sbitx_sound.c

#include <stdio.h>
#include <alsa/asoundlib.h>
#include <pthread.h>
#include <complex.h>
#include <fftw3.h>
#include <time.h>
#include "sound.h"
#include "wiringPi.h"
#include "sdr.h"


/* follows the tutorial at http://alsamodular.sourceforge.net/alsa_programming_howto.html
Next thing to try is http://www.saunalahti.fi/~s7l/blog/2005/08/21/Full%20Duplex%20ALSA

	We are using 4 bytes per sample, 
	each frame is consists of two channels of audio, hence 8 bytes 
  We are shooting for 1024x2 = 2048 samples per period. that is 8K
  At two periods in the buffer, the buffer has to be 16K

	To simply the work, we are picking up some settings for the Wolfson codec
	as it connects to a raspberry pi. These values are interdependent
	and they will work out of the box. It takes the guess work out of
	configuring the Raspberry Pi with Wolfson codec.
*/


/*
	MIXER api

	https://alsa.opensrc.org/HowTo_access_a_mixer_control

https://android.googlesource.com/platform/hardware/qcom/audio/+/jb-mr1-dev/alsa_sound/ALSAMixer.cpp

https://github.com/bear24rw/alsa-utils/blob/master/amixer/amixer.c

There are six kinds of controls:
	playback volume 
	playback switch
	playback enumeration
	capture volume
	capture switch
	capture enumeration

examples of using amixer to mute and unmute:
amixer -c 1  set 'Output Mixer Mic Sidetone' unmute
amixer -c 1  set 'Output Mixer Mic Sidetone' mute


examples of using sound_mixer function:
'Mic' 0/1 = mute/unmute the mic
'Line' 0/1= mute/unmute the line in
'Master' 0-100 controls the earphone volume only, line out remains unaffected
'Input Mux' 1/0 take the input either from the Mic or Line In


*/

void sound_volume(char *card_name, char *element, int volume)
{
    long min, max;
    snd_mixer_t *handle;
    snd_mixer_selem_id_t *sid;
		char *card;

		card = card_name;
    snd_mixer_open(&handle, 0);
    snd_mixer_attach(handle, card);
    snd_mixer_selem_register(handle, NULL, NULL);
    snd_mixer_load(handle);

    snd_mixer_selem_id_alloca(&sid);
    snd_mixer_selem_id_set_index(sid, 0);
    snd_mixer_selem_id_set_name(sid, element);
    snd_mixer_elem_t* elem = snd_mixer_find_selem(handle, sid);

    snd_mixer_selem_get_playback_volume_range(elem, &min, &max);
    snd_mixer_selem_set_playback_volume_all(elem, volume * max / 100);

    snd_mixer_close(handle);
}

void sound_mixer(char *card_name, char *element, int make_on)
{
    long min, max;
    snd_mixer_t *handle;
    snd_mixer_selem_id_t *sid;
    char *card = card_name;

    snd_mixer_open(&handle, 0);
    snd_mixer_attach(handle, card);
    snd_mixer_selem_register(handle, NULL, NULL);
    snd_mixer_load(handle);

    snd_mixer_selem_id_alloca(&sid);
    snd_mixer_selem_id_set_index(sid, 0);
    snd_mixer_selem_id_set_name(sid, element);
    snd_mixer_elem_t* elem = snd_mixer_find_selem(handle, sid);

