view src/mediastreamer/audiostream.c @ 12024:e67993da8a22

[gaim-migrate @ 14317] I strongly suspect CruiseControl is going to yell at me for this. A voice chat API, GUI + mediastreamer. This is what I'm using for Google Talk. This doesn't actually do anything at all. There's no code in the Jabber plugin yet to use this API (although it Works For Me). All it will do is compile and link. If you're lucky. To build this, you should install oRTP from Linphone, Speex and iLBC (also from linphone, I believe). To not build this, ./configure --disable-vv. Most of the configure.ac and Makefile.am hackery was lifted right out of Linphone with a few modifications. It seems to work if you have everything installed or if you --disable-vv. I haven't really tested not having everything installed and not --disabling-vv. It's kinda funky to include all of mediastreamer in the source tree like this, but linphone doesn't build it as a separate library. I'll probably wind up writing them a patch to build it as a .so so we can link it dynamically instead. This code certainly isn't finished. It'll adapt as I progress on the Google code, but it's certainly of more use here in CVS than in my personal tree. committer: Tailor Script <[e-mail]>

/*
  The mediastreamer library aims at providing modular media processing and I/O
	for linphone, but also for any telephony application.
  Copyright (C) 2001  Simon MORLAT [e-mail]
  										
  This library is free software; you can redistribute it and/or
  modify it under the terms of the GNU Lesser General Public
  License as published by the Free Software Foundation; either
  version 2.1 of the License, or (at your option) any later version.

  This library 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
  Lesser General Public License for more details.

  You should have received a copy of the GNU Lesser General Public
  License along with this library; if not, write to the Free Software
  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
*/


#include "mediastream.h"
#ifdef INET6
	#include <sys/types.h>
	#include <sys/socket.h>
	#include <netdb.h>
#endif


#define MAX_RTP_SIZE	1500

/* this code is not part of the library itself, it is part of the mediastream program */
void audio_stream_free(AudioStream *stream)
{
	RtpSession *s;
	RtpSession *destroyed=NULL;
	if (stream->rtprecv!=NULL) {
		s=ms_rtp_recv_get_session(MS_RTP_RECV(stream->rtprecv));
		if (s!=NULL){
			destroyed=s;
			rtp_session_destroy(s);
		}
		ms_filter_destroy(stream->rtprecv);
	}
	if (stream->rtpsend!=NULL) {
		s=ms_rtp_send_get_session(MS_RTP_SEND(stream->rtpsend));
		if (s!=NULL){
			if (s!=destroyed)
				rtp_session_destroy(s);
		}
		ms_filter_destroy(stream->rtpsend);
	}
	if (stream->soundread!=NULL) ms_filter_destroy(stream->soundread);
	if (stream->soundwrite!=NULL) ms_filter_destroy(stream->soundwrite);
	if (stream->encoder!=NULL) ms_filter_destroy(stream->encoder);
	if (stream->decoder!=NULL) ms_filter_destroy(stream->decoder);
	if (stream->timer!=NULL) ms_sync_destroy(stream->timer);
	g_free(stream);
}

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

static void on_dtmf_received(RtpSession *s,gint dtmf,gpointer user_data)
{
	AudioStream *stream=(AudioStream*)user_data;
	if (dtmf>15){
		g_warning("Unsupported telephone-event type.");
		return;
	}
	g_message("Receiving dtmf %c.",dtmf_tab[dtmf]);
	if (stream!=NULL){
		if (strcmp(stream->soundwrite->klass->name,"OssWrite")==0)
			ms_oss_write_play_dtmf(MS_OSS_WRITE(stream->soundwrite),dtmf_tab[dtmf]);
	}
}

static void on_timestamp_jump(RtpSession *s,guint32* ts, gpointer user_data)
{
	g_warning("The remote sip-phone has send data with a future timestamp: %u,"
			"resynchronising session.",*ts);
	rtp_session_reset(s);
}

static const char *ip4local="0.0.0.0";
static const char *ip6local="::";

const char *get_local_addr_for(const char *remote)
{
	const char *ret;
#ifdef INET6
	char num[8];
	struct addrinfo hints, *res0;
	int err;
	memset(&hints, 0, sizeof(hints));
	hints.ai_family = PF_UNSPEC;
	hints.ai_socktype = SOCK_DGRAM;
	err = getaddrinfo(remote,"8000", &hints, &res0);
	if (err!=0) {
		g_warning ("get_local_addr_for: %s", gai_strerror(err));
		return ip4local;
	}
	ret=(res0->ai_addr->sa_family==AF_INET6) ? ip6local : ip4local; 
	freeaddrinfo(res0);
#else
	ret=ip4local;
#endif
	return ret;
}

