創建gstreamer管道，目的是修改部分

Question

我正在處理音頻流，我希望能夠修改我正在流式傳輸的文件，以及我正在流式傳輸的目標。要做到這一點，我會修改我的filesrc的位置，或者我會修改我的udpsink的主機/端口。

我很難理解我需要知道的一切，以便將此管道連接在一起並進行播放。以前我硬編碼的一切，使用這條管道的GST管道分析工具：

filesrc location=/storage/sdcard0/Music/RunToTheHills.ogg ! oggdemux ! vorbisdec ! audioresample ! audioconvert ! audio/x-raw-int,channels=2,depth=16,width=16,rate=44100 ! rtpL16pay ! udpsink host=192.168.100.126 port=9001 

現在我想改變filesrc位置，並與上述主機的UDP /端口。

我的應用程序是一個使用NDK的Android應用程序。但是，這不應該影響設置管道所需的代碼。

這是我到目前爲止，這導致段錯誤。

我的數據結構：

/** 
* Structure to hold all the various variables we need. 
* This is handed to callbacks 
*/ 
typedef struct _CustomData { 
    jobject app; /* The Java app */ 
    GstElement *pipeline; /* gStreamer pipeline */ 
    GstElement *filesrc; /* Input file */ 
    GstPad *fileblock; /* Used to block filesrc */ 
    GstElement *ogg; /* Ogg demultiplexer */ 
    GstElement *vorbis; /* Vorbis decoder */ 
    GstElement *resample; 
    GstElement *convert; 
    GstCaps *caps; 
    GstElement *rtp; /* RTP packer */ 
    GstElement *udp; /* UDP sender */ 
    GMainContext *context; /* GLib Context */ 
    GMainLoop *main_loop; /* GLib main loop */ 
    gboolean initialised; /* True after initialisation */ 
    GstState state; /* Pipeline state */ 
    GstState target_state; /* What state we want to put the pipeline into */ 
    gint64 duration; /* Clip length */ 
    gint64 desired_position; /* Where we want to track to within the clip */ 
    GstClockTime last_seek_time; /* Used to throttle seeking */ 
    gboolean is_live; /* Live streams don't need buffering */ 
} CustomData; 

而且這裏是我創造的流水線：

data->pipeline = gst_pipeline_new("pipeline"); 

data->filesrc = gst_element_factory_make("filesrc", NULL); 
if (!data->filesrc) { 
    GST_ERROR("Failed to create filesrc."); 
    return NULL; 
} 
g_object_set(G_OBJECT(data->filesrc), "location", "/storage/sdcard0/Music/RunToTheHills.ogg", NULL); 

data->fileblock = gst_element_get_static_pad(data->filesrc, "src"); 

data->ogg = gst_element_factory_make("oggdemux", NULL); 
if (!data->ogg) { 
    GST_ERROR("Failed to create oggdemux."); 
    return NULL; 
} 

data->vorbis = gst_element_factory_make("vorbisdec", NULL); 
if (!data->vorbis) { 
    GST_ERROR("Failed to create vorbisdec."); 
    return NULL; 
} 

data->resample = gst_element_factory_make("audioresample", NULL); 
if (!data->resample) { 
    GST_ERROR("Failed to create audioresample."); 
    return NULL; 
} 

data->convert = gst_element_factory_make("audioconvert", NULL); 
if (!data->convert) { 
    GST_ERROR("Failed to create audioconvert."); 
    return NULL; 
} 

data->caps = gst_caps_new_simple("audio/x-raw-int", 
     "channels", G_TYPE_INT, 2, 
     "depth", G_TYPE_INT, 16, 
     "width", G_TYPE_INT, 16, 
     "rate", G_TYPE_INT, 44100); 

if (!data->caps) { 
    GST_ERROR("Failed to create caps"); 
    return NULL; 
} 

data->rtp = gst_element_factory_make("rtpL16pay", NULL); 
if (!data->rtp) { 
    GST_ERROR("Failed to create rtpL16pay."); 
    return NULL; 
} 

