jack2/linux/alsa/JackAlsaAdapter.h

/*
Copyright (C) 2008 Grame

This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.

This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
GNU General Public License for more details.

You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.

*/

#ifndef __JackAlsaAdapter__
#define __JackAlsaAdapter__

#include <math.h>
#include <limits.h>
#include <assert.h>
#include <alsa/asoundlib.h>
#include "JackAudioAdapterInterface.h"
#include "JackPlatformThread.h"
#include "JackError.h"
#include "jack.h"

namespace Jack
{

    //inline void* aligned_calloc(size_t nmemb, size_t size) { return (void*)((size_t)(calloc((nmemb * size) + 15, sizeof(char))) + 15 & ~15); }
    inline void* aligned_calloc(size_t nmemb, size_t size) { return (void*)calloc(nmemb, size); }
    
    #define max(x,y) (((x)>(y)) ? (x) : (y))
    #define min(x,y) (((x)<(y)) ? (x) : (y))

    #define check_error(err) if (err) { jack_error("%s:%d, alsa error %d : %s", __FILE__, __LINE__, err, snd_strerror(err)); return err; }
    #define check_error_msg(err,msg) if (err) { jack_error("%s:%d, %s : %s(%d)", __FILE__, __LINE__, msg, snd_strerror(err), err); return err; }
    #define display_error_msg(err,msg) if (err) { jack_error("%s:%d, %s : %s(%d)", __FILE__, __LINE__, msg, snd_strerror(err), err); }

    /**
     * A convenient class to pass parameters to AudioInterface
     */
    class AudioParam
    {
    
      public:
                
        const char*     fCardName;					
        unsigned int	fFrequency;
        int             fBuffering; 
        
        unsigned int	fSoftInputs;
        unsigned int	fSoftOutputs;
        
      public:
      
        AudioParam() : 
            fCardName("hw:0"),
            fFrequency(44100),
            fBuffering(512),
            fSoftInputs(2),
            fSoftOutputs(2)
        {}
        
        AudioParam(int input, int output, jack_nframes_t buffer_size, jack_nframes_t sample_rate) : 
            fCardName("hw:0"),
            fFrequency(sample_rate),
            fBuffering(buffer_size),
            fSoftInputs(input),
            fSoftOutputs(output)
        {}
        
        AudioParam&	cardName(const char* n)	{ fCardName = n; 	return *this; }
        AudioParam&	frequency(int f)	{ fFrequency = f; 	return *this; }
        AudioParam&	buffering(int fpb)	{ fBuffering = fpb; 	return *this; }
        AudioParam&	inputs(int n)		{ fSoftInputs = n; 	return *this; }
        AudioParam&	outputs(int n)		{ fSoftOutputs = n; 	return *this; }
    };

    /**
     * An ALSA audio interface
     */
    class AudioInterface : public AudioParam
    {
    
     public:
     
        snd_pcm_t*		fOutputDevice;		
        snd_pcm_t*		fInputDevice;			
        snd_pcm_hw_params_t* 	fInputParams;
        snd_pcm_hw_params_t* 	fOutputParams;
        
        snd_pcm_format_t 	fSampleFormat;
        snd_pcm_access_t 	fSampleAccess;
        
        unsigned int	fCardInputs;
        unsigned int	fCardOutputs;
        
        unsigned int	fChanInputs;
        unsigned int	fChanOutputs;
        
        // interleaved mode audiocard buffers
        void*		fInputCardBuffer;
        void*		fOutputCardBuffer;
        
        // non interleaved mode audiocard buffers
        void*		fInputCardChannels[256];
        void*		fOutputCardChannels[256];
        
        // non interleaved mod, floating point software buffers
        float*		fInputSoftChannels[256];
        float*		fOutputSoftChannels[256];

     public:
     
        const char*	cardName()	{ return fCardName;  	}
        int		frequency()		{ return fFrequency; 	}
        int		buffering()		{ return fBuffering;  	}
        
        float**		inputSoftChannels()	{ return fInputSoftChannels;	}
        float**		outputSoftChannels()	{ return fOutputSoftChannels;	}
        
        AudioInterface(const AudioParam& ap = AudioParam()) : AudioParam(ap)
        {
            fInputDevice 			= 0;
            fOutputDevice 			= 0;
            fInputParams			= 0;
            fOutputParams			= 0;
        }

