#include"libaudiomgr_linux.h"
#include "core_time.h"
#include <stdlib.h>
#include <string.h>


#ifndef RVC_PA_ADJUST_LATENCY_PROTOCOL_VERSION
#define RVC_PA_ADJUST_LATENCY_PROTOCOL_VERSION 13
#endif

#ifndef RVC_MAX_VOLUME
#define RVC_MAX_VOLUME 65536
#endif

static int sample_index = 0;

// From pulsecore/macro.h
#define pa_memzero(x,l) (memset((x), 0, (l)))
#define pa_zero(x) (pa_memzero(&(x), sizeof(x)))

static uint32_t latency_ms = 20; // requested initial latency in milisec: 0 use max
static pa_usec_t latency = 0; //real latency in usec (for timestamping)

static int sink_index = 0;
static int source_index = 0;


AudioMgrImpl::AudioMgrImpl(audiomgr_callback_t* pCallback)
{
	m_audio_context = NULL;
	memcpy(&m_callback, pCallback, sizeof(audiomgr_callback_t));
}

AudioMgrImpl::~AudioMgrImpl()
{
	if (NULL != m_audio_context){
		if (NULL != m_audio_context->list_input_devices) {
			free(m_audio_context->list_input_devices);
			m_audio_context->list_input_devices = NULL;
		}

		if (NULL != m_audio_context->list_output_devices) {
			free(m_audio_context->list_output_devices);
			m_audio_context->list_output_devices = NULL;
		}

		if (NULL != m_audio_context->capture_buff) {
			free(m_audio_context->capture_buff);
			m_audio_context->capture_buff = NULL;
		}

		free(m_audio_context);
		m_audio_context = NULL;
	}
}

int AudioMgrImpl::audio_mgr_destroy()
{
	delete this;
	return 0;
}

/*
 * init pulseaudio api
 * args:
 *    audio_ctx - pointer to audio context data
 *
 * asserts:
 *    audio_ctx is not null
 *
 * returns: error code (0 = E_OK)
 */
int AudioMgrImpl::audio_init_pulseaudio()
{
	/*assertions*/
	assert(NULL != m_audio_context);

	if (pa_get_devicelist() < 0)
	{
		audiolog( "pulse audio failed to get audio device list from pulse server\n");
		return -1;
	}

	return 0;
}

/*
 * pa_mainloop will call this function when it's ready to tell us
 *  about a source (input).
 *  Since we're not threading when listing devices,
 *  there's no need for mutexes on the devicelist structure
 * args:
 *    c - pointer to pulse context
 *    l - pointer to source info
 *    eol - end of list
 *    data - pointer to user data (audio context)
 *
 * asserts:
 *    none
 *
 * returns: none
 */
static void pa_sourcelist_cb(pa_context* c, const pa_source_info* l, int eol, void* data)
{
	rvc_audio_context_t* audio_ctx = (rvc_audio_context_t*)data;

	int channels = 1;

	/*
	* If eol is set to a positive number,
	* you're at the end of the list
	*/
	if (eol > 0)
		return;

	source_index++;

	if (l->sample_spec.channels < 1)
		channels = 1;
	else
		channels = l->sample_spec.channels;

	double my_latency = 0.0;
	int verbosity = 1;

	if (my_latency <= 0.0)
		my_latency = (double)latency_ms / 1000;

	if (l->monitor_of_sink == PA_INVALID_INDEX)
	{
		audio_ctx->num_input_dev++;
		/*add device to list*/
		audio_ctx->list_input_devices = (rvc_audio_device_t*)realloc(audio_ctx->list_input_devices, audio_ctx->num_input_dev * sizeof(rvc_audio_device_t));
		if (audio_ctx->list_input_devices == NULL)
		{
			exit(-1);
		}
		/*fill device data*/
		audio_ctx->list_input_devices[audio_ctx->num_input_dev - 1].id = l->index; /*saves dev id*/
		strncpy(audio_ctx->list_input_devices[audio_ctx->num_input_dev - 1].name, l->name, MAX_PATH-1);
		strncpy(audio_ctx->list_input_devices[audio_ctx->num_input_dev - 1].description, l->description, MAX_PATH-1);
		audio_ctx->list_input_devices[audio_ctx->num_input_dev - 1].channels = channels;
		audio_ctx->list_input_devices[audio_ctx->num_input_dev - 1].samprate = l->sample_spec.rate;
		audio_ctx->list_input_devices[audio_ctx->num_input_dev - 1].low_latency = my_latency; /*in seconds*/
		audio_ctx->list_input_devices[audio_ctx->num_input_dev - 1].high_latency = my_latency; /*in seconds*/
	}
}

void PaSetVolumeCallback(pa_context* c,
	int success,
	void* /*pThis*/) {
	if (!success) {
		//printf("failed to set volume\n");
	}
	else {
		//printf("success to set volume\n");
	}
}

static void pa_setsourcevolume_cb(pa_context* c, const pa_source_info* l, int eol, void* data)
{
	rvc_volume_set_t* audio_volume = (rvc_volume_set_t*)data;