data->udp = gst_element_factory_make("udpsink", NULL); 
if (!data->udp) { 
    GST_ERROR("Failed to create udpsink."); 
    return NULL; 
} 
g_object_set(G_OBJECT(data->udp), "host", "192.168.100.126", NULL); 
g_object_set(G_OBJECT(data->udp), "port", 9001, NULL); 


if (!data->ogg || !data->vorbis || !data->resample || !data->convert || !data->caps || !data->rtp || !data->udp) { 
    GST_ERROR("Unable to create all elements!"); 
    return NULL; 
} 

gst_bin_add_many(GST_BIN(data->pipeline), data->filesrc, data->ogg, data->vorbis, 
     data->resample, data->convert, data->caps, data->rtp, data->udp); 

/* Link all the elements together */ 
gst_element_link(data->filesrc, data->ogg); 
gst_element_link(data->ogg, data->vorbis); 
gst_element_link(data->vorbis, data->resample); 
gst_element_link(data->resample, data->convert); 
gst_element_link_filtered(data->convert, data->rtp, data->caps); 
gst_element_link(data->rtp, data->udp); 

能有人給我一些提示，以我哪裏錯了？

對於利息，這是我以前的工作流水線：

data->pipeline = gst_parse_launch("filesrc location=/storage/sdcard0/Music/RunToTheHills.ogg ! oggdemux ! vorbisdec ! audioresample ! audioconvert ! audio/x-raw-int,channels=2,depth=16,width=16,rate=44100 ! rtpL16pay ! udpsink host=192.168.100.126 port=9001", &error); 
if (error) { 
    gchar *message = g_strdup_printf("Unable to build pipeline: %s", error->message); 
    g_clear_error (&error); 
    set_ui_message(message, data); 
    g_free (message); 
    return NULL; 
} 

Answer 1

您不能簡單地oggdemux鏈接到vorbisdec，因爲分路具有有時墊。

您需要爲demux的「pad-added」信號添加處理函數，然後在那裏執行鏈接。

/* Connect to the pad-added signal */ 
g_signal_connect (data->ogg, "pad-added", G_CALLBACK (pad_added_handler), data); 

而且處理程序：

void on_pad_added (GstElement *src, GstPad *new_pad, CustomData *data) 
{ 
GstPad *sink_pad = gst_element_get_static_pad (data->vorbis, "sink"); 
GstPadLinkReturn ret; 
GstCaps *new_pad_caps = NULL; 
GstStructure *new_pad_struct = NULL; 
const gchar *new_pad_type = NULL; 

g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src)); 

/* If our converter is already linked, we have nothing to do here */ 
if (gst_pad_is_linked (sink_pad)) { 
g_print (" We are already linked. Ignoring.\n"); 
goto exit; 
} 

/* Check the new pad's type */ 
new_pad_caps = gst_pad_get_caps (new_pad); 
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0); 
new_pad_type = gst_structure_get_name (new_pad_struct); 
if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) { 
g_print (" It has type '%s' which is not raw audio. Ignoring.\n", new_pad_type); 
goto exit; 
} 

/* Attempt the link */ 
ret = gst_pad_link (new_pad, sink_pad); 
if (GST_PAD_LINK_FAILED (ret)) { 
g_print (" Type is '%s' but link failed.\n", new_pad_type); 
} else { 
g_print (" Link succeeded (type '%s').\n", new_pad_type); 
} 

exit: 
/* Unreference the new pad's caps, if we got them */ 
if (new_pad_caps != NULL) 
gst_caps_unref (new_pad_caps); 

/* Unreference the sink pad */ 
gst_object_unref (sink_pad); 
} 

而且，因爲你得到一個分段錯誤，我相信這是內存問題。你確定你正在使用CustomData結構嗎？我注意到你正在使用data->元素而不是data.element。