void create_duplex_rtpsession(RtpProfile *profile, int locport,char *remip,int remport,
				int payload,int jitt_comp,
			RtpSession **recvsend){
	RtpSession *rtpr;
	rtpr=rtp_session_new(RTP_SESSION_SENDRECV);
	rtp_session_max_buf_size_set(rtpr,MAX_RTP_SIZE);
	rtp_session_set_profile(rtpr,profile);
	rtp_session_set_local_addr(rtpr,get_local_addr_for(remip),locport);
	if (remport>0) rtp_session_set_remote_addr(rtpr,remip,remport);
	rtp_session_set_scheduling_mode(rtpr,0);
	rtp_session_set_blocking_mode(rtpr,0);
	rtp_session_set_payload_type(rtpr,payload);
	rtp_session_set_jitter_compensation(rtpr,jitt_comp);
	rtp_session_enable_adaptive_jitter_compensation(rtpr,TRUE);
	/*rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)on_timestamp_jump,NULL);*/
	*recvsend=rtpr;
}

void create_rtp_sessions(RtpProfile *profile, int locport,char *remip,int remport,
				int payload,int jitt_comp,
			RtpSession **recv, RtpSession **send){
	RtpSession *rtps,*rtpr;
	PayloadType *pt;
	/* creates two rtp filters to recv send streams (remote part)*/
	
	rtps=rtp_session_new(RTP_SESSION_SENDONLY);
	rtp_session_max_buf_size_set(rtps,MAX_RTP_SIZE);
	rtp_session_set_profile(rtps,profile);
#ifdef INET6
	rtp_session_set_local_addr(rtps,"::",locport+2);
#else
	rtp_session_set_local_addr(rtps,"0.0.0.0",locport+2);
#endif
	rtp_session_set_remote_addr(rtps,remip,remport);
	rtp_session_set_scheduling_mode(rtps,0);
	rtp_session_set_blocking_mode(rtps,0);
	rtp_session_set_payload_type(rtps,payload);
	rtp_session_set_jitter_compensation(rtps,jitt_comp);
	
	rtpr=rtp_session_new(RTP_SESSION_RECVONLY);
	rtp_session_max_buf_size_set(rtpr,MAX_RTP_SIZE);
	rtp_session_set_profile(rtpr,profile);
#ifdef INET6
	rtp_session_set_local_addr(rtpr,"::",locport);
#else
	rtp_session_set_local_addr(rtpr,"0.0.0.0",locport);
#endif
	rtp_session_set_scheduling_mode(rtpr,0);
	rtp_session_set_blocking_mode(rtpr,0);
	rtp_session_set_payload_type(rtpr,payload);
	rtp_session_set_jitter_compensation(rtpr,jitt_comp);
	rtp_session_signal_connect(rtpr,"telephone-event",(RtpCallback)on_dtmf_received,NULL);
	rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)on_timestamp_jump,NULL);
	*recv=rtpr;
	*send=rtps;
	
}


AudioStream * audio_stream_start_full(RtpProfile *profile, int locport,char *remip,int remport,
				int payload,int jitt_comp, gchar *infile, gchar *outfile, SndCard *playcard, SndCard *captcard)
{
	AudioStream *stream=g_new0(AudioStream,1);
	RtpSession *rtps,*rtpr;
	PayloadType *pt;
	
	//create_rtp_sessions(profile,locport,remip,remport,payload,jitt_comp,&rtpr,&rtps);
	
	create_duplex_rtpsession(profile,locport,remip,remport,payload,jitt_comp,&rtpr);
	rtp_session_signal_connect(rtpr,"telephone-event",(RtpCallback)on_dtmf_received,(gpointer)stream);
	rtps=rtpr;
	
	stream->rtpsend=ms_rtp_send_new();
	ms_rtp_send_set_session(MS_RTP_SEND(stream->rtpsend),rtps);
	stream->rtprecv=ms_rtp_recv_new();
	ms_rtp_recv_set_session(MS_RTP_RECV(stream->rtprecv),rtpr);
	
	
	/* creates the local part */
	if (infile==NULL) stream->soundread=snd_card_create_read_filter(captcard);
	else stream->soundread=ms_read_new(infile);
	if (outfile==NULL) stream->soundwrite=snd_card_create_write_filter(playcard);
	else stream->soundwrite=ms_write_new(outfile);
	
	/* creates the couple of encoder/decoder */
	pt=rtp_profile_get_payload(profile,payload);
	if (pt==NULL){
		g_error("audiostream.c: undefined payload type.");
		return NULL;
	}
	stream->encoder=ms_encoder_new_with_string_id(pt->mime_type);
	stream->decoder=ms_decoder_new_with_string_id(pt->mime_type);
	if ((stream->encoder==NULL) || (stream->decoder==NULL)){
		/* big problem: we have not a registered codec for this payload...*/
		audio_stream_free(stream);
		g_error("mediastream.c: No decoder available for payload %i.",payload);
		return NULL;
	}
	/* give the sound filters some properties */
	ms_filter_set_property(stream->soundread,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
	ms_filter_set_property(stream->soundwrite,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
	
	/* give the encoder/decoder some parameters*/
	ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
	ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_BITRATE,&pt->normal_bitrate);
	ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
	ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_BITRATE,&pt->normal_bitrate);
	
	ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FMTP, (void*)pt->fmtp);
	ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->fmtp);
	/* create the synchronisation source */
	stream->timer=ms_timer_new();
	