	/*
	* If eol is set to a positive number,
	* you're at the end of the list
	*/
	if (eol > 0)
		return;


	if (l->monitor_of_sink == PA_INVALID_INDEX)
	{
		if (strstr(l->description, audio_volume->strdevicename)) {
			pa_cvolume cvolumes;
			// Set the same volume for all channels
			pa_cvolume_set(&cvolumes, l->channel_map.channels, audio_volume->ivolume);
			pa_operation* paOperation = NULL;
			if (!(paOperation = pa_context_set_source_volume_by_index(c, l->index, &cvolumes, PaSetVolumeCallback, NULL))) {
				audio_volume->volume_flag = 1;
			}
			
			pa_operation_unref(paOperation);
		}
	}
}

static void pa_setsinkvolume_cb(pa_context* c, const pa_sink_info* l, int eol, void* data)
{
	rvc_volume_set_t* audio_volume = (rvc_volume_set_t*)data;

	/*
	* If eol is set to a positive number,
	* you're at the end of the list
	*/
	if (eol > 0)
		return;

	if (strstr(l->description, audio_volume->strdevicename)) {
		pa_cvolume cvolumes;
		// Set the same volume for all channels
		pa_cvolume_set(&cvolumes, l->channel_map.channels, audio_volume->ivolume);
		pa_operation* paOperation = NULL;
		if (!(paOperation = pa_context_set_sink_volume_by_index(c, l->index, &cvolumes, PaSetVolumeCallback, NULL))) {
			audio_volume->volume_flag = 1;
		}
		pa_operation_unref(paOperation);
	}
}

static void pa_sourcevolume_cb(pa_context* c, const pa_source_info* l, int eol, void* data)
{
	rvc_audio_volume_t* audio_volume = (rvc_audio_volume_t*)data;

	/*
	* If eol is set to a positive number,
	* you're at the end of the list
	*/
	if (eol > 0)
		return;

	if (l->monitor_of_sink == PA_INVALID_INDEX)
	{
		if (strstr(l->description, audio_volume->strdevicename)){
			audio_volume->cvolumes = l->volume;
			audio_volume->volume_flag = 1;
		}
	}
}


static void pa_sinkvolume_cb(pa_context* c, const pa_sink_info* l, int eol, void* userdata)
{
	rvc_audio_volume_t* audio_volume = (rvc_audio_volume_t*)userdata;
	/*
	 * If eol is set to a positive number,
	 * you're at the end of the list
	 */
	if (eol > 0)
		return;

	if (strstr(l->description, audio_volume->strdevicename)) {
		audio_volume->cvolumes = l->volume;
		audio_volume->volume_flag = 1;
	}
}

/*
 * pa_mainloop will call this function when it's ready to tell us
 *  about a source (input).
 *  This callback is pretty much identical to the previous
 *  but it will only print the output devices
 * args:
 *    c - pointer to pulse context
 *    l - pointer to sink info
 *    eol - end of list
 *    data - pointer to user data (audio context)
 *
 * asserts:
 *    none
 *
 * returns: none
 */

static void pa_sinklist_cb(pa_context* c, const pa_sink_info* l, int eol, void* userdata)
{
	rvc_audio_context_t* audio_ctx = (rvc_audio_context_t*)userdata;

	/*
	 * If eol is set to a positive number,
	 * you're at the end of the list
	 */
	if (eol > 0)
		return;

	sink_index++;

	double my_latency = 0.0;
	int verbosity = 1;

	if (my_latency <= 0.0)
		my_latency = (double)latency_ms / 1000;

	audio_ctx->num_output_dev++;
	/*add device to list*/
	audio_ctx->list_output_devices = (rvc_audio_device_t*)realloc(audio_ctx->list_output_devices, audio_ctx->num_output_dev * sizeof(rvc_audio_device_t));
	if (audio_ctx->list_output_devices == NULL)
	{
		printf("AUDIO: FATAL memory allocation failure (pa_sinklist_cb): %s\n", strerror(errno));
		exit(-1);
	}
	/*fill device data*/
	audio_ctx->list_output_devices[audio_ctx->num_output_dev - 1].id = l->index; /*saves dev id*/
	strncpy(audio_ctx->list_output_devices[audio_ctx->num_output_dev - 1].name, l->name, MAX_PATH-1);
	strncpy(audio_ctx->list_output_devices[audio_ctx->num_output_dev - 1].description, l->description, MAX_PATH-1);
	audio_ctx->list_output_devices[audio_ctx->num_output_dev - 1].channels = l->channel_map.channels;
	audio_ctx->list_output_devices[audio_ctx->num_output_dev - 1].samprate = l->sample_spec.rate;
	audio_ctx->list_output_devices[audio_ctx->num_output_dev - 1].low_latency = my_latency; /*in seconds*/
	audio_ctx->list_output_devices[audio_ctx->num_output_dev - 1].high_latency = my_latency; /*in seconds*/
}



