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/*
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The mediastreamer library aims at providing modular media processing and I/O
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for linphone, but also for any telephony application.
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Copyright (C) 2001 Simon MORLAT simon.morlat@linphone.org
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This library is free software; you can redistribute it and/or
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modify it under the terms of the GNU Lesser General Public
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License as published by the Free Software Foundation; either
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version 2.1 of the License, or (at your option) any later version.
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This library is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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Lesser General Public License for more details.
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You should have received a copy of the GNU Lesser General Public
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License along with this library; if not, write to the Free Software
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Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02111-1307 USA
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*/
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#include "mediastream.h"
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#ifdef INET6
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#include <sys/types.h>
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#include <sys/socket.h>
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#include <netdb.h>
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#endif
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#define MAX_RTP_SIZE 1500
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/* this code is not part of the library itself, it is part of the mediastream program */
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void audio_stream_free(AudioStream *stream)
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{
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RtpSession *s;
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RtpSession *destroyed=NULL;
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if (stream->rtprecv!=NULL) {
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s=ms_rtp_recv_get_session(MS_RTP_RECV(stream->rtprecv));
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if (s!=NULL){
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destroyed=s;
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rtp_session_destroy(s);
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}
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ms_filter_destroy(stream->rtprecv);
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}
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if (stream->rtpsend!=NULL) {
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s=ms_rtp_send_get_session(MS_RTP_SEND(stream->rtpsend));
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if (s!=NULL){
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if (s!=destroyed)
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rtp_session_destroy(s);
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}
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ms_filter_destroy(stream->rtpsend);
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}
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if (stream->soundread!=NULL) ms_filter_destroy(stream->soundread);
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if (stream->soundwrite!=NULL) ms_filter_destroy(stream->soundwrite);
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if (stream->encoder!=NULL) ms_filter_destroy(stream->encoder);
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if (stream->decoder!=NULL) ms_filter_destroy(stream->decoder);
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if (stream->timer!=NULL) ms_sync_destroy(stream->timer);
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g_free(stream);
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}
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static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
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static void on_dtmf_received(RtpSession *s,gint dtmf,gpointer user_data)
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{
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AudioStream *stream=(AudioStream*)user_data;
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if (dtmf>15){
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g_warning("Unsupported telephone-event type.");
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return;
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}
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g_message("Receiving dtmf %c.",dtmf_tab[dtmf]);
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if (stream!=NULL){
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if (strcmp(stream->soundwrite->klass->name,"OssWrite")==0)
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ms_oss_write_play_dtmf(MS_OSS_WRITE(stream->soundwrite),dtmf_tab[dtmf]);
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}
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}
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static void on_timestamp_jump(RtpSession *s,guint32* ts, gpointer user_data)
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{
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g_warning("The remote sip-phone has send data with a future timestamp: %u,"
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"resynchronising session.",*ts);
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rtp_session_reset(s);
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}
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static const char *ip4local="0.0.0.0";
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static const char *ip6local="::";
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const char *get_local_addr_for(const char *remote)
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{
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const char *ret;
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#ifdef INET6
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char num[8];
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struct addrinfo hints, *res0;
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int err;
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memset(&hints, 0, sizeof(hints));
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hints.ai_family = PF_UNSPEC;
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hints.ai_socktype = SOCK_DGRAM;
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err = getaddrinfo(remote,"8000", &hints, &res0);
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if (err!=0) {
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g_warning ("get_local_addr_for: %s", gai_strerror(err));
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return ip4local;
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}
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ret=(res0->ai_addr->sa_family==AF_INET6) ? ip6local : ip4local;
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freeaddrinfo(res0);
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#else
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ret=ip4local;
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#endif
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return ret;
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}
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void create_duplex_rtpsession(RtpProfile *profile, int locport,char *remip,int remport,
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int payload,int jitt_comp,
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RtpSession **recvsend){
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RtpSession *rtpr;
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rtpr=rtp_session_new(RTP_SESSION_SENDRECV);
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rtp_session_max_buf_size_set(rtpr,MAX_RTP_SIZE);
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rtp_session_set_profile(rtpr,profile);
