From 8e4fac6cf1cdb54a82906269a02aeece213c639a Mon Sep 17 00:00:00 2001
From: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
Date: Fri, 15 Feb 2008 17:25:57 -0500
Subject: [PATCH] Remove totally portaudio. alsa playback implemented
---
src/audio/audiolayer.cpp | 547 +++++++++++++++------------------------
src/audio/audiolayer.h | 239 +++++++++--------
src/managerimpl.cpp | 12 +-
3 files changed, 337 insertions(+), 461 deletions(-)
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index f1d990ff78..d892214c50 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -20,50 +20,35 @@
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
-#include <stdio.h>
-#include <stdlib.h>
+#include <cstdio>
+#include <cstdlib>
+#include <iostream>
#include "audiolayer.h"
#include "../global.h"
#include "../manager.h"
#include "../user_cfg.h"
-//#define SFL_TEST
-//#define SFL_TEST_SINE
+#define PCM_NAME_DEFAULT "plughw:0,0"
#ifdef SFL_TEST_SINE
#include <cmath>
#endif
-AudioLayer::AudioLayer(ManagerImpl* manager)
+ AudioLayer::AudioLayer(ManagerImpl* manager)
: _urgentRingBuffer(SIZEBUF)
, _mainSndRingBuffer(SIZEBUF)
, _micRingBuffer(SIZEBUF)
- , _defaultVolume(100)
- , _stream(NULL)
+ , _defaultVolume(100)
, _errorMessage("")
- , _manager(manager)
+ , _manager(manager)
{
_sampleRate = 8000;
-
+
_inChannel = 1; // don't put in stereo
_outChannel = 1; // don't put in stereo
_echoTesting = false;
- try {
- portaudio::AutoSystem autoSys;
- portaudio::System::initialize();
- }
- catch (const portaudio::PaException &e) {
- setErrorMessage(e.paErrorText());
- }
- catch (const portaudio::PaCppException &e) {
- setErrorMessage(e.what());
- } // std::runtime_error &e (e.what())
- catch (...) {
- setErrorMessage("Unknown type error in portaudio initialization");
- }
-
#ifdef SFL_TEST_SINE
leftPhase_ = 0;
tableSize_ = 200;
@@ -80,343 +65,247 @@ AudioLayer::AudioLayer(ManagerImpl* manager)
// Destructor
AudioLayer::~AudioLayer (void)
{
- stopStream();
- closeStream();
-
- try {
- portaudio::System::terminate();
- } catch (const portaudio::PaException &e) {
- _debug("! AL: Catch an exception when portaudio tried to terminate\n");
+ if(_playback_handle != NULL) {
+ close_alsa();
+ _playback_handle = NULL ;
}
#ifdef SFL_TEST_SINE
delete [] table_;
#endif
}
-void
-AudioLayer::closeStream (void)
+ void
+AudioLayer::close_alsa (void)
{
- ost::MutexLock guard(_mutex);
- if(_stream) {
- _stream->close();
- delete _stream; _stream = NULL;
- }
+ _debug(" Alsa close stream \n");
+ //ost::MutexLock guard(_mutex);
+ snd_pcm_close(_playback_handle);
}
bool
AudioLayer::hasStream(void) {
ost::MutexLock guard(_mutex);
- return (_stream!=0 ? true : false);
+ return (_playback_handle!=0 ? true : false);
}
-
-void
+ void
AudioLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize)
{
- closeStream();
_indexIn = indexIn;
_indexOut = indexOut;
_sampleRate = sampleRate;
_frameSize = frameSize;
- int portaudioFramePerBuffer = FRAME_PER_BUFFER; //=FRAME_PER_BUFFER; //= paFramesPerBufferUnspecified;
- //int portaudioFramePerBuffer = (int) (8000 * frameSize / 1000); //= paFramesPerBufferUnspecified;
-
- // Select default audio API
-// selectPreferedApi(paALSA, _indexIn, _indexOut);
-
- int nbDevice = getDeviceCount();
- _debug("Nb of audio devices: %i\n",nbDevice);
- if (nbDevice == 0) {
- _debug("Portaudio detect no sound card.");
- return;
- } else {
- _debug(" Setting audiolayer: device in=%2d, out=%2d\n", _indexIn, _indexOut);
- _debug(" : nb channel in=%2d, out=%2d\n", _inChannel, _outChannel);