        AudioInterface(int input, int output, jack_nframes_t buffer_size, jack_nframes_t sample_rate) : 
            AudioParam(input, output, buffer_size, sample_rate)
        {
            fInputCardBuffer = 0;
            fOutputCardBuffer = 0;

            for (int i = 0; i < 256; i++) {
                fInputCardChannels[i] = 0;
                fOutputCardChannels[i] = 0;
                fInputSoftChannels[i] = 0;
                fOutputSoftChannels[i] = 0;
            }
        }
        
        /**
         * Open the audio interface
         */
        int open()
        {
            int err;
            // allocation d'un stream d'entree et d'un stream de sortie
            err = snd_pcm_open(&fInputDevice,  fCardName, SND_PCM_STREAM_CAPTURE, 0); 	check_error(err)
            err = snd_pcm_open(&fOutputDevice, fCardName, SND_PCM_STREAM_PLAYBACK, 0); 	check_error(err)

            // recherche des parametres d'entree
            err = snd_pcm_hw_params_malloc(&fInputParams); 	check_error(err);
            setAudioParams(fInputDevice, fInputParams);
            snd_pcm_hw_params_get_channels(fInputParams, &fCardInputs);

            // recherche des parametres de sortie
            err = snd_pcm_hw_params_malloc(&fOutputParams); check_error(err)
            setAudioParams(fOutputDevice, fOutputParams);
            snd_pcm_hw_params_get_channels(fOutputParams, &fCardOutputs);

            jack_info("inputs : %ud, outputs : %ud", fCardInputs, fCardOutputs);

            // enregistrement des parametres d'entree-sortie
            err = snd_pcm_hw_params(fInputDevice,  fInputParams);	 	check_error (err);
            err = snd_pcm_hw_params(fOutputDevice, fOutputParams);	check_error (err);

            // allocation of alsa buffers
            if (fSampleAccess == SND_PCM_ACCESS_RW_INTERLEAVED) {
                fInputCardBuffer = aligned_calloc(interleavedBufferSize(fInputParams), 1);
                fOutputCardBuffer = aligned_calloc(interleavedBufferSize(fOutputParams), 1);
            } else {
                for (unsigned int i = 0; i < fCardInputs; i++) {
                    fInputCardChannels[i] = aligned_calloc(noninterleavedBufferSize(fInputParams), 1);
                }
                for (unsigned int i = 0; i < fCardOutputs; i++) {
                    fOutputCardChannels[i] = aligned_calloc(noninterleavedBufferSize(fOutputParams), 1);
                }
            }
            
            // allocation of floating point buffers needed by the dsp code
            fChanInputs = max(fSoftInputs, fCardInputs);		assert (fChanInputs < 256);
            fChanOutputs = max(fSoftOutputs, fCardOutputs);		assert (fChanOutputs < 256);

            for (unsigned int i = 0; i < fChanInputs; i++) {
                fInputSoftChannels[i] = (float*) aligned_calloc (fBuffering, sizeof(float));
                for (int j = 0; j < fBuffering; j++) {
                    fInputSoftChannels[i][j] = 0.0;
                }
            }

            for (unsigned int i = 0; i < fChanOutputs; i++) {
                fOutputSoftChannels[i] = (float*) aligned_calloc (fBuffering, sizeof(float));
                for (int j = 0; j < fBuffering; j++) {
                    fOutputSoftChannels[i][j] = 0.0;
                }
            }
            return 0;
        }
        
        int close()
        {
            snd_pcm_hw_params_free(fInputParams);
            snd_pcm_hw_params_free(fOutputParams);
            snd_pcm_close(fInputDevice);
            snd_pcm_close(fOutputDevice);
 
            for (unsigned int i = 0; i < fChanInputs; i++) {
                if (fInputSoftChannels[i])
                    free(fInputSoftChannels[i]);
            }

            for (unsigned int i = 0; i < fChanOutputs; i++) {
                if (fOutputSoftChannels[i])
                    free(fOutputSoftChannels[i]);
            }

            for (unsigned int i = 0; i < fCardInputs; i++) {
                if (fInputCardChannels[i])
                    free(fInputCardChannels[i]);
            }

            for (unsigned int i = 0; i < fCardOutputs; i++) {
                if (fOutputCardChannels[i])
                    free(fOutputCardChannels[i]);
            }

            if (fInputCardBuffer)
                free(fInputCardBuffer);
            if (fOutputCardBuffer)
                free(fOutputCardBuffer);
	           
            return 0;
        }
        
        int setAudioParams(snd_pcm_t* stream, snd_pcm_hw_params_t* params)
        {	
            int	err;

