啓動Linphone MediaStreamer2音頻和視頻流一起

Question

https://github.com/BelledonneCommunications/mediastreamer2 https://github.com/BelledonneCommunications/linphone-android

只使用mediastreamer2庫我能夠開始與遠程IP和指定端口，並呼籲各audiostream.c方法音頻呼叫。

我還需要開始視頻通話。所以初始化videostream.c和 使用其各自的方法，並提供表面渲染遠程和本地攝像頭的飼料。我能夠用遠程端口和ip給出成功啓動視頻流。

但問題是當我開始兩個流在一起。聲音停止，視頻流也停止。只有本地攝像頭供稿。

所以我有一個神奇的方法，爲我做這一切。如果我評論它的視頻部分。音頻通話工作正常，如果評論音頻部分，視頻通話工作正常。但是當我開始時。沒有聲音沒有流媒體。 但我們獲得AudioStream啓動成功並且VideoStream啓動成功登錄。

有人可以通過linphone的經驗幫助找出正確的方法順序嗎？或者我們做錯了什麼。這是我們的方法。

JNIEXPORT jint JNICALL Java_com_myapp_services_LinPhoneMSEngine_LinPhoneMSVE_1AudioStreamStartFull 
(JNIEnv *env, jclass self, jstring remote_ip, jint remote_port, jint localport, jint payloadindex, jboolean isLowEnd) 
{ 
int bRetVal = 0; 
MSVideoSize size = {320, 240}; 
char rtcp_tool[128]={0}; 
int ret; 
//jboolean copy; 
char cname[128]={0}; 
const char *cremote_ip; 

ortp_warning("Audio Stream Start Full"); 
LOGD("Audio Stream Start Full"); 

cremote_ip = (*env)->GetStringUTFChars(env, remote_ip, NULL); 
ortp_warning("Cremote_ip= %s", cremote_ip); 
LOGD("Cremote_ip= %s", cremote_ip); 

// ms_filter_enable_statistics(TRUE); 

veData->queue = ortp_ev_queue_new(); 
veData->soundCard = NULL; 

set_playback_device(); 


ortp_warning("sound: playback_dev_id: %s", ms_snd_card_get_string_id(veData->soundCard)); 
LOGD("sound: playback_dev_id: %s", ms_snd_card_get_string_id(veData->soundCard)); 

veData->CaptureCard = NULL; 
set_capture_device(); 

ortp_warning("sound: capture_dev_id: %s", ms_snd_card_get_string_id(veData->CaptureCard)); 
LOGD("sound: capture_dev_id: %s", ms_snd_card_get_string_id(veData->CaptureCard)); 


veData->audioStream = audio_stream_new(msFactory ,localport, localport + 1, false); 

audio_stream_enable_adaptive_bitrate_control(veData->audioStream, true); 
audio_stream_enable_adaptive_jittcomp(veData->audioStream, true); 
rtp_session_set_jitter_compensation(veData->audioStream->ms.sessions.rtp_session, 50); 

rtp_session_enable_rtcp_mux(veData->audioStream->ms.sessions.rtp_session, true); 
ret=AUDIO_STREAM_FEATURE_VOL_SND | \ 
      AUDIO_STREAM_FEATURE_VOL_RCV; 

if (!isLowEnd) 
{ 
    ret = ret | AUDIO_STREAM_FEATURE_EC | AUDIO_STREAM_FEATURE_EQUALIZER | AUDIO_STREAM_FEATURE_DTMF | AUDIO_STREAM_FEATURE_DTMF_ECHO; 
    audio_stream_set_features(veData->audioStream, ret); 

    ortp_warning("Setting Echo Canceller params"); 
    LOGD("Setting Echo Canceller params"); 

    rtp_session_enable_jitter_buffer(veData->audioStream->ms.sessions.rtp_session, TRUE); 

    audio_stream_set_echo_canceller_params(veData->audioStream, 60, 0, 128); 

    audio_stream_enable_gain_control(veData->audioStream, true); 
    audio_stream_enable_automatic_gain_control(veData->audioStream, true); 