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rtp_session_set_local_addr(rtpr,get_local_addr_for(remip),locport);
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if (remport>0) rtp_session_set_remote_addr(rtpr,remip,remport);
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rtp_session_set_scheduling_mode(rtpr,0);
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rtp_session_set_blocking_mode(rtpr,0);
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rtp_session_set_payload_type(rtpr,payload);
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rtp_session_set_jitter_compensation(rtpr,jitt_comp);
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rtp_session_enable_adaptive_jitter_compensation(rtpr,TRUE);
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/*rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)on_timestamp_jump,NULL);*/
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*recvsend=rtpr;
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}
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void create_rtp_sessions(RtpProfile *profile, int locport,char *remip,int remport,
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int payload,int jitt_comp,
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RtpSession **recv, RtpSession **send){
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RtpSession *rtps,*rtpr;
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PayloadType *pt;
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/* creates two rtp filters to recv send streams (remote part)*/
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rtps=rtp_session_new(RTP_SESSION_SENDONLY);
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rtp_session_max_buf_size_set(rtps,MAX_RTP_SIZE);
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rtp_session_set_profile(rtps,profile);
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#ifdef INET6
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rtp_session_set_local_addr(rtps,"::",locport+2);
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#else
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rtp_session_set_local_addr(rtps,"0.0.0.0",locport+2);
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#endif
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rtp_session_set_remote_addr(rtps,remip,remport);
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rtp_session_set_scheduling_mode(rtps,0);
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rtp_session_set_blocking_mode(rtps,0);
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rtp_session_set_payload_type(rtps,payload);
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rtp_session_set_jitter_compensation(rtps,jitt_comp);
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rtpr=rtp_session_new(RTP_SESSION_RECVONLY);
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rtp_session_max_buf_size_set(rtpr,MAX_RTP_SIZE);
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rtp_session_set_profile(rtpr,profile);
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#ifdef INET6
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rtp_session_set_local_addr(rtpr,"::",locport);
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#else
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rtp_session_set_local_addr(rtpr,"0.0.0.0",locport);
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#endif
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rtp_session_set_scheduling_mode(rtpr,0);
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rtp_session_set_blocking_mode(rtpr,0);
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rtp_session_set_payload_type(rtpr,payload);
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rtp_session_set_jitter_compensation(rtpr,jitt_comp);
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rtp_session_signal_connect(rtpr,"telephone-event",(RtpCallback)on_dtmf_received,NULL);
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rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)on_timestamp_jump,NULL);
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*recv=rtpr;
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*send=rtps;
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}
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AudioStream * audio_stream_start_full(RtpProfile *profile, int locport,char *remip,int remport,
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int payload,int jitt_comp, gchar *infile, gchar *outfile, SndCard *playcard, SndCard *captcard)
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{
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AudioStream *stream=g_new0(AudioStream,1);
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RtpSession *rtps,*rtpr;
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PayloadType *pt;
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/* //create_rtp_sessions(profile,locport,remip,remport,payload,jitt_comp,&rtpr,&rtps); */
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create_duplex_rtpsession(profile,locport,remip,remport,payload,jitt_comp,&rtpr);
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rtp_session_signal_connect(rtpr,"telephone-event",(RtpCallback)on_dtmf_received,(gpointer)stream);
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rtps=rtpr;
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stream->recv_session = rtpr;
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stream->send_session = rtps;
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stream->rtpsend=ms_rtp_send_new();
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ms_rtp_send_set_session(MS_RTP_SEND(stream->rtpsend),rtps);
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stream->rtprecv=ms_rtp_recv_new();
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ms_rtp_recv_set_session(MS_RTP_RECV(stream->rtprecv),rtpr);
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/* creates the local part */
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if (infile==NULL) stream->soundread=snd_card_create_read_filter(captcard);
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else stream->soundread=ms_read_new(infile);
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if (outfile==NULL) stream->soundwrite=snd_card_create_write_filter(playcard);
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else stream->soundwrite=ms_write_new(outfile);
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/* creates the couple of encoder/decoder */
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pt=rtp_profile_get_payload(profile,payload);
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if (pt==NULL){
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g_error("audiostream.c: undefined payload type.");
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return NULL;
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}
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stream->encoder=ms_encoder_new_with_string_id(pt->mime_type);
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stream->decoder=ms_decoder_new_with_string_id(pt->mime_type);
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if ((stream->encoder==NULL) || (stream->decoder==NULL)){
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/* big problem: we have not a registered codec for this payload...*/
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audio_stream_free(stream);
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g_error("mediastream.c: No decoder available for payload %i.",payload);
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return NULL;
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}
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/* give the sound filters some properties */
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ms_filter_set_property(stream->soundread,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
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ms_filter_set_property(stream->soundwrite,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
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/* give the encoder/decoder some parameters*/
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ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
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ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_BITRATE,&pt->normal_bitrate);
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ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FREQ,&pt->clock_rate);