/*		if (elem)
			puts("Element found.");	
	*/
    //find out if the his element is capture side or plaback
    if(snd_mixer_selem_has_capture_switch(elem)){
			//puts("this is a capture switch.");  
	  	snd_mixer_selem_set_capture_switch_all(elem, make_on);
		}
    else if (snd_mixer_selem_has_playback_switch(elem)){
		//	puts("this is a playback switch.");
			snd_mixer_selem_set_playback_switch_all(elem, make_on);
		}
    else if (snd_mixer_selem_has_playback_volume(elem)){
			//puts("this is  playback volume");
			long volume = make_on;
    	snd_mixer_selem_get_playback_volume_range(elem, &min, &max);
    	snd_mixer_selem_set_playback_volume_all(elem, volume * max / 100);
    }	
    else if (snd_mixer_selem_has_capture_volume(elem)){
		//	puts("this is a capture volume");
			long volume = make_on;
    	snd_mixer_selem_get_capture_volume_range(elem, &min, &max);
    	snd_mixer_selem_set_capture_volume_all(elem, volume * max / 100);
    }
		else if (snd_mixer_selem_is_enumerated(elem)){
//			puts("TBD: this is an enumerated capture element");
			snd_mixer_selem_set_enum_item(elem, 0, make_on);
		}
    snd_mixer_close(handle);
}

int rate = 96000; /* Sample rate */
static snd_pcm_uframes_t buff_size = 8192; /* Periodsize (bytes) */
static int n_periods_per_buffer = 2;       /* Number of periods */
//static int n_periods_per_buffer = 1024;       /* Number of periods */

static snd_pcm_t *pcm_play_handle=0;   	//handle for the pcm device
static snd_pcm_t *pcm_capture_handle=0;   	//handle for the pcm device
static snd_pcm_t *loopback_play_handle=0;   	//handle for the pcm device
static snd_pcm_t *loopback_capture_handle=0;   	//handle for the pcm device

static snd_pcm_stream_t play_stream = SND_PCM_STREAM_PLAYBACK;	//playback stream
static snd_pcm_stream_t capture_stream = SND_PCM_STREAM_CAPTURE;	//playback stream

static char	*pcm_play_name, *pcm_capture_name;
static snd_pcm_hw_params_t *hwparams;
static snd_pcm_sw_params_t *swparams;
static snd_pcm_hw_params_t *hloop_params;
static snd_pcm_sw_params_t *sloop_params;
static int exact_rate;   /* Sample rate returned by */
static int	sound_thread_continue = 0;
pthread_t sound_thread, loopback_thread;

#define LOOPBACK_LEVEL_DIVISOR 8				// Constant used to reduce audio level to the loopback channel (FLDIGI)
static int play_write_error = 0;				// count play channel write errors
static int loopback_write_error = 0;			// count loopback channel write errors
// Note: Error messages appear when the sbitx program is started from the command line

int use_virtual_cable = 0;

struct Queue qloop;

/* this function should be called just once in the application process.
Calling it frequently will result in more allocation of hw_params memory blocks
without releasing them.
The list of PCM devices available on any platform can be found by running
	aplay -L 
We have to pass the id of one of those devices to this function.
The sequence of the alsa functions must be maintained for this to work consistently

It returns a -1 if the device didn't open. The error message is on stderr.

IMPORTANT:
The sound is playback is carried on in a non-blocking way  

*/

int sound_start_play(char *device){
	//found out the correct device through aplay -L (for pcm devices)

	snd_pcm_hw_params_alloca(&hwparams);	//more alloc

	//puts a playback handle into the pointer to the pointer
	int e = snd_pcm_open(&pcm_play_handle, device, play_stream, SND_PCM_NONBLOCK);
	
	if (e < 0) {
		fprintf(stderr, "Error opening PCM playback device %s: %s\n", device, snd_strerror(e));
		return -1;
	}

	//fills up the hwparams with values, hwparams was allocated above
	e = snd_pcm_hw_params_any(pcm_play_handle, hwparams);

	if (e < 0) {
		fprintf(stderr, "*Error getting hw playback params (%d)\n", e);
		return(-1);
	}

	// set the pcm access to interleaved
	e = snd_pcm_hw_params_set_access(pcm_play_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (e < 0) {
		fprintf(stderr, "*Error setting playback access.\n");
		return(-1);
	}