} 
else 
{ 
    audio_stream_set_features(veData->audioStream, ret); 

    ortp_warning("No Echo Canceller params!!"); 
    LOGD("No Echo Canceller params!!"); 
    rtp_session_enable_jitter_buffer(veData->audioStream->ms.sessions.rtp_session, FALSE); 

} 

if(veData->audioStream == NULL){ 
    ortp_warning("AudioStream is Null"); 
    LOGD("AudioStream is Null"); 
    bRetVal = -1; 
    return -1; 
} 

audio_stream_play_received_dtmfs(veData->audioStream, true); 


snprintf(rtcp_tool,sizeof(rtcp_tool)-1,"%s-%s","Android","2.8.0"); 

snprintf(cname,sizeof(cname)-1,"%s-%d", cremote_ip, remote_port); 
ortp_warning("cname value: %s",cname); 
LOGD("cname value: %s",cname); 
audio_stream_prepare_sound(veData->audioStream, veData->soundCard, veData->CaptureCard); 

if(0== audio_stream_start_full(veData->audioStream,veData->prof, cremote_ip, remote_port, cremote_ip, remote_port + 1, 114, 50,NULL,NULL,veData->soundCard,veData->CaptureCard, !isLowEnd)) 
{ 
    veData->rtpSession = veData->audioStream->ms.sessions.rtp_session; 
    ortp_warning("AudioStreamStartFull Success"); 
    post_audio_config(veData->audioStream); 
    audio_stream_set_rtcp_information(veData->audioStream, cname, rtcp_tool); 
} 
else 
{ 
    ortp_warning("AudioStream start failed"); 
    bRetVal = -1; 
} 

// init video stream 
veData->videoStream = video_stream_new(msFactory, localport,localport+1,false); 

video_stream_enable_adaptive_bitrate_control(veData->videoStream, true); 
video_stream_enable_adaptive_jittcomp(veData->videoStream, true); 
rtp_session_enable_rtcp_mux(veData->videoStream->ms.sessions.rtp_session, true); 

video_stream_use_video_preset(veData->videoStream, "custom"); 
video_stream_set_sent_video_size(veData->videoStream, size); 
video_stream_set_preview_size(veData->videoStream, size); 
video_stream_enable_self_view(veData->videoStream, TRUE); 

ortp_message("Video Stream : [%p] & native window id : [%p]",veData->videoStream, veData->native_window_id); 
video_stream_set_native_window_id(veData->videoStream, veData->native_window_id); 

ortp_message("Video Stream : [%p] & preview window id : [%p]",veData->videoStream, veData->native_preview_window_id); 
video_stream_set_native_preview_window_id(veData->videoStream, veData->native_preview_window_id); 

video_stream_use_preview_video_window(veData->videoStream, TRUE); 
video_stream_set_device_rotation(veData->videoStream, 0); 

video_stream_set_fps(veData->videoStream, 10.0); 

// link audio with video 
audio_stream_link_video(veData->audioStream, veData->videoStream); 

ms_message("Setting webcam as %p", veData->msWebCam); 
if(bRetVal != -1 && video_stream_start(veData->videoStream, veData->prof, 
         cremote_ip, 
         remote_port, 
         cremote_ip, 
         remote_port + 1, 
         101, 
         60, 
         veData->msWebCam) >=0) { 

    ortp_warning("VideoStream started successfully"); 
    veData->rtpSession = veData->videoStream->ms.sessions.rtp_session; 

    video_stream_set_rtcp_information(veData->videoStream, cname,rtcp_tool); 
} 
else 
{ 
    ortp_warning("VideoStream start failed"); 
    bRetVal = -1; 

} 

(*env)->ReleaseStringUTFChars(env, remote_ip, cremote_ip); 


return bRetVal; 
} 

Answer 1

好的，最後在@ belledonne-communications的幫助下。我們發現我們在同一個端口上發送了兩個流。

這是不可能的。它需要在不同的端口上發送。我們糾正它，它工作。