/*
 * This callback gets called when our context changes state.
 *  We really only care about when it's ready or if it has failed
 * args:
 *    c -pointer to pulse context
 *    data - pointer to user data
 *
 * asserts:
 *    none
 *
 * retuns: none
 */
static void pa_state_cb(pa_context* c, void* data)
{
	pa_context_state_t state;
	int* pa_ready = (int*)data;
	state = pa_context_get_state(c);
	switch (state)
	{
		// These are just here for reference
	case PA_CONTEXT_UNCONNECTED:
		break;
	case PA_CONTEXT_CONNECTING:
	case PA_CONTEXT_AUTHORIZING:
	case PA_CONTEXT_SETTING_NAME:
		break;
	case PA_CONTEXT_FAILED:
	case PA_CONTEXT_TERMINATED:
		*pa_ready = 2;
		break;
	case PA_CONTEXT_READY:
		*pa_ready = 1;
		break;
	default:
		break;
	}
}


/*
 * clean up and disconnect
 * args:
 *    pa_ctx - pointer to pulse context
 *    pa_ml - pointer to pulse mainloop
 *
 * asserts:
 *    none
 *
 * returns:
 *    none
 */
static void finish(pa_context* pa_ctx, pa_mainloop* pa_ml)
{
	/* clean up and disconnect */
	pa_context_disconnect(pa_ctx);
	pa_context_unref(pa_ctx);
	pa_mainloop_free(pa_ml);
}

/*
 * iterate the main loop until all devices are listed
 * args:
 *    audio_ctx - pointer to audio context
 *
 * asserts:
 *    audio_ctx is not null
 *
 * returns: error code
 */
int AudioMgrImpl::pa_get_devicelist()
{
	/*assertions*/
	assert(m_audio_context != NULL);

	/* Define our pulse audio loop and connection variables */
	pa_mainloop* pa_ml;
	pa_mainloop_api* pa_mlapi;
	pa_operation* pa_op = NULL;
	pa_context* pa_ctx;

	/* We'll need these state variables to keep track of our requests */
	int state = 0;
	int pa_ready = 0;

	/* Create a mainloop API and connection to the default server */
	pa_ml = pa_mainloop_new();
	pa_mlapi = pa_mainloop_get_api(pa_ml);
	pa_ctx = pa_context_new(pa_mlapi, "getDevices");

	/* This function connects to the pulse server */
	if (pa_context_connect(pa_ctx, NULL, PA_CONTEXT_NOFLAGS, NULL) < 0)
	{
		audiolog("AUDIO: PULSE - unable to connect to server: pa_context_connect failed!");
		finish(pa_ctx, pa_ml);
		return -1;
	}

	/*
	 * This function defines a callback so the server will tell us
	 * it's state.
	 * Our callback will wait for the state to be ready.
	 * The callback will modify the variable to 1 so we know when we
	 * have a connection and it's ready.
	 * If there's an error, the callback will set pa_ready to 2
	 */
	pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);

	/*
	 * Now we'll enter into an infinite loop until we get the data
	 * we receive or if there's an error
	 */
	for (;;)
	{
		/*
		 * We can't do anything until PA is ready,
		 * so just iterate the mainloop and continue
		 */
		if (pa_ready == 0)
		{
			pa_mainloop_iterate(pa_ml, 1, NULL);
			continue;
		}
		///* We couldn't get a connection to the server, so exit out */
		if (pa_ready == 2)
		{
			finish(pa_ctx, pa_ml);
			return -1;
		}
		/*
		 * At this point, we're connected to the server and ready
		 * to make requests
		 */
		switch (state)
		{
			/* State 0: we haven't done anything yet */
		case 0:
			/*
			 * This sends an operation to the server.
			 * pa_sinklist_cb is our callback function and a pointer
			 * o our devicelist will be passed to the callback
			 * (audio_ctx) The operation ID is stored in the
			 * pa_op variable
			 */
			pa_op = pa_context_get_sink_info_list(
				pa_ctx,
				pa_sinklist_cb,
				(void*)m_audio_context);

			/* Update state for next iteration through the loop */
			state++;
			break;
		case 1:
			/*
			 * Now we wait for our operation to complete.
			 * When it's complete our pa_output_devicelist is
			 * filled out, and we move along to the next state
			 */
			if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE)
			{
				pa_operation_unref(pa_op);

