/* The load_voice method loads a sample into the driver's memory. The voice's
'streaming' field will be set to false for these voices, and it's
'buffer_size' field will be the total length in bytes of the sample data.
The voice's attached sample's looping mode should be honored, and loading
must fail if it cannot be. */
static int _openal_load_voice(ALLEGRO_VOICE *voice, const void *data)
{
ALLEGRO_AL_DATA *ex_data = voice->extra;
ALenum openal_err;
if (voice->attached_stream->loop != ALLEGRO_PLAYMODE_ONCE &&
voice->attached_stream->loop != ALLEGRO_PLAYMODE_LOOP) {
return 1;
}
ex_data->buffer_size = voice->buffer_size;
if (!ex_data->buffer_size) {
ALLEGRO_ERROR("Voice buffer and data buffer size mismatch\n");
return 1;
}
ex_data->num_buffers = 1;
alGenSources(1, &ex_data->source);
if ((openal_err = alGetError()) != AL_NO_ERROR) {
ALLEGRO_ERROR("Could not generate (voice) source: %s\n",
openal_get_err_str(openal_err));
return 1;
}
ex_data->buffers = al_malloc(sizeof(ALuint) * ex_data->num_buffers);
if (!ex_data->buffers) {
alSourcei(ex_data->source, AL_BUFFER, 0);
alDeleteSources(1, &ex_data->source);
ALLEGRO_ERROR("Could not allocate voice buffer memory\n");
return 1;
}
alGenBuffers(ex_data->num_buffers, ex_data->buffers);
if ((openal_err = alGetError()) != AL_NO_ERROR) {
alSourcei(ex_data->source, AL_BUFFER, 0);
alDeleteSources(1, &ex_data->source);
al_free(ex_data->buffers);
ex_data->buffers = NULL;
ALLEGRO_ERROR("Could not generate (voice) buffer: %s\n",
openal_get_err_str(openal_err));
return 1;
}
/* copies data into a buffer */
alBufferData(ex_data->buffers[0], ex_data->format,
data, ex_data->buffer_size, voice->frequency);
/* sets the buffer */
alSourcei(ex_data->source, AL_BUFFER, ex_data->buffers[0]);
/* Loop / no loop? */
alSourcei(ex_data->source, AL_LOOPING,
(voice->attached_stream->loop != ALLEGRO_PLAYMODE_ONCE));
/* make sure the volume is on */
alSourcef(ex_data->source, AL_GAIN, 1.0f);
if ((openal_err = alGetError()) != AL_NO_ERROR) {
alSourcei(ex_data->source, AL_BUFFER, 0);
alDeleteSources(1, &ex_data->source);
alDeleteBuffers(ex_data->num_buffers, ex_data->buffers);
al_free(ex_data->buffers);
ex_data->buffers = NULL;
ALLEGRO_ERROR("Could not attach voice source: %s\n",
openal_get_err_str(openal_err));
return 1;
}
return 0;
}
SoundSource::~SoundSource()
{
alCheck(alSourcei(m_source, AL_BUFFER, 0));
alCheck(alDeleteSources(1, &m_source));
}
SoundChannel::~SoundChannel()
{
#ifdef __DAVASOUND_AL__
AL_VERIFY(alDeleteSources(1, &source));
#endif //#ifdef __DAVASOUND_AL__
}
~OpenAlChannel() {
alDeleteSources(1, &_source);
alGetError(); // discard.
}
void Cyanide::audio_thread()
{
return;
const char *device_list, *output_device = NULL;
//void *audio_device = NULL;
bool call[MAX_CALLS] = {0}, preview = 0;
bool audio_filtering_enabled;
// bool groups_audio[MAX_NUM_GROUPS] = {0};
int perframe = (av_DefaultSettings.audio_frame_duration * av_DefaultSettings.audio_sample_rate) / 1000;
uint8_t buf[perframe * 2 * av_DefaultSettings.audio_channels], dest[perframe * 2 * av_DefaultSettings.audio_channels];
memset(buf, 0, sizeof(buf));
uint8_t audio_count = 0;
bool record_on = 0;
#ifdef AUDIO_FILTERING
qDebug() << "Audio Filtering enabled";
#ifdef ALC_LOOPBACK_CAPTURE_SAMPLES
qDebug() << "Echo cancellation enabled";
#endif
#endif
qDebug() << "frame size:" << perframe;
device_list = alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
if(device_list) {
output_device = device_list;
qDebug() << "Output Device List:";
while(*device_list) {
qDebug() << device_list;
//postmessage(NEW_AUDIO_OUT_DEVICE, 0, 0, (void*)device_list);
device_list += strlen(device_list) + 1;
}
}
device_out = alcOpenDevice(output_device);
if(!device_out) {
qDebug() << "alcOpenDevice() failed";
return;
}
int attrlist[] = { ALC_FREQUENCY, av_DefaultSettings.audio_sample_rate,
ALC_INVALID };
context = alcCreateContext(device_out, attrlist);
if(!alcMakeContextCurrent(context)) {
qDebug() << "alcMakeContextCurrent() failed";
alcCloseDevice(device_out);
return;
}
alGenSources(countof(source), source);
static ALuint ringSrc[MAX_CALLS];
alGenSources(MAX_CALLS, ringSrc);
/* Create buffer to store samples */
ALuint RingBuffer;
alGenBuffers(1, &RingBuffer);
{
float frequency1 = 441.f;
float frequency2 = 882.f;
int seconds = 4;
unsigned sample_rate = 22050;
size_t buf_size = seconds * sample_rate * 2; //16 bit (2 bytes per sample)
int16_t *samples = (int16_t*)malloc(buf_size * sizeof(int16_t));
if (!samples)
return;
/*Generate an electronic ringer sound that quickly alternates between two frequencies*/
int index = 0;
for(index = 0; index < buf_size; ++index) {
if ((index / (sample_rate)) % 4 < 2 ) {//4 second ring cycle, first 2 secondsring, the rest(2 seconds) is silence
if((index / 1000) % 2 == 1) {
samples[index] = 5000 * sin((2.0 * 3.1415926 * frequency1) / sample_rate * index); //5000=amplitude(volume level). It can be from zero to 32700
} else {
samples[index] = 5000 * sin((2.0 * 3.1415926 * frequency2) / sample_rate * index);
}
} else {
samples[index] = 0;
}
}
alBufferData(RingBuffer, AL_FORMAT_MONO16, samples, buf_size, sample_rate);
free(samples);
}
{
