diff --git a/CHANGELOG b/CHANGELOG
index f00b6a5a085a4d6033936353c41775eae42cf6a0..1f233a41fc90a6abe52053772c862a4f2a14694d 100644
--- a/CHANGELOG
+++ b/CHANGELOG
@@ -6,7 +6,12 @@ TODO::
LAST DONE:
CHANGES:
- - 2021.04.21 - Added ALSA tests in main.cpp and Launch configuration for Linux
+
+- ver. 0.20.005 - <b>2021.04.21</b> Changed:
+ - Added ALSA tests in main.cpp and Launch configuration for Linux
+ - Added to DSP_types.h a SOUND_object_t abstract class, a wrapper for different sound cards' API
+ - Added WMM_object_t abstract class, a wrapper for Windows MultiMedia API
+ - DSP::u::AudioOutput now uses SOUND_object_t (currently just WMM_object_t) for soundcard API
- ver. 0.20.004 - <b>2021.04.06</b> Changed:
- Fixed makefiles for Linux - install and examples compilation works
diff --git a/src/Main.cpp b/src/Main.cpp
index c6b39a83f6bc2c60d9101099257c031a30481e3a..d18918ac15e631b58c7d3a805e5ec0936d86714b 100644
--- a/src/Main.cpp
+++ b/src/Main.cpp
@@ -2909,7 +2909,7 @@ polling:
int test_ALSA() {
ALSA_object_t ALSA_object;
- ALSA_object.log_alsa_data();
+ ALSA_object.log_driver_data();
unsigned int Fs = 44100;
int rs = ALSA_object.open_alsa_device(SND_PCM_STREAM_PLAYBACK, 2, Fs);
diff --git a/src/Makefile.win b/src/Makefile.win
index 7333ccfdf5b66f39e1c834cea3b58ae972a37efb..295ad8175c528e1e3693b860bf2458f9215f1ec5 100644
--- a/src/Makefile.win
+++ b/src/Makefile.win
@@ -32,7 +32,7 @@ LINKER_FLAGS_release = $(comflag) -s -static-libgcc -static-libstdc++ -static
SOURCES_NAMES =
SOURCES_NAMES += DSP_AudioMixer.cpp DSP_Fourier.cpp DSP_misc.cpp DSP_clocks.cpp DSP_modules.cpp DSP_modules2.cpp
-SOURCES_NAMES += DSP_DOT.cpp DSP_modules_misc.cpp DSP_IO.cpp DSP_logstream.cpp
+SOURCES_NAMES += DSP_DOT.cpp DSP_modules_misc.cpp DSP_IO.cpp DSP_logstream.cpp WMM_support.cpp
SOURCES = $(addprefix $(SRC_CPP_SUBDIR)/,$(SOURCES_NAMES))
SOURCES_DBG =
diff --git a/src/cpp/ALSA_support.cpp b/src/cpp/ALSA_support.cpp
index 92be453cb781b76855493562edda4bcdb793b689..0632be7a8b7c8524e0915054e3ef21794d75cf74 100644
--- a/src/cpp/ALSA_support.cpp
+++ b/src/cpp/ALSA_support.cpp
@@ -24,7 +24,7 @@ ALSA_object_t::~ALSA_object_t()
}
-void ALSA_object_t::log_alsa_data() {
+void ALSA_object_t::log_driver_data() {
int val;
DSP::log << "ALSA library version: " << SND_LIB_VERSION_STR << endl;
diff --git a/src/cpp/DSP_IO.cpp b/src/cpp/DSP_IO.cpp
index 3a40d1a2f78e2db9deb0a9a693198419d08ce914..a356b67c064d05481daac179906fbf850b8abcf2 100644
--- a/src/cpp/DSP_IO.cpp
+++ b/src/cpp/DSP_IO.cpp
@@ -3607,56 +3607,6 @@ uint32_t DSP::f::GetAudioBufferSize(const unsigned long &SamplingFreq, const DSP
}
#ifdef WINMMAPI
- void CALLBACK DSP::u::AudioOutput::waveOutProc(HWAVEOUT hwo, UINT uMsg,
- uint32_t dwInstance, uint32_t dwParam1, uint32_t dwParam2)
- {
- UNUSED_ARGUMENT(hwo);
- UNUSED_ARGUMENT(uMsg);
- UNUSED_ARGUMENT(dwInstance);
- UNUSED_ARGUMENT(dwParam1);
- UNUSED_ARGUMENT(dwParam2);
- #ifdef __DEBUG__
- #ifdef AUDIO_DEBUG_MESSAGES_ON
- // MMRESULT result;
- DSP::u::AudioOutput *Current;
- bool AllDone;
- int ind;
- string tekst;
-
- Current = AudioObjects[dwInstance];
-
- switch (uMsg)
- {
- case WOM_OPEN:
- DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second
- << "WOM_OPEN(" << (int)dwInstance << ")" << endl;
- break;
- case WOM_CLOSE:
- DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second
- << "WOM_CLOSE(" << (int)dwInstance << ")" << endl;
- break;
- case WOM_DONE:
- DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second
- << "WOM_DONE(" << (int)dwInstance << ")" << endl;
-
- if (Current->StopPlaying)
- {
- DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second << "StopPlaying is set" << endl;
- return;
- }
- else
- {
- AllDone=true;
- for (ind=0; ind < DSP_NoOfAudioOutputBuffers; ind++)
- AllDone &= (Current->waveHeaderOut[ind].dwFlags & WHDR_DONE);
- if (AllDone)
- DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second << "All buffers had been used - nothing to play" << endl;
- }
- break;
- }
- #endif
- #endif
- }
//! \bug allow user to select number of internal buffers
void CALLBACK DSP::u::AudioInput::waveInProc_short(HWAVEIN hwi, UINT uMsg,
@@ -3949,14 +3899,14 @@ void DSP::u::AudioOutput::Init(unsigned long SamplingFreq,
unsigned int WaveOutDevNo)
{
#ifdef WINMMAPI
- MMRESULT result;
- DWORD_PTR Callback;
-
- //Rezerwacja pamiÄci dla formatu WAVE
- // WAVEFORMATEX wfx; //to wymaga korekty