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ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_BITRATE,&pt->normal_bitrate);
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ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FMTP, (void*)pt->fmtp);
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ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->fmtp);
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/* create the synchronisation source */
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stream->timer=ms_timer_new();
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/* and then connect all */
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ms_filter_add_link(stream->soundread,stream->encoder);
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ms_filter_add_link(stream->encoder,stream->rtpsend);
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ms_filter_add_link(stream->rtprecv,stream->decoder);
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ms_filter_add_link(stream->decoder,stream->soundwrite);
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ms_sync_attach(stream->timer,stream->soundread);
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ms_sync_attach(stream->timer,stream->rtprecv);
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/* and start */
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ms_start(stream->timer);
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return stream;
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}
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static int defcard=0;
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void audio_stream_set_default_card(int cardindex){
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defcard=cardindex;
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}
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AudioStream * audio_stream_start_with_files(RtpProfile *prof,int locport,char *remip,
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int remport,int profile,int jitt_comp,gchar *infile, gchar*outfile)
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{
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return audio_stream_start_full(prof,locport,remip,remport,profile,jitt_comp,infile,outfile,NULL,NULL);
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}
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AudioStream * audio_stream_start(RtpProfile *prof,int locport,char *remip,int remport,int profile,int jitt_comp)
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{
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SndCard *sndcard;
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sndcard=snd_card_manager_get_card(snd_card_manager,defcard);
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return audio_stream_start_full(prof,locport,remip,remport,profile,jitt_comp,NULL,NULL,sndcard,sndcard);
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}
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AudioStream *audio_stream_start_with_sndcards(RtpProfile *prof,int locport,char *remip,int remport,int profile,int jitt_comp,SndCard *playcard, SndCard *captcard)
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{
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g_return_val_if_fail(playcard!=NULL,NULL);
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g_return_val_if_fail(captcard!=NULL,NULL);
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return audio_stream_start_full(prof,locport,remip,remport,profile,jitt_comp,NULL,NULL,playcard,captcard);
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}
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void audio_stream_set_rtcp_information(AudioStream *st, const char *cname){
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if (st->send_session!=NULL){
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rtp_session_set_source_description(st->send_session,cname,NULL,NULL,NULL, NULL,"linphone",
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"This is free software (GPL) !");
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}
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}
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void audio_stream_stop(AudioStream * stream)
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{
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ms_stop(stream->timer);
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ortp_global_stats_display();
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ms_sync_detach(stream->timer,stream->soundread);
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ms_sync_detach(stream->timer,stream->rtprecv);
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ms_filter_remove_links(stream->soundread,stream->encoder);
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ms_filter_remove_links(stream->encoder,stream->rtpsend);
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ms_filter_remove_links(stream->rtprecv,stream->decoder);
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ms_filter_remove_links(stream->decoder,stream->soundwrite);
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audio_stream_free(stream);
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}
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RingStream * ring_start(gchar *file,gint interval,SndCard *sndcard)
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{
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return ring_start_with_cb(file,interval,sndcard,NULL,NULL);
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}
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RingStream * ring_start_with_cb(gchar *file,gint interval,SndCard *sndcard, MSFilterNotifyFunc func,gpointer user_data)
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{
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RingStream *stream;
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int tmp;
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g_return_val_if_fail(sndcard!=NULL,NULL);
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stream=g_new0(RingStream,1);
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stream->source=ms_ring_player_new(file,interval);
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if (stream->source==NULL) {
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g_warning("Could not create ring player. Probably the ring file (%s) does not exist.",file);
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return NULL;
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}
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if (func!=NULL) ms_filter_set_notify_func(MS_FILTER(stream->source),func,user_data);
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stream->sndwrite=snd_card_create_write_filter(sndcard);
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ms_filter_get_property(stream->source,MS_FILTER_PROPERTY_FREQ,&tmp);
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ms_filter_set_property(stream->sndwrite,MS_FILTER_PROPERTY_FREQ,&tmp);
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ms_filter_get_property(stream->source,MS_FILTER_PROPERTY_CHANNELS,&tmp);
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ms_filter_set_property(stream->sndwrite,MS_FILTER_PROPERTY_CHANNELS,&tmp);
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stream->timer=ms_timer_new();
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ms_filter_add_link(stream->source,stream->sndwrite);
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ms_sync_attach(stream->timer,stream->source);
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ms_start(stream->timer);
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return stream;
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}
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void ring_stop(RingStream *stream)
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{
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ms_stop(stream->timer);
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ms_sync_detach(stream->timer,stream->source);
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ms_sync_destroy(stream->timer);
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ms_filter_remove_links(stream->source,stream->sndwrite);
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|
|
|
ms_filter_destroy(stream->source);
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|
|
|
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);
|
|
|
|
}
|