				/*
				 * Now we perform another operation to get the
				 * source(input device) list just like before.
				 * This time we pass a pointer to our input structure
				 */
				pa_op = pa_context_get_source_info_list(
					pa_ctx,
					pa_sourcelist_cb,
					(void*)m_audio_context);
				/* Update the state so we know what to do next */
				state++;
			}
			break;
		case 2:
			if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE)
			{
				/*
				 * Now we're done,
				 * clean up and disconnect and return
				 */
				pa_operation_unref(pa_op);
				finish(pa_ctx, pa_ml);
				return 0;
			}
			break;
		default:
			/* We should never see this state */
			audiolog("AUDIO: Pulse audio in state %d", state);
			return -1;
		}
		/*
		 * Iterate the main loop and go again.  The second argument is whether
		 * or not the iteration should block until something is ready to be
		 * done.  Set it to zero for non-blocking.
		 */
		pa_mainloop_iterate(pa_ml, 1, NULL);
	}

	finish(pa_ctx, pa_ml);

	return 0;
}


/*
 * get the number of available audio devices
 *
  * args:
 *    binput - is input device
 * asserts:
 *   audio_ctx is not null
 *
 * returns: number of listed audio devices
 */
int AudioMgrImpl::audio_get_device_count(bool binput)
{
	/*assertions*/
	assert(m_audio_context != NULL);
	if (binput){
		return m_audio_context->num_input_dev;
	}
	else {
		return m_audio_context->num_output_dev;
	}

	
}


/*
 * get the audio device referenced by index
 * args:
 *   index - index of audio device
 *
 * asserts:
 *   audio_ctx is not null
 *
 * returns: audio device referenced by index
 */
rvc_audio_device_t* AudioMgrImpl::audio_get_input_device(int index)
{
	/*assertions*/
	assert(m_audio_context != NULL);

	if (index >= m_audio_context->num_input_dev){
		index = m_audio_context->num_input_dev - 1;
	}

	if (index < 0){
		index = 0;
	}

	return &m_audio_context->list_input_devices[index];
}


/*
 * get the output audio device referenced by index
 * args:
 *   index - index of output audio device
 *
 * asserts:
 *   audio_ctx is not null
 *
 * returns: output audio device referenced by index
 */
rvc_audio_device_t* AudioMgrImpl::audio_get_output_device( int index)
{
	/*assertions*/
	assert(m_audio_context != NULL);

	if (index >= m_audio_context->num_output_dev){
		index = m_audio_context->num_output_dev - 1;
	}

	if (index < 0){
		index = 0;
	}

	return &m_audio_context->list_output_devices[index];
}


int AudioMgrImpl::audio_get_device_name(char* pstrbuf, size_t ulen, bool binput, int index)
{
	int iret = -1;

	//模拟立体声
	const char stranalogstereo[] = { 0x20,0xe6,0xa8,0xa1,0xe6,0x8b,0x9f,0xe7,0xab,0x8b,0xe4,0xbd,0x93,0xe5,0xa3,0xb0,0 };
	//数字立体声
	const char strdigitalstereo[] = { 0x20,0xe6,0x95,0xb0,0xe5,0xad,0x97,0xe7,0xab,0x8b,0xe4,0xbd,0x93,0xe5,0xa3,0xb0,0 };
	//立体声
	const char strstereo[] = { 0x20,0xe7,0xab,0x8b,0xe4,0xbd,0x93,0xe5,0xa3,0xb0,0 };
	//多声道
	const char strmultistereo[] = { 0x20,0xe5,0xa4,0x9a,0xe5,0xa3,0xb0,0xe9,0x81,0x93,0 };

	rvc_audio_device_t* audio_device = NULL;
	if (binput){
		audio_device = audio_get_input_device(index);
	}
	else{
		audio_device = audio_get_output_device(index);
	}

	size_t unamelen = strlen(audio_device->description);
	if (ulen > unamelen){
		memcpy(pstrbuf, audio_device->description, ulen);
		char* pindex = NULL;
		if ((pindex = strstr(pstrbuf, stranalogstereo))||(pindex = strstr(pstrbuf, strdigitalstereo))){
			*pindex = 0;
		}
		if ((pindex = strstr(pstrbuf, strstereo)) || (pindex = strstr(pstrbuf, strmultistereo))) {
			*pindex = 0;
		}

		iret = 0;
	}

	return iret;
}


int AudioMgrImpl::audio_get_device_id(const char* pstrname, bool binput)
{
	int iret = -1;
	if (NULL == pstrname){
		return iret;
	}

	int icount = audio_get_device_count(binput);
	if (icount > 0){
		for (int index = 0; index < icount; index++){
			rvc_audio_device_t* audio_device = NULL;
			if (binput) {
				audio_device = audio_get_input_device(index);
			}
			else {
				audio_device = audio_get_output_device(index);
			}
			if (NULL != audio_device){
				if (strstr(audio_device->description, pstrname)){
					iret = index;
					break;
				}
			}
		}
	}

	return iret;
}

rvc_audio_device_t* AudioMgrImpl::audio_get_device_infos(bool binput, int index)
{
	rvc_audio_device_t* audio_device = NULL;
	if (binput) {
		audio_device = audio_get_input_device(index);
	}
	else {
		audio_device = audio_get_output_device(index);
	}

	return audio_device;
}

int AudioMgrImpl::audio_get_device_volume(int* ivolume, const char* pstrname, bool binput)
{
	audiopulse_get_device_volume(ivolume, pstrname, binput);
	float fvol = (*ivolume) * 1000 / RVC_MAX_VOLUME;
	int ivol = fvol;
	int ilast = ivol % 10;
	int inum = ivol / 10;
	if (ilast >= 5) {
		inum++;
	}

