myvtm/Other/libaudioframework/audiortp.c

#include "precompile.h"
#include "audiortp.h"
#include "audioframe.h"
#include "audiostream.h"
#include "audiocontext.h"
#include "audiolog.h"
#include "other/jbuf.h"

#include <assert.h>

#include "memutil.h"

#define CIRC_LEN(rd, wr, n)	(((wr)+(n)-(rd))%(n))
#define CIRC_ADD(old, val, n) (((old)+(n)+(val))%(n))
#define CIRC_SUB(old, val, n) (((old)+(n)-(val))%(n))
#define CIRC_INC(i, n)	CIRC_ADD(i, 1, n)
#define CIRC_DEC(i, n)	CIRC_SUB(i, 1, n)
#define CIRC_IS_EMPTY(rd, wr) ((rd) == (wr))

#define JB_MAX		10
#define DTMF_DURATION	1600
#define DTMF_DURATION_STEP	320
#define DTMF_VOLUME		10

static const char digitmap[16] = { '0', '1', '2', '3', '4', '5', '6', '7', '8', '9',
'*', '#','A', 'B', 'C', 'D'};

static __inline int digit2event(int digit) {
	int event;
	switch (digit) {
			case '0':case '1':case '2':case '3':case '4':
			case '5':case '6':case '7':case '8':case '9':
				event = digit - '0';
				break;
			case '*':
				event = 10;
				break;
			case '#':
				event = 11;
				break;
			case 'a':case 'b':case 'c':case 'd':
				event = 12 + digit - 'a';
				break;
			default:
				return -1;
	}
	return event;
}

static int send_dtmf_digit(rtp_session_t *sess, rtp_dtmf_event *digit, unsigned dtmf_pt)
{
	rtp_state *st = rtp_session_get_rtp_state(sess);
	rtp_session_send(sess, 0, dtmf_pt, digit->e, 0, (const char*)digit, sizeof(rtp_dtmf_event));
	if (digit->e) {
		rtp_session_send(sess, 0, dtmf_pt, digit->e, 0, (const char*)digit, sizeof(rtp_dtmf_event));
		rtp_session_send(sess, 0, dtmf_pt, digit->e, 0, (const char*)digit, sizeof(rtp_dtmf_event));
	}
	return 0;
}

static int pt2bitsample(int pt)
{
	switch (pt) {
	case RTP_PT_PCM16:
		return 16;
	case RTP_PT_PCM8:
	case RTP_PT_PCMA:
	case RTP_PT_PCMU:
		return 8;
	case RTP_PT_G729:
		return 1;
	default:
		break;
	}
	assert(0);
	return 0;
}

static apr_status_t read_frame(void *self, audioframe_t *frame)
{
	audiortp_t *audiortp = CONTAINING_RECORD(self, audiortp_t, base);
	char buf[SUGGEST_FRAME_SIZE];
	unsigned short seq;
	unsigned mark;	
	unsigned ts;
	unsigned pt;
	int i;
	int max_recv_cnt = 5; // max 
	int new_dtmf = 0;

	audiortp->m_rtp_tick++;

