are coming in as U� (always with the � symbol.) is this a posix problem? Here is the code I was using:
int32_t
SocketUdpClient::recv(std::string *data)
{
sf::Socket::Status recvStatus = sf::Socket::Error;
if( m_bOpen && data != NULL )
{
char buf[4096];
std::size_t bytesRead;
sf::IpAddress senderIp;
uint16_t senderPort;
recvStatus = m_socket.receive( buf, 4096, bytesRead, senderIp, senderPort );
if (recvStatus == sf::Socket::Done)
{
std::cout << "Received " << bytesRead << " bytes from " << senderIp << " on port " << senderPort << std::endl;
std::cout << buf <<std::endl;
m_senderIp = senderIp.toString();
if (m_listenAll)
{
m_peerIp = senderIp.toString();
printf("SocketUdpClient::recvfrom m_senderIp = %s, m_peerIp = %s\n", m_senderIp.c_str(), m_peerIp.c_str());
}
if (m_senderIp.compare(m_peerIp) == 0)
{
*data = std::string(buf, bytesRead);
return bytesRead;
//return senderIp.toString();
}
}
else if (recvStatus == sf::Socket::NotReady)
{
return 0;
}
}
return -1;
}
Change this:
recvStatus = m_socket.receive( buf, 4096, bytesRead, senderIp, senderPort );
if (recvStatus == sf::Socket::Done)
{
std::cout << "Received " << bytesRead << " bytes from " << senderIp << " on port " << senderPort << std::endl;
std::cout << buf <<std::endl;
To this:
bytesRead = 0; // always a good idea
recvStatus = m_socket.receive( buf, 4095 /* 1 less than buffer size*/, bytesRead, senderIp, senderPort );
if (recvStatus == sf::Socket::Done)
{
std::cout << "Received " << bytesRead << " bytes from " << senderIp << " on port " << senderPort << std::endl;
buf[bytesRead] = '\0'; // null terminate
std::cout << buf <<std::endl;

audio do not stop recording after pause ffmpeg c++

I am developing an application that record the screen and the audio from microphone. I implemented the pause function stopping video and audio thread on a condition variable, resuming them with a notify on the same condition variable. This is done in captureAudio(), in the main while. In this way works on macOS and linux, where I use avfoudation and alsa respectively, but on windows, with dshow, keep recording audio during the pause, when the thread is waiting on the condition variable. Does anybody know how can I fix this behaviour?
#include "ScreenRecorder.h"
using namespace std;
ScreenRecorder::ScreenRecorder() : pauseCapture(false), stopCapture(false), started(false), activeMenu(true) {
avcodec_register_all();
avdevice_register_all();
width = 1920;
height = 1200;
}
ScreenRecorder::~ScreenRecorder() {
if (started) {
value = av_write_trailer(outAVFormatContext);
if (value < 0) {
cerr << "Error in writing av trailer" << endl;
exit(-1);
}
avformat_close_input(&inAudioFormatContext);
if(inAudioFormatContext == nullptr){
cout << "inAudioFormatContext close successfully" << endl;
}
else{
cerr << "Error: unable to close the inAudioFormatContext" << endl;
exit(-1);
//throw "Error: unable to close the file";
}
avformat_free_context(inAudioFormatContext);
if(inAudioFormatContext == nullptr){
cout << "AudioFormat freed successfully" << endl;
}
else{
cerr << "Error: unable to free AudioFormatContext" << endl;
exit(-1);
}
avformat_close_input(&pAVFormatContext);
if (pAVFormatContext == nullptr) {
cout << "File close successfully" << endl;
}
else {
cerr << "Error: unable to close the file" << endl;
exit(-1);
//throw "Error: unable to close the file";
}
avformat_free_context(pAVFormatContext);
if (pAVFormatContext == nullptr) {
cout << "VideoFormat freed successfully" << endl;
}
else {
cerr << "Error: unable to free VideoFormatContext" << endl;
exit(-1);
}
}
}
/*==================================== VIDEO ==============================*/
int ScreenRecorder::openVideoDevice() throw() {
value = 0;
options = nullptr;
pAVFormatContext = nullptr;
pAVFormatContext = avformat_alloc_context();
string dimension = to_string(width) + "x" + to_string(height);
av_dict_set(&options, "video_size", dimension.c_str(), 0); //option to set the dimension of the screen section to record
#ifdef _WIN32
pAVInputFormat = av_find_input_format("gdigrab");
if (avformat_open_input(&pAVFormatContext, "desktop", pAVInputFormat, &options) != 0) {
cerr << "Couldn't open input stream" << endl;
exit(-1);
}
#elif defined linux
int offset_x = 0, offset_y = 0;
string url = ":0.0+" + to_string(offset_x) + "," + to_string(offset_y); //custom string to set the start point of the screen section
pAVInputFormat = av_find_input_format("x11grab");
value = avformat_open_input(&pAVFormatContext, url.c_str(), pAVInputFormat, &options);
if (value != 0) {
cerr << "Error in opening input device (video)" << endl;
exit(-1);
}
#else
value = av_dict_set(&options, "pixel_format", "0rgb", 0);
if (value < 0) {
cerr << "Error in setting pixel format" << endl;
exit(-1);
}
value = av_dict_set(&options, "video_device_index", "1", 0);
if (value < 0) {
cerr << "Error in setting video device index" << endl;
exit(-1);
}
pAVInputFormat = av_find_input_format("avfoundation");