	*ivolume = inum;

	return 0;
}

int AudioMgrImpl::audiopulse_get_device_volume(int* ivolume, const char* pstrname, bool binput)
{
	int iret = -1;

	pa_mainloop* pa_ml;
	pa_mainloop_api* pa_mlapi;
	pa_operation* pa_op = NULL;
	pa_context* pa_ctx;

	/* We'll need these state variables to keep track of our requests */
	int state = 0;
	int pa_ready = 0;

	rvc_audio_volume_t rvc_volume = { 0 };
	strcpy(rvc_volume.strdevicename, pstrname);

	/* Create a mainloop API and connection to the default server */
	pa_ml = pa_mainloop_new();
	pa_mlapi = pa_mainloop_get_api(pa_ml);
	pa_ctx = pa_context_new(pa_mlapi, "get audio volume");

	/* This function connects to the pulse server */
	if (pa_context_connect(pa_ctx, NULL, PA_CONTEXT_NOFLAGS, NULL) < 0)
	{
		audiolog("AUDIO: PULSE - unable to connect to server: pa_context_connect failed");
		finish(pa_ctx, pa_ml);
		return -1;
	}

	/*
	 * This function defines a callback so the server will tell us
	 * it's state.
	 * Our callback will wait for the state to be ready.
	 * The callback will modify the variable to 1 so we know when we
	 * have a connection and it's ready.
	 * If there's an error, the callback will set pa_ready to 2
	 */
	pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);

	/*
	 * Now we'll enter into an infinite loop until we get the data
	 * we receive or if there's an error
	 */
	for (;;)
	{
		/*
		 * We can't do anything until PA is ready,
		 * so just iterate the mainloop and continue
		 */
		if (pa_ready == 0)
		{
			pa_mainloop_iterate(pa_ml, 1, NULL);
			continue;
		}
		///* We couldn't get a connection to the server, so exit out */
		if (pa_ready == 2)
		{
			finish(pa_ctx, pa_ml);
			return -1;
		}
		/*
		 * At this point, we're connected to the server and ready
		 * to make requests
		 */

		if (0 == state){
			if (binput) {
				pa_op = pa_context_get_source_info_list(
					pa_ctx,
					pa_sourcevolume_cb,
					(void*)& rvc_volume);
			}
			else
			{
				pa_op = pa_context_get_sink_info_list(
					pa_ctx,
					pa_sinkvolume_cb,
					(void*)& rvc_volume);
			}
			state++;
		}

		if (rvc_volume.volume_flag){
			*ivolume = rvc_volume.cvolumes.values[0];
			pa_operation_unref(pa_op);
			break;
		}
		if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE)
		{
			/*
			 * Now we're done,
			 * clean up and disconnect and return
			 */
			pa_operation_unref(pa_op);
			finish(pa_ctx, pa_ml);
			return 0;
		}
		/*
		 * Iterate the main loop and go again.  The second argument is whether
		 * or not the iteration should block until something is ready to be
		 * done.  Set it to zero for non-blocking.
		 */
		pa_mainloop_iterate(pa_ml, 1, NULL);
	}
	finish(pa_ctx, pa_ml);
	return iret;
}


int AudioMgrImpl::audio_set_device_volume(int ivolume, const char* pstrname, bool binput)
{
	int ivol = ivolume * RVC_MAX_VOLUME / 100;
	return audiopulse_set_device_volume(ivol, pstrname, binput);
}

int AudioMgrImpl::audiopulse_set_device_volume(int ivolume, const char* pstrname, bool binput)
{
	int iret = -1;

	pa_mainloop* pa_ml;
	pa_mainloop_api* pa_mlapi;
	pa_operation* pa_op = NULL;
	pa_context* pa_ctx;

	/* We'll need these state variables to keep track of our requests */
	int state = 0;
	int pa_ready = 0;

	rvc_volume_set_t rvc_volume = { 0 };
	strcpy(rvc_volume.strdevicename, pstrname);
	rvc_volume.ivolume = ivolume;

	/* Create a mainloop API and connection to the default server */
	pa_ml = pa_mainloop_new();
	pa_mlapi = pa_mainloop_get_api(pa_ml);
	pa_ctx = pa_context_new(pa_mlapi, "set audio volume");

	/* This function connects to the pulse server */
	if (pa_context_connect(pa_ctx, NULL, PA_CONTEXT_NOFLAGS, NULL) < 0)
	{
		audiolog("AUDIO: PULSE - unable to connect to server: pa_context_connect failed");
		finish(pa_ctx, pa_ml);
		return -1;
	}