  /* Set sample format */
	e = snd_pcm_hw_params_set_format(pcm_play_handle, hwparams, SND_PCM_FORMAT_S32_LE);
	if (e < 0) {
		fprintf(stderr, "*Error setting plyaback format.\n");
		return(-1);
	}


	/* Set sample rate. If the exact rate is not supported */
	/* by the hardware, use nearest possible rate.         */ 
	exact_rate = rate;
	e = snd_pcm_hw_params_set_rate_near(pcm_play_handle, hwparams, &exact_rate, 0);
	if ( e< 0) {
		fprintf(stderr, "Error setting playback rate.\n");
		return(-1);
	}
	if (rate != exact_rate)
		fprintf(stderr, "*The playback rate %d changed to %d Hz\n", rate, exact_rate);
/*	else
		fprintf(stderr, "Playback sampling rate is set to %d\n", exact_rate);
*/

	/* Set number of channels */
	if ((e = snd_pcm_hw_params_set_channels(pcm_play_handle, hwparams, 2)) < 0) {
		fprintf(stderr, "*Error setting playback channels.\n");
		return(-1);
	}


	// frame = bytes_per_sample x n_channel
	// period = frames transfered at a time (160 for voip, etc.)
	// we use two periods per buffer.
	if ((e = snd_pcm_hw_params_set_periods(pcm_play_handle, hwparams, n_periods_per_buffer, 0)) < 0) {
		fprintf(stderr, "*Error setting playback periods.\n");
		return(-1);
	}


	// the buffer size is each periodsize x n_periods
	snd_pcm_uframes_t  n_frames= (buff_size  * n_periods_per_buffer)/8;
	//printf("trying for buffer size of %ld\n", n_frames);
	e = snd_pcm_hw_params_set_buffer_size_near(pcm_play_handle, hwparams, &n_frames);
	if (e < 0) {
		    fprintf(stderr, "*Error setting playback buffersize.\n");
		    return(-1);
	}

	if (snd_pcm_hw_params(pcm_play_handle, hwparams) < 0) {
		fprintf(stderr, "*Error setting playback HW params.\n");
		return(-1);
	}
//	puts("All hw params set to play sound");

	return 0;
}


int sound_start_loopback_capture(char *device){

	snd_pcm_hw_params_alloca(&hloop_params);
	//printf ("opening audio tx stream to %s\n", device); 
	int e = snd_pcm_open(&loopback_capture_handle, device, capture_stream, 0);
	
	if (e < 0) {
		fprintf(stderr, "Err: Opening loop capture  %s: %s\n", device, snd_strerror(e));
		return -1;
	}

	e = snd_pcm_hw_params_any(loopback_capture_handle, hloop_params);

	if (e < 0) {
		fprintf(stderr, "*Error setting capture access (%d)\n", e);
		return(-1);
	}

	e = snd_pcm_hw_params_set_access(loopback_capture_handle, hloop_params, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (e < 0) {
		fprintf(stderr, "*Error setting capture access.\n");
		return(-1);
	}

  /* Set sample format */
	e = snd_pcm_hw_params_set_format(loopback_capture_handle, hloop_params, SND_PCM_FORMAT_S32_LE);
	if (e < 0) {
		fprintf(stderr, "*Error setting loopback capture format.\n");
		return(-1);
	}

	/* Set sample rate. If the exact rate is not supported */
	/* by the hardware, use nearest possible rate.         */ 
	exact_rate = 48000;
	//printf("Setting loopback capture rate to %d\n", exact_rate);
	e = snd_pcm_hw_params_set_rate_near(loopback_capture_handle, hloop_params, &exact_rate, 0);
	if ( e< 0) {
		fprintf(stderr, "*Error setting loopback capture rate.\n");
		return(-1);
	}

	if (48000 != exact_rate)
		fprintf(stderr, "#The loopback capture rate set to %d Hz\n", exact_rate);

	/* Set number of channels */
	if ((e = snd_pcm_hw_params_set_channels(loopback_capture_handle, hloop_params, 2)) < 0) {
		fprintf(stderr, "*Error setting loopback capture channels.\n");
		return(-1);
	}