unsigned int i;
for (i = 0; i < MAX_CALLS; ++i) {
alSourcei(ringSrc[i], AL_LOOPING, AL_TRUE);
alSourcei(ringSrc[i], AL_BUFFER, RingBuffer);
}
}
#ifdef AUDIO_FILTERING
Filter_Audio *f_a = NULL;
#endif
while(loop == LOOP_RUN || loop == LOOP_SUSPEND) {
#ifdef AUDIO_FILTERING
if (!f_a && audio_filtering_enabled) {
f_a = new_filter_audio(av_DefaultSettings.audio_sample_rate);
if (!f_a) {
audio_filtering_enabled = 0;
qDebug() << "filter audio failed";
} else {
qDebug() << "filter audio on";
}
} else if (f_a && !audio_filtering_enabled) {
kill_filter_audio(f_a);
f_a = NULL;
qDebug() << "filter audio off";
}
#else
if (audio_filtering_enabled)
audio_filtering_enabled = 0;
#endif
bool sleep = 1;
if(record_on) {
ALint samples;
alcGetIntegerv(device_in, ALC_CAPTURE_SAMPLES, sizeof(samples), &samples);
if(samples >= perframe) {
alcCaptureSamples(device_in, buf, perframe);
if (samples >= perframe * 2) {
sleep = 0;
}
}
}
#ifdef AUDIO_FILTERING
#ifdef ALC_LOOPBACK_CAPTURE_SAMPLES
if (f_a && audio_filtering_enabled) {
ALint samples;
alcGetIntegerv(device_out, ALC_LOOPBACK_CAPTURE_SAMPLES, sizeof(samples), &samples);
if(samples >= perframe) {
int16_t buffer[perframe];
alcCaptureSamplesLoopback(device_out, buffer, perframe);
pass_audio_output(f_a, buffer, perframe);
set_echo_delay_ms(f_a, 5);
if (samples >= perframe * 2) {
sleep = 0;
}
}
}
#endif
#endif
#ifdef AUDIO_FILTERING
if (f_a && filter_audio(f_a, (int16_t*)buf, perframe) == -1) {
qDebug() << "filter audio error";
}
#endif
if(preview) {
audio_play(0, (int16_t*)buf, perframe, av_DefaultSettings.audio_channels, av_DefaultSettings.audio_sample_rate);
}
int i;
for(i = 0; i < MAX_CALLS; i++) {
if(call[i]) {
int r;
if((r = toxav_prepare_audio_frame(toxav, i, dest, sizeof(dest), (const int16_t*)buf, perframe)) < 0) {
qDebug() << "toxav_prepare_audio_frame error" << r;
continue;
}
if((r = toxav_send_audio(toxav, i, dest, r)) < 0) {
qDebug() << "toxav_send_audio error" << r;
}
}
}
if (sleep) {
usleep(5000);
}
}
#ifdef AUDIO_FILTERING
kill_filter_audio(f_a);
#endif
//missing some cleanup ?
alDeleteSources(MAX_CALLS, ringSrc);
alDeleteSources(countof(source), source);
alDeleteBuffers(1, &RingBuffer);
if(device_in) {
if(record_on) {
alcCaptureStop(device_in);
}
alcCaptureCloseDevice(device_in);
}
alcMakeContextCurrent(NULL);
alcDestroyContext(context);
alcCloseDevice(device_out);
}
int
main() {
char input[INPUT_BUFFER];
char *words[MAX_WORDS];
int count_of_words, signal_index, sleep_usec, count_of_processed, count_of_queued;
int i;
ALint state;
ALshort signals[BUFFER];
ALCdevice *device;
ALCcontext *context;
ALuint buffer, source;
sleep_usec = (int)(((1.0 / SAMPLING_FREQUENCY) * MAX_WORDS / 2 ) * 1000 * 1000);
device = alcOpenDevice(NULL);
context = alcCreateContext(device, NULL);
alcMakeContextCurrent(context);
alGenSources(1, &source);
alSourcei(source, AL_SOURCE_TYPE, AL_STREAMING);
while (fgets(input, INPUT_BUFFER, stdin) != NULL) {
puts("### got input");
count_of_words = split_string(input, words);
signal_index = 0;
for (i = 0; i < count_of_words; i++) {
signals[signal_index++] = (ALshort)atoi(words[i]);
}
if (alGetSourcei(source, AL_BUFFERS_QUEUED, &count_of_queued), count_of_queued < COUNT_OF_BUFFERS) {
alGenBuffers(1, &buffer);
}
else {
while (alGetSourcei(source, AL_BUFFERS_PROCESSED, &count_of_processed), count_of_processed == 0) {
usleep(sleep_usec);
}
for (i = 0; i < count_of_processed; i++) {
alSourceUnqueueBuffers(source, 1, &buffer);
}
}
alBufferData(buffer, AL_FORMAT_MONO16, signals, sizeof(ALshort) * signal_index, SAMPLING_FREQUENCY);
alSourceQueueBuffers(source, 1, &buffer);
if (alGetSourcei(source, AL_SOURCE_STATE, &state), state != AL_PLAYING) {
puts("--- start playing");
alGetSourcei(source, AL_BUFFERS_PROCESSED, &count_of_processed);
for (i = 0; i < count_of_processed - 1; i++) {
printf("--- unqueue: %d\n", i);
alSourceUnqueueBuffers(source, 1, &buffer);
}
alSourcePlay(source);
}
}
alSourceStop(source);
alDeleteSources(1, &source);
alDeleteBuffers(1, &buffer);
alcMakeContextCurrent(NULL);
alcDestroyContext(context);
alcCloseDevice(device);
return 0;
}
int main(void)
{
/*Отсюда нужно две функции -
1) получение в виде массива (строки) данных с микрофона за время необработки
2)Проигрвание строки равной значению
*/
const ALCchar * devices;
const ALCchar * ptr;
ALCdevice * mainDev;
ALCcontext * mainContext;
ALCdevice * captureDev;
ALubyte captureBuffer[1048576];
ALubyte *captureBufPtr;
ALint samplesAvailable;
ALint samplesCaptured;
time_t currentTime;
time_t lastTime;
ALuint buffer;
ALuint source;
ALint playState;
int i;
// Print the list of capture devices
printf("Available playback devices:\n");
devices = alcGetString(NULL, ALC_DEVICE_SPECIFIER);
ptr = devices;
//while (ptr[0] != NULL)
while (*ptr)
{
printf(" %s\n", ptr);
ptr += strlen(ptr) + 1;
}
// Open a playback device and create a context first
printf("Opening playback device:\n");
mainDev = alcOpenDevice(NULL);
if (mainDev == NULL)
{
printf("Unable to open playback device!\n");
exit(1);
}
devices = alcGetString(mainDev, ALC_DEVICE_SPECIFIER);
printf(" opened device '%s'\n", devices);
mainContext = alcCreateContext(mainDev, NULL);