- PCMWAVEFORMAT wfx;
-
- unsigned long ind;
+ // MMRESULT result;
+ // DWORD_PTR Callback;
+ //
+ // //Rezerwacja pamiÄci dla formatu WAVE
+ // // WAVEFORMATEX wfx; //to wymaga korekty
+ // PCMWAVEFORMAT wfx;
+ //
+ // unsigned long ind;
#else
UNUSED_ARGUMENT(WaveOutDevNo);
#endif
@@ -3989,10 +3939,10 @@ void DSP::u::AudioOutput::Init(unsigned long SamplingFreq,
ClockGroups.AddInput2Group("input", Input(temp));
}
- #ifdef WINMMAPI
- //! \bug in Debug mode this callback does nothing so it would be better just not use it
- Callback = (DWORD_PTR)(&DSP::u::AudioOutput::waveOutProc);
- #endif
+// #ifdef WINMMAPI
+// //! \bug in Debug mode this callback does nothing so it would be better just not use it
+// Callback = (DWORD_PTR)(&DSP::u::AudioOutput::waveOutProc);
+// #endif
Current_CallbackInstance=Next_CallbackInstance;
Next_CallbackInstance++;
AudioObjects.resize(Current_CallbackInstance+1);
@@ -4000,96 +3950,28 @@ void DSP::u::AudioOutput::Init(unsigned long SamplingFreq,
StopPlaying = false;
- switch (BitPrec)
- {
- case 8:
- OutSampleType=DSP::e::SampleType::ST_uchar;
- break;
- case 16:
- OutSampleType=DSP::e::SampleType::ST_short;
- break;
- default:
- OutSampleType=DSP::e::SampleType::ST_short;
- BitPrec=16;
- break;
- }
-
audio_outbuffer_size = DSP::f::GetAudioBufferSize(SamplingFreq, DSP::e::AudioBufferType::output);
- #ifdef WINMMAPI
- //Wypeniamy struktur wfx
- wfx.wf.wFormatTag=WAVE_FORMAT_PCM;
- wfx.wf.nChannels=(uint16_t)NoOfInputs;
- wfx.wf.nSamplesPerSec=(UINT)SamplingFreq;
- wfx.wBitsPerSample=BitPrec;
- wfx.wf.nAvgBytesPerSec=wfx.wf.nSamplesPerSec*(wfx.wBitsPerSample/8);
- wfx.wf.nBlockAlign=(uint16_t)(wfx.wf.nChannels*(wfx.wBitsPerSample/8));
+ snd_object.select_device_by_number(WaveOutDevNo); // use default device
+ snd_object.open_PCM_device_4_output(NoOfInputs, BitPrec, SamplingFreq, audio_outbuffer_size);
- if (WaveOutDevNo >= (UINT)waveOutGetNumDevs())
- result=waveOutOpen(&hWaveOut,
- WAVE_MAPPER, //&DeviceID,
- (WAVEFORMATEX *)(&wfx),
- Callback,
- Current_CallbackInstance, //CallbackInstance,
- CALLBACK_FUNCTION | WAVE_ALLOWSYNC | WAVE_FORMAT_DIRECT //| WAVE_MAPPED //CALLBACK_NULL
- );
- else
- result=waveOutOpen(&hWaveOut,
- WaveOutDevNo, //&DeviceID,
- (WAVEFORMATEX *)(&wfx),
- Callback,
- Current_CallbackInstance, //CallbackInstance,
- CALLBACK_FUNCTION | WAVE_ALLOWSYNC | WAVE_FORMAT_DIRECT //| WAVE_MAPPED //CALLBACK_NULL
- );
- if (DSP::f::AudioCheckError(result) == false)
- { // everything is ok
- waveHeaderOut.resize(DSP::NoOfAudioOutputBuffers);
- WaveOutBufferLen=wfx.wf.nBlockAlign*audio_outbuffer_size;
- WaveOutBuffers.resize(DSP::NoOfAudioOutputBuffers);
- for (ind=0; ind< DSP::NoOfAudioOutputBuffers; ind++)
- {
- WaveOutBuffers[ind].clear();
- WaveOutBuffers[ind].resize(WaveOutBufferLen, 0);
- }
+ OutBufferLen=NoOfInputs*audio_outbuffer_size;
+ OutBuffer.clear(); OutBuffer.resize(OutBufferLen, 0.0);
+ BufferIndex=0;
- OutBufferLen=NoOfInputs*audio_outbuffer_size;
- OutBuffer.clear(); OutBuffer.resize(OutBufferLen, 0.0);
- BufferIndex=0;
+ // #ifdef WINMMAPI
- for (ind=0; ind< DSP::NoOfAudioOutputBuffers; ind++)
- {
- waveHeaderOut[ind].lpData=(char *)(WaveOutBuffers[ind].data());
- waveHeaderOut[ind].dwBufferLength=OutBufferLen*(BitPrec/8); //sizeof(short);
- waveHeaderOut[ind].dwFlags= 0; // WHDR_BEGINLOOP | WHDR_ENDLOOP;
- waveHeaderOut[ind].dwLoops=0;
-
- result=waveOutPrepareHeader(hWaveOut,
- &(waveHeaderOut[ind]), sizeof(WAVEHDR));
- DSP::f::AudioCheckError(result);
- waveHeaderOut[ind].dwFlags= WHDR_DONE; // WHDR_BEGINLOOP | WHDR_ENDLOOP;
- }
- }
- else
- { //error while creating audio output
- waveHeaderOut.clear();
- WaveOutBuffers.clear();
- WaveOutBufferLen = 0;
- OutBuffer.clear();
- OutBufferLen = 0;
- BufferIndex=0;
- }
+ // #else
- #else
+ // WaveOutBufferLen = 0;
+ // WaveOutBuffers.clear();
+ // OutBuffer.clear(); OutBufferLen = 0;
+ // BufferIndex=0;
- WaveOutBufferLen = 0;
- WaveOutBuffers.clear();
- OutBuffer.clear(); OutBufferLen = 0;
- BufferIndex=0;
+ // #endif // WINMMAPI
- #endif // WINMMAPI
-
- IsPlayingNow = false;
- NextBufferInd=0;
+ // IsPlayingNow = false;
+ //NextBufferInd=0;
Execute_ptr = &InputExecute;
}
@@ -4294,50 +4176,52 @@ void DSP::u::AudioInput::Init(DSP::Clock_ptr ParentClock,
DSP::u::AudioOutput::~AudioOutput()
{
- #ifdef WINMMAPI
- MMRESULT result;
- unsigned long ind;
- #endif
+ // #ifdef WINMMAPI
+ // MMRESULT result;