if (avformat_open_input(&pAVFormatContext, "Capture screen 0:none", pAVInputFormat, &options) != 0) { //TODO trovare un modo per selezionare sempre lo schermo (forse "Capture screen 0")
cerr << "Error in opening input device" << endl;
exit(-1);
}
#endif
//set frame per second
value = av_dict_set(&options, "framerate", "30", 0);
if (value < 0) {
cerr << "Error in setting dictionary value (setting framerate)" << endl;
exit(-1);
}
value = av_dict_set(&options, "preset", "medium", 0);
if (value < 0) {
cerr << "Error in setting dictionary value (setting preset value)" << endl;
exit(-1);
}
/*
value = av_dict_set(&options, "vsync", "1", 0);
if(value < 0){
cerr << "Error in setting dictionary value (setting vsync value)" << endl;
exit(-1);
}
*/
value = av_dict_set(&options, "probesize", "60M", 0);
if (value < 0) {
cerr << "Error in setting probesize value" << endl;
exit(-1);
}
//get video stream infos from context
value = avformat_find_stream_info(pAVFormatContext, nullptr);
if (value < 0) {
cerr << "Error in retrieving the stream info" << endl;
exit(-1);
}
VideoStreamIndx = -1;
for (int i = 0; i < pAVFormatContext->nb_streams; i++) {
if (pAVFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) {
VideoStreamIndx = i;
break;
}
}
if (VideoStreamIndx == -1) {
cerr << "Error: unable to find video stream index" << endl;
exit(-2);
}
pAVCodecContext = pAVFormatContext->streams[VideoStreamIndx]->codec;
pAVCodec = avcodec_find_decoder(pAVCodecContext->codec_id/*params->codec_id*/);
if (pAVCodec == nullptr) {
cerr << "Error: unable to find decoder video" << endl;
exit(-1);
}
cout << "Insert height and width [h w]: "; //custom screen dimension to record
cin >> h >> w;*/
return 0;
}
/*========================================== AUDIO ============================*/
int ScreenRecorder::openAudioDevice() {
audioOptions = nullptr;
inAudioFormatContext = nullptr;
inAudioFormatContext = avformat_alloc_context();
value = av_dict_set(&audioOptions, "sample_rate", "44100", 0);
if (value < 0) {
cerr << "Error: cannot set audio sample rate" << endl;
exit(-1);
}
value = av_dict_set(&audioOptions, "async", "1", 0);
if (value < 0) {
cerr << "Error: cannot set audio sample rate" << endl;
exit(-1);
}
#if defined linux
audioInputFormat = av_find_input_format("alsa");
value = avformat_open_input(&inAudioFormatContext, "hw:0", audioInputFormat, &audioOptions);
if (value != 0) {
cerr << "Error in opening input device (audio)" << endl;
exit(-1);
}
#endif
#if defined _WIN32
audioInputFormat = av_find_input_format("dshow");
value = avformat_open_input(&inAudioFormatContext, "audio=Microfono (Realtek(R) Audio)", audioInputFormat, &audioOptions);
if (value != 0) {
cerr << "Error in opening input device (audio)" << endl;
exit(-1);
}
#endif
value = avformat_find_stream_info(inAudioFormatContext, nullptr);
if (value != 0) {
cerr << "Error: cannot find the audio stream information" << endl;
exit(-1);
}
audioStreamIndx = -1;
for (int i = 0; i < inAudioFormatContext->nb_streams; i++) {
if (inAudioFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
audioStreamIndx = i;
break;
}
}
if (audioStreamIndx == -1) {
cerr << "Error: unable to find audio stream index" << endl;
exit(-2);
}
}
int ScreenRecorder::initOutputFile() {
value = 0;
outAVFormatContext = nullptr;
outputAVFormat = av_guess_format(nullptr, "output.mp4", nullptr);
if (outputAVFormat == nullptr) {
cerr << "Error in guessing the video format, try with correct format" << endl;
exit(-5);
}
avformat_alloc_output_context2(&outAVFormatContext, outputAVFormat, outputAVFormat->name, "..\\media\\output.mp4");
if (outAVFormatContext == nullptr) {
cerr << "Error in allocating outAVFormatContext" << endl;
exit(-4);
}
/*===========================================================================*/
this->generateVideoStream();
this->generateAudioStream();
//create an empty video file
if (!(outAVFormatContext->flags & AVFMT_NOFILE)) {
if (avio_open2(&outAVFormatContext->pb, "..\\media\\output.mp4", AVIO_FLAG_WRITE, nullptr, nullptr) < 0) {
cerr << "Error in creating the video file" << endl;
exit(-10);
}
}
if (outAVFormatContext->nb_streams == 0) {
cerr << "Output file does not contain any stream" << endl;
exit(-11);
}
value = avformat_write_header(outAVFormatContext, &options);
if (value < 0) {
cerr << "Error in writing the header context" << endl;
exit(-12);
}
return 0;
}
/*=================================== VIDEO ==================================*/
void ScreenRecorder::generateVideoStream() {
//Generate video stream
videoSt = avformat_new_stream(outAVFormatContext, nullptr);
if (videoSt == nullptr) {
cerr << "Error in creating AVFormatStream" << endl;
exit(-6);
}
outVideoCodec = avcodec_find_encoder(AV_CODEC_ID_MPEG4); //AV_CODEC_ID_MPEG4
if (outVideoCodec == nullptr) {
cerr << "Error in finding the AVCodec, try again with the correct codec" << endl;
exit(-8);
}
avcodec_alloc_context3(outAVCodec)
outVideoCodecContext = avcodec_alloc_context3(outVideoCodec);