	/* read packet from net */
	for (i = 0 ; i < max_recv_cnt; ++i) {
		int ret = rtp_session_recv_hook(audiortp->m_rtpsession, &pt, &mark, &ts, &seq, buf, sizeof(buf), audiortp->m_on_recv_hook, audiortp->m_hook_data);
		if (ret <= 0)
			break;
		audiortp->m_last_rtp_tick = audiortp->m_rtp_tick;
		if (pt == audiortp->m_recv_dtmf_pt) { /* dtmf */
			if (ret == sizeof(rtp_dtmf_event)) {
				int digit[2];
				int digit_cnt = 0;
				int j;
				rtp_dtmf_event *dtmf = (rtp_dtmf_event *)&buf[0];
				if (ts != audiortp->m_recving_dtmf_ts) {
					audiortp->m_recving_dtmf_ts = ts;
					if (!audiortp->m_recving_dtmf_event.e)
						digit[digit_cnt++] = audiortp->m_recving_dtmf_event.event;
					audiortp->m_recving_dtmf_event.event = dtmf->event;
					audiortp->m_recving_dtmf_event.e = dtmf->e;
					if (audiortp->m_recving_dtmf_event.e)
						digit[digit_cnt++] = audiortp->m_recving_dtmf_event.event;
				} else {
					if (dtmf->e) { /* end bit set */
						if (!audiortp->m_recving_dtmf_event.e) {
							audiortp->m_recving_dtmf_event.e = 1;
							audiortp->m_recving_dtmf_event.event = dtmf->event;
							digit[digit_cnt++] = audiortp->m_recving_dtmf_event.event;
						} else {
							if (audiortp->m_recving_dtmf_event.event == 0xff) {
								audiortp->m_recving_dtmf_event.event = dtmf->event;
								digit[digit_cnt++] = audiortp->m_recving_dtmf_event.event;
							}
						}
					} else {
						audiortp->m_recving_dtmf_event.event = dtmf->event;
					}
				}
				for (j = 0; j < digit_cnt; ++j) {
					if (digit[j] < sizeof(digitmap)) {
						int ch = digitmap[digit[j]];
						audiostream_raise_event(self, STREAM_EVT_RTP_DTMF, digit[j], 0);
						EnterCriticalSection(&audiortp->m_dtmf_lock);
						if (audiortp->m_recv_dtmf_cnt < MAX_DTMF)
							audiortp->m_recv_dtmf[audiortp->m_recv_dtmf_cnt++] = digit[j];
						LeaveCriticalSection(&audiortp->m_dtmf_lock);
					}
				}
				if (digit_cnt > 0)
					new_dtmf = digit[digit_cnt-1]; // the last one
			}
		} else if (pt == audiortp->m_recv_pt) { /* audio frame */
			if (ret == audiortp->m_recv_psize) {
				int time_span = (audiortp->m_recv_clock / 1000 * audiortp->m_recv_ptime);
				jbuf_put_frame2(audiortp->m_jitterbuf, &buf[0], ret, 0, ts / time_span, NULL);
			}
		} else {
			break;
		}
		audiortp->m_last_rtp_tick = audiortp->m_rtp_tick;
	}

	if (audiortp->m_timeout > 0) {
		unsigned int diffms = (audiortp->m_rtp_tick - audiortp->m_last_rtp_tick) * FRAME_TIME;
		if (diffms >= (unsigned)audiortp->m_timeout) {
			audiostream_raise_event(self, STREAM_EVT_RTP_TIMEOUT, 0, 0);
		}
	}

	{
		char frm_type;
		jbuf_get_frame(audiortp->m_jitterbuf, frame->buffer, &frm_type);
		if (frm_type == JB_NORMAL_FRAME) {
			frame->size = audiortp->m_jbuf_psize;
		} else {
			frame->size = 0;
		}
	}

	frame->dtmf = new_dtmf;

	return APR_SUCCESS;
}

static apr_status_t write_frame(void *self, const audioframe_t *frame)
{
	audiortp_t *audiortp = CONTAINING_RECORD(self, audiortp_t, base);
	int dtmf = 0;

	EnterCriticalSection(&audiortp->m_dtmf_lock);
	if (frame->dtmf && CIRC_LEN(audiortp->m_send_dtmf_rd, audiortp->m_send_dtmf_wr, MAX_DTMF) < MAX_DTMF-1) {
		rtp_dtmf_event *dtmf = &audiortp->m_send_dtmf[audiortp->m_send_dtmf_wr];
		dtmf->event = (unsigned char)digit2event(frame->dtmf);
		dtmf->e_vol = DTMF_VOLUME;
		dtmf->p = 0;
		dtmf->e = 0;
		dtmf->duration = 0;
		audiortp->m_send_dtmf_wr = CIRC_INC(audiortp->m_send_dtmf_wr, MAX_DTMF);
	}
	if (!CIRC_IS_EMPTY(audiortp->m_send_dtmf_rd, audiortp->m_send_dtmf_wr)) {
		rtp_dtmf_event *digit = &audiortp->m_send_dtmf[audiortp->m_send_dtmf_rd];
		send_dtmf_digit(audiortp->m_rtpsession, digit, audiortp->m_send_dtmf_pt);
		dtmf = 1;
		digit->duration += DTMF_DURATION_STEP;
		if (!digit->e) {
			if (digit->duration == DTMF_DURATION) {
				digit->e = 1;
			}
		} else {
			audiortp->m_send_dtmf_rd = CIRC_INC(audiortp->m_send_dtmf_rd, MAX_DTMF);
			rtp_session_advance_timestamp(audiortp->m_rtpsession, DTMF_DURATION);
		}
	}
	LeaveCriticalSection(&audiortp->m_dtmf_lock);