	//printf("%d: set the #channels\n", __LINE__, 2);
	/* Set number of periods. Periods used to be called fragments. */ 
	if ((e = snd_pcm_hw_params_set_periods(loopback_capture_handle, hloop_params, n_periods_per_buffer, 0)) < 0) {
		fprintf(stderr, "*Error setting loopback capture periods.\n");
		return(-1);
	}

	// the buffer size is each periodsize x n_periods
	snd_pcm_uframes_t  n_frames= (buff_size  * n_periods_per_buffer)/ 8;
	//printf("trying for buffer size of %ld\n", n_frames);
	e = snd_pcm_hw_params_set_buffer_size_near(loopback_capture_handle, hloop_params, &n_frames);
	if (e < 0) {
		    fprintf(stderr, "*Error setting loopback capture buffersize.\n");
		    return(-1);
	}

	//printf("%d: set buffer to \n", __LINE__, n_frames);
	if (snd_pcm_hw_params(loopback_capture_handle, hloop_params) < 0) {
		fprintf(stderr, "*Error setting capture HW params.\n");
		return(-1);
	}

	//printf("%d: set  hwparams\n", __LINE__);
	/* set some parameters in the driver to handle the latencies */
	snd_pcm_sw_params_malloc(&sloop_params);
	if((e = snd_pcm_sw_params_current(loopback_capture_handle, sloop_params)) < 0){
		fprintf(stderr, "Error getting current loopback capture sw params : %s\n", snd_strerror(e));
		return (-1);
	}
	
	if ((e = snd_pcm_sw_params_set_start_threshold(loopback_capture_handle, sloop_params, 15)) < 0){
		fprintf(stderr, "Unable to set threshold mode for loopback capture\n");
	} 
	
	if ((e = snd_pcm_sw_params_set_stop_threshold(loopback_capture_handle, sloop_params, 1)) < 0){

		fprintf(stderr, "Unable to set stop threshold for loopback  capture\n");
	}
	return 0;
}

/*
The capture is opened in a blocking mode, the read function will block until 
there are enough samples to return a block.
This ensures that the blocks are returned in perfect timing with the codec's clock
Once you process these captured samples and send them to the playback device, you
just wait for the next block to arrive 
*/

int sound_start_capture(char *device){
	snd_pcm_hw_params_alloca(&hwparams);

	int e = snd_pcm_open(&pcm_capture_handle, device,  	capture_stream, 0);
	
	if (e < 0) {
		fprintf(stderr, "Error opening PCM capture device %s: %s\n", pcm_capture_name, snd_strerror(e));
		return -1;
	}

	e = snd_pcm_hw_params_any(pcm_capture_handle, hwparams);

	if (e < 0) {
		fprintf(stderr, "*Error setting capture access (%d)\n", e);
		return(-1);
	}

	e = snd_pcm_hw_params_set_access(pcm_capture_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (e < 0) {
		fprintf(stderr, "*Error setting capture access.\n");
		return(-1);
	}

  /* Set sample format */
	e = snd_pcm_hw_params_set_format(pcm_capture_handle, hwparams, SND_PCM_FORMAT_S32_LE);
	if (e < 0) {
		fprintf(stderr, "*Error setting capture format.\n");
		return(-1);
	}


	/* Set sample rate. If the exact rate is not supported */
	/* by the hardware, use nearest possible rate.         */ 
	exact_rate = rate;
	e = snd_pcm_hw_params_set_rate_near(pcm_capture_handle, hwparams, &exact_rate, 0);
	if ( e< 0) {
		fprintf(stderr, "*Error setting capture rate.\n");
		return(-1);
	}

	if (rate != exact_rate)
		fprintf(stderr, "#The capture rate %d changed to %d Hz\n", rate, exact_rate);