if (mainContext == NULL)
{
printf("Unable to create playback context!\n");
exit(1);
}
printf(" created playback context\n");
// Make the playback context current
alcMakeContextCurrent(mainContext);
alcProcessContext(mainContext);
// Print the list of capture devices
printf("Available capture devices:\n");
devices = alcGetString(NULL, ALC_CAPTURE_DEVICE_SPECIFIER);
ptr = devices;
//while (ptr[0] != NULL)
while (*ptr)
{
printf(" %s\n", ptr);
ptr += strlen(ptr) + 1;
}
// Open the default device
printf("Opening capture device:\n");
captureDev = alcCaptureOpenDevice(NULL, 8000, AL_FORMAT_MONO16, 800);
if (captureDev == NULL)
{
printf(" Unable to open device!\n");
exit(1);
}
devices = alcGetString(captureDev, ALC_CAPTURE_DEVICE_SPECIFIER);
printf(" opened device %s\n", devices);
// Wait for three seconds to prompt the user
for (i = 3; i > 0; i--)
{
printf("Starting capture in %d...\r", i);
fflush(stdout);
lastTime = time(NULL);
currentTime = lastTime;
while (currentTime == lastTime)
{
currentTime = time(NULL);
usleep(100000);
}
}
printf("Starting capture NOW!\n");
fflush(stdout);
lastTime = currentTime;
// Capture (roughly) five seconds of audio
alcCaptureStart(captureDev);
samplesCaptured = 0;
captureBufPtr = captureBuffer;
while (currentTime < (lastTime + 5))
{
// Get the number of samples available
alcGetIntegerv(captureDev, ALC_CAPTURE_SAMPLES, 1, &samplesAvailable);
// Copy the samples to our capture buffer
if (samplesAvailable > 0)
{
alcCaptureSamples(captureDev, captureBufPtr, samplesAvailable);
samplesCaptured += samplesAvailable;
printf("Captured %d samples (adding %d)\r", samplesCaptured,
samplesAvailable);
fflush(stdout);
// Advance the buffer (two bytes per sample * number of samples)
captureBufPtr += samplesAvailable * 2;
}
// Wait for a bit
usleep(10000);
// Update the clock
currentTime = time(NULL);
}
printf("\nPausing capture.\n");
alcCaptureStop(captureDev);
// Wait for three seconds to prompt the user
for (i = 3; i > 0; i--)
{
printf("Resuming capture in %d...\r", i);
fflush(stdout);
lastTime = time(NULL);
currentTime = lastTime;
while (currentTime == lastTime)
{
currentTime = time(NULL);
usleep(100000);
}
}
printf("Resuming capture NOW!\n");
fflush(stdout);
lastTime = currentTime;
// Capture (roughly) five seconds of audio
alcCaptureStart(captureDev);
while (currentTime < (lastTime + 5))
{
// Get the number of samples available
alcGetIntegerv(captureDev, ALC_CAPTURE_SAMPLES, 1, &samplesAvailable);
// Copy the samples to our capture buffer
if (samplesAvailable > 0)
{
alcCaptureSamples(captureDev, captureBufPtr, samplesAvailable);
samplesCaptured += samplesAvailable;
printf("Captured %d samples (adding %d)\r", samplesCaptured,
samplesAvailable);
fflush(stdout);
// Advance the buffer (two bytes per sample * number of samples)
captureBufPtr += samplesAvailable * 2;
}
// Wait for a bit
usleep(10000);
// Update the clock
currentTime = time(NULL);
}
printf("\nDone capturing.\n");
alcCaptureStop(captureDev);
// Play back the captured data
printf("Starting playback...\n");
fflush(stdout);
// Generate an OpenAL buffer for the captured data
alGenBuffers(1, &buffer);
alGenSources(1, &source);
alBufferData(buffer, AL_FORMAT_MONO16, captureBuffer,samplesCaptured*2, 8000);
alSourcei(source, AL_BUFFER, buffer);
alSourcePlay(source);
// Wait for the source to stop playing
playState = AL_PLAYING;
while (playState == AL_PLAYING)
{
printf(" source %d is playing...\r", source);
fflush(stdout);
alGetSourcei(source, AL_SOURCE_STATE, &playState);
usleep(100000);
}
printf("\nDone with playback.\n");
fflush(stdout);
// Shut down OpenAL
alDeleteSources(1, &source);
alDeleteBuffers(1, &buffer);
alcMakeContextCurrent(NULL);
alcCloseDevice(mainDev);
alcCaptureCloseDevice(captureDev);
}
~source() {
SET_CONTEXT(ctx_);
alSourcePlay(src_);
alDeleteSources(1, &src_);
}
void Audio::deleteSoundSource(const SoundSourceID source){
if (soundSources[source].sound != SOUND_NONE){
alDeleteSources(1, &soundSources[source].source);
soundSources[source].sound = SOUND_NONE;
}
}
void ALDevice::Stop(ALuint Source)
{
alSourceStop(Source);
alDeleteSources(1, &Source);
}
Sound::~Sound() {
alDeleteSources(1, &source);
}
AudioSource::~AudioSource()
{
alDeleteSources(1, &m_ID);
}
int main(int argc, char *argv[])
{
enum WaveType wavetype = WT_Sine;
ALuint source, buffer;
ALint last_pos, num_loops;
ALint max_loops = 4;
ALint srate = -1;
ALint tone_freq = 1000;
ALCint dev_rate;
ALenum state;
int i;
for(i = 1;i < argc;i++)
{
if(strcmp(argv[i], "-h") == 0 || strcmp(argv[i], "--help") == 0)
{
fprintf(stderr, "OpenAL Tone Generator\n"
"\n"
"Usage: %s <options>\n"
"\n"
"Available options:\n"
" --help/-h This help text\n"
" -t <seconds> Time to play a tone (default 5 seconds)\n"
" --waveform/-w <type> Waveform type: sine (default), square, sawtooth,\n"
" triangle, impulse\n"
" --freq/-f <hz> Tone frequency (default 1000 hz)\n"
" --srate/-s <sample rate> Sampling rate (default output rate)\n",
argv[0]
);
return 1;
}
else if(i+1 < argc && strcmp(argv[i], "-t") == 0)
{
i++;
max_loops = atoi(argv[i]) - 1;
}
else if(i+1 < argc && (strcmp(argv[i], "--waveform") == 0 || strcmp(argv[i], "-w") == 0))