- _debug(" : sample rate=%5d, format=%s\n", _sampleRate, SFLPortaudioFormatString);
- _debug(" : frame per buffer=%d\n", portaudioFramePerBuffer);
+
+ _debug(" Setting audiolayer: device in=%2d, out=%2d\n", _indexIn, _indexOut);
+ _debug(" : nb channel in=%2d, out=%2d\n", _inChannel, _outChannel);
+ _debug(" : sample rate=%5d, format=%s\n", _sampleRate, SFLPortaudioFormatString);
+ _debug(" : frame per buffer=%d\n", FRAME_PER_BUFFER);
+
+ // Name of the PCM device
+ // TODO: Must be dynamic
+
+ // TODO: capture
+ // Capture stream
+ snd_pcm_stream_t capture_stream = SND_PCM_STREAM_CAPTURE;
+
+ // Open the playback device
+ open_playback_device(PCM_NAME_DEFAULT);
+
+ // Set up the parameters required to open a device:
+ init_hw_parameters();
+
+ device_info();
+
+ //ost::MutexLock guard(_mutex);
+}
+
+void
+AudioLayer::open_playback_device(std::string pcm_name)
+{
+ // Playback stream
+ snd_pcm_stream_t playback_stream = SND_PCM_STREAM_PLAYBACK;
+
+ if(snd_pcm_open(&_playback_handle, pcm_name.c_str(), playback_stream, SND_PCM_NONBLOCK) < 0){
+ _debug(" Error while opening PCM device %s\n", pcm_name.c_str());
}
+ else
+ _debug(" Device %s successfully opened. \n", pcm_name.c_str());
+}
- try {
- // Set up the parameters required to open a (Callback)Stream:
- portaudio::DirectionSpecificStreamParameters
- inParams(portaudio::System::instance().deviceByIndex(_indexIn),
- _inChannel, SFLPortaudioFormat, true,
- portaudio::System::instance().deviceByIndex(_indexIn).defaultLowInputLatency(),
- NULL);
-
- portaudio::DirectionSpecificStreamParameters outParams(
- portaudio::System::instance().deviceByIndex(_indexOut),
- _outChannel, SFLPortaudioFormat, true,
- portaudio::System::instance().deviceByIndex(_indexOut).defaultLowOutputLatency(),
- NULL);
-
- // like audacity
- // DON'T USE paFramesPerBufferUnspecified, it's 32, instead of 160 for FRAME_PER_BUFFER
- // DON'T USE paDitherOff or paClipOff,
- // FRAME_PER_BUFFER | paFramesPerBufferUnspecified
- // paNoFlag | paClipOff || paDitherOff | paPrimeOutputBuffersUsingStreamCallback | paNeverDropInput
-
- portaudio::StreamParameters const params(
- inParams,
- outParams,
- _sampleRate, portaudioFramePerBuffer, paClipOff);
-
- // Create (and open) a new Stream, using the AudioLayer::audioCallback
- ost::MutexLock guard(_mutex);
-#ifdef SFL_TEST
- _stream = new portaudio::MemFunCallbackStream<AudioLayer>(params, *this, &AudioLayer::miniAudioCallback);
-#else
- _stream = new portaudio::MemFunCallbackStream<AudioLayer>(params,*this, &AudioLayer::audioCallback);
-#endif
-
+void
+AudioLayer::init_hw_parameters( )
+{
+ int periods = 2;
+ snd_pcm_uframes_t periodSize = 8192;
+ unsigned int sr = 44100; //(unsigned int)sampleRate;
+
+ // Information about the hardware
+ snd_pcm_hw_params_t *hwparams;
+ // Allocate the struct on the stack
+ snd_pcm_hw_params_alloca( &hwparams );
+ // Allocate memory space
+ if(snd_pcm_hw_params_malloc( &hwparams) < 0)
+ _debug(" can not allocate hardware paramater structure. \n");
+ // Init hwparams with full configuration space
+ if(snd_pcm_hw_params_any(_playback_handle, hwparams) < 0){
+ _debug(" Can not configure this PCM device .\n");
+ }
+ // Set access type
+ if(snd_pcm_hw_params_set_access( _playback_handle , hwparams , SND_PCM_ACCESS_RW_INTERLEAVED ) <0){
+ _debug(" Error setting access. \n");
}
- catch (const portaudio::PaException &e) {
- setErrorMessage(e.paErrorText());
- _debug("Portaudio openDevice error: %s\n", e.paErrorText());