+ // unsigned long ind;
+ // #endif
if (OutBufferLen != 0)
{ // if device was opened successfully
StopPlaying=true;
- #ifdef WINMMAPI
- result = waveOutReset(hWaveOut);
- DSP::f::AudioCheckError(result);
- for (ind=0; ind< DSP::NoOfAudioOutputBuffers; ind++)
- {
- result=waveOutUnprepareHeader(hWaveOut,
- &(waveHeaderOut[ind]), sizeof(WAVEHDR));
- DSP::f::AudioCheckError(result);
- }
-
- #ifdef AUDIO_DEBUG_MESSAGES_ON
- DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Closing DSP::u::AudioOutput" << endl;
- #endif
- result=waveOutClose(hWaveOut);
- while (result==WAVERR_STILLPLAYING)
- {
- // #ifdef WINBASEAPI
- DSP::f::Sleep(100);
- // #else
- // sleep(100);
- // #endif
- #ifdef AUDIO_DEBUG_MESSAGES_ON
- DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Closing DSP::u::AudioOutput" << endl;
- #endif
- result=waveOutClose(hWaveOut);
- }
- DSP::f::AudioCheckError(result);
- #endif
-
-
- // 2) Free buffers
- WaveOutBuffers.clear();
- #ifdef WINMMAPI
- waveHeaderOut.clear();
- #endif
+ snd_object.close_PCM_device();
+
+ // #ifdef WINMMAPI
+ // result = waveOutReset(hWaveOut);
+ // DSP::f::AudioCheckError(result);
+ // for (ind=0; ind< DSP::NoOfAudioOutputBuffers; ind++)
+ // {
+ // result=waveOutUnprepareHeader(hWaveOut,
+ // &(waveHeaderOut[ind]), sizeof(WAVEHDR));
+ // DSP::f::AudioCheckError(result);
+ // }
+
+ // #ifdef AUDIO_DEBUG_MESSAGES_ON
+ // DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Closing DSP::u::AudioOutput" << endl;
+ // #endif
+ // result=waveOutClose(hWaveOut);
+ // while (result==WAVERR_STILLPLAYING)
+ // {
+ // // #ifdef WINBASEAPI
+ // DSP::f::Sleep(100);
+ // // #else
+ // // sleep(100);
+ // // #endif
+ // #ifdef AUDIO_DEBUG_MESSAGES_ON
+ // DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Closing DSP::u::AudioOutput" << endl;
+ // #endif
+ // result=waveOutClose(hWaveOut);
+ // }
+ // DSP::f::AudioCheckError(result);
+ // #endif
+
+
+ // // 2) Free buffers
+ // WaveOutBuffers.clear();
+ // #ifdef WINMMAPI
+ // waveHeaderOut.clear();
+ // #endif
OutBuffer.clear();
}
@@ -4416,114 +4300,116 @@ DSP::u::AudioInput::~AudioInput()
void DSP::u::AudioOutput::FlushBuffer(void)
{
- #ifdef WINMMAPI
- MMRESULT result;
- uint8_t *temp8;
- short *temp16;
- DSP::Float_ptr Sample;
- short Znak;
- uint32_t ind;
-
- // ************************************************** //
- // Send buffer to the audio device
-
- #ifdef AUDIO_DEBUG_MESSAGES_ON
- DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Flushing output buffer" << endl;
- #endif
-
- while (1)
- {
- if (waveHeaderOut[NextBufferInd].dwFlags & WHDR_DONE)
- {
- result=waveOutUnprepareHeader(hWaveOut,
- &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
- DSP::f::AudioCheckError(result);
-
- Sample=OutBuffer.data();
- // ************************************************** //
- // Converts samples format to the one suitable for the audio device
- switch (OutSampleType)
- {
- case DSP::e::SampleType::ST_uchar:
- temp8=(uint8_t *)(WaveOutBuffers[NextBufferInd].data());
- for (ind=0; ind<OutBufferLen; ind++)
- {
- if (*Sample < 0)
- Znak=-1;
- else
- Znak=1;
-
- *Sample*=127;
- if ((*Sample)*Znak > 127)
- *temp8=(unsigned char)(128+Znak*127);
- else
- *temp8=(unsigned char)(128+*Sample+Znak*0.5);
-
- Sample++;
- temp8++;
- }
- break;
- case DSP::e::SampleType::ST_short:
- temp16=(short *)(WaveOutBuffers[NextBufferInd].data());
- for (ind=0; ind<OutBufferLen; ind++)
- {
- if (*Sample < 0)
- Znak=-1;
- else
- Znak=1;
-
- *Sample*=SHRT_MAX;
- if ((*Sample)*Znak > SHRT_MAX)
- *temp16=(short)(Znak*SHRT_MAX);
- else
- *temp16=(short)(*Sample+Znak*0.5);
- Sample++;
- temp16++;
- }
- break;
- default:
- break;
- }
-
- result=waveOutPrepareHeader(hWaveOut,
- &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
- DSP::f::AudioCheckError(result);
-
- if (IsPlayingNow == false)
- {
- if (NextBufferInd == 1)
- {
- for (ind=0; ind < DSP::NoOfAudioOutputBuffers-1; ind++) //one spare buffer
- {
- result=waveOutWrite(hWaveOut,
- &(waveHeaderOut[ind]), sizeof(WAVEHDR));
- DSP::f::AudioCheckError(result);
- }
- IsPlayingNow = true;
- }
-
- }
- else
- {
- result=waveOutWrite(hWaveOut,
- &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
- DSP::f::AudioCheckError(result);
- }
- NextBufferInd++;
- NextBufferInd %= DSP::NoOfAudioOutputBuffers;
-
- break;
- }
- else
- {
- // Sleep(10);
- #ifdef AUDIO_DEBUG_MESSAGES_ON
- DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Waiting for free output buffer" << endl;
- #endif
- DSP::f::Sleep(0);
- }
- }
- #endif // WINMMAPI
+ snd_object.append_playback_buffer(OutBuffer);
+
+ // #ifdef WINMMAPI
+ // MMRESULT result;
+ // uint8_t *temp8;
+ // short *temp16;
+ // DSP::Float_ptr Sample;
+ // short Znak;
+ // uint32_t ind;
+
+ // // ************************************************** //
+ // // Send buffer to the audio device
+
+ // #ifdef AUDIO_DEBUG_MESSAGES_ON
+ // DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Flushing output buffer" << endl;
+ // #endif
+
+ // while (1)
+ // {
+ // if (waveHeaderOut[NextBufferInd].dwFlags & WHDR_DONE)
+ // {
+ // result=waveOutUnprepareHeader(hWaveOut,
+ // &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
+ // DSP::f::AudioCheckError(result);
+
+ // Sample=OutBuffer.data();
+ // // ************************************************** //
+ // // Converts samples format to the one suitable for the audio device
+ // switch (OutSampleType)
+ // {
+ // case DSP::e::SampleType::ST_uchar:
+ // temp8=(uint8_t *)(WaveOutBuffers[NextBufferInd].data());
+ // for (ind=0; ind<OutBufferLen; ind++)
+ // {
+ // if (*Sample < 0)
+ // Znak=-1;
+ // else
+ // Znak=1;
+
+ // *Sample*=127;
+ // if ((*Sample)*Znak > 127)
+ // *temp8=(unsigned char)(128+Znak*127);
+ // else
+ // *temp8=(unsigned char)(128+*Sample+Znak*0.5);
+
+ // Sample++;
+ // temp8++;
+ // }
+ // break;
+ // case DSP::e::SampleType::ST_short:
+ // temp16=(short *)(WaveOutBuffers[NextBufferInd].data());
+ // for (ind=0; ind<OutBufferLen; ind++)
+ // {
+ // if (*Sample < 0)
+ // Znak=-1;
+ // else
+ // Znak=1;
+
+ // *Sample*=SHRT_MAX;
+ // if ((*Sample)*Znak > SHRT_MAX)
+ // *temp16=(short)(Znak*SHRT_MAX);
+ // else
+ // *temp16=(short)(*Sample+Znak*0.5);
+ // Sample++;
+ // temp16++;
+ // }
+ // break;
+ // default:
+ // break;
+ // }
+
+ // result=waveOutPrepareHeader(hWaveOut,
+ // &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
+ // DSP::f::AudioCheckError(result);
+
+ // if (IsPlayingNow == false)
+ // {
+ // if (NextBufferInd == 1)
+ // {
+ // for (ind=0; ind < DSP::NoOfAudioOutputBuffers-1; ind++) //one spare buffer
+ // {
+ // result=waveOutWrite(hWaveOut,
+ // &(waveHeaderOut[ind]), sizeof(WAVEHDR));
+ // DSP::f::AudioCheckError(result);
+ // }
+ // IsPlayingNow = true;
+ // }
+
+ // }
+ // else
+ // {
+ // result=waveOutWrite(hWaveOut,
+ // &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
+ // DSP::f::AudioCheckError(result);
+ // }
+ // NextBufferInd++;
+ // NextBufferInd %= DSP::NoOfAudioOutputBuffers;
+
+ // break;
+ // }
+ // else
+ // {
+ // // Sleep(10);
+ // #ifdef AUDIO_DEBUG_MESSAGES_ON
+ // DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Waiting for free output buffer" << endl;
+ // #endif
+ // DSP::f::Sleep(0);
+ // }
+ // }
+ // #endif // WINMMAPI
}
void DSP::u::AudioOutput::InputExecute(INPUT_EXECUTE_ARGS)
diff --git a/src/cpp/WMM_support.cpp b/src/cpp/WMM_support.cpp
new file mode 100644
index 0000000000000000000000000000000000000000..f53a2bc1bace805fe4288b6ca7d4d80facd1b4a7
--- /dev/null
+++ b/src/cpp/WMM_support.cpp
@@ -0,0 +1,360 @@
+/*! \file WMM_support.cpp
+ * WMM support code file
+ *
+ * \author Marek Blok
+ */
+
+#include <WMM_support.h>
+
+#include <DSP_lib.h> // for logging
+
+DSP::WMM_object_t::WMM_object_t()
+{
+ is_device_openned = false;
+ WaveOutDevNo = UINT_MAX;
+
+ NextBufferInd=0;
+ IsPlayingNow = false;
+};
+DSP::WMM_object_t::~WMM_object_t()
+{
+ if (is_device_openned) {
+ close_PCM_device();
+ }
+}
+
+void CALLBACK DSP::WMM_object_t::waveOutProc(HWAVEOUT hwo, UINT uMsg,
+ uint32_t dwInstance, uint32_t dwParam1, uint32_t dwParam2)
+{
+ UNUSED_ARGUMENT(hwo);
+ UNUSED_ARGUMENT(uMsg);
+ UNUSED_ARGUMENT(dwInstance);
+ UNUSED_ARGUMENT(dwParam1);
+ UNUSED_ARGUMENT(dwParam2);
+#ifdef __DEBUG__
+#ifdef AUDIO_DEBUG_MESSAGES_ON
+// MMRESULT result;
+ DSP::u::AudioOutput *Current;
+ bool AllDone;
+ int ind;
+ string tekst;
+
+ Current = AudioObjects[dwInstance];
+
+ switch (uMsg)
+ {
+ case WOM_OPEN:
+ DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second
+ << "WOM_OPEN(" << (int)dwInstance << ")" << endl;
+ break;
+ case WOM_CLOSE:
+ DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second
+ << "WOM_CLOSE(" << (int)dwInstance << ")" << endl;
+ break;
+ case WOM_DONE:
+ DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second
