view src/mediastreamer/audiostream.c @ 12116:e75ef7aa913e

[gaim-migrate @ 14416] " This patch implements a replacement for the queuing system from 1.x. It also obsoletes a previous patch [#1338873] I submitted to prioritize the unseen states in gtk conversations. The attached envelope.png is ripped from the msgunread.png already included in gaim. It should be dropped in the pixmaps directory (Makefile.am is updated accordingly in this patch). The two separate queuing preferences from 1.x, queuing messages while away and queuing all new messages (from docklet), are replaced with a single 3-way preference for conversations. The new preference is "Hide new IM conversations". This preference can be set to never, away and always. When a gtk conversation is created, it may be placed in a hidden conversation window instead of being placed normally. This decision is based upon the preference and possibly the away state of the account the conversation is being created for. This *will* effect conversations the user explicitly requests to be created, so in these cases the caller must be sure to present the conversation to the user, using gaim_gtkconv_present_conversation(). This is done already in gtkdialogs.c which handles creating conversations requested by the user from gaim proper (menus, double-clicking on budy in blist, etc.). The main advantage to not queuing messages is that the conversations exist, the message is written to the conversation (and logged if appropriate) and the unseen state is set on the conversation. This means no additional features are needed to track whether there are queued messages or not, just use the unseen state on conversations. Since conversations may not be visible (messages "queued"), gaim proper needs some notification that there are messages waiting. I opted for a menutray icon that shows up when an im conversation has an unseen message. Clicking this icon will focus (and show if hidden) the first conversation with an unseen message. This is essentially the same behavior of the docklet in cvs right now, except that the icon is only visible when there is a conversation with an unread message. The api that is added is flexible enough to allow either the docklet or the new blist menutray icon to be visible for conversations of any/all types and for unseen messages >= any state. Currently they are set to only IM conversations and only unseen states >= TEXT (system messages and no log messages will not trigger blinking the docklet or showing the blist tray icon), but these could be made preferences relatively easily in the future. Other plugins could probably benefit as well: gaim_gtk_conversations_get_first_unseen(). There is probably some limit to comment size, so I'll stop rambling now. If anyone has more questions/comments, catch me in #gaim, here or on gaim-devel." 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
	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;
	/* 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);

	/* not sure what this should be returning, nothing in mediastreamer calls
	 * it directly, assuming 0 is okay here. -- Gary
	 */
	return 0;
}