+ // Set sample format
+ if(snd_pcm_hw_params_set_format( _playback_handle, hwparams , SND_PCM_FORMAT_S16_LE) < 0 ){
+ _debug(" Error setting format. \n");
}
- catch (const portaudio::PaCppException &e) {
- setErrorMessage(e.what());
- _debug("Portaudio openDevice error: %s\n", e.what());
- } // std::runtime_error &e (e.what())
- catch (...) {
- setErrorMessage("Unknown type error in portaudio openDevice");
- _debug("Portaudio openDevice: unknown error\n");
+ // Set sample rate
+ if(snd_pcm_hw_params_set_rate_near( _playback_handle, hwparams, &sr, 0 ) <0 ) {
+ _debug(" Error setting sample rate. \n");
}
+ if(_sampleRate != sr)
+ _debug(" The rate %i Hz is not supported by your hardware.\n ==> Using %i Hz instead. \n", _sampleRate, sr);
+ // Set number of channels
+ if (snd_pcm_hw_params_set_channels( _playback_handle, hwparams, _inChannel) < 0)
+ _debug(" Error setting channels. \n");
+ // Set number of periods
+ if (snd_pcm_hw_params_set_periods( _playback_handle, hwparams, periods, 0) < 0)
+ _debug(" Error setting periods. \n ");
+ // Set buffer size
+ if(snd_pcm_hw_params_set_buffer_size( _playback_handle, hwparams, (periodSize * periods) >>2) < 0 )
+ _debug(" Error setting buffer size. \n ");
+ // Apply hardware parameters to the device
+ if(snd_pcm_hw_params( _playback_handle, hwparams) < 0 )
+ _debug(" Error setting HW params. \n ");
}
int
-AudioLayer::getDeviceCount()
+AudioLayer::device_info( void )
{
- return portaudio::System::instance().deviceCount();
+ /*snd_pcm_info_t *info;
+ if(snd_pcm_info( _playback_handle , info) <0 ) {
+ _debug(" Can not gather infos on opened device. \n");
+ return 0;
+ }
+ else{*/
+ return 1;
+ // }
}
-/**
- * Checks if ALSA is supported and selects compatible devices
- * Write changes to configuration file if necessary
- */
-void
-AudioLayer::selectPreferedApi(PaHostApiTypeId apiTypeID, int& outputDeviceIndex, int& inputDeviceIndex)
-{
- // Create iterators
- portaudio::System::HostApiIterator hostApiIter, hostApiIterEnd;
- hostApiIter = portaudio::System::instance().hostApisBegin();
- hostApiIterEnd = portaudio::System::instance().hostApisEnd();
-
- // Loop all Api
- for(; hostApiIter != hostApiIterEnd; hostApiIter++)
- {
- // If prefered Api is found, see if devices are compatible
- if(hostApiIter->typeId() == apiTypeID)
- {
- bool compatibleInputDevice = false;
- bool compatibleOutputDevice = false;
-
- // Create device iterators
- portaudio::System::DeviceIterator deviceIter, deviceIterEnd;
- deviceIter = hostApiIter->devicesBegin();
- deviceIterEnd = hostApiIter->devicesEnd();
-
- // Loop all devices
- for(; deviceIter != deviceIterEnd; deviceIter++)
- {
- // If we found our input device and it is not only an output device
- if(deviceIter->index() == inputDeviceIndex && !deviceIter->isOutputOnlyDevice())
- compatibleInputDevice = true;
- // If we found our output device and it is not only an input device
- if(deviceIter->index() == outputDeviceIndex && !deviceIter->isInputOnlyDevice())
- compatibleOutputDevice = true;
- }
-
- // Select default device of prefered API if compatible device was not found
- // and write changes to configuration file
- if(!compatibleOutputDevice)
- {
- outputDeviceIndex = hostApiIter->defaultOutputDevice().index();
- _manager->setConfig(AUDIO, DRIVER_NAME_OUT, outputDeviceIndex);
- }
- if(!compatibleInputDevice)
- {
- inputDeviceIndex = hostApiIter->defaultInputDevice().index();
- _manager->setConfig(AUDIO, DRIVER_NAME_IN, inputDeviceIndex);
- }
- }
- }
-}
-/**