if (outVideoCodecContext == nullptr) {
cerr << "Error in allocating the codec context" << endl;
exit(-7);
}
//set properties of the video file (stream)
outVideoCodecContext = videoSt->codec;
outVideoCodecContext->codec_id = AV_CODEC_ID_MPEG4;
outVideoCodecContext->codec_type = AVMEDIA_TYPE_VIDEO;
outVideoCodecContext->pix_fmt = AV_PIX_FMT_YUV420P;
outVideoCodecContext->bit_rate = 10000000;
outVideoCodecContext->width = width;
outVideoCodecContext->height = height;
outVideoCodecContext->gop_size = 10;
outVideoCodecContext->global_quality = 500;
outVideoCodecContext->max_b_frames = 2;
outVideoCodecContext->time_base.num = 1;
outVideoCodecContext->time_base.den = 30;
outVideoCodecContext->bit_rate_tolerance = 400000;
if (outVideoCodecContext->codec_id == AV_CODEC_ID_H264) {
av_opt_set(outVideoCodecContext->priv_data, "preset", "slow", 0);
}
if (outAVFormatContext->oformat->flags & AVFMT_GLOBALHEADER) {
outVideoCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
value = avcodec_open2(outVideoCodecContext, outVideoCodec, nullptr);
if (value < 0) {
cerr << "Error in opening the AVCodec" << endl;
exit(-9);
}
outVideoStreamIndex = -1;
for (int i = 0; i < outAVFormatContext->nb_streams; i++) {
if (outAVFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_UNKNOWN) {
outVideoStreamIndex = i;
}
}
if (outVideoStreamIndex < 0) {
cerr << "Error: cannot find a free stream index for video output" << endl;
exit(-1);
}
avcodec_parameters_from_context(outAVFormatContext->streams[outVideoStreamIndex]->codecpar, outVideoCodecContext);
}
/*=============================== AUDIO ==================================*/
void ScreenRecorder::generateAudioStream() {
AVCodecParameters* params = inAudioFormatContext->streams[audioStreamIndx]->codecpar;
inAudioCodec = avcodec_find_decoder(params->codec_id);
if (inAudioCodec == nullptr) {
cerr << "Error: cannot find the audio decoder" << endl;
exit(-1);
}
inAudioCodecContext = avcodec_alloc_context3(inAudioCodec);
if (avcodec_parameters_to_context(inAudioCodecContext, params) < 0) {
cout << "Cannot create codec context for audio input" << endl;
}
value = avcodec_open2(inAudioCodecContext, inAudioCodec, nullptr);
if (value < 0) {
cerr << "Error: cannot open the input audio codec" << endl;
exit(-1);
}
//Generate audio stream
outAudioCodecContext = nullptr;
outAudioCodec = nullptr;
int i;
AVStream* audio_st = avformat_new_stream(outAVFormatContext, nullptr);
if (audio_st == nullptr) {
cerr << "Error: cannot create audio stream" << endl;
exit(1);
}
outAudioCodec = avcodec_find_encoder(AV_CODEC_ID_AAC);
if (outAudioCodec == nullptr) {
cerr << "Error: cannot find requested encoder" << endl;
exit(1);
}
outAudioCodecContext = avcodec_alloc_context3(outAudioCodec);
if (outAudioCodecContext == nullptr) {
cerr << "Error: cannot create related VideoCodecContext" << endl;
exit(1);
}
if ((outAudioCodec)->supported_samplerates) {
outAudioCodecContext->sample_rate = (outAudioCodec)->supported_samplerates[0];
for (i = 0; (outAudioCodec)->supported_samplerates[i]; i++) {
if ((outAudioCodec)->supported_samplerates[i] == inAudioCodecContext->sample_rate)
outAudioCodecContext->sample_rate = inAudioCodecContext->sample_rate;
}
}
outAudioCodecContext->codec_id = AV_CODEC_ID_AAC;
outAudioCodecContext->sample_fmt = (outAudioCodec)->sample_fmts ? (outAudioCodec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
outAudioCodecContext->channels = inAudioCodecContext->channels;
outAudioCodecContext->channel_layout = av_get_default_channel_layout(outAudioCodecContext->channels);
outAudioCodecContext->bit_rate = 96000;
outAudioCodecContext->time_base = { 1, inAudioCodecContext->sample_rate };
outAudioCodecContext->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
if ((outAVFormatContext)->oformat->flags & AVFMT_GLOBALHEADER) {
outAudioCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
if (avcodec_open2(outAudioCodecContext, outAudioCodec, nullptr) < 0) {
cerr << "error in opening the avcodec" << endl;
exit(1);
}
//find a free stream index
outAudioStreamIndex = -1;
for (i = 0; i < outAVFormatContext->nb_streams; i++) {
if (outAVFormatContext->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_UNKNOWN) {
outAudioStreamIndex = i;
}
}
if (outAudioStreamIndex < 0) {
cerr << "Error: cannot find a free stream for audio on the output" << endl;
exit(1);
}
avcodec_parameters_from_context(outAVFormatContext->streams[outAudioStreamIndex]->codecpar, outAudioCodecContext);
}
int ScreenRecorder::init_fifo()
{
/* Create the FIFO buffer based on the specified output sample format. */
if (!(fifo = av_audio_fifo_alloc(outAudioCodecContext->sample_fmt,
outAudioCodecContext->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
int ScreenRecorder::add_samples_to_fifo(uint8_t** converted_input_samples, const int frame_size) {
int error;
/* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples. */
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
/* Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void**)converted_input_samples, frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
int ScreenRecorder::initConvertedSamples(uint8_t*** converted_input_samples,
AVCodecContext* output_codec_context,
int frame_size) {
int error;
/* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = (uint8_t**)calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
if (av_samples_alloc(*converted_input_samples, nullptr,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0) < 0) {
exit(1);
}
return 0;
}
static int64_t pts = 0;
void ScreenRecorder::captureAudio() {
int ret;
AVPacket* inPacket, * outPacket;
AVFrame* rawFrame, * scaledFrame;
uint8_t** resampledData;
init_fifo();
//allocate space for a packet
inPacket = (AVPacket*)av_malloc(sizeof(AVPacket));
if (!inPacket) {
cerr << "Cannot allocate an AVPacket for encoded video" << endl;
exit(1);
}
av_init_packet(inPacket);
//allocate space for a packet
rawFrame = av_frame_alloc();
if (!rawFrame) {
cerr << "Cannot allocate an AVPacket for encoded video" << endl;
exit(1);
}
scaledFrame = av_frame_alloc();
if (!scaledFrame) {
cerr << "Cannot allocate an AVPacket for encoded video" << endl;
exit(1);
}
outPacket = (AVPacket*)av_malloc(sizeof(AVPacket));
if (!outPacket) {
cerr << "Cannot allocate an AVPacket for encoded video" << endl;
exit(1);
}
//init the resampler
SwrContext* resampleContext = nullptr;
resampleContext = swr_alloc_set_opts(resampleContext,
av_get_default_channel_layout(outAudioCodecContext->channels),
outAudioCodecContext->sample_fmt,
outAudioCodecContext->sample_rate,
av_get_default_channel_layout(inAudioCodecContext->channels),
inAudioCodecContext->sample_fmt,
inAudioCodecContext->sample_rate,
0,
nullptr);
if (!resampleContext) {
cerr << "Cannot allocate the resample context" << endl;
exit(1);
}
if ((swr_init(resampleContext)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(&resampleContext);
exit(1);
}
while (true) {
if (pauseCapture) {
cout << "Pause audio" << endl;
}
cv.wait(ul, [this]() { return !pauseCapture; });
if (stopCapture) {
break;
}
ul.unlock();
if (av_read_frame(inAudioFormatContext, inPacket) >= 0 && inPacket->stream_index == audioStreamIndx) {
//decode audio routing
av_packet_rescale_ts(outPacket, inAudioFormatContext->streams[audioStreamIndx]->time_base, inAudioCodecContext->time_base);
if ((ret = avcodec_send_packet(inAudioCodecContext, inPacket)) < 0) {
cout << "Cannot decode current audio packet " << ret << endl;
continue;
}
while (ret >= 0) {
ret = avcodec_receive_frame(inAudioCodecContext, rawFrame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
cerr << "Error during decoding" << endl;
exit(1);
}
if (outAVFormatContext->streams[outAudioStreamIndex]->start_time <= 0) {
outAVFormatContext->streams[outAudioStreamIndex]->start_time = rawFrame->pts;
}
initConvertedSamples(&resampledData, outAudioCodecContext, rawFrame->nb_samples);
swr_convert(resampleContext,
resampledData, rawFrame->nb_samples,
(const uint8_t**)rawFrame->extended_data, rawFrame->nb_samp
add_samples_to_fifo(resampledData, rawFrame->nb_samples);
//raw frame ready
av_init_packet(outPacket);
outPacket->data = nullptr;
outPacket->size = 0;
const int frame_size = FFMAX(av_audio_fifo_size(fifo), outAudioCodecContext->frame_size);
scaledFrame = av_frame_alloc();
if (!scaledFrame) {
cerr << "Cannot allocate an AVPacket for encoded video" << endl;
exit(1);
}
scaledFrame->nb_samples = outAudioCodecContext->frame_size;
scaledFrame->channel_layout = outAudioCodecContext->channel_layout;
scaledFrame->format = outAudioCodecContext->sample_fmt;
scaledFrame->sample_rate = outAudioCodecContext->sample_rate;
av_frame_get_buffer(scaledFrame, 0);
while (av_audio_fifo_size(fifo) >= outAudioCodecContext->frame_size) {
ret = av_audio_fifo_read(fifo, (void**)(scaledFrame->data), outAudioCodecContext->frame_size);
scaledFrame->pts = pts;
pts += scaledFrame->nb_samples;
if (avcodec_send_frame(outAudioCodecContext, scaledFrame) < 0) {
cout << "Cannot encode current audio packet " << endl;
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_packet(outAudioCodecContext, outPacket);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
cerr << "Error during encoding" << endl;
exit(1);
}
av_packet_rescale_ts(outPacket, outAudioCodecContext->time_base, outAVFormatContext->streams[outAudioStreamIndex]->time_base);
outPacket->stream_index = outAudioStreamIndex;
write_lock.lock();
if (av_write_frame(outAVFormatContext, outPacket) != 0)
{
cerr << "Error in writing audio frame" << endl;
}
write_lock.unlock();
av_packet_unref(outPacket);
}
ret = 0;
}
av_frame_free(&scaledFrame);
av_packet_unref(outPacket);
}
}
}
}
int ScreenRecorder::captureVideoFrames() {
int64_t pts = 0;
int flag;
int frameFinished = 0;
bool endPause = false;
int numPause = 0;
ofstream outFile{ "..\\media\\log.txt", ios::out };
int frameIndex = 0;
value = 0;
pAVPacket = (AVPacket*)av_malloc(sizeof(AVPacket));
