diff options
Diffstat (limited to 'kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/audiostream.c')
| -rw-r--r-- | kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/audiostream.c | 343 |
1 files changed, 343 insertions, 0 deletions
diff --git a/kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/audiostream.c b/kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/audiostream.c new file mode 100644 index 00000000..f4ff4867 --- /dev/null +++ b/kopete/protocols/jabber/jingle/libjingle/talk/third_party/mediastreamer/audiostream.c @@ -0,0 +1,343 @@ +/* + 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 + 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->recv_session = rtpr; + stream->send_session = rtps; + 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", + "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); +} |