	if (dtmf) {
		/* because we send dtmf, so ignore this frame */
		return 0;
	}

	if (frame->size < (unsigned long)audiortp->m_send_psize) {
		if (frame->size) {
			memcpy(audiortp->m_SendSample.buffer + audiortp->m_SendSample.size, frame->buffer, frame->size);
			audiortp->m_SendSample.size += frame->size;
		} else {
			char tmp[160];
			int tmp_cnt = FRAME_TIME * audiortp->m_send_clock / 1000 * audiortp->m_send_samplebit / 8;
			memset(tmp, 0, tmp_cnt);
			memcpy(audiortp->m_SendSample.buffer + audiortp->m_SendSample.size, tmp, tmp_cnt);
			audiortp->m_SendSample.size += tmp_cnt;
		}
		if (audiortp->m_SendSample.size >= (unsigned long)audiortp->m_send_psize) {
			//printf("send out %d\n", audiortp->m_send_psize);
			rtp_session_send_hook(audiortp->m_rtpsession, 0, 
				audiortp->m_send_pt, 0, audiortp->m_send_ptime*audiortp->m_send_clock/1000, 
				audiortp->m_SendSample.buffer, audiortp->m_send_psize, audiortp->m_on_send_hook, audiortp->m_hook_data);
			if (audiortp->m_SendSample.size > (unsigned long)audiortp->m_send_psize) {
				audiortp->m_SendSample.size -= audiortp->m_send_psize;
				memmove(audiortp->m_SendSample.buffer, 
					audiortp->m_SendSample.buffer+audiortp->m_send_psize, 
					audiortp->m_SendSample.size);
			} else {
				audiortp->m_SendSample.size = 0;
			}
		}
	} else {
		int delta_ts = audiortp->m_send_ptime*audiortp->m_send_clock/1000;
		rtp_session_send_hook(audiortp->m_rtpsession, 0, 
			audiortp->m_send_pt, 0, delta_ts, frame->buffer, audiortp->m_send_psize, audiortp->m_on_send_hook, audiortp->m_hook_data);
		if (frame->size > (unsigned long)audiortp->m_send_psize) {
			audiortp->m_SendSample.size = frame->size-audiortp->m_send_psize;
			memcpy(audiortp->m_SendSample.buffer, 
				frame->buffer+audiortp->m_send_psize, 
				audiortp->m_SendSample.size);			
		}
	}

	return APR_SUCCESS;
}

static audiostream_vtbl_t g_stream_vtbl = {
	&read_frame,
	&write_frame,
};

void audiortp_destroy(audiortp_t *self)
{
	audiortp_t *audiortp = (audiortp_t *)self;
	DeleteCriticalSection(&audiortp->m_dtmf_lock);
}

apr_status_t audiortp_create(apr_pool_t *pool, audioengine_t *engine, rtp_session_t *sess, audiortp_t **p_audiortp)
{
	audiortp_t *audiortp;

	audiortp = apr_palloc(pool, sizeof(audiortp_t));
	memset(audiortp, 0, sizeof(audiortp_t));
	audiostream_init(engine, &g_stream_vtbl, &audiortp->base);
	audiortp->m_rtpsession = sess;
	audiortp->m_SendSample.buffer = apr_palloc(pool, SUGGEST_FRAME_SIZE);
	audiortp->m_SendSample.size = 0;
	audiortp->m_SendSample.dtmf = 0;
	audiortp->m_RecvSample.buffer = apr_palloc(pool, SUGGEST_FRAME_SIZE);
	audiortp->m_RecvSample.size = 0;
	audiortp->m_RecvSample.dtmf = 0;
	audiortp->m_send_ptime = 0;
	audiortp->m_recv_ptime = 0;
	audiortp->m_send_psize = 0;
	audiortp->m_recv_psize = 0;
	audiortp->m_send_samplebit = 0;
	audiortp->m_recv_samplebit = 0;
	audiortp->m_timeout = -1;
	audiortp->m_send_pt = -1;
	audiortp->m_recv_pt = -1;
	audiortp->m_send_clock = 8000;
	audiortp->m_recv_clock = 8000;
	audiortp->m_send_dtmf_rd = 0;
	audiortp->m_send_dtmf_wr = 0;
	audiortp->m_recv_dtmf_cnt = 0;
	audiortp->m_recving_dtmf_ts = 0;
	audiortp->m_recving_dtmf_event.e = 1;
	audiortp->m_recving_dtmf_event.event = 0xff;
	audiortp->m_rtp_tick = 0;
	audiortp->m_last_rtp_tick = 0;
	audiortp->m_jitterbuf = 0;
	audiortp->m_on_recv_hook = NULL;
	audiortp->m_on_send_hook = NULL;
	audiortp->m_hook_data = NULL;
	InitializeCriticalSection(&audiortp->m_dtmf_lock);
	*p_audiortp = audiortp;
	return APR_SUCCESS;
}