if (pAVPacket == nullptr) {
cerr << "Error in allocating AVPacket" << endl;
exit(-1);
}
pAVFrame = av_frame_alloc();
if (pAVFrame == nullptr) {
cerr << "Error: unable to alloc the AVFrame resources" << endl;
exit(-1);
}
outFrame = av_frame_alloc();
if (outFrame == nullptr) {
cerr << "Error: unable to alloc the AVFrame resources for out frame" << endl;
exit(-1);
}
int videoOutBuffSize;
int nBytes = av_image_get_buffer_size(outVideoCodecContext->pix_fmt, outVideoCodecContext->width, outVideoCodecContext->height, 32);
uint8_t* videoOutBuff = (uint8_t*)av_malloc(nBytes);
if (videoOutBuff == nullptr) {
cerr << "Error: unable to allocate memory" << endl;
exit(-1);
}
value = av_image_fill_arrays(outFrame->data, outFrame->linesize, videoOutBuff, AV_PIX_FMT_YUV420P, outVideoCodecContext->width, outVideoCodecContext->height, 1);
if (value < 0) {
cerr << "Error in filling image array" << endl;
}
SwsContext* swsCtx_;
if (avcodec_open2(pAVCodecContext, pAVCodec, nullptr) < 0) {
cerr << "Could not open codec" << endl;
exit(-1);
}
swsCtx_ = sws_getContext(pAVCodecContext->width, pAVCodecContext->height, pAVCodecContext->pix_fmt, outVideoCodecContext->width, outVideoCodecContext->height, outVideoCodecContext->pix_fmt, SWS_BICUBIC,
nullptr, nullptr, nullptr);
AVPacket outPacket;
int gotPicture;
time_t startTime;
time(&startTime);
while (true) {
if (pauseCapture) {
cout << "Pause" << endl;
outFile << "/////////////////// Pause ///////////////////" << endl;
cout << "outVideoCodecContext->time_base: " << outVideoCodecContext->time_base.num << ", " << outVideoCodecContext->time_base.den << endl;
}
cv.wait(ul, [this]() { return !pauseCapture; }); //pause capture (not busy waiting)
if (endPause) {
endPause = false;
}
if (stopCapture) //check if the capture has to stop
break;
ul.unlock();
if (av_read_frame(pAVFormatContext, pAVPacket) >= 0 && pAVPacket->stream_index == VideoStreamIndx) {
av_packet_rescale_ts(pAVPacket, pAVFormatContext->streams[VideoStreamIndx]->time_base, pAVCodecContext->time_base);
value = avcodec_decode_video2(pAVCodecContext, pAVFrame, &frameFinished, pAVPacket);
if (value < 0) {
cout << "Unable to decode video" << endl;
}
if (frameFinished) { //frame successfully decoded
//sws_scale(swsCtx_, pAVFrame->data, pAVFrame->linesize, 0, pAVCodecContext->height, outFrame->data, outFrame->linesize);
av_init_packet(&outPacket);
outPacket.data = nullptr;
outPacket.size = 0;
if (outAVFormatContext->streams[outVideoStreamIndex]->start_time <= 0) {
outAVFormatContext->streams[outVideoStreamIndex]->start_time = pAVFrame->pts;
}
//disable warning on the console
outFrame->width = outVideoCodecContext->width;
outFrame->height = outVideoCodecContext->height;
outFrame->format = outVideoCodecContext->pix_fmt;
sws_scale(swsCtx_, pAVFrame->data, pAVFrame->linesize, 0, pAVCodecContext->height, outFrame->data, outFrame->linesize);
avcodec_encode_video2(outVideoCodecContext, &outPacket, outFrame, &gotPicture);
if (gotPicture) {
if (outPacket.pts != AV_NOPTS_VALUE) {
outPacket.pts = av_rescale_q(outPacket.pts, videoSt->codec->time_base, videoSt->time_base);
}
if (outPacket.dts != AV_NOPTS_VALUE) {
outPacket.dts = av_rescale_q(outPacket.dts, videoSt->codec->time_base, videoSt->time_base);
}
//cout << "Write frame " << j++ << " (size = " << outPacket.size / 1000 << ")" << endl;
//cout << "(size = " << outPacket.size << ")" << endl;
//av_packet_rescale_ts(&outPacket, outVideoCodecContext->time_base, outAVFormatContext->streams[outVideoStreamIndex]->time_base);
//outPacket.stream_index = outVideoStreamIndex;
outFile << "outPacket->duration: " << outPacket.duration << ", " << "pAVPacket->duration: " << pAVPacket->duration << endl;
outFile << "outPacket->pts: " << outPacket.pts << ", " << "pAVPacket->pts: " << pAVPacket->pts << endl;
outFile << "outPacket.dts: " << outPacket.dts << ", " << "pAVPacket->dts: " << pAVPacket->dts << endl;
time_t timer;
double seconds;
mu.lock();
if (!activeMenu) {
time(&timer);
seconds = difftime(timer, startTime);
int h = (int)(seconds / 3600);
int m = (int)(seconds / 60) % 60;
int s = (int)(seconds) % 60;
std::cout << std::flush << "\r" << std::setw(2) << std::setfill('0') << h << ':'
<< std::setw(2) << std::setfill('0') << m << ':'
<< std::setw(2) << std::setfill('0') << s << std::flush;
}
mu.unlock();
write_lock.lock();
if (av_write_frame(outAVFormatContext, &outPacket) != 0) {
cerr << "Error in writing video frame" << endl;
}
write_lock.unlock();
av_packet_unref(&outPacket);
}
av_packet_unref(&outPacket);
av_free_packet(pAVPacket); //avoid memory saturation
}
}
}
outFile.close();
av_free(videoOutBuff);
return 0;
}
I resolved this problem performing an avformat_close_input(&inAudioFormatContext) before enter in pause, and an avformat_open_input(&inAudioFormatContext, "audio=Microfono (Realtek(R) Audio)", audioInputFormat, &audioOptions) after resume the recording. In this way the final file seems well syncronized with video.