            // set params record with initial values
            err = snd_pcm_hw_params_any	( stream, params ); 	
            check_error_msg(err, "unable to init parameters")

            // set alsa access mode (and fSampleAccess field) either to non interleaved or interleaved
                    
            err = snd_pcm_hw_params_set_access (stream, params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
            if (err) {
                err = snd_pcm_hw_params_set_access (stream, params, SND_PCM_ACCESS_RW_INTERLEAVED );
                check_error_msg(err, "unable to set access mode neither to non-interleaved or to interleaved");
            }
            snd_pcm_hw_params_get_access(params, &fSampleAccess);  

            // search for 32-bits or 16-bits format
            err = snd_pcm_hw_params_set_format (stream, params, SND_PCM_FORMAT_S32);
            if (err) {
                err = snd_pcm_hw_params_set_format (stream, params, SND_PCM_FORMAT_S16);
                check_error_msg(err, "unable to set format to either 32-bits or 16-bits");
            }
            snd_pcm_hw_params_get_format(params, &fSampleFormat);
            // set sample frequency
            snd_pcm_hw_params_set_rate_near (stream, params, &fFrequency, 0); 

            // set period and period size (buffering)
            err = snd_pcm_hw_params_set_period_size	(stream, params, fBuffering, 0); 	
            check_error_msg(err, "period size not available");
            
            err = snd_pcm_hw_params_set_periods (stream, params, 2, 0); 			
            check_error_msg(err, "number of periods not available");
            return 0;
        }

        ssize_t interleavedBufferSize(snd_pcm_hw_params_t* params)
        {
            _snd_pcm_format 	format;  	snd_pcm_hw_params_get_format(params, &format);
            snd_pcm_uframes_t 	psize;		snd_pcm_hw_params_get_period_size(params, &psize, NULL);
            unsigned int 		channels; 	snd_pcm_hw_params_get_channels(params, &channels);
            ssize_t bsize = snd_pcm_format_size(format, psize * channels);
            return bsize;
        }

        ssize_t noninterleavedBufferSize(snd_pcm_hw_params_t* params)
        {
            _snd_pcm_format 	format;  	snd_pcm_hw_params_get_format(params, &format);
            snd_pcm_uframes_t 	psize;		snd_pcm_hw_params_get_period_size(params, &psize, NULL);
            ssize_t bsize = snd_pcm_format_size(format, psize);
            return bsize;
        }

        /**
         * Read audio samples from the audio card. Convert samples to floats and take 
         * care of interleaved buffers
         */
        int read()
        {
            if (fSampleAccess == SND_PCM_ACCESS_RW_INTERLEAVED) {
                
                int count = snd_pcm_readi(fInputDevice, fInputCardBuffer, fBuffering); 	
                if (count<0) { 
                    display_error_msg(count, "reading samples");
                    int err = snd_pcm_prepare(fInputDevice);	
                    check_error_msg(err, "preparing input stream");
                }
                
                if (fSampleFormat == SND_PCM_FORMAT_S16) {

                    short* 	buffer16b = (short*) fInputCardBuffer;
                    for (int s = 0; s < fBuffering; s++) {
                        for (unsigned int c = 0; c < fCardInputs; c++) {
                            fInputSoftChannels[c][s] = float(buffer16b[c + s*fCardInputs])*(1.0/float(SHRT_MAX));
                        }
                    }

                } else { // SND_PCM_FORMAT_S32

                    long* 	buffer32b = (long*) fInputCardBuffer;
                    for (int s = 0; s < fBuffering; s++) {
                        for (unsigned int c = 0; c < fCardInputs; c++) {
                            fInputSoftChannels[c][s] = float(buffer32b[c + s*fCardInputs])*(1.0/float(LONG_MAX));
                        }
                    }
                }
                