{
i++;
if(strcmp(argv[i], "sine") == 0)
wavetype = WT_Sine;
else if(strcmp(argv[i], "square") == 0)
wavetype = WT_Square;
else if(strcmp(argv[i], "sawtooth") == 0)
wavetype = WT_Sawtooth;
else if(strcmp(argv[i], "triangle") == 0)
wavetype = WT_Triangle;
else if(strcmp(argv[i], "impulse") == 0)
wavetype = WT_Impulse;
else
fprintf(stderr, "Unhandled waveform: %s\n", argv[i]);
}
else if(i+1 < argc && (strcmp(argv[i], "--freq") == 0 || strcmp(argv[i], "-f") == 0))
{
i++;
tone_freq = atoi(argv[i]);
if(tone_freq < 1)
{
fprintf(stderr, "Invalid tone frequency: %s (min: 1hz)\n", argv[i]);
tone_freq = 1;
}
}
else if(i+1 < argc && (strcmp(argv[i], "--srate") == 0 || strcmp(argv[i], "-s") == 0))
{
i++;
srate = atoi(argv[i]);
if(srate < 40)
{
fprintf(stderr, "Invalid sample rate: %s (min: 40hz)\n", argv[i]);
srate = 40;
}
}
}
InitAL();
if(!alIsExtensionPresent("AL_EXT_FLOAT32"))
{
fprintf(stderr, "Required AL_EXT_FLOAT32 extension not supported on this device!\n");
CloseAL();
return 1;
}
{
ALCdevice *device = alcGetContextsDevice(alcGetCurrentContext());
alcGetIntegerv(device, ALC_FREQUENCY, 1, &dev_rate);
assert(alcGetError(device)==ALC_NO_ERROR && "Failed to get device sample rate");
}
if(srate < 0)
srate = dev_rate;
/* Load the sound into a buffer. */
buffer = CreateWave(wavetype, tone_freq, srate);
if(!buffer)
{
CloseAL();
return 1;
}
printf("Playing %dhz %s-wave tone with %dhz sample rate and %dhz output, for %d second%s...\n",
tone_freq, GetWaveTypeName(wavetype), srate, dev_rate, max_loops+1, max_loops?"s":"");
fflush(stdout);
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
alSourcei(source, AL_BUFFER, buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound for a while. */
num_loops = 0;
last_pos = 0;
alSourcei(source, AL_LOOPING, (max_loops > 0) ? AL_TRUE : AL_FALSE);
alSourcePlay(source);
do {
ALint pos;
al_nssleep(10000000);
alGetSourcei(source, AL_SAMPLE_OFFSET, &pos);
alGetSourcei(source, AL_SOURCE_STATE, &state);
if(pos < last_pos && state == AL_PLAYING)
{
++num_loops;
if(num_loops >= max_loops)
alSourcei(source, AL_LOOPING, AL_FALSE);
printf("%d...\n", max_loops - num_loops + 1);
fflush(stdout);
}
last_pos = pos;
} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
/* All done. Delete resources, and close OpenAL. */
alDeleteSources(1, &source);
alDeleteBuffers(1, &buffer);
/* Close up OpenAL. */
CloseAL();
return 0;
}
/* The start_voice should, surprise, start the voice. For streaming voices, it
should start polling the device and call _al_voice_update for audio data.
For non-streaming voices, it should resume playing from the last set
position */
static int _openal_start_voice(ALLEGRO_VOICE *voice)
{
ALLEGRO_AL_DATA *ex_data = voice->extra;
ALenum openal_err;
/* playing a sample instead of a stream */
if (!voice->is_streaming) {
alSourcePlay(ex_data->source);
if ((openal_err = alGetError()) != AL_NO_ERROR) {
ALLEGRO_ERROR("Could not start voice: %s\n",
openal_get_err_str(openal_err));
return 1;
}
ALLEGRO_INFO("Starting voice\n");
return 0;
}
{
ex_data->buffer_size = voice->buffer_size;
if (!ex_data->buffer_size) {
switch (ex_data->format) {
case AL_FORMAT_STEREO16:
ex_data->buffer_size = preferred_frag_size * 4;
break;
case AL_FORMAT_STEREO8:
case AL_FORMAT_MONO16:
ex_data->buffer_size = preferred_frag_size * 2;
break;
default:
ex_data->buffer_size = preferred_frag_size;
break;
}
}
ex_data->num_buffers = voice->num_buffers;
if (!ex_data->num_buffers)
ex_data->num_buffers = preferred_buf_count;
alGenSources(1, &ex_data->source);
if (alGetError() != AL_NO_ERROR)
return 1;
ex_data->buffers = al_malloc(sizeof(ALuint) * ex_data->num_buffers);
if (!ex_data->buffers) {
alSourcei(ex_data->source, AL_BUFFER, 0);
alDeleteSources(1, &ex_data->source);
return 1;
}
alGenBuffers(ex_data->num_buffers, ex_data->buffers);
if (alGetError() != AL_NO_ERROR) {
alSourcei(ex_data->source, AL_BUFFER, 0);
alDeleteSources(1, &ex_data->source);
al_free(ex_data->buffers);
ex_data->buffers = NULL;
return 1;
}
alSourcef(ex_data->source, AL_GAIN, 1.0f);
if (alGetError() != AL_NO_ERROR) {
alSourcei(ex_data->source, AL_BUFFER, 0);
alDeleteSources(1, &ex_data->source);
alDeleteBuffers(ex_data->num_buffers, ex_data->buffers);
al_free(ex_data->buffers);
ex_data->buffers = NULL;
return 1;
}
ex_data->stopped = false;
ex_data->thread = al_create_thread(_openal_update, (void *)voice);
al_start_thread(ex_data->thread);
}
ALLEGRO_INFO("Starting voice\n");
return 0;
}
virtual ~Sound(void)
{
stop();
alDeleteSources(1, &_source);
alDeleteBuffers(1, &_buffer);
}
AudioPlayer::~AudioPlayer() {
pause();
if(source)alDeleteSources(1, &source);
}
SoundSource::~SoundSource()
{
alDeleteSources(1, &mALSource);
SoundGeneral::checkAlError("Deleting sound source.");
}
static void _cleanup(music_context_t* context) {
alSourceStop(context->source);
alDeleteSources(1, &context->source);
alDeleteBuffers(NUM_BUFFERS, context->buffers);
ov_clear(&context->oggStream);
}
sound::~sound()
{
// Clean up buffers and sources
alDeleteSources(1, &m_uiSource);
alDeleteBuffers(1, &m_uiBuffer);
}
void Sound3D::KillALData()
{
//alDeleteBuffers(1, &Buffer);