Playing a stereo .wav file with PortAudio & sndfile, output is fuzzy and pitched down / slowed

I've been writing some code to play back a stereo .wav file in c++ using PortAudio and sndfile, but the output sound is fuzzy and pitched down (pitch down is less of an issue for me but it may be part of the problem). It almost seems like its playing partially garbage data with the fuzz, but I don't believe there's any variable I'm using that this could happen where I'm not clearing the data first. I've tried to follow some of PortAudios examples with stereo playback but due to the input being from a .wav file rather than generated I've not been able to follow it perfectly. I've also compiled and ran some PortAudio examples (that use stereo sound) and those work fine. I'm not sure where the issue is.
Audio.h
struct AudioFile {
SNDFILE* file = nullptr;
SF_INFO info;
int buffer_size = 512;
int readHead = 0;
sf_count_t count = 1;
};
/*
Class for handling basic audio functions
*/
class Audio {
protected:
public:
/// Constructor
Audio();
/// Destructor
~Audio();
/// Load an audio file
AudioFile loadFile(const char* path);
/// Play an audio file
void playFile(AudioFile* file);
};
Audio.cpp
/// Audio constructor
Audio::Audio() {
PaError err = Pa_Initialize();
if (err != paNoError) std::cerr << "PAError: " << err << std::endl;
#ifdef DEBUG
std::cout << "Initialising PortAudio" << std::endl;
std::cout << "----------------------" << std::endl;
std::cout << "Version: " << Pa_GetVersion << std::endl;
std::cout << "Devices:" << std::endl;
std::cout << "----------------------" << std::endl;
int numDevices = Pa_GetDeviceCount();
for (int i=0; i < numDevices; i++) {
auto deviceInfo = Pa_GetDeviceInfo(i);
std::cout << "Name: " << deviceInfo->name << std::endl;
std::cout << "HostApi: " << deviceInfo->hostApi << std::endl;
std::cout << "SampleRate: " << deviceInfo->defaultSampleRate << std::endl;
std::cout << "InputChannels: " << deviceInfo->maxInputChannels << std::endl;
std::cout << "OutputChannels: " << deviceInfo->maxOutputChannels << std::endl;
std::cout << "----------------------" << std::endl;
}
#endif
}
Audio::~Audio() {
PaError err = Pa_Terminate();
if (err != paNoError) std::cerr << "PAError: " << err << std::endl;
}
/* Loads an audiofile */
AudioFile Audio::loadFile(const char* path) {
AudioFile file;
::memset(&file.info, 0, sizeof(file.info));
file.file = sf_open(path, SFM_READ, &file.info);
return file;
}
static int patestCallback(const void* inputBuffer, void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags, void* userData) {
/// Prevent warnings
(void)inputBuffer;
(void)timeInfo;
(void)statusFlags;
/// an AudioFile gets passed as userData
AudioFile* file = (AudioFile*)userData;
float* out = (float*)outputBuffer;
sf_seek(file->file, file->readHead, SF_SEEK_SET);
auto data = std::make_unique<float[]>(framesPerBuffer);
file->count = sf_read_float(file->file, data.get(), framesPerBuffer);
for (int i = 0; i < framesPerBuffer; i++) {
*out++ = data[i];
}
file->readHead += file->buffer_size;
if (file->count > 0) return paContinue;
else return paComplete;
}
void Audio::playFile(AudioFile* file) {
PaStream* stream = nullptr;
PaStreamParameters params;
params.device = Pa_GetDefaultOutputDevice();
params.channelCount = file->info.channels;
params.sampleFormat = paFloat32;
params.suggestedLatency =
Pa_GetDeviceInfo(params.device)->defaultLowOutputLatency;
params.hostApiSpecificStreamInfo = nullptr;
/// Check if params work
PaError err = Pa_IsFormatSupported(nullptr, &params, file->info.samplerate);
if (err != paFormatIsSupported) {
std::cerr << "PAError: " << Pa_GetErrorText(err) << std::endl;
return;
}
err = Pa_OpenStream(&stream, nullptr, &params, file->info.samplerate,
file->buffer_size * params.channelCount, paClipOff,
&patestCallback, file);
if (err != paNoError) std::cerr << "PAError: " << Pa_GetErrorText(err) << std::endl;
err = Pa_StartStream(stream);
if (err != paNoError)
std::cerr << "PAError: " << Pa_GetErrorText(err) << std::endl;
/// wait until file finishes playing
while (file->count > 0) {}
err = Pa_StopStream(stream);
if (err != paNoError)
std::cerr << "PAError: " << Pa_GetErrorText(err) << std::endl;
err = Pa_CloseStream(stream);
if (err != paNoError)
std::cerr << "PAError: " << Pa_GetErrorText(err) << std::endl;
}
I've also tried without the data pointer (using this does seem to produce a cleaner, but still fuzzy sound) and passing the audio file by value into the playFile function. Any help is appreciated.
Ended up figuring it out, I had one primary issue, here:
err = Pa_OpenStream(&stream, nullptr, &params, file->info.samplerate,
file->buffer_size * params.channelCount, paClipOff,
&patestCallback, file);
I gave the Pa_OpenStream the buffersize * the number of channels, however I should've just been giving it the buffersize, and make the channel adjustment to the framesPerBuffer in the callback function directly:
static int patestCallback(const void* inputBuffer, void* outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags, void* userData) {
/// Prevent warnings
(void)inputBuffer;
(void)timeInfo;
(void)statusFlags;
/// an AudioFile gets passed as userData
velox::AudioFile* file = (velox::AudioFile*)userData;
float* out = (float*)outputBuffer;
sf_seek(file->file, file->readHead, SF_SEEK_SET);
auto data = std::make_unique<float[]>(framesPerBuffer * file->info.channels);
file->count = sf_read_float(file->file, data.get(),
framesPerBuffer * file->info.channels);
for (int i = 0; i < framesPerBuffer * file->info.channels; i++) {
*out++ = data[i];
}
file->readHead += file->buffer_size;
if (file->count > 0) return paContinue;
else return paComplete;
}
This change fixed both the pitch and the fuzz, which makes sense in hindsight.