	/* Set number of channels */
	if ((e = snd_pcm_hw_params_set_channels(pcm_capture_handle, hwparams, 2)) < 0) {
		fprintf(stderr, "*Error setting capture channels.\n");
		return(-1);
	}


	/* Set number of periods. Periods used to be called fragments. */ 
	if ((e = snd_pcm_hw_params_set_periods(pcm_capture_handle, hwparams, n_periods_per_buffer, 0)) < 0) {
		fprintf(stderr, "*Error setting capture periods.\n");
		return(-1);
	}


	// the buffer size is each periodsize x n_periods
	snd_pcm_uframes_t  n_frames= (buff_size  * n_periods_per_buffer)/ 8;
	//printf("trying for buffer size of %ld\n", n_frames);
	e = snd_pcm_hw_params_set_buffer_size_near(pcm_play_handle, hwparams, &n_frames);
	if (e < 0) {
		    fprintf(stderr, "*Error setting capture buffersize.\n");
		    return(-1);
	}

	if (snd_pcm_hw_params(pcm_capture_handle, hwparams) < 0) {
		fprintf(stderr, "*Error setting capture HW params.\n");
		return(-1);
	}

	return 0;
}

int sound_start_loopback_play(char *device){
	//found out the correct device through aplay -L (for pcm devices)

	snd_pcm_hw_params_alloca(&hwparams);	//more alloc

	//printf ("opening audio rx stream to %s\n", device); 
	int e = snd_pcm_open(&loopback_play_handle, device, play_stream, SND_PCM_NONBLOCK);
	
	if (e < 0) {
		fprintf(stderr, "Error opening loopback playback device %s: %s\n", device, snd_strerror(e));
		return -1;
	}

	e = snd_pcm_hw_params_any(loopback_play_handle, hwparams);

	if (e < 0) {
		fprintf(stderr, "*Error getting loopback playback params (%d)\n", e);
		return(-1);
	}

	e = snd_pcm_hw_params_set_access(loopback_play_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (e < 0) {
		fprintf(stderr, "*Error setting loopback access.\n");
		return(-1);
	}

  /* Set sample format */
	e = snd_pcm_hw_params_set_format(loopback_play_handle, hwparams, SND_PCM_FORMAT_S32_LE);
	if (e < 0) {
		fprintf(stderr, "*Error setting loopback format.\n");
		return(-1);
	}

	/* Set sample rate. If the exact rate is not supported */
	/* by the hardware, use nearest possible rate.         */ 
	exact_rate = 48000;
	e = snd_pcm_hw_params_set_rate_near(loopback_play_handle, hwparams, &exact_rate, 0);
	if ( e< 0) {
		fprintf(stderr, "Error setting playback rate.\n");
		return(-1);
	}
	if (48000 != exact_rate)
		fprintf(stderr, "*The loopback playback rate %d changed to %d Hz\n", rate, exact_rate);


	/* Set number of channels */
	if ((e = snd_pcm_hw_params_set_channels(loopback_play_handle, hwparams, 2)) < 0) {
		fprintf(stderr, "*Error setting playback channels.\n");
		return(-1);
	}


	// frame = bytes_per_sample x n_channel
	// period = frames transfered at a time (160 for voip, etc.)
	// we use two periods per buffer.
	if ((e = snd_pcm_hw_params_set_periods(loopback_play_handle, hwparams, 8, 0)) < 0) {
		fprintf(stderr, "*Error setting playback periods.\n");
		return(-1);
	}


	// the buffer size is each periodsize x n_periods
	snd_pcm_uframes_t  n_frames= (buff_size  * n_periods_per_buffer)/8;
	//lets pump it up to see if we can reduce the dropped frames
	n_frames *= 4;
	//printf("trying for loopback buffer size of %ld\n", n_frames);
	e = snd_pcm_hw_params_set_buffer_size_near(loopback_play_handle, hwparams, &n_frames);
	if (e < 0) {
		    fprintf(stderr, "*Error setting loopback playback buffersize.\n");
		    return(-1);
	}