alDeleteSources(1, &Source1);
}
LLAudioChannelOpenAL::~LLAudioChannelOpenAL()
{
cleanup();
alDeleteSources(1, &mALSource);
}
ActiveSound::~ActiveSound()
{
delete mOgg;
clear();
alDeleteSources(1,&mSource);
}
int main(int argc, char *argv[])
{
ALuint freq;
ALenum format;
ALvoid *data;
ALsizei i, size;
thread_id thread1, thread2;
status_t status;
/* listener parameters */
ALfloat listenerOrientation[] = { 0.0f, 0.0f, 1.0f, 0.0f, 1.0f, 0.0f };
ALfloat listenerPosition[] = { 0.0f, 0.0f, 0.0f };
ALfloat listenerVelocity[] = { 0.0f, 0.0f, 0.0f };
/* source parameters */
ALfloat sourcePosition[] = { 0.0f, 0.0f, 1.0f };
ALfloat sourceVelocity[] = { 0.0f, 0.0f, 0.0f };
ALfloat sourcePitch = 1.0f;
ALfloat sourceGain = 1.0f;
/* initialize */
print("Main: initialize");
alInit((ALint *) &argc, (ALubyte **) argv);
/* create context */
print("Main: create context");
context = alcCreateContext(22050, AL_FORMAT_STEREO16, 2048);
/* lock the context */
print("Main: make current");
alcMakeCurrent(context);
/* create buffers and sources */
if (alGenBuffers(kNumBuffers, buffer) != kNumBuffers)
quit("Can't create buffers");
if (alGenSources(kNumSources, source) != kNumSources)
quit("Can't create sources");
/* load buffers with data */
alutLoadWAV(kWaveFileName, &format, &data, &size, &freq);
for (i = 0; i < kNumBuffers; i++) {
alBufferData(buffer[i], format, data, size, freq);
}
free(data);
/* initialize listener */
alListenerfv(AL_POSITION, listenerPosition);
alListenerfv(AL_VELOCITY, listenerVelocity);
alListenerfv(AL_ORIENTATION, listenerOrientation);
/* initialize sources */
for (i = 0; i < kNumSources; i++) {
alSourcefv(source[i], AL_POSITION, sourcePosition);
alSourcefv(source[i], AL_VELOCITY, sourceVelocity);
alSourcef(source[i], AL_PITCH, sourcePitch);
alSourcef(source[i], AL_GAIN, sourceGain);
alSourcei(source[i], AL_BUFFER, buffer[i % kNumBuffers]);
alSourcei(source[i], AL_LOOPING, AL_TRUE);
}
/* start the sources */
print("Main: play");
for (i = 0; i < kNumSources; i++)
alSourcePlay(source[i]);
/* release the context */
print("Main: release current");
alcMakeCurrent(NULL);
/* spawn two threads */
print("Main: spawn thread 1");
thread1 = spawn_thread(threadFunc1, "thread 1", B_NORMAL_PRIORITY, NULL);
print("Main: spawn thread 2");
thread2 = spawn_thread(threadFunc2, "thread 2", B_NORMAL_PRIORITY, NULL);
/* resume the threads */
print("Main: resume thread 1");
resume_thread(thread1);
print("Main: resume thread 2");
resume_thread(thread2);
/* acquire context, snooze and release context */
print("Main: make current");
alcMakeCurrent(context);
print("Main: snooze...");
snooze(500000);
print("Main: release current");
alcMakeCurrent(NULL);
/* wait until the threads end */
print("Main: wait thread 1");
wait_for_thread(thread1, &status);
if (status != 0)
print("Main: thread 1 failed?");
print("Main: wait thread 2");
wait_for_thread(thread2, &status);
if (status != 0)
print("Main: thread 2 failed?");
/* acquire the context */
print("Main: make current");
alcMakeCurrent(context);
/* delete buffers and sources */
print("Main: delete sources");
alDeleteSources(kNumSources, source);
print("Main: delete buffers");
alDeleteBuffers(kNumBuffers, buffer);
/* release the context */
print("Main: release current");
alcMakeCurrent(NULL);
/* shutdown */
print("Main: delete context");
alcDeleteContext(context);
/* bye */
print("Main: bye");
alExit();
return 0;
}
/** Plays the audio data from the given file
* \param fileHandle,volume,onFinished,user_data see sound_PlayStream()
* \param streamBufferSize the size to use for the decoded audio buffers
* \param buffer_count the amount of audio buffers to use
* \see sound_PlayStream() for details about the rest of the function
* parameters and other details.
*/
AUDIO_STREAM* sound_PlayStreamWithBuf(PHYSFS_file* fileHandle, float volume, void (*onFinished)(void*), void* user_data, size_t streamBufferSize, unsigned int buffer_count)
{
#if !defined(WZ_NOSOUND)
AUDIO_STREAM* stream;
ALuint* buffers = alloca(sizeof(ALuint) * buffer_count);
ALint error;
unsigned int i;
if ( !openal_initialized )
{
debug(LOG_WARNING, "OpenAL isn't initialized, not creating an audio stream");
return NULL;
}
stream = malloc(sizeof(AUDIO_STREAM));
if (stream == NULL)
{
debug(LOG_FATAL, "sound_PlayStream: Out of memory");
abort();
return NULL;
}
// Clear error codes
alGetError();
// Retrieve an OpenAL sound source
alGenSources(1, &(stream->source));
error = sound_GetError();
if (error != AL_NO_ERROR)
{
// Failed to create OpenAL sound source, so bail out...
debug(LOG_SOUND, "alGenSources failed, most likely out of sound sources");
free(stream);
return NULL;
}
stream->fileHandle = fileHandle;
stream->decoder = sound_CreateOggVorbisDecoder(stream->fileHandle, false);
if (stream->decoder == NULL)
{
debug(LOG_ERROR, "sound_PlayStream: Failed to open audio file for decoding");
free(stream);
return NULL;
}
stream->volume = volume;
stream->bufferSize = streamBufferSize;
alSourcef(stream->source, AL_GAIN, stream->volume);
// HACK: this is a workaround for a bug in the 64bit implementation of OpenAL on GNU/Linux
// The AL_PITCH value really should be 1.0.
