void SawtoothSoundComponent_HW::init(){
m_FrequencyIn_1_Port->registerCallback(ICallbackPtr(new OnFrequencyChange(*this)));
/* initialize reconos */
if(slot.isValid()){
// TODO: Warum werden hier zwei message boxen benötigt!?
/* initialize message boxes with 1 data word */
mbox_init(&m_CtrlStart, 1);
mbox_init(&m_CtrlStop, 1);
m_HWTParams[0] = (uint32_t) m_LocalBuffer;
m_HWTParams[1] = 0;
m_HWTParams[2] = (uint32_t) (getPhaseIncrement(440) * SOUNDGATES_FIXED_PT_SCALE);
m_ReconOSResource[0].type = RECONOS_TYPE_MBOX;
m_ReconOSResource[0].ptr = &m_CtrlStart;
m_ReconOSResource[1].type = RECONOS_TYPE_MBOX;
m_ReconOSResource[1].ptr = &m_CtrlStop;
reconos_hwt_setresources(&m_ReconOSThread, &m_ReconOSResource[0], 2);
reconos_hwt_setinitdata(&m_ReconOSThread, (void *) &m_HWTParams[0]);
reconos_hwt_create(&m_ReconOSThread, slot.getSlot(), NULL);
m_SoundOut_1_Port->clearWriteBuffer();
}
}
void SawtoothSoundComponent::init()
{
m_SoundOut_1_Port->init();
m_FrequencyIn_1_Port->registerCallback(
ICallbackPtr(new OnFrequencyChange(*this)));
}
void TemplateSoundComponent::init()
{
// You can init() sound output ports to clear their buffers
m_TemplateSoundOut_2_Port->init();
// Control values can trigger a callback function when they change.
m_TemplateControlIn_1_Port->registerCallback(ICallbackPtr(new OnChange(*this)));
}
void LoggingComponent::init()
{
// Control values can trigger a callback function when they change.
m_Trigger_2_Port->registerCallback(ICallbackPtr(new OnChange(*this)));
}
void MulCCComponent::init(void)
{
m_ControlIn_1_Port->registerCallback(ICallbackPtr(new OnChange1(*this)));
m_ControlIn_2_Port->registerCallback(ICallbackPtr(new OnChange2(*this)));
}
void SamplePlayer_SW::init() {
AVPacket avpkt;
ReSampleContext* resmplCtx;
int compressed_bytes_consumed = 0;
int compressed_file_size = 0;
uint8_t* l_ResampleBuffer;
initCodec(); /* Initialize audio codec */
av_init_packet(&avpkt); /* Initialize packet */
// resmplCtx = av_audio_resample_init(m_CodecCtx->channels,
// m_CodecCtx->channels,
// Synthesizer::config::samplerate,
// m_CodecCtx->sample_rate,
// AV_SAMPLE_FMT_S32,
// m_CodecCtx->sample_fmt,
// 16, 10, 0, 1.0);
m_DecompressedDataBuffer = new uint8_t[AVCODEC_MAX_AUDIO_FRAME_SIZE * MAX_PLAYBACK_LENGTH_SEC];
memset(m_DecompressedDataBuffer, 0 ,AVCODEC_MAX_AUDIO_FRAME_SIZE * MAX_PLAYBACK_LENGTH_SEC);
LOG_DEBUG("Decoding audio file");
LOG_DEBUG("Sample format:" << av_get_sample_fmt_name(m_CodecCtx->sample_fmt));
while (!av_read_frame(m_ContainerCtx, &avpkt)) {
int got_frame = 0;
if (!m_Decoded_Frame) {
if (!(m_Decoded_Frame = avcodec_alloc_frame())) { /* allocate once */
LOG_ERROR("Could not allocate audio frame");
exit(1);
}
}else{
avcodec_get_frame_defaults(m_Decoded_Frame);
}
compressed_bytes_consumed = avcodec_decode_audio4(m_CodecCtx, m_Decoded_Frame, &got_frame, &avpkt);
if (compressed_bytes_consumed < 0) {
LOG_ERROR("Error while decoding audio frame");
exit(1);
}
if (got_frame) {
int samples_written = 0;
int data_size = 0;
size_t resampled_Bytes = 0;
/* if a frame has been decoded, output it */
data_size = av_samples_get_buffer_size(NULL, m_CodecCtx->channels, m_Decoded_Frame->nb_samples, m_CodecCtx->sample_fmt, 1);
int32_t* buf = (int32_t*) (m_DecompressedDataBuffer + m_DecompressedDataSize);
switch(m_CodecCtx->sample_fmt){
case AV_SAMPLE_FMT_NONE:
case AV_SAMPLE_FMT_U8:
// for(int i = 0; i < m_Decoded_Frame->nb_samples * m_CodecCtx->channels; i++){
// buf[i] = (int32_t)( ((int8_t*) m_Decoded_Frame->data[0])[i] << 24 );
// }
// break;
//FIXME: convert other sample formats
case AV_SAMPLE_FMT_S16:
for(int i = 0; i < m_Decoded_Frame->nb_samples * m_CodecCtx->channels; i++){
buf[i] = *(((short*)m_Decoded_Frame->data[0]) + i) << 16;
}
m_DecompressedDataSize += m_Decoded_Frame->nb_samples * m_CodecCtx->channels * av_get_bytes_per_sample(AV_SAMPLE_FMT_S32);
break;
case AV_SAMPLE_FMT_S32:
memcpy(m_DecompressedDataBuffer + m_DecompressedDataSize, m_Decoded_Frame->data[0], data_size);
m_DecompressedDataSize += data_size;
break;
case AV_SAMPLE_FMT_FLT:
case AV_SAMPLE_FMT_DBL:
case AV_SAMPLE_FMT_U8P:
case AV_SAMPLE_FMT_S16P:
case AV_SAMPLE_FMT_S32P:
case AV_SAMPLE_FMT_FLTP:
case AV_SAMPLE_FMT_DBLP:
default:
LOG_ERROR("Sample format is not supported");
}
}
}
char* writeLeftPortPtr = m_SoundOut_Left_1_Port->getWriteBuffer();
// char* writeRightPortPtr = m_SoundOut_Right_2_Port->getWriteBuffer();
memset(writeLeftPortPtr, 0, Synthesizer::config::bytesPerBlock);
// memset(writeRightPortPtr, 0, Synthesizer::config::bytesPerBlock);
m_Trigger_1_Port->registerCallback(ICallbackPtr(new OnValueChange<int, ControlPortPtr>(m_DoPlayback, m_Trigger_1_Port)));
// audio_resample_close(resmplCtx);
// av_free(m_CodecCtx);
// av_free(m_Decoded_Frame);
// avformat_free_context(m_ContainerCtx);
}
void SubSCComponent::init(void)
{
m_ControlIn_2_Port->registerCallback(ICallbackPtr(new OnChange(*this)));
}