+ << "WOM_DONE(" << (int)dwInstance << ")" << endl;
+
+ if (Current->StopPlaying)
+ {
+ DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second << "StopPlaying is set" << endl;
+ return;
+ }
+ else
+ {
+ AllDone=true;
+ for (ind=0; ind < DSP_NoOfAudioOutputBuffers; ind++)
+ AllDone &= (Current->waveHeaderOut[ind].dwFlags & WHDR_DONE);
+ if (AllDone)
+ DSP::log << "DSP::u::AudioOutput::waveOutProc" << DSP::e::LogMode::second << "All buffers had been used - nothing to play" << endl;
+ }
+ break;
+ }
+#endif
+#endif
+}
+
+
+void DSP::WMM_object_t::log_driver_data() {
+ /*
+ int val;
+
+ DSP::log << "WMM library version: " << SND_LIB_VERSION_STR << endl;
+
+ DSP::log << endl;
+ DSP::log << "PCM stream types:" << endl;
+ for (val = 0; val <= SND_PCM_STREAM_LAST; val++)
+ DSP::log << " " << snd_pcm_stream_name((snd_pcm_stream_t)val) << endl;
+
+ DSP::log << endl;
+ DSP::log << "PCM access types:" << endl;
+ for (val = 0; val <= SND_PCM_ACCESS_LAST; val++)
+ DSP::log << " " << snd_pcm_access_name((snd_pcm_access_t)val) << endl;
+
+ DSP::log << endl;
+ DSP::log << "PCM formats:" << endl;
+ for (val = 0; val <= SND_PCM_FORMAT_LAST; val++)
+ if (snd_pcm_format_name((snd_pcm_format_t)val) != NULL)
+ DSP::log << " " << snd_pcm_format_name((snd_pcm_format_t)val) <<
+ "(" << snd_pcm_format_description((snd_pcm_format_t)val) << ")" << endl;
+
+ DSP::log << endl;
+ DSP::log << "PCM subformats:" << endl;
+ for (val = 0; val <= SND_PCM_SUBFORMAT_LAST; val++)
+ DSP::log << " " << snd_pcm_subformat_name((snd_pcm_subformat_t)val) <<
+ "(" << snd_pcm_subformat_description((snd_pcm_subformat_t)val) << ")" << endl;
+
+ DSP::log << endl;
+ DSP::log << "PCM states:" << endl;
+ for (val = 0; val <= SND_PCM_STATE_LAST; val++)
+ DSP::log << " " << snd_pcm_state_name((snd_pcm_state_t)val) << endl;
+ */
+}
+
+// Callback = (DWORD_PTR)(&DSP::u::AudioOutput::waveOutProc);
+
+bool DSP::WMM_object_t::select_device_by_number(const unsigned int &device_number) {
+ WaveOutDevNo = device_number;
+
+ return WaveOutDevNo;
+}
+
+bool DSP::WMM_object_t::is_device_playing(void) {
+ return IsPlayingNow;
+}
+
+long DSP::WMM_object_t::open_PCM_device_4_output(const int &no_of_channels, int no_of_bits, const long &sampling_rate, const long &audio_outbuffer_size) {
+ if (is_device_openned)
+ {
+ DSP::log << "DSP::WMM_object_t::open_PCM_device_4_output" << DSP::e::LogMode::second << "Device has been already opened: closing device before reopening" << endl;
+ close_PCM_device();
+ }
+
+ switch (no_of_bits)
+ {
+ case 8:
+ OutSampleType=DSP::e::SampleType::ST_uchar;
+ break;
+ case 16:
+ OutSampleType=DSP::e::SampleType::ST_short;
+ break;
+ default:
+ OutSampleType=DSP::e::SampleType::ST_short;
+ no_of_bits=16;
+ break;
+ }
+
+ //Wypeniamy struktur wfx
+ wfx.wf.wFormatTag=WAVE_FORMAT_PCM;
+ wfx.wf.nChannels=(uint16_t)no_of_channels;
+ wfx.wf.nSamplesPerSec=(UINT)sampling_rate;
+ wfx.wBitsPerSample=(uint16_t)no_of_bits;
+ wfx.wf.nAvgBytesPerSec=wfx.wf.nSamplesPerSec*(wfx.wBitsPerSample/8);
+ wfx.wf.nBlockAlign=(uint16_t)(wfx.wf.nChannels*(wfx.wBitsPerSample/8));
+
+ if (WaveOutDevNo >= (UINT)waveOutGetNumDevs())
+ result=waveOutOpen(&hWaveOut,
+ WAVE_MAPPER, //&DeviceID,
+ (WAVEFORMATEX *)(&wfx),
+ DWORD_PTR(NULL), //Callback,
+ 0, //Current_CallbackInstance,
+ WAVE_ALLOWSYNC | WAVE_FORMAT_DIRECT //| WAVE_MAPPED //CALLBACK_NULL
+ //CALLBACK_FUNCTION | WAVE_ALLOWSYNC | WAVE_FORMAT_DIRECT //| WAVE_MAPPED //CALLBACK_NULL
+ );
+ else
+ result=waveOutOpen(&hWaveOut,
+ WaveOutDevNo, //&DeviceID,
+ (WAVEFORMATEX *)(&wfx),
+ DWORD_PTR(NULL), //Callback,
+ 0, // Current_CallbackInstance, //CallbackInstance,
+ WAVE_ALLOWSYNC | WAVE_FORMAT_DIRECT //| WAVE_MAPPED //CALLBACK_NULL
+// CALLBACK_FUNCTION | WAVE_ALLOWSYNC | WAVE_FORMAT_DIRECT //| WAVE_MAPPED //CALLBACK_NULL
+ );
+
+ if (DSP::f::AudioCheckError(result) == false)
+ { // everything is ok
+ waveHeaderOut.resize(DSP::NoOfAudioOutputBuffers);
+ WaveOutBufferLen=wfx.wf.nBlockAlign*audio_outbuffer_size;
+ WaveOutBuffers.resize(DSP::NoOfAudioOutputBuffers);
+ for (unsigned int ind=0; ind< DSP::NoOfAudioOutputBuffers; ind++)
+ {
+ WaveOutBuffers[ind].clear();
+ WaveOutBuffers[ind].resize(WaveOutBufferLen, 0);
+ }
+
+
+ for (unsigned int ind=0; ind< DSP::NoOfAudioOutputBuffers; ind++)
+ {
+ waveHeaderOut[ind].lpData=(char *)(WaveOutBuffers[ind].data());