	/*
	 * This function defines a callback so the server will tell us
	 * it's state.
	 * Our callback will wait for the state to be ready.
	 * The callback will modify the variable to 1 so we know when we
	 * have a connection and it's ready.
	 * If there's an error, the callback will set pa_ready to 2
	 */
	pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);

	/*
	 * Now we'll enter into an infinite loop until we get the data
	 * we receive or if there's an error
	 */
	for (;;)
	{
		/*
		 * We can't do anything until PA is ready,
		 * so just iterate the mainloop and continue
		 */
		if (pa_ready == 0)
		{
			pa_mainloop_iterate(pa_ml, 1, NULL);
			continue;
		}
		///* We couldn't get a connection to the server, so exit out */
		if (pa_ready == 2)
		{
			finish(pa_ctx, pa_ml);
			return -1;
		}
		/*
		 * At this point, we're connected to the server and ready
		 * to make requests
		 */

		if (0 == state) {
			if (binput) {
				pa_op = pa_context_get_source_info_list(
					pa_ctx,
					pa_setsourcevolume_cb,
					(void*)&rvc_volume);
			}
			else
			{
				pa_op = pa_context_get_sink_info_list(
					pa_ctx,
					pa_setsinkvolume_cb,
					(void*)&rvc_volume);
			}
			state++;
		}

		if (rvc_volume.volume_flag) {
			pa_operation_unref(pa_op);
			break;
		}

		if (pa_operation_get_state(pa_op) == PA_OPERATION_DONE)
		{
			/*
			 * Now we're done,
			 * clean up and disconnect and return
			 */
			pa_operation_unref(pa_op);
			finish(pa_ctx, pa_ml);
			return 0;
		}
		/*
		 * Iterate the main loop and go again.  The second argument is whether
		 * or not the iteration should block until something is ready to be
		 * done.  Set it to zero for non-blocking.
		 */
		pa_mainloop_iterate(pa_ml, 1, NULL);
	}
	finish(pa_ctx, pa_ml);
	return iret;
}


int AudioMgrImpl::audio_mgr_initialize()
{
	int iret = -1;
	m_audio_context = (rvc_audio_context_t*)calloc(1, sizeof(rvc_audio_context_t));

	if (NULL == m_audio_context){
        audiolog("%s:%d couldn't allocate audio context.", __FUNCTION__, __LINE__);
		return iret;
	}


	iret = audio_init_pulseaudio();

	/*force a valid number of channels*/
	if (m_audio_context->channels > 2) {
		m_audio_context->channels = 2;
	}
		
	return iret;
}

/*
 * close and clean audio context for pulse audio api
 *
 * asserts:
 *   none
 *
 * returns: none
 */
int AudioMgrImpl::audio_close_pulseaudio()
{
	int iret = -1;
	if (m_audio_context == NULL){
		return iret;
	}

	if (m_audio_context->stream_flag == AUDIO_STRM_ON) {
		stop_audio_capture();
	}
		
	if (NULL != m_audio_context->list_input_devices) {
		free(m_audio_context->list_input_devices);
		m_audio_context->list_input_devices = NULL;
	}

	if (NULL != m_audio_context->list_output_devices){
		free(m_audio_context->list_output_devices);
		m_audio_context->list_output_devices = NULL;
	}

	if (NULL != m_audio_context->capture_buff) {
		free(m_audio_context->capture_buff);
		m_audio_context->capture_buff = NULL;
	}
	
	free(m_audio_context);
	m_audio_context = NULL;

	iret = 0;

	return iret;
}

/*
 * stop and join the main loop iteration thread
 *
 * asserts:
 *   audio_ctx is not null
 *
 * returns: error code
 */
int AudioMgrImpl::stop_audio_capture()
{
	/*assertions*/
	assert(m_audio_context != NULL);
	m_audio_context->stream_flag = AUDIO_STRM_OFF;
	if (0 != __THREAD_JOIN(m_readthread)){
        audiolog("read thread join failed!");
	}
	return 0;
}

int AudioMgrImpl::audio_mgr_terminate()
{
	int iret = -1;
	/*assertions*/
	assert(m_audio_context != NULL);

	/* thread must be joined before destroying the mutex
		 * so no need to unlock before destroying it
	*/

	iret = audio_close_pulseaudio();

	return iret;
}


/*
 * update pulse audio latency
 * args:
 *    s - pointer to pa_stream
 *
 * asserts:
 *    none
 *
 * returns:none
 */
static void get_latency(pa_stream* s)
{
	pa_usec_t l;
	int negative;

	pa_stream_get_timing_info(s);

	if (pa_stream_get_latency(s, &l, &negative) != 0)
	{
		return;
	}

	latency = l; /*can only be negative in monitoring streams*/
}

/*
 * audio record callback
 * args:
 *   s - pointer to pa_stream
 *   length - buffer length
 *   data - pointer to user data
 *
 * asserts:
 *   none
 *
 * returns: none
 */
static void stream_request_cb(pa_stream* s, size_t length, void* data)
{
	rvc_audio_context_t* audio_ctx = (rvc_audio_context_t*)data;

	if (audio_ctx->channels == 0){
		return;
	}

	if (audio_ctx->samprate == 0){
		return;
	}

	int64_t ts = 0;

	while (pa_stream_readable_size(s) > 0)
	{
		const void* inputBuffer;
		size_t length;