- * Get list of audio devices index supported by api
- * and corresponding to IO device type
- */
-std::vector<std::string>
-AudioLayer::getAudioDeviceList(PaHostApiTypeId apiTypeID, int ioDeviceMask)
+ int
+AudioLayer::getDeviceCount()
{
- std::vector<std::string> v;
-
- // Create api iterators
- portaudio::System::HostApiIterator hostApiIter, hostApiIterEnd;
- hostApiIter = portaudio::System::instance().hostApisBegin();
- hostApiIterEnd = portaudio::System::instance().hostApisEnd();
-
- // Loop all Api
- for(; hostApiIter != hostApiIterEnd; hostApiIter++)
- {
- // If prefered Api is found, use this one
- if(hostApiIter->typeId() == apiTypeID)
- break;
- }
- // If none was found, use first api
- if(hostApiIter == hostApiIterEnd)
- hostApiIter = portaudio::System::instance().hostApisBegin();
-
- // Create device iterators
- portaudio::System::DeviceIterator deviceIter, deviceIterEnd;
- deviceIter = hostApiIter->devicesBegin();
- deviceIterEnd = hostApiIter->devicesEnd();
-
- // For each device supported by api
- for(; deviceIter != deviceIterEnd; deviceIter++)
- {
- // Check for input or output capabality
- if ( (ioDeviceMask == InputDevice && !deviceIter->isOutputOnlyDevice()) ||
- (ioDeviceMask == OutputDevice && !deviceIter->isInputOnlyDevice()) ||
- deviceIter->isFullDuplexDevice())
- {
- char id[10];
- sprintf(id, "%d", deviceIter->index());
- v.push_back(id);
- }
- }
- return v;
+ // TODO: everything
+ return 1;
}
-/**
- * Get audio device by index if it supports input output device mask
- */
-AudioDevice*
-AudioLayer::getAudioDeviceInfo(int index, int ioDeviceMask)
+ void
+AudioLayer::startStream(void)
{
- try {
- portaudio::System& sys = portaudio::System::instance();
- portaudio::Device& device = sys.deviceByIndex(index);
- int deviceIsSupported = false;
-
- if (ioDeviceMask == InputDevice && !device.isOutputOnlyDevice()) {
- deviceIsSupported = true;
- } else if (ioDeviceMask == OutputDevice && !device.isInputOnlyDevice()) {
- deviceIsSupported = true;
- } else if (device.isFullDuplexDevice()) {
- deviceIsSupported = true;
- }
-
- if (deviceIsSupported) {
- AudioDevice* audiodevice = new AudioDevice(index, device.hostApi().name(), device.name());
- if (audiodevice) {
- audiodevice->setRate(device.defaultSampleRate());
- }
- return audiodevice;
- }
- } catch (...) {
- return 0;
- }
- return 0;
+ snd_pcm_prepare( _playback_handle );
+ _debug(" ALSA start stream. \n");
+ if( snd_pcm_state( _playback_handle ) == SND_PCM_STATE_PREPARED )
+ snd_pcm_start( _playback_handle );
+ else
+ _debug(" Device not ready to start. \n");
+ //playSinusWave();
}
-
-void
-AudioLayer::startStream(void)
+ void
+AudioLayer::stopStream(void)
{
- try {
- ost::MutexLock guard(_mutex);
- if (_stream && !_stream->isActive()) {
- _debug("- AL Action: Starting sound stream\n");
- _stream->start();
- } else {
- _debug ("* AL Info: Stream doesn't exist or is already active\n");
- }
- } catch (const portaudio::PaException &e) {
- _debugException("! AL: Portaudio error: error on starting audiolayer stream");
- throw;
- } catch(...) {
- _debugException("! AL: Stream start error");
- throw;
+ _debug(" Alsa stop stream. \n");
+ ost::MutexLock guard(_mutex);
+ if( isStreamActive() ){
+ // tells the device to stop reading
+ snd_pcm_drop(_playback_handle);
+ _mainSndRingBuffer.flush();
+ _urgentRingBuffer.flush();
+ _micRingBuffer.flush();
}
+ else
+ _debug(" Device not running. \n" );
}
-
-void
-AudioLayer::stopStream(void)
+
+ void
+AudioLayer::playSinusWave( void )
{