+ waveHeaderOut[ind].dwBufferLength=no_of_channels*audio_outbuffer_size*(no_of_bits/8); //sizeof(short);
+ waveHeaderOut[ind].dwFlags= 0; // WHDR_BEGINLOOP | WHDR_ENDLOOP;
+ waveHeaderOut[ind].dwLoops=0;
+
+ result=waveOutPrepareHeader(hWaveOut,
+ &(waveHeaderOut[ind]), sizeof(WAVEHDR));
+ DSP::f::AudioCheckError(result);
+ waveHeaderOut[ind].dwFlags= WHDR_DONE; // WHDR_BEGINLOOP | WHDR_ENDLOOP;
+ }
+ }
+ else
+ { //error while creating audio output
+ waveHeaderOut.clear();
+ WaveOutBuffers.clear();
+ WaveOutBufferLen = 0;
+ }
+
+ return sampling_rate;
+}
+
+long DSP::WMM_object_t::open_PCM_device_4_input(const int &no_of_channels, int no_of_bits, const long &sampling_rate, const long &audio_outbuffer_size) {
+ return -1;
+}
+
+
+bool DSP::WMM_object_t::close_PCM_device(void) {
+ result = waveOutReset(hWaveOut);
+ DSP::f::AudioCheckError(result);
+
+ for (unsigned int ind=0; ind< DSP::NoOfAudioOutputBuffers; ind++)
+ {
+ result=waveOutUnprepareHeader(hWaveOut,
+ &(waveHeaderOut[ind]), sizeof(WAVEHDR));
+ DSP::f::AudioCheckError(result);
+ }
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Closing DSP::u::AudioOutput" << endl;
+ #endif
+ result=waveOutClose(hWaveOut);
+ while (result==WAVERR_STILLPLAYING)
+ {
+ DSP::f::Sleep(100);
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "DSP::u::AudioOutput" << DSP::e::LogMode::second << "Closing DSP::u::AudioOutput" << endl;
+ #endif
+ result=waveOutClose(hWaveOut);
+ }
+ DSP::f::AudioCheckError(result);
+
+ // 2) Free buffers
+ WaveOutBuffers.clear();
+ #ifdef WINMMAPI
+ waveHeaderOut.clear();
+ #endif
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "WMM PCM sound closed" << endl;
+ #endif
+
+ return true; // device has been closed
+}
+
+
+long DSP::WMM_object_t::append_playback_buffer(DSP::Float_vector &float_buffer) {
+ uint8_t *temp8;
+ short *temp16;
+ short Znak;
+ uint32_t ind;
+
+ // ************************************************** //
+ // Send float_buffer to the audio device
+
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "DSP::WMM_object_t::append_playback_buffer" << DSP::e::LogMode::second << "Flushing output buffer" << endl;
+ #endif
+
+ while (1)
+ {
+ if (waveHeaderOut[NextBufferInd].dwFlags & WHDR_DONE)
+ {
+ result=waveOutUnprepareHeader(hWaveOut,
+ &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
+ DSP::f::AudioCheckError(result);
+
+ DSP::Float_ptr Sample=float_buffer.data();
+ // ************************************************** //
+ // Converts samples format to the one suitable for the audio device
+ switch (OutSampleType)
+ {
+ case DSP::e::SampleType::ST_uchar:
+ temp8=(uint8_t *)(WaveOutBuffers[NextBufferInd].data());
+ for (ind=0; ind<float_buffer.size(); ind++)
+ {
+ if (*Sample < 0)
+ Znak=-1;
+ else
+ Znak=1;
+
+ *Sample*=127;
+ if ((*Sample)*Znak > 127)
+ *temp8=(unsigned char)(128+Znak*127);
+ else
+ *temp8=(unsigned char)(128+*Sample+Znak*0.5);
+
+ Sample++;
+ temp8++;
+ }
+ break;
+ case DSP::e::SampleType::ST_short:
+ temp16=(short *)(WaveOutBuffers[NextBufferInd].data());
+ for (ind=0; ind<float_buffer.size(); ind++)
+ {
+ if (*Sample < 0)
+ Znak=-1;
+ else
+ Znak=1;
+
+ *Sample*=SHRT_MAX;
+ if ((*Sample)*Znak > SHRT_MAX)
+ *temp16=(short)(Znak*SHRT_MAX);
+ else
+ *temp16=(short)(*Sample+Znak*0.5);
+ Sample++;
+ temp16++;
+ }
+ break;
+ default:
+ break;
+ }
+
+ result=waveOutPrepareHeader(hWaveOut,
+ &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
+ DSP::f::AudioCheckError(result);
+
+ if (IsPlayingNow == false)
+ {
+ if (NextBufferInd == 1)
+ {
+ for (ind=0; ind < DSP::NoOfAudioOutputBuffers-1; ind++) //one spare buffer
+ {
+ result=waveOutWrite(hWaveOut,
+ &(waveHeaderOut[ind]), sizeof(WAVEHDR));
+ DSP::f::AudioCheckError(result);
+ }
+ IsPlayingNow = true;
+ }
+
+ }
+ else
+ {
+ result=waveOutWrite(hWaveOut,
+ &(waveHeaderOut[NextBufferInd]), sizeof(WAVEHDR));
+ DSP::f::AudioCheckError(result);
+ }
+ NextBufferInd++;
+ NextBufferInd %= DSP::NoOfAudioOutputBuffers;
+
+ break;
+ }
+ else
+ {
+ // Sleep(10);
+ #ifdef AUDIO_DEBUG_MESSAGES_ON
+ DSP::log << "DSP::WMM_object_t::append_playback_buffer" << DSP::e::LogMode::second << "Waiting for free output buffer" << endl;
+ #endif
+ DSP::f::Sleep(0);
+ }
+ }
+
+ return long(float_buffer.size());
+}
\ No newline at end of file
diff --git a/src/include/ALSA_support.h b/src/include/ALSA_support.h
index 1dd3993c8f2169520c5831e4b51b353c7605881c..cf7084b2ca81cff2fce3a277f7db52d1b217a756 100644