            } else if (fSampleAccess == SND_PCM_ACCESS_RW_NONINTERLEAVED) {
                
                int count = snd_pcm_readn(fInputDevice, fInputCardChannels, fBuffering); 	
                if (count < 0) { 
                    display_error_msg(count, "reading samples");
                    int err = snd_pcm_prepare(fInputDevice);	
                    check_error_msg(err, "preparing input stream");
                }
                
                if (fSampleFormat == SND_PCM_FORMAT_S16) {

                    for (unsigned int c = 0; c < fCardInputs; c++) {
                        short* 	chan16b = (short*) fInputCardChannels[c];
                        for (int s = 0; s < fBuffering; s++) {
                            fInputSoftChannels[c][s] = float(chan16b[s])*(1.0/float(SHRT_MAX));
                        }
                    }

                } else { // SND_PCM_FORMAT_S32

                    for (unsigned int c = 0; c < fCardInputs; c++) {
                        long* 	chan32b = (long*) fInputCardChannels[c];
                        for (int s = 0; s < fBuffering; s++) {
                            fInputSoftChannels[c][s] = float(chan32b[s])*(1.0/float(LONG_MAX));
                        }
                    }
                }
                
            } else {
                check_error_msg(-10000, "unknow access mode");
            }

            return 0;
        }

        /**
         * write the output soft channels to the audio card. Convert sample 
         * format and interleaves buffers when needed
         */
        int write()
        {
            recovery:
                    
            if (fSampleAccess == SND_PCM_ACCESS_RW_INTERLEAVED) {
                
                if (fSampleFormat == SND_PCM_FORMAT_S16) {

                    short* buffer16b = (short*) fOutputCardBuffer;
                    for (int f = 0; f < fBuffering; f++) {
                        for (unsigned int c = 0; c < fCardOutputs; c++) {
                            float x = fOutputSoftChannels[c][f];
                            buffer16b[c + f * fCardOutputs] = short(max(min(x, 1.0), -1.0) * float(SHRT_MAX));
                        }
                    }

                } else { // SND_PCM_FORMAT_S32

                    long* buffer32b = (long*) fOutputCardBuffer;
                    for (int f = 0; f < fBuffering; f++) {
                        for (unsigned int c = 0; c < fCardOutputs; c++) {
                            float x = fOutputSoftChannels[c][f];
                            buffer32b[c + f * fCardOutputs] = long( max(min(x, 1.0), -1.0) * float(LONG_MAX));
                        }
                    }
                }

                int count = snd_pcm_writei(fOutputDevice, fOutputCardBuffer, fBuffering); 	
                if (count < 0) { 
                    display_error_msg(count, "w3"); 
                    int err = snd_pcm_prepare(fOutputDevice);	
                    check_error_msg(err, "preparing output stream");
                    goto recovery;
                }
                
                
            } else if (fSampleAccess == SND_PCM_ACCESS_RW_NONINTERLEAVED) {
                
                if (fSampleFormat == SND_PCM_FORMAT_S16) {

                    for (unsigned int c = 0; c < fCardOutputs; c++) {
                        short* chan16b = (short*) fOutputCardChannels[c];
                        for (int f = 0; f < fBuffering; f++) {
                            float x = fOutputSoftChannels[c][f];
                            chan16b[f] = short(max(min(x,1.0), -1.0) * float(SHRT_MAX)) ;
                        }
                    }

                } else { // SND_PCM_FORMAT_S32

                    for (unsigned int c = 0; c < fCardOutputs; c++) {
                        long* chan32b = (long*) fOutputCardChannels[c];
                        for (int f = 0; f < fBuffering; f++) {
                            float x = fOutputSoftChannels[c][f];
                            chan32b[f] = long(max(min(x,1.0),-1.0) * float(LONG_MAX)) ;
                        }
                    }
                }

                int count = snd_pcm_writen(fOutputDevice, fOutputCardChannels, fBuffering); 	
                if (count<0) { 
                    display_error_msg(count, "w3"); 
                    int err = snd_pcm_prepare(fOutputDevice);	
                    check_error_msg(err, "preparing output stream");
                    goto recovery;
                }
                
            } else {
                check_error_msg(-10000, "unknow access mode");
            }

            return 0;
        }
      
        /**
         *  print short information on the audio device
         */
        int shortinfo()
        {
            int	err;
            snd_ctl_card_info_t* card_info;
            snd_ctl_t* ctl_handle;
            err = snd_ctl_open(&ctl_handle, fCardName, 0);	check_error(err);
            snd_ctl_card_info_alloca(&card_info);
            err = snd_ctl_card_info(ctl_handle, card_info);	check_error(err);
            jack_info("%s|%d|%d|%d|%d|%s", 
                    snd_ctl_card_info_get_driver(card_info),
                    fCardInputs, fCardOutputs,
                    fFrequency, fBuffering,
                    snd_pcm_format_name((_snd_pcm_format)fSampleFormat));
        }
                        