- _debug("- AL Action: Stopping sound stream\n");
- try {
- ost::MutexLock guard(_mutex);
- if (_stream && !_stream->isStopped()) {
- _stream->stop();
- _mainSndRingBuffer.flush();
- _urgentRingBuffer.flush();
- _micRingBuffer.flush();
+ unsigned char* data;
+ int pcmreturn, l1, l2;
+ short s1, s2;
+ int frames;
+
+ data = (unsigned char*)malloc(8192);
+ frames = 8192 >> 2;
+ for(l1 = 0; l1 < 100; l1++) {
+ for(l2 = 0; l2 < frames; l2++) {
+ s1 = (l2 % 128) * 100 - 5000;
+ s2 = (l2 % 256) * 100 - 5000;
+ data[4*l2] = (unsigned char)s1;
+ data[4*l2+1] = s1 >> 8;
+ data[4*l2+2] = (unsigned char)s2;
+ data[4*l2+3] = s2 >> 8;
+ }
+ /* Write num_frames frames from buffer data to the PCM device pointed to by pcm_handle */
+ /* writei for interleaved */
+ /* writen for non-interleaved */
+ while ((pcmreturn = snd_pcm_writei(_playback_handle, data, frames)) < 0) {
+ snd_pcm_prepare(_playback_handle);
+ _debug("<<<<<<<<<<<<<<< Buffer Underrun >>>>>>>>>>>>>>>\n");
}
- } catch (const portaudio::PaException &e) {
- _debugException("! AL: Portaudio error: stoping audiolayer stream failed");
- throw;
- } catch(...) {
- _debugException("! AL: Stream stop error");
- throw;
}
}
-void
+
+
+//TODO
+ void
AudioLayer::sleep(int msec)
{
- if (_stream) {
- portaudio::System::instance().sleep(msec);
- }
}
-bool
+//TODO
+ bool
AudioLayer::isStreamActive (void)
{
ost::MutexLock guard(_mutex);
- try {
- if(_stream && _stream->isActive()) {
- return true;
- }
- } catch (const portaudio::PaException &e) {
- _debugException("! AL: Portaudio error: isActive returned an error");
- }
- return false;
+ if(snd_pcm_state(_playback_handle) == SND_PCM_STATE_RUNNING)
+ return true;
+ else
+ return false;
}
-int
+ int
AudioLayer::putMain(void* buffer, int toCopy)
{
- ost::MutexLock guard(_mutex);
- if (_stream) {
- int a = _mainSndRingBuffer.AvailForPut();
- if ( a >= toCopy ) {
- return _mainSndRingBuffer.Put(buffer, toCopy, _defaultVolume);
- } else {
- _debug("Chopping sound, Ouch! RingBuffer full ?\n");
- return _mainSndRingBuffer.Put(buffer, a, _defaultVolume);
- }
- }
+ play_alsa(buffer, toCopy);
+ //ost::MutexLock guard(_mutex);
+ /*int a = _mainSndRingBuffer.AvailForPut();
+ if ( a >= toCopy ) {
+ return _mainSndRingBuffer.Put(buffer, toCopy, _defaultVolume);
+ } else {
+ _debug("Chopping sound, Ouch! RingBuffer full ?\n");
+ return _mainSndRingBuffer.Put(buffer, a, _defaultVolume);
+ }*/
return 0;
}
-void
+ void
AudioLayer::flushMain()
{
ost::MutexLock guard(_mutex);
_mainSndRingBuffer.flush();
}
-int
+ int
AudioLayer::putUrgent(void* buffer, int toCopy)
{
ost::MutexLock guard(_mutex);
- if (_stream) {
+ if ( hasStream() ) {
int a = _urgentRingBuffer.AvailForPut();
if ( a >= toCopy ) {
return _urgentRingBuffer.Put(buffer, toCopy, _defaultVolume);
@@ -427,44 +316,40 @@ AudioLayer::putUrgent(void* buffer, int toCopy)
return 0;
}
-int
+ int
AudioLayer::canGetMic()
{
- if (_stream) {
+ if ( hasStream() ) {
return _micRingBuffer.AvailForGet();
} else {
return 0;
}
}
-int
+ int
AudioLayer::getMic(void *buffer, int toCopy)
{
- if(_stream) {
+ if( hasStream() ) {
return _micRingBuffer.Get(buffer, toCopy, 100);
} else {
return 0;
}
}
-void
+ void
AudioLayer::flushMic()
{
_micRingBuffer.flush();
}
-bool
+ bool
AudioLayer::isStreamStopped (void)
{
ost::MutexLock guard(_mutex);
- try {
- if(_stream && _stream->isStopped()) {
- return true;
- }
- } catch (const portaudio::PaException &e) {
- _debugException("! AL: Portaudio error: isStopped returned an exception");
- }
- return false;
