Commit 006d694f authored by Sandrine Avakian's avatar Sandrine Avakian

Merge branch 'master' of git.linphone.org:mediastreamer2 into dev_msfactory

# Conflicts:
#	src/voip/audiostream.c
parents 9abf28b2 7b1b79e2
......@@ -121,6 +121,11 @@ include(CheckIncludeFile)
include(CheckLibraryExists)
include(CMakePushCheckState)
if(MSVC AND MSVC_VERSION LESS 1800)
set(MSVC_INCLUDE_DIR "${CMAKE_CURRENT_SOURCE_DIR}/include/MSVC")
list(APPEND CMAKE_REQUIRED_INCLUDES "${MSVC_INCLUDE_DIR}")
endif()
find_package(Threads)
check_include_file(sys/shm.h HAVE_SYS_SHM_H)
......@@ -379,7 +384,7 @@ endif()
if(PCAP_FOUND)
include_directories(${PCAP_INCLUDE_DIRS})
endif()
if(MSVC)
if(MSVC_INCLUDE_DIR)
include_directories(${MSVC_INCLUDE_DIR})
endif()
......
......@@ -525,7 +525,8 @@ static int ms_dtls_srtp_rtp_process_on_receive(struct _RtpTransportModifier *t,
}
if (ctx->role != MSDtlsSrtpRoleIsServer) { /* close the connection only if we are client, if we are server, the client may ask again for last packets */
ret = ssl_close_notify( &(ctx->rtp_dtls_context->ssl) );
/*FireFox version 43 requires DTLS channel to be kept openned, probably a bug in FireFox ret = ssl_close_notify( &(ctx->rtp_dtls_context->ssl) );*/
}
}
......@@ -596,7 +597,7 @@ static int ms_dtls_srtp_rtcp_process_on_receive(struct _RtpTransportModifier *t,
ms_dtls_srtp_check_channels_status(ctx);
}
}
ret = ssl_close_notify( &(ctx->rtcp_dtls_context->ssl) );
/*FireFox version 43 requires DTLS channel to be kept openned, probably a bug in FireFox ret = ssl_close_notify( &(ctx->rtcp_dtls_context->ssl) );*/
}
return 0;
......
......@@ -1084,14 +1084,14 @@ int audio_stream_start_from_io(AudioStream *stream, RtpProfile *profile, const c
if (decoder_have_plc == 0) {
stream->plc = ms_factory_create_filter(stream->ms.factory, MS_GENERIC_PLC_ID);
}
if (stream->plc) {
ms_filter_call_method(stream->plc, MS_FILTER_SET_NCHANNELS, &nchannels);
ms_filter_call_method(stream->plc, MS_FILTER_SET_SAMPLE_RATE, &sample_rate);
/*as first rough approximation, a codec without PLC capabilities has no VAD/DTX builtin, thus setup generic confort noise if possible*/
setup_generic_confort_noise(stream);
}
if (stream->plc) {
ms_filter_call_method(stream->plc, MS_FILTER_SET_NCHANNELS, &nchannels);
ms_filter_call_method(stream->plc, MS_FILTER_SET_SAMPLE_RATE, &sample_rate);
}
/*as first rough approximation, a codec without PLC capabilities has no VAD/DTX builtin, thus setup generic confort noise if possible*/
setup_generic_confort_noise(stream);
} else {
stream->plc = NULL;
}
......
......@@ -208,9 +208,9 @@ static void setup_media_streams(MediastreamDatas *args)
/*create the rtp session */
ortp_init();
if (args->is_verbose) {
ortp_set_log_level_mask(ORTP_DEBUG | ORTP_MESSAGE | ORTP_WARNING | ORTP_ERROR | ORTP_FATAL);
ortp_set_log_level_mask(ORTP_LOG_DOMAIN, ORTP_DEBUG | ORTP_MESSAGE | ORTP_WARNING | ORTP_ERROR | ORTP_FATAL);
} else {
ortp_set_log_level_mask(ORTP_MESSAGE | ORTP_WARNING | ORTP_ERROR | ORTP_FATAL);
ortp_set_log_level_mask(ORTP_LOG_DOMAIN, ORTP_MESSAGE | ORTP_WARNING | ORTP_ERROR | ORTP_FATAL);
}
rtp_profile_set_payload(&av_profile, 110, &payload_type_speex_nb);
......
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