        /**
         *  print more detailled information on the audio device
         */
        int longinfo()
        {
            int	err;
            snd_ctl_card_info_t* card_info;
            snd_ctl_t* ctl_handle;

            jack_info("Audio Interface Description :");
            jack_info("Sampling Frequency : %d, Sample Format : %s, buffering : %d", 
                    fFrequency, snd_pcm_format_name((_snd_pcm_format)fSampleFormat), fBuffering);
            jack_info("Software inputs : %2d, Software outputs : %2d", fSoftInputs, fSoftOutputs);
            jack_info("Hardware inputs : %2d, Hardware outputs : %2d", fCardInputs, fCardOutputs);
            jack_info("Channel inputs  : %2d, Channel outputs  : %2d", fChanInputs, fChanOutputs);
            
            // affichage des infos de la carte
            err = snd_ctl_open (&ctl_handle, fCardName, 0);		check_error(err);
            snd_ctl_card_info_alloca (&card_info);
            err = snd_ctl_card_info(ctl_handle, card_info);		check_error(err);
            printCardInfo(card_info);

            // affichage des infos liees aux streams d'entree-sortie
            if (fSoftInputs > 0)	printHWParams(fInputParams);
            if (fSoftOutputs > 0)	printHWParams(fOutputParams);
            return 0;
        }
        
        void printCardInfo(snd_ctl_card_info_t*	ci)
        {
            jack_info("Card info (address : %p)", ci);
            jack_info("\tID         = %s", snd_ctl_card_info_get_id(ci));
            jack_info("\tDriver     = %s", snd_ctl_card_info_get_driver(ci));
            jack_info("\tName       = %s", snd_ctl_card_info_get_name(ci));
            jack_info("\tLongName   = %s", snd_ctl_card_info_get_longname(ci));
            jack_info("\tMixerName  = %s", snd_ctl_card_info_get_mixername(ci));
            jack_info("\tComponents = %s", snd_ctl_card_info_get_components(ci));
            jack_info("--------------");
        }

        void printHWParams(snd_pcm_hw_params_t* params)
        {
            jack_info("HW Params info (address : %p)\n", params);
    #if 0
            jack_info("\tChannels    = %d", snd_pcm_hw_params_get_channels(params));
            jack_info("\tFormat      = %s", snd_pcm_format_name((_snd_pcm_format)snd_pcm_hw_params_get_format(params)));
            jack_info("\tAccess      = %s", snd_pcm_access_name((_snd_pcm_access)snd_pcm_hw_params_get_access(params)));
            jack_info("\tRate        = %d", snd_pcm_hw_params_get_rate(params, NULL));
            jack_info("\tPeriods     = %d", snd_pcm_hw_params_get_periods(params, NULL));
            jack_info("\tPeriod size = %d", (int)snd_pcm_hw_params_get_period_size(params, NULL));
            jack_info("\tPeriod time = %d", snd_pcm_hw_params_get_period_time(params, NULL));
            jack_info("\tBuffer size = %d", (int)snd_pcm_hw_params_get_buffer_size(params));
            jack_info("\tBuffer time = %d", snd_pcm_hw_params_get_buffer_time(params, NULL));
    #endif
            jack_info("--------------");
        }
    };

	class JackAlsaAdapter : public JackAudioAdapterInterface, public JackRunnableInterface
	{
    
	    private:
        	
            JackThread fThread;
            AudioInterface fAudioInterface;
       
	    public:
        
            JackAlsaAdapter(int input, int output, jack_nframes_t buffer_size, jack_nframes_t sample_rate)
                :JackAudioAdapterInterface(input, output, buffer_size, sample_rate)
                ,fThread(this), fAudioInterface(input, output, buffer_size, sample_rate)	
            {}
           
            ~JackAlsaAdapter()
            {}
            
            virtual int Open();
            virtual int Close();
             
            virtual int SetBufferSize(jack_nframes_t buffer_size);
            
            virtual bool Init();
            virtual bool Execute();
             
   	};
}

#endif