+ if(snd_pcm_state( _playback_handle ) == SND_PCM_STATE_XRUN)
+ return true;
+ else
+ return false;
}
void
@@ -475,19 +360,14 @@ AudioLayer::toggleEchoTesting() {
int
AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
- unsigned long framesPerBuffer,
- const PaStreamCallbackTimeInfo *timeInfo,
- PaStreamCallbackFlags statusFlags) {
+ unsigned long framesPerBuffer){
- (void) timeInfo;
- (void) statusFlags;
-
SFLDataFormat *in = (SFLDataFormat *) inputBuffer;
SFLDataFormat *out = (SFLDataFormat *) outputBuffer;
if (_echoTesting) {
memcpy(out, in, framesPerBuffer*sizeof(SFLDataFormat));
- return paContinue;
+ return framesPerBuffer*sizeof(SFLDataFormat);
}
int toGet;
@@ -519,9 +399,9 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
if (toGet) {
- _mainSndRingBuffer.Get(out, toGet, spkrVolume);
+ _mainSndRingBuffer.Get(out, toGet, spkrVolume);
} else {
- bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
+ bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
}
}
}
@@ -532,37 +412,24 @@ AudioLayer::audioCallback (const void *inputBuffer, void *outputBuffer,
//_debug("AL: Nb sample: %d char, [0]=%f [1]=%f [2]=%f\n", toPut, in[0], in[1], in[2]);
_micRingBuffer.Put(in, toPut, micVolume);
- return paContinue;
+ return toPut;
}
-int
-AudioLayer::miniAudioCallback (const void *inputBuffer, void *outputBuffer,
- unsigned long framesPerBuffer,
- const PaStreamCallbackTimeInfo *timeInfo,
- PaStreamCallbackFlags statusFlags) {
- (void) timeInfo;
- (void) statusFlags;
-
- _debug("mini audio callback!!\n");
-#ifdef SFL_TEST_SINE
- assert(outputBuffer != NULL);
-
- float *out = static_cast<float *>(outputBuffer);
- for (unsigned int i = 0; i < framesPerBuffer; ++i) {
- out[i] = table_[leftPhase_];
- leftPhase_ += 1;
- if (leftPhase_ >= tableSize_) {
- leftPhase_ -= tableSize_;
- }
- }
-#else
- SFLDataFormat *out = (SFLDataFormat *) outputBuffer;
- AudioLoop* tone = _manager->getTelephoneTone();
- if ( tone != 0) {
- //_debug("Frames Per Buffer: %d\n", framesPerBuffer);
- tone->getNext(out, framesPerBuffer, 100);
- }
-#endif
- return paContinue;
+
+ int
+AudioLayer::play_alsa(void* buffer, int length)
+{
+ if(_playback_handle == NULL)
+ return 0;
+
+ snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _playback_handle, length);
+ snd_pcm_prepare( _playback_handle );
+ if( snd_pcm_writei( _playback_handle, buffer, frames) < 0){
+ snd_pcm_prepare( _playback_handle );
+ _debug(" Buffer underrun!!!\n");
+ return 0;
+ }
+ return 1;
}
+
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 667b6eee5e..973eb6856f 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -25,13 +25,12 @@
#include <cc++/thread.h> // for ost::Mutex
-#include "portaudiocpp/PortAudioCpp.hxx"
-
#include "../global.h"
#include "ringbuffer.h"
#include "audiodevice.h"
#include <vector>
+#include <alsa/asoundlib.h>
#define FRAME_PER_BUFFER 160
@@ -39,119 +38,129 @@ class RingBuffer;
class ManagerImpl;
class AudioLayer {
- public:
- AudioLayer(ManagerImpl* manager);
- ~AudioLayer(void);
-
- /*
- * @param indexIn
- * @param indexOut
- * @param sampleRate
- * @param frameSize
- */
- void openDevice(int, int, int, int);
- void startStream(void);
- void stopStream(void);
- void sleep(int);
- bool hasStream(void);
- bool isStreamActive(void);
- bool isStreamStopped(void);
-
- void flushMain();
- int putMain(void* buffer, int toCopy);
- int putUrgent(void* buffer, int toCopy);
- int canGetMic();
- int getMic(void *, int);
- void flushMic();
-
- int audioCallback (const void *, void *, unsigned long,