		/*read from stream*/
		if (pa_stream_peek(s, &inputBuffer, &length) < 0){
			return;
		}

		if (length == 0){
			return; /*buffer is empty*/
		}

		get_latency(s);

		ts = ns_time_monotonic() - (latency * 1000);

		if (audio_ctx->last_ts <= 0)
			audio_ctx->last_ts = ts;

		audio_ctx->audio_param.on_audio_callback(inputBuffer, length, audio_ctx->audio_param.user_data);

		pa_stream_drop(s); /*clean the samples*/
	}
}


/*
 * Iterate the main loop while recording is on.
 * This function runs under it's own thread called by audio_pulse_start
 * args:
 *   data - pointer to user data (audio context)
 *
 * asserts:
 *   data is not null
 *
 * returns: pointer to error code
 */
void* pulse_read_audio(void* data)
{
	AudioMgrImpl* audio_mgr = (AudioMgrImpl*)data;
	assert(audio_mgr != NULL);

	rvc_audio_context_t* audio_ctx = audio_mgr->audio_get_context();
	/*assertions*/
	assert(audio_ctx != NULL);
		
	pa_mainloop* pa_ml;
	pa_mainloop_api* pa_mlapi;
	pa_buffer_attr bufattr;
	pa_stream* recordstream = NULL; // pulse audio stream
	pa_context* pa_ctx; //pulse context
	pa_sample_spec ss;
	pa_stream_flags_t flags = PA_STREAM_NOFLAGS;
	int32_t pastream_flag = (int32_t)PA_STREAM_NOFLAGS;

	int r;
	int pa_ready = 0;

	/* Create a mainloop API and connection to the default server */
	pa_ml = pa_mainloop_new();
	pa_mlapi = pa_mainloop_get_api(pa_ml);
	pa_ctx = pa_context_new(pa_mlapi, "rvc pulse API");

	if (pa_context_connect(pa_ctx, NULL, PA_CONTEXT_NOFLAGS, NULL) < 0)
	{
		audio_mgr->audiolog("AUDIO: PULSE - unable to connect to server: pa_context_connect failed.");
		finish(pa_ctx, pa_ml);
		return ((void*)-1);
	}

	/*
	 * This function defines a callback so the server will tell us it's state.
	 * Our callback will wait for the state to be ready.  The callback will
	 * modify the variable to 1 so we know when we have a connection and it's
	 * ready.
	 * If there's an error, the callback will set pa_ready to 2
	 */
	pa_context_set_state_callback(pa_ctx, pa_state_cb, &pa_ready);

	/*
	 * This function defines a time event callback (called every TIME_EVENT_USEC)
	 */
	 //pa_context_rttime_new(pa_ctx, pa_rtclock_now() + TIME_EVENT_USEC, time_event_callback, NULL);

	 /*
	  * We can't do anything until PA is ready, so just iterate the mainloop
	  * and continue
	  */
	while (pa_ready == 0){
		pa_mainloop_iterate(pa_ml, 1, NULL);
	}
	if (pa_ready == 2){
		finish(pa_ctx, pa_ml);
		return ((void*)-1);
	}

	/* set the sample spec (frame rate, channels and format) */
	ss.rate = audio_ctx->samprate;
	ss.channels = audio_ctx->channels;
	ss.format = audio_ctx->eformat;

	recordstream = pa_stream_new(pa_ctx, "Record", &ss, NULL);
	if (!recordstream)
		audio_mgr->audiolog("AUDIO: (pulse audio) pa_stream_new failed (chan:%d rate:%d)",
			ss.channels, ss.rate);

	/* define the callbacks */
	pa_stream_set_read_callback(recordstream, stream_request_cb, (void*)audio_ctx);

