gstreamer - remplace le récepteur audio au moment de l'exécution

Je souhaite supprimer le récepteur alsa et en créer un nouveau lors de l'exécution. L'ancien élément et le nouvel élément ont tous deux été supprimés et créés avec succès. Et j'essaie de reprendre l'audio, mais l'audio n'a pas été joué. Veuillez indiquer la pièce qui ne va pas.gstreamer - remplace le récepteur audio au moment de l'exécution

#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); 
}

Si l'audio ne sonne pas, recherchez ci-dessous pour le lire.

gst_element_seek_simple (pipeline, GST_FORMAT_TIME, gst_flags, 0);
gst_element_set_state (pipeline, GST_STATE_PLAYING);

Je ne connais pas la raison exacte.

Source

2017-09-15 user8616541

gstreamer - remplace le récepteur audio au moment de l'exécution

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