- const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
- int miniAudioCallback (const void *, void *, unsigned long,
- const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
-
- void setErrorMessage(const std::string& error) { _errorMessage = error; }
- std::string getErrorMessage() { return _errorMessage; }
-
- /**
- * Get the sample rate of audiolayer
- * accessor only
- */
- int getIndexIn() { return _indexIn; }
- int getIndexOut() { return _indexOut; }
- unsigned int getSampleRate() { return _sampleRate; }
- unsigned int getFrameSize() { return _frameSize; }
- int getDeviceCount();
-
-
- // NOW
- void selectPreferedApi(PaHostApiTypeId apiTypeID, int& outputDeviceIndex, int& inputDeviceIndex);
-
- std::vector<std::string> getAudioDeviceList(PaHostApiTypeId apiTypeID, int ioDeviceMask);
-
-
- AudioDevice* getAudioDeviceInfo(int index, int ioDeviceMask);
-
- enum IODEVICE {InputDevice=0x01, OutputDevice=0x02 };
-
- /**
- * Toggle echo testing on/off
- */
- void toggleEchoTesting();
-
- private:
- void closeStream (void);
- RingBuffer _urgentRingBuffer;
- RingBuffer _mainSndRingBuffer;
- RingBuffer _micRingBuffer;
- ManagerImpl* _manager; // augment coupling, reduce indirect access
- // a audiolayer can't live without manager
-
- portaudio::MemFunCallbackStream<AudioLayer> *_stream;
-
- /**
- * Portaudio indexes of audio devices on which stream has been opened
- */
- int _indexIn;
- int _indexOut;
-
- /**
- * Sample Rate SFLphone should send sound data to the sound card
- * The value can be set in the user config file- now: 44100HZ
- */
- unsigned int _sampleRate;
-
- /**
- * Length of the sound frame we capture or read in ms
- * The value can be set in the user config file - now: 20ms
- */
- unsigned int _frameSize;
-
- /**
- * Input channel (mic) should be 1 mono
- */
- unsigned int _inChannel; // mic
-
- /**
- * Output channel (stereo) should be 1 mono
- */
- unsigned int _outChannel; // speaker
-
- /**
- * Default volume for incoming RTP and Urgent sounds.
- */
- unsigned short _defaultVolume; // 100
-
- /**
- * Echo testing or not
- */
- bool _echoTesting;
-
- std::string _errorMessage;
- ost::Mutex _mutex;
-
- float *table_;
- int tableSize_;
- int leftPhase_;
+ public:
+ AudioLayer(ManagerImpl* manager);
+ ~AudioLayer(void);
+
+ /*
+ * @param indexIn
+ * @param indexOut
+ * @param sampleRate
+ * @param frameSize
+ */
+ void openDevice(int, int, int, int);
+ void startStream(void);
+ void stopStream(void);
+ void sleep(int);
+ bool hasStream(void);
+ bool isStreamActive(void);
+ bool isStreamStopped(void);
+
+ void flushMain();
+ int putMain(void* buffer, int toCopy);
+ int putUrgent(void* buffer, int toCopy);
+ int canGetMic();
+ int getMic(void *, int);
+ void flushMic();
+
+ int audioCallback (const void *, void *, unsigned long);
+ //int miniAudioCallback (const void *, void *, unsigned long,
+ //const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
+
+ void setErrorMessage(const std::string& error) { _errorMessage = error; }
+ std::string getErrorMessage() { return _errorMessage; }
+
+ /**
+ * Get the sample rate of audiolayer
+ * accessor only
+ */
+ int getIndexIn() { return _indexIn; }
+ int getIndexOut() { return _indexOut; }
+ unsigned int getSampleRate() { return _sampleRate; }
+ unsigned int getFrameSize() { return _frameSize; }
+ int getDeviceCount();
+
+ void playSinusWave();
+
+
+ // NOW
+ //void selectPreferedApi(PaHostApiTypeId apiTypeID, int& outputDeviceIndex, int& inputDeviceIndex);
