gstreamer - 런타임에 오디오 싱크 교체

alsa 싱크를 삭제하고 런타임에 새 싱크를 만들고 싶습니다. 이전 요소와 새 요소가 모두 삭제되고 성공적으로 만들어졌습니다. 그리고 오디오를 다시 시작하려고 시도했지만 오디오가 재생되지 않았습니다. 잘못된 부분을 알려주십시오.gstreamer - 런타임에 오디오 싱크 교체

#include <stdio.h> 
#include <unistd.h> 
#include <glib.h> 
#include <gst/gst.h> 
#include <pthread.h> 

GstElement *pipeline, *src_, *conv_, *alsaaudiosink, *volume_, *fake_sink_; 
gint64 pos, len; 
GstPad *src_pad; 
GstFormat fmt = GST_FORMAT_TIME; 
GstSeekFlags gst_flags = (GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT); 

void OnNewPadAdded (GstElement* src, GstPad* new_pad, gpointer data); 
GstBusSyncReply GstreamerMessageHandler(GstBus* gst_bus, GstMessage* message, gpointer data); 


GstBusSyncReply GstreamerMessageHandler(GstBus* gst_bus, GstMessage* message, gpointer data) { 
    printf("GstreamerMessageHandler\n"); 
    switch(GST_MESSAGE_TYPE(message)) { 
    case GST_MESSAGE_EOS: 
     break; 
    case GST_MESSAGE_BUFFERING: 
     break; 
    case GST_MESSAGE_DURATION: 
     break; 
    case GST_MESSAGE_ERROR: 
     break; 
    case GST_MESSAGE_STATE_CHANGED: 
     break; 
    case GST_MESSAGE_STREAM_STATUS: 
    default: 
     break; 
    } 
    return GST_BUS_DROP; 
} 

void OnNewPadAdded (GstElement* src, GstPad* new_pad, gpointer data) { 
    GstPad *sink_pad_audio = gst_element_get_static_pad (conv_, "sink"); 

    if (GST_PAD_IS_LINKED (sink_pad_audio)) { 
    g_object_unref (sink_pad_audio); 
    return; 
    } 

    gst_pad_link (new_pad, sink_pad_audio); 
    g_object_unref (sink_pad_audio); 
} 

void reset_fake_sink() { 
//remove fakesink 
    gst_element_unlink(conv_, fake_sink_);  
    gst_bin_remove (GST_BIN (pipeline), fake_sink_); 
    gst_element_set_state (fake_sink_, GST_STATE_NULL); 
    gst_object_unref (GST_OBJECT (fake_sink_));  
} 

void create_alsa_sink() { 
    printf("create alsa\n"); 
    alsaaudiosink = gst_element_factory_make ("alsasink", "sink"); 
    // g_object_set(alsaaudiosink, "device", "AdevWebMainOut", NULL); 
    gst_bin_add (GST_BIN (pipeline), alsaaudiosink); 
    gst_element_link(conv_, alsaaudiosink); 
    gst_element_sync_state_with_parent (alsaaudiosink); 
    gst_element_sync_state_with_parent (conv_); 
} 

void remove_alsa_sink() { 
    printf("remove alsa\n"); 
    gst_element_unlink(conv_, alsaaudiosink);  
    gst_bin_remove (GST_BIN (pipeline), alsaaudiosink); 
    gst_element_set_state (alsaaudiosink, GST_STATE_NULL); 
    gst_object_unref (GST_OBJECT (alsaaudiosink));  
    gst_element_sync_state_with_parent (conv_); 
} 

int main (int argc, char *argv[]) { 

    gst_init(&argc, &argv); 

    if (argc != 2) { 
    printf("%s <mp3 filename>\n", __FUNCTION__, argv[0]); 
    exit (-1); 
    } 

    src_ = gst_element_factory_make ("uridecodebin", "source"); 
    conv_ = gst_element_factory_make ("audioconvert", "convert"); 
    fake_sink_ = gst_element_factory_make ("fakesink", NULL); 

    pipeline = gst_pipeline_new ("pipeline"); 

    if (!pipeline || !src_ || !conv_ || !fake_sink_) { 
    return -1; 
    } 

    gst_bin_add_many (GST_BIN (pipeline), src_, conv_, fake_sink_, NULL); 

    if (!gst_element_link (conv_, fake_sink_)) { 
     gst_object_unref(pipeline); 
     return -1; 
    } 

// Bus - Sync. 
    GstBus* gst_bus; 
    gst_bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); 

    gst_bus_set_sync_handler(gst_bus, GstreamerMessageHandler, NULL); 
    gst_object_unref(gst_bus); 
    g_signal_connect (src_, "pad-added", G_CALLBACK (OnNewPadAdded), NULL); 

    g_object_set (src_, "uri", argv[1], NULL); 

    gst_element_set_state(pipeline, GST_STATE_PAUSED); 

    //remove fakesink 
    reset_fake_sink(); 

    //create new sink 
    create_alsa_sink(); 

    gst_element_set_state (pipeline, GST_STATE_PLAYING); 

    static int is_quiting = 0; 

    while (1) { 
    if(!(gst_bin_iterate_elements (GST_BIN (pipeline)))) break; 
    printf(" q:Quit, press'c' : Pause -> remove sink -> create sink -> Play\n"); 
    char ch =0 ; 
    ch=getchar(); 
    switch (ch) { 
    case 'c': 
     /* pause */ 
     gst_element_set_state (pipeline, GST_STATE_PAUSED); 
     remove_alsa_sink(); 
     create_alsa_sink(); 
     gst_element_set_state (pipeline, GST_STATE_PLAYING); 
    break; 
    case 'q': 
     is_quiting = 1; 
    break; 

    default: 

    break; 

    } 

    if(is_quiting) break; 
    } 
    /* stop the pipeline */ 
    gst_element_set_state (pipeline, GST_STATE_NULL); 
    /* we don't need a reference to these objects anymore */ 
    gst_object_unref (GST_OBJECT (pipeline)); 
    /* unreffing the pipeline unrefs the contained elements as well */ 
    exit (0); 
}

오디오가 출력되지 않으면 아래를 검색하여 재생하십시오.

gst_element_seek_simple (파이프 라인, GST_FORMAT_TIME, gst_flags, 0);
gst_element_set_state (파이프 라인, GST_STATE_PLAYING);

나는 정확한 이유를 모른다.

출처

2017-09-15 user8616541

gstreamer - 런타임에 오디오 싱크 교체

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