alSourcef(stream->source, AL_PITCH, 1.001f);
// Create some OpenAL buffers to store the decoded data in
alGenBuffers(buffer_count, buffers);
sound_GetError();
// Fill some buffers with audio data
for (i = 0; i < buffer_count; ++i)
{
// Decode some audio data
soundDataBuffer* soundBuffer = sound_DecodeOggVorbis(stream->decoder, stream->bufferSize);
// If we actually decoded some data
if (soundBuffer && soundBuffer->size > 0)
{
// Determine PCM data format
ALenum format = (soundBuffer->channelCount == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
// Copy the audio data into one of OpenAL's own buffers
alBufferData(buffers[i], format, soundBuffer->data, soundBuffer->size, soundBuffer->frequency);
sound_GetError();
// Clean up our memory
free(soundBuffer);
}
else
{
// If no data has been decoded we're probably at the end of our
// stream. So cleanup the excess stuff here.
// First remove the data buffer itself
free(soundBuffer);
// Then remove OpenAL's buffers
alDeleteBuffers(buffer_count - i, &buffers[i]);
sound_GetError();
break;
}
}
// Bail out if we didn't fill any buffers
if (i == 0)
{
debug(LOG_ERROR, "Failed to fill buffers with decoded audio data!");
// Destroy the decoder
sound_DestroyOggVorbisDecoder(stream->decoder);
// Destroy the OpenAL source
alDeleteSources(1, &stream->source);
// Free allocated memory
free(stream);
return NULL;
}
// Attach the OpenAL buffers to our OpenAL source
// (i = the amount of buffers we worked on in the above for-loop)
alSourceQueueBuffers(stream->source, i, buffers);
sound_GetError();
// Start playing the source
alSourcePlay(stream->source);
sound_GetError();
// Set callback info
stream->onFinished = onFinished;
stream->user_data = user_data;
// Prepend this stream to the linked list
stream->next = active_streams;
active_streams = stream;
return stream;
#else
return NULL;
#endif
}
void KillALData()
{
alDeleteBuffers(1, &Buffer_dungeonWAV);
alDeleteSources(1, &Source_dungeonWAV);
alutExit();
}
Source::~Source()
{
#ifdef ENABLE_OPENAL
alDeleteSources(1, &source);
#endif
};
void audio_SourceCommon_free(audio_SourceCommon *source) {
audio_SourceCommon_stop(source);
alDeleteSources(1, &source->source);
}
void sound_destroy_source(sound_source_t source) {
sound_stop(source);
alDeleteSources(1, &source);
}
int main (int argc, const char * argv[]) {
MyLoopPlayer player;
// convert to an OpenAL-friendly format and read into memory
CheckError(loadLoopIntoBuffer(&player),
"Couldn't load loop into buffer") ;
// set up OpenAL buffer
ALCdevice* alDevice = alcOpenDevice(NULL);
CheckALError ("Couldn't open AL device"); // default device
ALCcontext* alContext = alcCreateContext(alDevice, 0);
CheckALError ("Couldn't open AL context");
alcMakeContextCurrent (alContext);
CheckALError ("Couldn't make AL context current");
ALuint buffers[1];
alGenBuffers(1, buffers);
CheckALError ("Couldn't generate buffers");
alBufferData(*buffers,
AL_FORMAT_MONO16,
player.sampleBuffer,
player.bufferSizeBytes,
player.dataFormat.mSampleRate);
// AL copies the samples, so we can free them now
free(player.sampleBuffer);
// set up OpenAL source
alGenSources(1, player.sources);
CheckALError ("Couldn't generate sources");
alSourcei(player.sources[0], AL_LOOPING, AL_TRUE);
CheckALError ("Couldn't set source looping property");
alSourcef(player.sources[0], AL_GAIN, AL_MAX_GAIN);
CheckALError("Couldn't set source gain");
updateSourceLocation(player);
CheckALError ("Couldn't set initial source position");
// connect buffer to source
alSourcei(player.sources[0], AL_BUFFER, buffers[0]);
CheckALError ("Couldn't connect buffer to source");
// set up listener
alListener3f (AL_POSITION, 0.0, 0.0, 0.0);
CheckALError("Couldn't set listner position");
// ALfloat listenerOrientation[6]; // 3 vectors: forward x,y,z components, then up x,y,z
// listenerOrientation[2] = -1.0;
// listenerOrientation[0] = listenerOrientation [1] = 0.0;
// listenerOrientation[3] = listenerOrientation [4] = listenerOrientation[5] = 0.0;
// alListenerfv (AL_ORIENTATION, listenerOrientation);
// start playing
// alSourcePlayv (1, player.sources);
alSourcePlay(player.sources[0]);
CheckALError ("Couldn't play");
// and wait
printf("Playing...\n");
time_t startTime = time(NULL);
do
{
// get next theta
updateSourceLocation(player);
CheckALError ("Couldn't set looping source position");
CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.1, false);
} while (difftime(time(NULL), startTime) < RUN_TIME);
// cleanup:
alSourceStop(player.sources[0]);
alDeleteSources(1, player.sources);
alDeleteBuffers(1, buffers);
alcDestroyContext(alContext);
alcCloseDevice(alDevice);
printf ("Bottom of main\n");
}
void utox_audio_thread(void *args) {
ToxAV *av = args;
const char *device_list, *output_device = NULL;
void *audio_device = NULL;
_Bool call[MAX_CALLS] = {0}, preview = 0;
_Bool groups_audio[MAX_NUM_GROUPS] = {0};
int perframe = (UTOX_DEFAULT_FRAME_A * UTOX_DEFAULT_SAMPLE_RATE_A) / 1000;
uint8_t buf[perframe * 2 * UTOX_DEFAULT_AUDIO_CHANNELS]; //, dest[perframe * 2 * UTOX_DEFAULT_AUDIO_CHANNELS];
memset(buf, 0, sizeof(buf));
uint8_t audio_count = 0;
_Bool record_on = 0;
#ifdef AUDIO_FILTERING
debug("Audio Filtering");
#ifdef ALC_LOOPBACK_CAPTURE_SAMPLES
debug(" and Echo cancellation");
#endif
debug(" enabled in this build\n");
#endif
debug("frame size: %u\n", perframe);
device_list = alcGetString(NULL, ALC_CAPTURE_DEVICE_SPECIFIER);
if (device_list) {
audio_device = (void*)device_list;
debug("uTox audio input device list:\n");
while(*device_list) {
debug("\t%s\n", device_list);
postmessage(AUDIO_IN_DEVICE, UI_STRING_ID_INVALID, 0, (void*)device_list);
device_list += strlen(device_list) + 1;
}
}
postmessage(AUDIO_IN_DEVICE, STR_AUDIO_IN_NONE, 0, NULL);
audio_detect();
if (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT")) {
device_list = alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER);
} else {
device_list = alcGetString(NULL, ALC_DEVICE_SPECIFIER);
}
if(device_list) {
output_device = device_list;
debug("uTox audio output device list:\n");
while(*device_list) {
debug("\t%s\n", device_list);
postmessage(AUDIO_OUT_DEVICE, 0, 0, (void*)device_list);
device_list += strlen(device_list) + 1;
}
}
device_out = alcOpenDevice(output_device);
if(!device_out) {
debug("alcOpenDevice() failed\n");
return;
}
int attrlist[] = { ALC_FREQUENCY, UTOX_DEFAULT_SAMPLE_RATE_A,
ALC_INVALID
};
context = alcCreateContext(device_out, attrlist);
if(!alcMakeContextCurrent(context)) {
debug("alcMakeContextCurrent() failed\n");
alcCloseDevice(device_out);
return;
}
alGenSources(countof(source), source);
/* TODO hacky fix. This source list should be a VLA with a way to link sources to friends.