--- a/src/include/ALSA_support.h
+++ b/src/include/ALSA_support.h
@@ -13,30 +13,34 @@
/* All of the ALSA library API is defined in this header */
#include <alsa/asoundlib.h>
-#include <DSP_lib.h> // for logging
+#include <DSP_lib.h> // for logging and types
-class ALSA_object_t {
-private:
- snd_pcm_t *alsa_handle;
- snd_pcm_hw_params_t *hw_params;
+namespace DSP {
+
+ class ALSA_object_t : public DSP::SOUND_object_t {
+ private:
+ snd_pcm_t *alsa_handle;
+ snd_pcm_hw_params_t *hw_params;
-public:
+ public:
- //! log basic ALSA information
- void log_alsa_data();
+ //! log basic ALSA information
+ void log_driver_data();
- //! open default PCM device and return 1 on success or negative error code
- /*! stream_type = SND_PCM_STREAM_PLAYBACK or SND_PCM_STREAM_CAPTURE
- */
- int open_alsa_device(snd_pcm_stream_t stream_type, int no_of_channels, unsigned int &sampling_rate);
- void close_alsa_device(bool do_drain = false, bool use_log = false);
+ //! open default PCM device and return 1 on success or negative error code
+ /*! stream_type = SND_PCM_STREAM_PLAYBACK or SND_PCM_STREAM_CAPTURE
+ */
+ int open_alsa_device(snd_pcm_stream_t stream_type, int no_of_channels, unsigned int &sampling_rate);
+ void close_alsa_device(bool do_drain = false, bool use_log = false);
- void get_params(snd_pcm_uframes_t &frames, unsigned int &period_time);
- snd_pcm_sframes_t pcm_writei(const void *buffer, snd_pcm_uframes_t &frames);
+ void get_params(snd_pcm_uframes_t &frames, unsigned int &period_time);
+ snd_pcm_sframes_t pcm_writei(const void *buffer, snd_pcm_uframes_t &frames);
- //! object constructor
- ALSA_object_t();
- ~ALSA_object_t();
-};
+ //! object constructor
+ ALSA_object_t();
+ ~ALSA_object_t();
+ };
+
+}
#endif // ALSA_support_H
diff --git a/src/include/DSP_IO.h b/src/include/DSP_IO.h
index 74bf78f6d999c6a5c81c729e7800abf39e4420c2..76cb695a23034dd8da0476a1e82c740607f016f6 100644
--- a/src/include/DSP_IO.h
+++ b/src/include/DSP_IO.h
@@ -23,8 +23,7 @@
////#ifdef __CYGWIN__
//#ifdef WIN32
#if defined(WIN32) || defined(WIN64)
- #include <windef.h>
- #include <mmsystem.h>
+ #include <WMM_support.h>
#else
#ifdef LINUX
//! \TODO adapt for linux
@@ -834,21 +833,22 @@ class DSP::u::AudioOutput : public DSP::Block
private:
DSP::T_WAVEchunk WAVEchunk;
#ifdef WINMMAPI
- HWAVEOUT hWaveOut;
+ //HWAVEOUT hWaveOut;
+ DSP::WMM_object_t snd_object;
#endif
- //! Index of the buffer which must be used next time
- unsigned long NextBufferInd;
- //! Type of samples in WaveOutBuffers
- DSP::e::SampleType OutSampleType;
- #ifdef WINMMAPI
- std::vector<WAVEHDR> waveHeaderOut;
- #endif
- uint32_t WaveOutBufferLen; // in bytes
- //! Buffers for audio samples prepared for playing
- std::vector<std::vector<uint8_t>> WaveOutBuffers;
-
- //! size of the buffers used internally with WMM driver
+ // //! Index of the buffer which must be used next time
+ // unsigned long NextBufferInd;
+ // //! Type of samples in WaveOutBuffers
+ // DSP::e::SampleType OutSampleType;
+ //#ifdef WINMMAPI
+ // std::vector<WAVEHDR> waveHeaderOut;
+ //#endif
+ // uint32_t WaveOutBufferLen; // in bytes
+ // //! Buffers for audio samples prepared for playing
+ // std::vector<std::vector<uint8_t>> WaveOutBuffers;
+
+ //! size of the buffers used internally with audio driver
uint32_t audio_outbuffer_size;
//! in samples times number of channels
@@ -861,7 +861,7 @@ class DSP::u::AudioOutput : public DSP::Block
/*! saturation logic should be implemented */
void FlushBuffer(void);
- bool IsPlayingNow;
+ //bool IsPlayingNow;
bool StartAudio(void);
bool StopAudio(void);
@@ -877,10 +877,10 @@ class DSP::u::AudioOutput : public DSP::Block
//! (returns number of samples read per channel)
uint32_t GetAudioSegment(void);
- #ifdef WINMMAPI
- static void CALLBACK waveOutProc(HWAVEOUT hwo, UINT uMsg,
- uint32_t dwInstance, uint32_t dwParam1, uint32_t dwParam2);
- #endif
+ //#ifdef WINMMAPI
+ // static void CALLBACK waveOutProc(HWAVEOUT hwo, UINT uMsg,
+ // uint32_t dwInstance, uint32_t dwParam1, uint32_t dwParam2);
+ //#endif
void Init(unsigned long SamplingFreq,
unsigned int InputsNo=1, //just one channel
diff --git a/src/include/DSP_lib.h b/src/include/DSP_lib.h
index e6d13c603eadb5a1ea9684e7e610c5ee8d5d0135..d89b1f4e1ca42b3fc9c7f767cca699242e4cf1bc 100644
--- a/src/include/DSP_lib.h
+++ b/src/include/DSP_lib.h
@@ -11,7 +11,7 @@