apr_status_t audiortp_set_param(audiortp_t *audiortp, int flag, const void *ptr)
{
	audiortp_t *rtp = (audiortp_t *)audiortp;
	switch (flag) {
	case AUDIO_RTP_FLAG_RECV_PTIME:
		rtp->m_recv_ptime = *(const int *)ptr;
		break;
	case AUDIO_RTP_FLAG_SEND_PTIME:
		rtp->m_send_ptime = *(const int *)ptr;
		break;
	case AUDIO_RTP_FLAG_RECV_PT:
		rtp->m_recv_pt = *(const int *)ptr;
		break;
	case AUDIO_RTP_FLAG_SEND_PT:
		rtp->m_send_pt = *(const int*)ptr;
		break;
	case AUDIO_RTP_FLAG_TIMEOUT:
		rtp->m_timeout = *(const int *)ptr;
		break;
	case AUDIO_RTP_FLAG_SEND_DTMF:
		rtp->m_send_dtmf_pt = *(const int *)ptr;
		break;
	case AUDIO_RTP_FLAG_RECV_DTMF:
		rtp->m_recv_dtmf_pt = *(const int *)ptr;
		break;
	case AUDIO_RTP_FLAG_SEND_CLOCK:
		rtp->m_send_clock = *(const int*)ptr;
		break;
	case AUDIO_RTP_FLAG_RECV_CLOCK:
		rtp->m_recv_clock = *(const int*)ptr;
		break;
	case AUDIO_RTP_FLAG_SEND_HOOK:
		rtp->m_on_send_hook = /**(void (**)(const char*,int,void*))*/(void*)ptr;
		break;
	case AUDIO_RTP_FLAG_RECV_HOOK:
		rtp->m_on_recv_hook = /**(void (**)(const char*,int,void*))*/(void*)ptr;
		break;
	case AUDIO_RTP_FLAG_HOOK_ARG:
		rtp->m_hook_data = /**(void**)*/(void*)ptr;
		break;
	default:
		return APR_BADARG;
	}
	return APR_SUCCESS;
}

apr_status_t audiortp_get_param(audiortp_t *audiortp, int flag, void *ptr)
{
	audiortp_t *rtp = (audiortp_t *)audiortp;
	switch (flag) {
	case AUDIO_RTP_FLAG_RECV_PTIME:
		*(int *)ptr = rtp->m_recv_ptime;
		break;
	case AUDIO_RTP_FLAG_SEND_PTIME:
		*(int *)ptr = rtp->m_send_ptime;
		break;
	case AUDIO_RTP_FLAG_RECV_PT:
		*(int *)ptr = rtp->m_recv_pt;
		break;
	case AUDIO_RTP_FLAG_SEND_PT:
		*(int*)ptr = rtp->m_send_pt;
		break;
	case AUDIO_RTP_FLAG_TIMEOUT:
		*(int *)ptr = rtp->m_timeout;
		break;
	case AUDIO_RTP_FLAG_SEND_DTMF:
		*(int *)ptr = rtp->m_send_dtmf_pt;
		break;
	case AUDIO_RTP_FLAG_RECV_DTMF:
		*(int *)ptr = rtp->m_recv_dtmf_pt;
		break;
	case AUDIO_RTP_FLAG_SEND_CLOCK:
		*(int*)ptr = rtp->m_send_clock;
		break;
	case AUDIO_RTP_FLAG_RECV_CLOCK:
		*(int*)ptr = rtp->m_recv_clock;
		break;
	case AUDIO_RTP_FLAG_SEND_HOOK:
		*(void (**)(const char*,int,void*))ptr = rtp->m_on_send_hook;
		break;
	case AUDIO_RTP_FLAG_RECV_HOOK:
		*(void (**)(const char*,int,void*))ptr = rtp->m_on_recv_hook;
		break;
	case AUDIO_RTP_FLAG_HOOK_ARG:
		*(void**)ptr = rtp->m_hook_data;
		break;
	default:
		return APR_BADARG;
	}
	return APR_SUCCESS;
}