* NO SRSLY don't leave this like this! */
static ALuint ringSrc[UTOX_MAX_NUM_FRIENDS];
alGenSources(UTOX_MAX_NUM_FRIENDS, ringSrc);
/* Create buffer to store samples */
ALuint RingBuffer;
alGenBuffers(1, &RingBuffer);
{ /* wrapped to keep this data on the stack... I think... */
float frequency1 = 441.f;
float frequency2 = 882.f;
int seconds = 4;
unsigned sample_rate = 22050;
size_t buf_size = seconds * sample_rate * 2; //16 bit (2 bytes per sample)
int16_t *samples = malloc(buf_size * sizeof(int16_t));
if (!samples)
return;
/*Generate an electronic ringer sound that quickly alternates between two frequencies*/
int index = 0;
for(index = 0; index < buf_size; ++index) {
if ((index / (sample_rate)) % 4 < 2 ) {//4 second ring cycle, first 2 secondsring, the rest(2 seconds) is silence
if((index / 1000) % 2 == 1) {
samples[index] = 5000 * sin((2.0 * 3.1415926 * frequency1) / sample_rate * index); //5000=amplitude(volume level). It can be from zero to 32700
} else {
samples[index] = 5000 * sin((2.0 * 3.1415926 * frequency2) / sample_rate * index);
}
} else {
samples[index] = 0;
}
}
alBufferData(RingBuffer, AL_FORMAT_MONO16, samples, buf_size, sample_rate);
free(samples);
}
{
unsigned int i;
for (i = 0; i < UTOX_MAX_NUM_FRIENDS; ++i) {
alSourcei(ringSrc[i], AL_LOOPING, AL_TRUE);
alSourcei(ringSrc[i], AL_BUFFER, RingBuffer);
}
}
Filter_Audio *f_a = NULL;
audio_thread_init = 1;
int16_t *preview_buffer = NULL;
unsigned int preview_buffer_index = 0;
#define PREVIEW_BUFFER_SIZE (UTOX_DEFAULT_SAMPLE_RATE_A / 2)
while(1) {
if(audio_thread_msg) {
TOX_MSG *m = &audio_msg;
if(!m->msg) {
break;
}
switch(m->msg) {
case AUDIO_SET_INPUT: {
audio_device = m->data;
if(record_on) {
alccapturestop(device_in);
alccaptureclose(device_in);
}
if(audio_count) {
device_in = alcopencapture(audio_device);
if(!device_in) {
record_on = 0;
} else {
alccapturestart(device_in);
record_on = 1;
}
}
debug("set audio in\n");
break;
}
case AUDIO_SET_OUTPUT: {
output_device = m->data;
ALCdevice *device = alcOpenDevice(output_device);
if(!device) {
debug("alcOpenDevice() failed\n");
break;
}
ALCcontext *con = alcCreateContext(device, NULL);
if(!alcMakeContextCurrent(con)) {
debug("alcMakeContextCurrent() failed\n");
alcCloseDevice(device);
break;
}
alcDestroyContext(context);
alcCloseDevice(device_out);
context = con;
device_out = device;
alGenSources(countof(source), source);
alGenSources(MAX_CALLS, ringSrc);
Tox *tox = toxav_get_tox(av);
uint32_t num_chats = tox_count_chatlist(tox);
if (num_chats != 0) {
int32_t chats[num_chats];
uint32_t max = tox_get_chatlist(tox, chats, num_chats);
unsigned int i;
for (i = 0; i < max; ++i) {
if (tox_group_get_type(tox, chats[i]) == TOX_GROUPCHAT_TYPE_AV) {
GROUPCHAT *g = &group[chats[i]];
alGenSources(g->peers, g->source);
}
}
}
debug("set audio out\n");
break;
}
case AUDIO_PREVIEW_START: {
preview = 1;
audio_count++;
preview_buffer = calloc(PREVIEW_BUFFER_SIZE, 2);
preview_buffer_index = 0;
if(!record_on) {
device_in = alcopencapture(audio_device);
if(device_in) {
alccapturestart(device_in);
record_on = 1;
debug("Starting Audio Preview\n");
}
}
break;
}
case AUDIO_START: {
audio_count++;
if(!record_on) {
device_in = alcopencapture(audio_device);
if(device_in) {
alccapturestart(device_in);
record_on = 1;
debug("Listening to audio\n");
yieldcpu(20);
}
}
break;
}
case GROUP_AUDIO_CALL_START: {
break; // TODO, new groups API
audio_count++;
groups_audio[m->param1] = 1;
if(!record_on) {
device_in = alcopencapture(audio_device);
if(device_in) {
alccapturestart(device_in);
record_on = 1;
debug("Starting Audio GroupCall\n");
}
}
break;
}
case AUDIO_PREVIEW_END: {
preview = 0;
audio_count--;
free(preview_buffer);
preview_buffer = NULL;
if(!audio_count && record_on) {
alccapturestop(device_in);
alccaptureclose(device_in);
record_on = 0;
debug("Audio Preview Stopped\n");
}
break;
}
case AUDIO_END: {
if(!call[m->param1]) {
break;
}
call[m->param1] = 0;
audio_count--;
if(!audio_count && record_on) {
alccapturestop(device_in);
alccaptureclose(device_in);
record_on = 0;
debug("stop\n");
}
break;
}
case GROUP_AUDIO_CALL_END: {
break; // TODO, new groups API
if(!groups_audio[m->param1]) {
break;
}
audio_count--;
groups_audio[m->param1] = 0;
