--- ./talk/third_party/mediastreamer/audiostream.c.orig Thu Mar 16 18:43:07 2006 +++ ./talk/third_party/mediastreamer/audiostream.c Fri Apr 21 10:56:34 2006 @@ -29,6 +29,8 @@ #define MAX_RTP_SIZE 1500 +#define rtp_session_max_buf_size_set(session, bufsize) (rtp_session_set_recv_buf_size(session, bufsize)) + /* this code is not part of the library itself, it is part of the mediastream program */ void audio_stream_free(AudioStream *stream) { @@ -118,7 +120,8 @@ 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_send_payload_type(rtpr,payload); + rtp_session_set_recv_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);*/ @@ -143,7 +146,8 @@ 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_send_payload_type(rtps,payload); + rtp_session_set_recv_payload_type(rtps,payload); rtp_session_set_jitter_compensation(rtps,jitt_comp); rtpr=rtp_session_new(RTP_SESSION_RECVONLY); @@ -156,9 +160,9 @@ #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_send_payload_type(rtpr,payload); + rtp_session_set_recv_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; @@ -179,8 +183,6 @@ 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(); @@ -217,8 +219,8 @@ 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); + ms_filter_set_property(stream->encoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->send_fmtp); + ms_filter_set_property(stream->decoder,MS_FILTER_PROPERTY_FMTP,(void*)pt->recv_fmtp); /* create the synchronisation source */ stream->timer=ms_timer_new(); @@ -340,4 +342,5 @@ { ms_rtp_send_dtmf(MS_RTP_SEND(stream->rtpsend), dtmf); ms_oss_write_play_dtmf(MS_OSS_WRITE(stream->soundwrite),dtmf); + return 0; }