	//printf("loopback playback buffer size is set to %d\n", n_frames);

	if (snd_pcm_hw_params(loopback_play_handle, hwparams) < 0) {
		fprintf(stderr, "*Error setting loopback playback HW params.\n");
		return(-1);
	}

	return 0;
}

// this is only a test process to be substituted to try loopback 
// it was used to debug timing errors
void sound_process2(int32_t *input_i, int32_t *input_q, int32_t *output_i, int32_t *output_q, int n_samples){
 
	for (int i= 0; i < n_samples; i++){
		output_i[i] = input_q[i];
		output_q[i] = 0;
	}	
}

//check that we haven't free()-ed up the hwparams block
//don't call this function at all until that is fixed
//you don't have to call it anyway
void sound_stop(){
	snd_pcm_drop(pcm_play_handle);
	snd_pcm_drain(pcm_play_handle);

	snd_pcm_drop(pcm_capture_handle);
	snd_pcm_drain(pcm_capture_handle);
}

static int count = 0;
static struct timespec gettime_now;
static long int last_time = 0;
static long int last_sec = 0;
static int nframes = 0;
int32_t	resample_in[10000];
int32_t	resample_out[10000];

int last_second = 0;
int nsamples = 0;
int	played_samples = 0;
int sound_loop(){
	int32_t		*line_in, *line_out, *data_in, *data_out, 
						*input_i, *output_i, *input_q, *output_q;
  int pcmreturn, i, j, loopreturn;
  short s1, s2;
  int frames;

	//we allocate enough for two channels of int32_t sized samples	
  data_in = (int32_t *)malloc(buff_size * 2);
  line_in = (int32_t *)malloc(buff_size * 2);
  line_out = (int32_t *)malloc(buff_size * 2);
  data_out = (int32_t *)malloc(buff_size * 2);
  input_i = (int32_t *)malloc(buff_size * 2);
  output_i = (int32_t *)malloc(buff_size * 2);
  input_q = (int32_t *)malloc(buff_size * 2);
  output_q = (int32_t *)malloc(buff_size * 2);

  frames = buff_size / 8;
	
  snd_pcm_prepare(pcm_play_handle);
  snd_pcm_prepare(loopback_play_handle);
  snd_pcm_writei(pcm_play_handle, data_out, frames);
  snd_pcm_writei(pcm_play_handle, data_out, frames);

	//Note: the virtual cable samples queue should be flushed at the start of tx
 	qloop.stall = 1;

  while(sound_thread_continue) {

		//restart the pcm capture if there is an error reading the samples
		//this is opened as a blocking device, hence we derive accurate timing 

		last_time = gettime_now.tv_nsec/1000;


		while ((pcmreturn = snd_pcm_readi(pcm_capture_handle, data_in, frames)) < 0){
			snd_pcm_prepare(pcm_capture_handle);
			//putchar('=');
		}
		i = 0; 
		j = 0;
		int ret_card = pcmreturn;
    if (use_virtual_cable){

			//printf(" we have %d in qloop, writing now\n", q_length(&qloop));
			// if don't we have enough to last two iterations loop back...
			if (q_length(&qloop) < pcmreturn){
//				puts(" skipping");
				continue;
			}
	
			//copy 1024 samples from the queue.
			i = 0;
			j = 0;

			
			for (int samples  = 0; samples < 1024; samples++){
				int32_t s = q_read(&qloop);
				input_i[j] = input_q[j] = s;
				j++; 
			}
			played_samples += 1024;
		}
		else {
			while (i < ret_card){
				input_i[i] = data_in[j++]/2;
				input_q[i] = data_in[j++]/2;
				i++;
			}
		}

		//printf("%d %ld %d\n", count++, nsamples, pcmreturn);
			
		sound_process(input_i, input_q, output_i, output_q, ret_card);

		i = 0; 
		j = 0;	
		while (i < ret_card){
			data_out[j++] = output_i[i];
			data_out[j++] = output_q[i++];
		}