if(!audio_count && record_on) {
alccapturestop(device_in);
alccaptureclose(device_in);
record_on = 0;
debug("stop\n");
}
break;
}
case AUDIO_PLAY_RINGTONE: {
if(!audible_notifications_enabled) {
break;
}
alSourcePlay(ringSrc[m->param1]);
break;
}
case AUDIO_STOP_RINGTONE: {
ALint state;
alGetSourcei(ringSrc[m->param1], AL_SOURCE_STATE, &state);
if(state == AL_PLAYING) {
alSourceStop(ringSrc[m->param1]);
}
break;
}
}
audio_thread_msg = 0;
}
// TODO move this code to filter_audio.c
#ifdef AUDIO_FILTERING
if (!f_a && audio_filtering_enabled) {
f_a = new_filter_audio(UTOX_DEFAULT_SAMPLE_RATE_A);
if (!f_a) {
audio_filtering_enabled = 0;
debug("filter audio failed\n");
} else {
debug("filter audio on\n");
}
} else if (f_a && !audio_filtering_enabled) {
kill_filter_audio(f_a);
f_a = NULL;
debug("filter audio off\n");
}
#else
if (audio_filtering_enabled) {
audio_filtering_enabled = 0;
}
#endif
_Bool sleep = 1;
if(record_on) {
ALint samples;
_Bool frame = 0;
/* If we have a device_in we're on linux so we can just call OpenAL, otherwise we're on something else so
* we'll need to call audio_frame() to add to the buffer for us. */
if (device_in == (void*)1) {
frame = audio_frame((void*)buf);
if (frame) {
/* We have an audio frame to use, continue without sleeping. */
sleep = 0;
}
} else {
alcGetIntegerv(device_in, ALC_CAPTURE_SAMPLES, sizeof(samples), &samples);
if(samples >= perframe) {
alcCaptureSamples(device_in, buf, perframe);
frame = 1;
if (samples >= perframe * 2) {
sleep = 0;
}
}
}
#ifdef AUDIO_FILTERING
#ifdef ALC_LOOPBACK_CAPTURE_SAMPLES
if (f_a && audio_filtering_enabled) {
alcGetIntegerv(device_out, ALC_LOOPBACK_CAPTURE_SAMPLES, sizeof(samples), &samples);
if(samples >= perframe) {
int16_t buffer[perframe];
alcCaptureSamplesLoopback(device_out, buffer, perframe);
pass_audio_output(f_a, buffer, perframe);
set_echo_delay_ms(f_a, UTOX_DEFAULT_FRAME_A);
if (samples >= perframe * 2) {
sleep = 0;
}
}
}
#endif
#endif
if (frame) {
_Bool voice = 1;
#ifdef AUDIO_FILTERING
if (f_a) {
int ret = filter_audio(f_a, (int16_t*)buf, perframe);
if (ret == -1) {
debug("filter audio error\n");
}
if (ret == 0) {
voice = 0;
}
}
#endif
/* If push to talk, we don't have to do anything */
if (!check_ptt_key()) {
voice = 0; //PTT is up, send nothing.
}
if (preview) {
if (preview_buffer_index + perframe > PREVIEW_BUFFER_SIZE) {
preview_buffer_index = 0;
}
sourceplaybuffer(0, preview_buffer + preview_buffer_index, perframe, UTOX_DEFAULT_AUDIO_CHANNELS, UTOX_DEFAULT_SAMPLE_RATE_A);
if (voice) {
memcpy(preview_buffer + preview_buffer_index, buf, perframe * sizeof(int16_t));
} else {
memset(preview_buffer + preview_buffer_index, 0, perframe * sizeof(int16_t));
}
preview_buffer_index += perframe;
}
if (voice) {
int i, active_call_count = 0;
for(i = 0; i < UTOX_MAX_NUM_FRIENDS; i++) {
if( UTOX_SEND_AUDIO(i) ) {
active_call_count++;
TOXAV_ERR_SEND_FRAME error = 0;
toxav_audio_send_frame(av, friend[i].number, (const int16_t *)buf, perframe, UTOX_DEFAULT_AUDIO_CHANNELS, UTOX_DEFAULT_SAMPLE_RATE_A, &error);
if (error) {
debug("toxav_send_audio error friend == %i, error == %i\n", i, error);
} else {
// debug("Send a frame to friend %i\n",i);
if (active_call_count >= UTOX_MAX_CALLS) {
debug("We're calling more peers than allowed by UTOX_MAX_CALLS, This is a bug\n");
break;
}
}
}
}
// TODO REMOVED until new groups api can be implemented.
/*Tox *tox = toxav_get_tox(av);
uint32_t num_chats = tox_count_chatlist(tox);
if (num_chats != 0) {
int32_t chats[num_chats];
uint32_t max = tox_get_chatlist(tox, chats, num_chats);
for (i = 0; i < max; ++i) {
if (groups_audio[chats[i]]) {
toxav_group_send_audio(tox, chats[i], (int16_t *)buf, perframe, UTOX_DEFAULT_AUDIO_CHANNELS, UTOX_DEFAULT_SAMPLE_RATE_A);
}
}
}*/
}
}
}
if (sleep) {
yieldcpu(5);
}
}
utox_filter_audio_kill(f_a);
//missing some cleanup ?
alDeleteSources(MAX_CALLS, ringSrc);
alDeleteSources(countof(source), source);
alDeleteBuffers(1, &RingBuffer);
if(device_in) {
if(record_on) {
alcCaptureStop(device_in);
}
alcCaptureCloseDevice(device_in);
}
alcMakeContextCurrent(NULL);
alcDestroyContext(context);
alcCloseDevice(device_out);
audio_thread_msg = 0;
audio_thread_init = 0;
debug("UTOX AUDIO:\tClean thread exit!\n");
}