apr_status_t audiortp_init(audiortp_t *ar)
{
	audiortp_t *audiortp = (audiortp_t *)ar;
	audiortp->m_send_samplebit = pt2bitsample(audiortp->m_send_pt);
	audiortp->m_recv_samplebit = pt2bitsample(audiortp->m_recv_pt);
	audiortp->m_send_psize = audiortp->m_send_ptime * audiortp->m_send_clock / 1000 * audiortp->m_send_samplebit / 8;
	audiortp->m_recv_psize = audiortp->m_recv_ptime * audiortp->m_recv_clock / 1000 * audiortp->m_recv_samplebit / 8;
	audiortp->m_jbuf_ptime = audiortp->m_recv_ptime;
	audiortp->m_jbuf_psize = audiortp->m_jbuf_ptime * audiortp->m_recv_clock / 1000 * audiortp->m_recv_samplebit / 8;
	jbuf_create(audiortp->m_jbuf_psize, audiortp->m_jbuf_ptime, JB_MAX, &audiortp->m_jitterbuf);
	jbuf_set_adaptive(audiortp->m_jitterbuf, 0, 7, JB_MAX);
	DEBUG_TRACE("audiortp init, send_psize:%d, recv_psize:%d", audiortp->m_send_psize, audiortp->m_recv_psize);
	return APR_SUCCESS;
}

apr_status_t audiortp_term(audiortp_t *self)
{
	audiortp_t *audiortp = (audiortp_t *)self;
	jbuf_destroy(audiortp->m_jitterbuf);
	audiortp->m_jitterbuf = 0;
	return APR_SUCCESS;
}

apr_status_t audiortp_send_dtmf(audiortp_t *self, const char *digits, unsigned cnt)
{
	audiortp_t *audiortp = (audiortp_t *)self;
	unsigned count = 0;

	if (!(audiostream_get_direction(&audiortp->base)&STREAM_DIR_WRITE)) {
		AUDIO_LOG_ERROR("audio rtp cannot used for send!");
		return APR_EGENERAL;
	}

	if (!digits || !cnt)
		return -1;
	if (!(audiostream_get_direction(&audiortp->base)&STREAM_DIR_WRITE))
		return -1;
	EnterCriticalSection(&audiortp->m_dtmf_lock);
	while (CIRC_LEN(audiortp->m_send_dtmf_rd, audiortp->m_send_dtmf_wr, MAX_DTMF) < MAX_DTMF-1 && count < cnt) {
		rtp_dtmf_event *dtmf = &audiortp->m_send_dtmf[audiortp->m_send_dtmf_wr];
		dtmf->event = (unsigned char)digit2event(digits[count]);
		dtmf->e_vol = DTMF_VOLUME;
		dtmf->p = 0;
		dtmf->e = 0;
		dtmf->duration = 0;
		audiortp->m_send_dtmf_wr = CIRC_INC(audiortp->m_send_dtmf_wr, MAX_DTMF);
		count ++;
	}
	LeaveCriticalSection(&audiortp->m_dtmf_lock);

	return APR_SUCCESS;
}

apr_status_t audiortp_recv_dtmf(audiortp_t *self, char *digits, unsigned *cnt)
{
	audiortp_t *audiortp = (audiortp_t *)self;
	unsigned count = 0;
	int i = 0;

	if (!(audiostream_get_direction(&audiortp->base)&STREAM_DIR_READ)) {
		AUDIO_LOG_ERROR("audio rtp cannot used for recv!");
		return APR_EGENERAL;
	}

	if (!digits)
		return APR_BADARG;

	EnterCriticalSection(&audiortp->m_dtmf_lock);
	while (count < *cnt && i < audiortp->m_recv_dtmf_cnt) {
		digits[count++] = audiortp->m_recv_dtmf[i++];
	}
	if (i < audiortp->m_recv_dtmf_cnt)
		memmove(&audiortp->m_recv_dtmf[0], &audiortp->m_recv_dtmf[i], audiortp->m_recv_dtmf_cnt-i);
	audiortp->m_recv_dtmf_cnt -= i;
	*cnt = count;
	LeaveCriticalSection(&audiortp->m_dtmf_lock);

	return APR_SUCCESS;
}

apr_status_t audiortp_reset_jitter(audiortp_t *audiortp)
{
	jbuf_reset(audiortp->m_jitterbuf);
	return APR_SUCCESS;
}