+
+ //std::vector<std::string> getAudioDeviceList(PaHostApiTypeId apiTypeID, int ioDeviceMask);
+
+
+ //AudioDevice* getAudioDeviceInfo(int index, int ioDeviceMask);
+
+ enum IODEVICE {InputDevice=0x01, OutputDevice=0x02 };
+
+ /**
+ * Toggle echo testing on/off
+ */
+ void toggleEchoTesting();
+
+ private:
+ void close_alsa (void);
+ int play_alsa( void* , int );
+ int device_info( void );
+ void init_hw_parameters( void );
+ void open_playback_device( std::string );
+ RingBuffer _urgentRingBuffer;
+ RingBuffer _mainSndRingBuffer;
+ RingBuffer _micRingBuffer;
+ ManagerImpl* _manager; // augment coupling, reduce indirect access
+ // a audiolayer can't live without manager
+
+ //portaudio::MemFunCallbackStream<AudioLayer> *_stream;
+
+ snd_pcm_t* _playback_handle;
+ snd_pcm_state_t* playback_state;
+ snd_pcm_t* _capture_handle;
+
+ int playback_callback(snd_pcm_sframes_t nframes);
+ /**
+ * Portaudio indexes of audio devices on which stream has been opened
+ */
+ int _indexIn;
+ int _indexOut;
+
+ /**
+ * Sample Rate SFLphone should send sound data to the sound card
+ * The value can be set in the user config file- now: 44100HZ
+ */
+ unsigned int _sampleRate;
+
+ /**
+ * Length of the sound frame we capture or read in ms
+ * The value can be set in the user config file - now: 20ms
+ */
+ unsigned int _frameSize;
+
+ /**
+ * Input channel (mic) should be 1 mono
+ */
+ unsigned int _inChannel; // mic
+
+ /**
+ * Output channel (stereo) should be 1 mono
+ */
+ unsigned int _outChannel; // speaker
+
+ /**
+ * Default volume for incoming RTP and Urgent sounds.
+ */
+ unsigned short _defaultVolume; // 100
+
+ /**
+ * Echo testing or not
+ */
+ bool _echoTesting;
+
+ std::string _errorMessage;
+ ost::Mutex _mutex;
+
+ float *table_;
+ int tableSize_;
+ int leftPhase_;
};
#endif // _AUDIO_LAYER_H_
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index ecad7b0880..9f9d8baf1a 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -1236,7 +1236,7 @@ std::vector<std::string>
ManagerImpl::getAudioOutputDeviceList(void)
{
_debug("Get audio output device list");
- return _audiodriver->getAudioDeviceList(paALSA, _audiodriver->OutputDevice);
+ //return _audiodriver->getAudioDeviceList(paALSA, _audiodriver->OutputDevice);
}
/**
@@ -1257,7 +1257,7 @@ std::vector<std::string>
ManagerImpl::getAudioInputDeviceList(void)
{
_debug("Get audio input device list");
- return _audiodriver->getAudioDeviceList(paALSA, _audiodriver->InputDevice);
+ //return _audiodriver->getAudioDeviceList(paALSA, _audiodriver->InputDevice);
}
/**
@@ -1296,7 +1296,7 @@ std::vector<std::string>
ManagerImpl::getAudioDeviceDetails(const int index)
{
_debug("Get audio input device list");
- std::vector<std::string> v;
+ /*std::vector<std::string> v;
try
{
@@ -1313,7 +1313,7 @@ ManagerImpl::getAudioDeviceDetails(const int index)
v.push_back(answer);
}
catch (...) {}
- return v;
+ return v;*/
}
@@ -1684,7 +1684,7 @@ ManagerImpl::getAudioDeviceList(const std::string& sequenceId, int ioDeviceMask)
TokenList tk;
AudioDevice* device = 0;
int nbDevice = audiolayer->getDeviceCount();
-
+ /*
for (int index = 0; index < nbDevice; index++ ) {
device = audiolayer->getAudioDeviceInfo(index, ioDeviceMask);
if (device != 0) {
@@ -1698,7 +1698,7 @@ ManagerImpl::getAudioDeviceList(const std::string& sequenceId, int ioDeviceMask)
// don't forget to delete it after
delete device; device = 0;
}
- }
+ }*/
returnValue = true;
std::ostringstream rate;
--
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