#define DSP_VER_MAJOR 0
#define DSP_VER_MINOR 20
-#define DSP_VER_BUILD 4 // !!! without zeroes before, else this will be treated as octal number
+#define DSP_VER_BUILD 5 // !!! without zeroes before, else this will be treated as octal number
#define DSP_VER_YEAR 2021
#define DSP_VER DSP_VER_MAJOR.DSP_VER_MINOR.DSP_VER_BUILD
diff --git a/src/include/DSP_types.h b/src/include/DSP_types.h
index ce092f4f8ef9ddd7d30f343338159dafe4a17f03..c6b5e3d859e9efccf2c2f50bf2ba3d04868da90f 100644
--- a/src/include/DSP_types.h
+++ b/src/include/DSP_types.h
@@ -528,6 +528,31 @@ namespace DSP {
#endif
+namespace DSP {
+ //! Base class for classes implementing sound card support for DSP::u::AudioInput and DSP::u::AudioOutput
+ /*! \TODO convert WMM support in DSP::u::AudioInput and DSP::u::AudioOutput into support through WMM_object_t
+ */
+ class SOUND_object_t {
+
+ public:
+ virtual void log_driver_data() = 0;
+
+ virtual bool select_device_by_number(const unsigned int &device_number=UINT_MAX) = 0;
+
+ virtual long open_PCM_device_4_output(const int &no_of_channels, int no_of_bits, const long &sampling_rate, const long &audio_outbuffer_size) = 0;
+ virtual long open_PCM_device_4_input(const int &no_of_channels, int no_of_bits, const long &sampling_rate, const long &audio_outbuffer_size) = 0;
+ virtual bool close_PCM_device(void) = 0;
+
+ //! returns true is the playback is on
+ virtual bool is_device_playing(void) = 0;
+
+ //! \note values stored in float_buffer might be altered
+ long append_playback_buffer(DSP::Float_vector &float_buffer);
+
+ virtual ~SOUND_object_t() {};
+ };
+}
+
#include <DSP_DOT.h>
diff --git a/src/include/WMM_support.h b/src/include/WMM_support.h
new file mode 100644
index 0000000000000000000000000000000000000000..5706e99e88c2c54ebeaef4e4d21d3bce2bab37d0
--- /dev/null
+++ b/src/include/WMM_support.h
@@ -0,0 +1,83 @@
+/*! \file WMM_support.h
+ * WMM support header file
+ *
+ * \author Marek Blok
+ */
+//---------------------------------------------------------------------------
+#ifndef WMM_support_H
+#define WMM_support_H
+
+/* inluce WMM API */
+////#ifdef __CYGWIN__
+//#ifdef WIN32
+#if defined(WIN32) || defined(WIN64)
+ #include <windef.h>
+ #include <mmsystem.h>
+#else
+ #error NO WIN32
+#endif
+
+#include <DSP_types.h> // for types
+
+namespace DSP {
+ class WMM_object_t : public DSP::SOUND_object_t {
+ private:
+ bool is_device_openned;
+
+ HWAVEOUT hWaveOut;
+
+ MMRESULT result;
+ DWORD_PTR Callback;
+
+ //Rezerwacja pamiÄci dla formatu WAVE
+ // WAVEFORMATEX wfx; //to wymaga korekty
+ PCMWAVEFORMAT wfx;
+
+ unsigned int WaveOutDevNo; // device numer used in next open operations
+
+ std::vector<WAVEHDR> waveHeaderOut;
+ uint32_t WaveOutBufferLen; // in bytes
+ //! Buffers for audio samples prepared for playing
+ std::vector<std::vector<uint8_t>> WaveOutBuffers;
+ //! Type of samples in WaveOutBuffers
+ DSP::e::SampleType OutSampleType;
+ //! Index of the buffer which must be used next time
+ unsigned long NextBufferInd;
+
+ bool IsPlayingNow;
+
+ static void CALLBACK waveOutProc(HWAVEOUT hwo, UINT uMsg,
+ uint32_t dwInstance, uint32_t dwParam1, uint32_t dwParam2);
+
+
+ public:
+
+ //! log basic WMM information
+ void log_driver_data();
+
+ // returns false if device is already opened
+ bool select_device_by_number(const unsigned int &device_number=UINT_MAX);
+
+ //! opens default PCM device for playback and returns selected sampling rate on success or negative error code
+ long open_PCM_device_4_output(const int &no_of_channels, int no_of_bits, const long &sampling_rate, const long &audio_outbuffer_size);
+ //! opens default PCM device for capture and returns selected sampling rate on success or negative error code
+ long open_PCM_device_4_input(const int &no_of_channels, int no_of_bits, const long &sampling_rate, const long &audio_outbuffer_size);
+
+ bool close_PCM_device(void);
+
+ //! returns true is the playback is on
+ bool is_device_playing(void);
+
+ //void get_params(snd_pcm_uframes_t &frames, unsigned int &period_time);
+ //snd_pcm_sframes_t pcm_writei(const void *buffer, snd_pcm_uframes_t &frames);
+ //! \note values stored in float_buffer might be altered
+ long append_playback_buffer(DSP::Float_vector &float_buffer);
+
+ //! object constructor \TODO check use of virtual in constructor and destructor
+ WMM_object_t();
+ ~WMM_object_t();
+ };
+
+}
+
+#endif // WMM_support_H