/*
	// This is the original pcm play write routine, now commented out.
    while ((pcmreturn = snd_pcm_writei(pcm_play_handle, 
			data_out, frames)) < 0) {
       snd_pcm_prepare(pcm_play_handle);
    }
*/

	// This is the new pcm play write routine

	int framesize = ret_card;
	int offset = 0;
		
	while(framesize > 0)
	{
		pcmreturn = snd_pcm_writei(pcm_play_handle, data_out + offset, framesize);
		if((pcmreturn < 0) && (pcmreturn != -11))	// also ignore "temporarily unavailable" errors
		{
			// Handle an error condition from the snd_pcm_writei function
//			printf("Play PCM Write Error: %s  count = %d\n",snd_strerror(pcmreturn), play_write_error++);
			snd_pcm_prepare(pcm_play_handle);		
		}
		
		if(pcmreturn >= 0)
		{
			// Calculate remaining number of samples to be sent and new position in sample array.
			// If all the samples were processed by the snd_pcm_writei function then framesize will be
			// zero and the while() loop will end.
			framesize -= pcmreturn;
			offset += (pcmreturn * 2);
		}
	}
	// End of new pcm play write routine


	//decimate the line out to half, ie from 96000 to 48000
	//play the received data (from left channel) to both of line out
		
	int jj = 0;
	int ii = 0;
	while (ii < ret_card){
		line_out[jj++] = output_i[ii] / LOOPBACK_LEVEL_DIVISOR;  // Left Channel. Reduce audio level to FLDIGI a bit
		line_out[jj++] = output_i[ii] / LOOPBACK_LEVEL_DIVISOR;  // Right Channel. Note: FLDIGI does not use the this channel.
		// The right channel can be used to output other integer values such as AGC, for capture by an
		// application such as audacity.
		ii += 2;	// Skip a pair of samples to account for the 96K sample to 48K sample rate change.
	}

/*
	// This is the original pcm loopback write routine, now commented out.
    while((pcmreturn = snd_pcm_writei(loopback_play_handle, 
			 line_out, jj)) < 0){
			 //printf("loopback rx error: %s\n", snd_strerror(pcmreturn));
       snd_pcm_prepare(loopback_play_handle);
			//puts("preparing loopback");
    }
*/    

	// This is the new pcm loopback write routine
	framesize = (ret_card + 1) /2;		// only writing half the number of samples because of the slower channel rate
	offset = 0;

	while(framesize > 0)
	{
		pcmreturn = snd_pcm_writei(loopback_play_handle, line_out + offset, framesize);
		if(pcmreturn < 0)
		{
//			printf("Loopback PCM Write Error: %s  count = %d\n",snd_strerror(pcmreturn), loopback_write_error++);
			// Handle an error condition from the snd_pcm_writei function
			snd_pcm_prepare(loopback_play_handle);
		}
		
		if(pcmreturn >= 0)
		{
			// Calculate remaining number of samples to be sent and new position in sample array.
			// If all the samples were processed by the snd_pcm_writei function then framesize will be
			// zero and the while() loop will end.	
			framesize -= pcmreturn;
			offset += (pcmreturn * 2);
		}
	}
	// End of new pcm loopback write routine	
	
    
		//played_samples += pcmreturn;
  }
	//fclose(pf);
  printf("********Ending sound thread\n");
}


int loopback_loop(){
	int32_t		*line_in, *line_out, *data_in, *data_out, 
						*input_i, *output_i, *input_q, *output_q;
  int pcmreturn, i, j, loopreturn;
  short s1, s2;
  int frames;

