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c170c3911b
current CVS of net/linphone http://savannah.gnu.org/projects/linphone/) library [1] o Pacify gcc: add missing "return" values to non-void functions o This last round of patches get voice communications to "realtime" responsiveness levels. Previously, there were a couple seconds of latency on communications which made them a patience test. - Particularly, the change on talk/third_party/mediastreamer/msrtprecv.c where we now rely on the rtp session's send_pt payload instead of the recv_pt one o Bump PORTREVISION Obtained from: linphone's CVS [1]
69 lines
2.9 KiB
C
69 lines
2.9 KiB
C
--- ./talk/third_party/mediastreamer/audiostream.c.orig Thu Mar 16 18:43:07 2006
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+++ ./talk/third_party/mediastreamer/audiostream.c Fri Apr 21 10:56:34 2006
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@@ -29,6 +29,8 @@
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#define MAX_RTP_SIZE 1500
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+#define rtp_session_max_buf_size_set(session, bufsize) (rtp_session_set_recv_buf_size(session, bufsize))
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+
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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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@@ -118,7 +120,8 @@
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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_send_payload_type(rtpr,payload);
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+ rtp_session_set_recv_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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@@ -143,7 +146,8 @@
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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_send_payload_type(rtps,payload);
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+ rtp_session_set_recv_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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@@ -156,9 +160,9 @@
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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_send_payload_type(rtpr,payload);
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+ rtp_session_set_recv_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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@@ -179,8 +183,6 @@
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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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@@ -217,8 +219,8 @@
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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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+ ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->send_fmtp);
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+ ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->recv_fmtp);
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/* create the synchronisation source */
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stream->timer=ms_timer_new();
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@@ -340,4 +342,5 @@
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{
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ms_rtp_send_dtmf(MS_RTP_SEND(stream->rtpsend), dtmf);
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ms_oss_write_play_dtmf(MS_OSS_WRITE(stream->soundwrite),dtmf);
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+ return 0;
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}
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