Adding metadata informations with ffmpeg

I am trying to read and most important to write metadata into file using ffmpeg. But I am getting sigseg.
int main(int argc, char **argv) {
av_register_all();
AVFormatContext* ctx;
std::string path("/home/stefan/test_track.mp3");
ctx = avformat_alloc_context();
if (avformat_open_input(&ctx, path.c_str(), 0, 0) < 0)
std::cout << "error1" << std::endl;
if (avformat_find_stream_info(ctx, 0) < 0)
std::cout << "error2" << std::endl;
AVDictionaryEntry *tag = nullptr;
tag = av_dict_get(ctx->metadata, "artist", tag, AV_DICT_IGNORE_SUFFIX);
std::cout << tag->key << " : " << tag->value << std::endl;
av_dict_set(&ctx->metadata, "TPFL", "testtest", 0);
std::cout << "test!" << std::endl;
int status = avformat_write_header(ctx, &ctx->metadata);
if(status == 0)
std::cout << "test1" << std::endl;
return 0;
}
I also tried to make a entire copy of AVDictionary and than saving it with new addition of field I want to save. But still sigseg.
What am I missing?
You need to use another AVFormatContext to write your output.
In your example, just add some meta data and copy the codec, so the muxing steps for ffmpeg library is
create the desired output format context, avformat_alloc_output_context2
add streams to the output format context, avformat_new_stream
add some custom meta data and write header
use av_write_frame to write the encoded data
write trailer
Below is a workable example with small modification of your code.
#include <iostream>
#ifdef __cplusplus
extern "C" {
#endif
#include <libavformat/avformat.h>
#ifdef __cplusplus
}
#endif
int main(int argc, char **argv) {
av_register_all();
avcodec_register_all();
AVFormatContext* ctx;
std::string path("./input.mp3");
ctx = avformat_alloc_context();
if (avformat_open_input(&ctx, path.c_str(), 0, 0) < 0)
std::cout << "error1" << std::endl;
if (avformat_find_stream_info(ctx, 0) < 0)
std::cout << "error2" << std::endl;
AVDictionaryEntry *tag = nullptr;
tag = av_dict_get(ctx->metadata, "artist", tag, AV_DICT_IGNORE_SUFFIX);
std::cout << tag->key << " : " << tag->value << std::endl;
av_dict_set(&ctx->metadata, "TPFL", "testtest", 0);
std::cout << "test!" << std::endl;
int status;
AVFormatContext* ofmt_ctx;
AVOutputFormat* ofmt = av_guess_format("mp3", "./out.mp3", NULL);
status = avformat_alloc_output_context2(&ofmt_ctx, ofmt, "mp3", "./out.mp3");
if (status < 0) {
std::cerr << "could not allocate output format" << std::endl;
return 0;
}
int audio_stream_index = 0;
for (unsigned i = 0; i < ctx->nb_streams; i++) {
if (ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
audio_stream_index = i;
const AVCodec *c = avcodec_find_encoder(ctx->streams[i]->codecpar->codec_id);
if (c) {
AVStream *ostream = avformat_new_stream(ofmt_ctx, c);
avcodec_parameters_copy(ostream->codecpar, ctx->streams[i]->codecpar);
ostream->codecpar->codec_tag = 0;
}
break;
}
}
av_dict_set(&ofmt_ctx->metadata, "TPFL", "testtest", 0);
av_dump_format(ofmt_ctx, 0, "./out.mp3", 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
avio_open(&ofmt_ctx->pb, "./out.mp3", AVIO_FLAG_WRITE);
}
if (avformat_init_output(ofmt_ctx, NULL) == AVSTREAM_INIT_IN_WRITE_HEADER) {
status = avformat_write_header(ofmt_ctx, NULL);
}
AVPacket *pkt = av_packet_alloc();
av_init_packet(pkt);
pkt->data = NULL;
pkt->size = 0;
while (av_read_frame(ctx, pkt) == 0) {
if (pkt->stream_index == audio_stream_index) {
// this is optional, we are copying the stream
av_packet_rescale_ts(pkt, ctx->streams[audio_stream_index]->time_base,
ofmt_ctx->streams[audio_stream_index]->time_base);
av_write_frame(ofmt_ctx, pkt);
}
}
av_packet_free(&pkt);
av_write_trailer(ofmt_ctx);
avformat_close_input(&ctx);
avformat_free_context(ofmt_ctx);
avformat_free_context(ctx);
if(status == 0)
std::cout << "test1" << std::endl;
return 0;
}
After done, you can use ffprobe output.mp3 to check the written meta data.
Input #0, mp3, from 'out.mp3':
Metadata:
TPFL : testtest
encoder : Lavf57.75.100
Duration: 00:00:08.75, start: 0.011995, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
encoder : Lavf

Binaural beats in C++

I am trying to build a brain device using EEG input and outputting light pulses and binaural beats in "close to real-time" on a Raspberry PI. The light-output is no problem using WiringPi, but any audio output seems to be a major hurdle. The math of calculating a sinewave for a buffer is straightforward, but playing two frequencies on two channels via any standard libraries seems to be a very complicated process, and I can't come up with any relevant examples. I have successfully opened and closed an ALSA device thanks to this tutorial which complicates my fairly simple code tremendously but appears necessary for ALSA. I would be incredibly grateful if someone could show me the easiest method for playing two different calculated tones on left and right channels. The code below is the simplest ALSA playback example I could find.