	// Set properties of the record buffer
	pa_zero(bufattr);
	/* optimal value for all is (uint32_t)-1   ~= 2 sec */
	bufattr.maxlength = (uint32_t)-1;
	bufattr.prebuf = (uint32_t)-1;
	bufattr.minreq = (uint32_t)-1;

	if (audio_ctx->latency > 0){
		bufattr.fragsize = bufattr.tlength = pa_usec_to_bytes((audio_ctx->latency * 1000) * PA_USEC_PER_MSEC, &ss);
		uint32_t uvsersion = pa_context_get_protocol_version(pa_ctx);
		if (uvsersion >= RVC_PA_ADJUST_LATENCY_PROTOCOL_VERSION) {
			pastream_flag |= PA_STREAM_ADJUST_LATENCY;
		}
		//audio_mgr->audiolog("pa protocol version is %d.", uvsersion);
	}
	else {
		bufattr.fragsize = bufattr.tlength = (uint32_t)-1;
	}
		
	pastream_flag |= PA_STREAM_INTERPOLATE_TIMING;
	pastream_flag |= PA_STREAM_AUTO_TIMING_UPDATE;

	char* dev = audio_ctx->list_input_devices[audio_ctx->device].name;
	//audio_mgr->audiolog("AUDIO: (pulse audio) connecting to device %s (channels %d rate %d)", dev, ss.channels, ss.rate);
	r = pa_stream_connect_record(recordstream, dev, &bufattr, (pa_stream_flags_t)pastream_flag);
	if (r < 0)
	{
		//audio_mgr->audiolog("AUDIO: (pulse audio) skip latency adjustment");
		/*
		 * Old pulse audio servers don't like the ADJUST_LATENCY flag,
		 * so retry without that
		 */
		r = pa_stream_connect_record(recordstream, dev, &bufattr,
			pa_stream_flags_t((int32_t)PA_STREAM_INTERPOLATE_TIMING |
			(int32_t)PA_STREAM_AUTO_TIMING_UPDATE));
	}

	if (r < 0)
	{
		audio_mgr->audiolog("AUDIO: (pulse audio) pa_stream_connect_record failed");
		finish(pa_ctx, pa_ml);
		return ((void*)-1);
	}

	get_latency(recordstream);

	/*
	 * Iterate the main loop while streaming.  The second argument is whether
	 * or not the iteration should block until something is ready to be
	 * done.  Set it to zero for non-blocking.
	 */

	while (audio_ctx->stream_flag == AUDIO_STRM_ON){
		pa_mainloop_iterate(pa_ml, 1, NULL);
	}
	pa_stream_set_read_callback(recordstream, NULL, NULL);
	//audio_mgr->audiolog("AUDIO: (pulse audio) stream terminated(%i)", audio_ctx->stream_flag);

	pa_stream_disconnect(recordstream);
	pa_stream_unref(recordstream);
	finish(pa_ctx, pa_ml);
	return ((void*)0);
}


int AudioMgrImpl::set_audio_capture_params(audiocap_param_t* param)
{
	assert(param != NULL);

	audio_set_pulseaudio_device(param->ideviceid);
	audio_set_pulsecap_params(param);

	return 0;
}


/*
 * set audio device
 *   index - device index to set
 *
 * asserts:
 *   audio_ctx is not null
 *
 * returns: none
 */
int AudioMgrImpl::audio_set_pulseaudio_device(int index)
{
	/*assertions*/
	assert(m_audio_context != NULL);

	if (index >= m_audio_context->num_input_dev) {
		m_audio_context->device = m_audio_context->num_input_dev - 1;
	}
	else if (index >= 0) {
		m_audio_context->device = index;
	}
	


	return 0;
}


int AudioMgrImpl::audio_set_pulsecap_params(audiocap_param_t* param)
{
	assert(param != NULL);

	m_audio_context->channels = param->ichannels;

	if (m_audio_context->channels > 2) {
		m_audio_context->channels = 2;/*limit it to stereo input*/
	}

	m_audio_context->samprate = param->isamprate;
	m_audio_context->eformat = pa_sample_format_t(param->isampleformat);
	m_audio_context->audio_param.on_audio_callback = param->on_audio_callback;
	m_audio_context->audio_param.user_data = param->user_data;
	m_audio_context->latency = param->flatency;
}
/*
 * set the current latency
 *
 * asserts:
 *   audio_ctx is not null
 *
 * returns: none
 */
void AudioMgrImpl::audio_set_latency(double latency)
{
	/*assertions*/
	assert(m_audio_context != NULL);

	m_audio_context->latency = latency;
}

rvc_audio_context_t* AudioMgrImpl::audio_get_context()
{
	return m_audio_context;
}

audiomgr_callback_t* AudioMgrImpl::audio_get_callback()
{
	return &m_callback;
}


int AudioMgrImpl::start_audio_capture()
{
	/*assertions*/
	assert(m_audio_context != NULL);

	m_audio_context->stream_flag = AUDIO_STRM_ON;
	/* start audio capture thread */
	if (__THREAD_CREATE(&m_readthread, pulse_read_audio, this)){
		audiolog("AUDIO: (pulse audio) read thread creation failed");
		m_audio_context->stream_flag = AUDIO_STRM_OFF;
		return (-1);
	}

	return 0;
}

void AudioMgrImpl::audiolog(const char* fmt, ...)
{
	if (m_callback.debug) {
		va_list arg;
		va_start(arg, fmt);
		if(*m_callback.debug){
			(*m_callback.debug)(m_callback.user_data, fmt, arg);
		}
		va_end(arg);
	}
}