	/* and then connect all */
	ms_filter_add_link(stream->soundread,stream->encoder);
	ms_filter_add_link(stream->encoder,stream->rtpsend);
	ms_filter_add_link(stream->rtprecv,stream->decoder);
	ms_filter_add_link(stream->decoder,stream->soundwrite);
	
	ms_sync_attach(stream->timer,stream->soundread);
	ms_sync_attach(stream->timer,stream->rtprecv);
	
	/* and start */
	ms_start(stream->timer);
	
	return stream;
}

static int defcard=0;

void audio_stream_set_default_card(int cardindex){
	defcard=cardindex;
}

AudioStream * audio_stream_start_with_files(RtpProfile *prof,int locport,char *remip,
		int remport,int profile,int jitt_comp,gchar *infile, gchar*outfile)
{
	return audio_stream_start_full(prof,locport,remip,remport,profile,jitt_comp,infile,outfile,NULL,NULL);
}

AudioStream * audio_stream_start(RtpProfile *prof,int locport,char *remip,int remport,int profile,int jitt_comp)
{
	SndCard *sndcard;
	sndcard=snd_card_manager_get_card(snd_card_manager,defcard);
	return audio_stream_start_full(prof,locport,remip,remport,profile,jitt_comp,NULL,NULL,sndcard,sndcard);
}

AudioStream *audio_stream_start_with_sndcards(RtpProfile *prof,int locport,char *remip,int remport,int profile,int jitt_comp,SndCard *playcard, SndCard *captcard)
{
	g_return_val_if_fail(playcard!=NULL,NULL);
	g_return_val_if_fail(captcard!=NULL,NULL);
	return audio_stream_start_full(prof,locport,remip,remport,profile,jitt_comp,NULL,NULL,playcard,captcard);
}

void audio_stream_set_rtcp_information(AudioStream *st, const char *cname){
	if (st->send_session!=NULL){
		rtp_session_set_source_description(st->send_session,cname,NULL,NULL,NULL,NULL,"linphone-" "2.0.0", // SME
											"This is free software (GPL) !");
	}
}

void audio_stream_stop(AudioStream * stream)
{
	
	ms_stop(stream->timer);
	ortp_global_stats_display();
	ms_sync_detach(stream->timer,stream->soundread);
	ms_sync_detach(stream->timer,stream->rtprecv);
	
	ms_filter_remove_links(stream->soundread,stream->encoder);
	ms_filter_remove_links(stream->encoder,stream->rtpsend);
	ms_filter_remove_links(stream->rtprecv,stream->decoder);
	ms_filter_remove_links(stream->decoder,stream->soundwrite);
	
	audio_stream_free(stream);
}

RingStream * ring_start(gchar *file,gint interval,SndCard *sndcard)
{
   return ring_start_with_cb(file,interval,sndcard,NULL,NULL);
}

RingStream * ring_start_with_cb(gchar *file,gint interval,SndCard *sndcard, MSFilterNotifyFunc func,gpointer user_data)
{
	RingStream *stream;
	int tmp;
	g_return_val_if_fail(sndcard!=NULL,NULL);
	stream=g_new0(RingStream,1);
	stream->source=ms_ring_player_new(file,interval);
	if (stream->source==NULL) {
		g_warning("Could not create ring player. Probably the ring file (%s) does not exist.",file);
		return NULL;
	}
  if (func!=NULL) ms_filter_set_notify_func(MS_FILTER(stream->source),func,user_data);
	stream->sndwrite=snd_card_create_write_filter(sndcard);
	ms_filter_get_property(stream->source,MS_FILTER_PROPERTY_FREQ,&tmp);
	ms_filter_set_property(stream->sndwrite,MS_FILTER_PROPERTY_FREQ,&tmp);
	ms_filter_get_property(stream->source,MS_FILTER_PROPERTY_CHANNELS,&tmp);
	ms_filter_set_property(stream->sndwrite,MS_FILTER_PROPERTY_CHANNELS,&tmp);
	stream->timer=ms_timer_new();
	ms_filter_add_link(stream->source,stream->sndwrite);
	ms_sync_attach(stream->timer,stream->source);
	ms_start(stream->timer);
	return stream;
}

void ring_stop(RingStream *stream)
{
	ms_stop(stream->timer);
	ms_sync_detach(stream->timer,stream->source);
	ms_sync_destroy(stream->timer);
	ms_filter_remove_links(stream->source,stream->sndwrite);
	ms_filter_destroy(stream->source);
	ms_filter_destroy(stream->sndwrite);
	g_free(stream);
}

/* returns the latency in samples if the audio device with id dev_id is openable in full duplex mode, else 0 */
gint test_audio_dev(int dev_id)
{
	gint err;
	SndCard *sndcard=snd_card_manager_get_card(snd_card_manager,dev_id);
	if (sndcard==NULL) return -1;
	err=snd_card_probe(sndcard,16,0,8000);
	return err;  /* return latency in number of sample */
}

gint audio_stream_send_dtmf(AudioStream *stream, gchar dtmf)
{
	ms_rtp_send_dtmf(MS_RTP_SEND(stream->rtpsend), dtmf);
	ms_oss_write_play_dtmf(MS_OSS_WRITE(stream->soundwrite),dtmf);
}