	//we allocate enough for two channels of int32_t sized samples	
  data_in = (int32_t *)malloc(buff_size * 2);
  frames = buff_size / 8;
  snd_pcm_prepare(loopback_capture_handle);

  while(sound_thread_continue) {

		//restart the pcm capture if there is an error reading the samples
		//this is opened as a blocking device, hence we derive accurate timing 

		last_time = gettime_now.tv_nsec/1000;

		while ((pcmreturn = snd_pcm_readi(loopback_capture_handle, data_in, frames/2)) < 0){
			snd_pcm_prepare(loopback_capture_handle);
			//putchar('=');
		}
		i = 0; 
		j = 0;
		int ret_card = pcmreturn;

		//fill up a local buffer, take only the left channel	
		i = 0; 
		j = 0;	
		for (int i = 0; i < pcmreturn; i++){
			q_write(&qloop, data_in[j]/64);
			q_write(&qloop, data_in[j]/64);
			j += 2;
		}
		nsamples += j;

		clock_gettime(CLOCK_MONOTONIC, &gettime_now);
		if (gettime_now.tv_sec != last_sec){
			if(use_virtual_cable)
//			printf("######sampling rate %d/%d\n", played_samples, nsamples);
			last_sec = gettime_now.tv_sec;
			nsamples = 0;
			played_samples = 0;
			count = 0;
		}

  }
  printf("********Ending loopback thread\n");
}

/*
We process the sound in a background thread.
It will call the user-supplied function sound_process()  
*/
void *sound_thread_function(void *ptr){
	char *device = (char *)ptr;
	struct sched_param sch;

	//switch to maximum priority
	sch.sched_priority = sched_get_priority_max(SCHED_FIFO);
	pthread_setschedparam(sound_thread, SCHED_FIFO, &sch);
	
	int i = 0;
	for (i = 0; i < 10; i++){
		if (!sound_start_play(device))
			break;
			delay(1000); //wait for the sound system to bootup
			printf("Retrying the sound system %d\n", i);
	}
	if (i == 10){
	 fprintf(stderr, "*Error opening play device");
		return NULL;
	}

	if (sound_start_capture(device)){
		fprintf(stderr, "*Error opening capture device");
		return NULL;
	}

//  printf("opening loopback on plughw:1,0 sound card\n");	
	if(sound_start_loopback_play("plughw:1,0")){
		fprintf(stderr, "*Error opening loopback play device");
		return NULL;
	}
	sound_thread_continue = 1;
	sound_loop();
	sound_stop();
}

void *loopback_thread_function(void *ptr){
	struct sched_param sch;

	//switch to maximum priority
	sch.sched_priority = sched_get_priority_max(SCHED_FIFO);
	pthread_setschedparam(loopback_thread, SCHED_FIFO, &sch);
//	printf("loopback thread is %x\n", loopback_thread);
//  printf("opening loopback on plughw:1,0 sound card\n");	

	if (sound_start_loopback_capture("plughw:2,1")){
		fprintf(stderr, "*Error opening loopback capture device");
		return NULL;
	}
	sound_thread_continue = 1;
	loopback_loop();
	sound_stop();
}

int sound_thread_start(char *device){
	q_init(&qloop, 10240);
 	qloop.stall = 1;

	pthread_create( &sound_thread, NULL, sound_thread_function, (void*)device);
	sleep(1);
	pthread_create( &loopback_thread, NULL, loopback_thread_function, (void*)device);
}

void sound_thread_stop(){
	sound_thread_continue = 0;
}

void sound_input(int loop){
  if (loop){
    use_virtual_cable = 1;
	}
  else{
    use_virtual_cable = 0;
	}
}

//demo, uncomment it to test it out
/*
void sound_process(int32_t *input_i, int32_t *input_q, int32_t *output_i, int32_t *output_q, int n_samples){
 
	for (int i= 0; i < n_samples; i++){
		output_i[i] = input_i[i];
		output_q[i] = input_q[i];
	}	
}

void main(int argc, char **argv){
	sound_thread_start("plughw:0,0");
	sleep(10);
	sound_thread_stop();
	sleep(10);
}
*/