#include <alsa/asoundlib.h>
#include <iostream>
using namespace std;
// Globals are generally a bad idea in code. We're using one here to keep it simple.
snd_pcm_t * _soundDevice;
bool Init(const char *name)
{
int i;
int err;
snd_pcm_hw_params_t *hw_params;
if( name == NULL )
{
// Try to open the default device
err = snd_pcm_open( &_soundDevice, "plughw:0,0", SND_PCM_STREAM_PLAYBACK, 0 );
}
else
{
// Open the device we were told to open.
err = snd_pcm_open (&_soundDevice, name, SND_PCM_STREAM_PLAYBACK, 0);
}
// Check for error on open.
if( err < 0 )
{
cout << "Init: cannot open audio device " << name << " (" << snd_strerror (err) << ")" << endl;
return false;
}
else
{
cout << "Audio device opened successfully." << endl;
}
// Allocate the hardware parameter structure.
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{
cout << "Init: cannot allocate hardware parameter structure (" << snd_strerror (err) << ")" << endl;
return false;
}
if ((err = snd_pcm_hw_params_any (_soundDevice, hw_params)) < 0)
{
cout << "Init: cannot initialize hardware parameter structure (" << snd_strerror (err) << ")" << endl;
return false;
}
// Enable resampling.
unsigned int resample = 1;
err = snd_pcm_hw_params_set_rate_resample(_soundDevice, hw_params, resample);
if (err < 0)
{
cout << "Init: Resampling setup failed for playback: " << snd_strerror(err) << endl;
return err;
}
// Set access to RW interleaved.
if ((err = snd_pcm_hw_params_set_access (_soundDevice, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
cout << "Init: cannot set access type (" << snd_strerror (err) << ")" << endl;
return false;
}
if ((err = snd_pcm_hw_params_set_format (_soundDevice, hw_params, SND_PCM_FORMAT_S16_LE)) < 0)
{
cout << "Init: cannot set sample format (" << snd_strerror (err) << ")" << endl;
return false;
}
// Set channels to stereo (2).
if ((err = snd_pcm_hw_params_set_channels (_soundDevice, hw_params, 2)) < 0)
{
cout << "Init: cannot set channel count (" << snd_strerror (err) << ")" << endl;
return false;
}
// Set sample rate.
unsigned int actualRate = 44100;
if ((err = snd_pcm_hw_params_set_rate_near (_soundDevice, hw_params, &actualRate, 0)) < 0)
{
cout << "Init: cannot set sample rate to 44100. (" << snd_strerror (err) << ")" << endl;
return false;
}
if( actualRate < 44100 )
{
cout << "Init: sample rate does not match requested rate. (" << "44100 requested, " << actualRate << " acquired)" << endl;
}
// Apply the hardware parameters that we've set.
if ((err = snd_pcm_hw_params (_soundDevice, hw_params)) < 0)
{
cout << "Init: cannot set parameters (" << snd_strerror (err) << ")" << endl;
return false;
}
else
{
cout << "Audio device parameters have been set successfully." << endl;
}
// Get the buffer size.
snd_pcm_uframes_t bufferSize;
snd_pcm_hw_params_get_buffer_size( hw_params, &bufferSize );
// If we were going to do more with our sound device we would want to store
// the buffer size so we know how much data we will need to fill it with.
cout << "Init: Buffer size = " << bufferSize << " frames." << endl;
// Display the bit size of samples.
cout << "Init: Significant bits for linear samples = " << snd_pcm_hw_params_get_sbits(hw_params) << endl;
// Free the hardware parameters now that we're done with them.
snd_pcm_hw_params_free (hw_params);
// Prepare interface for use.
if ((err = snd_pcm_prepare (_soundDevice)) < 0)
{
cout << "Init: cannot prepare audio interface for use (" << snd_strerror (err) << ")" << endl;
return false;
}
else
{
cout << "Audio device has been prepared for use." << endl;
}
return true;
}
bool UnInit()
{
snd_pcm_close (_soundDevice);
cout << "Audio device has been uninitialized." << endl;
return true;
}
int main( char *argc, int argv )
{
Init(NULL);
UnInit();
return 0;
}
Use some up-to-date example, like this:
#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
static const char *device = "default";
unsigned short buffer[2 * 24000];
int main(void)
{
int err;
snd_pcm_t *handle;
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
printf("open error: %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
if ((err = snd_pcm_set_params(handle,
SND_PCM_FORMAT_S16,
SND_PCM_ACCESS_RW_INTERLEAVED,
2, /* channels */
48000, /* rate */
1,
500000)) < 0) { /* buffer: 0.5 sec */
printf("open error: %s\n", snd_strerror(err));
exit(EXIT_FAILURE);
}
for (;;) {
for (int i = 0; i < 24000; i++) {
buffer[2 * i + 0] = 32767 * sin(...); /* left channel */
buffer[2 * i + 1] = 32767 * sin(...); /* right channel */
}
snd_pcm_sframes_t frames = snd_pcm_writei(handle, buffer, 24000);
if (frames < 0)
frames = snd_pcm_recover(handle, frames, 0);
if (frames < 0) {
printf("snd_pcm_writei failed: %s\n", snd_strerror(err));
break;
}
}
snd_pcm_close(handle);
return 0;
}