Mercurial > pidgin.yaz
view src/mediastreamer/audiostream.c @ 13194:7099a443131d
[gaim-migrate @ 15557]
The other half of sf patch #1414103, from Sadrul Habib Chowdhury
"Make the initial selection in the substatus-
edit dialog correct (currently it always opens with the
last status in the list selected)."
committer: Tailor Script <tailor@pidgin.im>
author | Mark Doliner <mark@kingant.net> |
---|---|
date | Thu, 09 Feb 2006 00:00:08 +0000 |
parents | 1c771536a032 |
children |
line wrap: on
line source
/* 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 simon.morlat@linphone.org 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; }