Gstreamer rtsp应用程序的音频和视频
我正在尝试开发一个管道应用程序：
gst-launch-1.0 rtspsrc location =“rtsp：//192.168.3.30：8554 / rajvi”latency = 0 name = demux demux。 ！ 队列！ rtpmp4gdepay！ aacparse！ avdec_aac！ audioconvert！ audioresample！ autoaudiosink demux。 ！ 队列！ rtph264depay！ h264parse！ omxh264dec！ videoconvert！ videoscale！ video / x-raw，width = 176，height = 144！ ximagesink
以下是我已经实现的代码：#include
static void onPadAdded（GstElement * element，GstPad * pad，gpointer data）{gchar * name;
name = gst_pad_get_name(pad);
g_print("A new pad %s was createdn", name);
GstCaps * p_caps = gst_pad_get_pad_template_caps (pad);
gchar * description = gst_caps_to_string(p_caps);
g_free(description);
GstElement *depay = GST_ELEMENT(data);
if(gst_element_link_pads(element, name, depay, "sink") == 0)
{
g_print("cb_new_rtspsrc_pad : failed to link elements n");
}
g_free(name);
}
int main(int argc, char *argv[]) {
GstElement *source, *videosink, *audio, *video, *convert, *pipeline, *audioDepay, *audioQueue, *videoQueue,
*audioParse, *audioDecode, *audioConvert, *audioResample, *audioSink, *videoDepay, *videoParser, *videoDecode, *videoConvert, *videoScale, *videoSink;
GstCaps *capsFilter;
GstBus *bus;
GstMessage *msg;
GstPad *pad;
GstPad *sinkpad,*ghost_sinkpad;
gboolean link_ok;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create Elements */
pipeline = gst_pipeline_new("rtsp-pipeline");
source = gst_element_factory_make ("rtspsrc", "source");
/*audio bin*/
audioQueue = gst_element_factory_make ("queue", "audio-queue");
audioDepay = gst_element_factory_make ("rtpmp4gdepay", "audio-depayer");
audioParse = gst_element_factory_make ("aacparse", "audio-parser");
audioDecode = gst_element_factory_make ("avdec_aac", "audio-decoder");
audioConvert = gst_element_factory_make ("audioconvert", "aconv");
audioResample = gst_element_factory_make ("audioresample", "audio-resample");
audioSink = gst_element_factory_make ("autoaudiosink", "audiosink");
if (!audioQueue || !audioDepay || !audioParse || !audioConvert || !audioResample || !audioSink)
{
g_printerr("Cannot create audio elements n");
return 0;
g_object_set(source, "location", "rtsp://192.168.3.30:8554/rajvi", NULL);
g_object_set(source, "latency", 0, NULL);
g_signal_connect(G_OBJECT(source), "pad-added", G_CALLBACK(onPadAdded), audioDepay);
gst_bin_add_many(GST_BIN(pipeline), source, audioQueue, audioDepay, audioParse, audioDecode,
audioConvert, audioResample, audioSink, NULL);
if (!gst_element_link_many(audioQueue, audioDepay, audioParse, audioDecode, audioConvert, audioResample, audioSink, NULL))
{
g_printerr("Error linking fields ...1 n");
return 0;
}
video = gst_bin_new ("videobin");
videoQueue = gst_element_factory_make ("queue", "video-queue");
videoDepay= gst_element_factory_make ("rtph264depay", "video-depayer");
videoParser = gst_element_factory_make ("h264parse", "video-parser");
videoDecode = gst_element_factory_make ("omxh264dec", "video-decoder");
videoConvert = gst_element_factory_make("videoconvert", "convert");
videoScale = gst_element_factory_make("videoscale", "video-scale");
videoSink = gst_element_factory_make("ximagesink", "video-sink");
capsFilter = gst_caps_new_simple("video/x-raw",
"width", G_TYPE_INT, 176,
"height", G_TYPE_INT, 144,
NULL);
if (!videoQueue || !videoDepay || !videoParser || !videoDecode || !videoConvert || !videoScale || !videoSink || !capsFilter)
{
g_printerr("Cannot create video elements n");
return 0;
}
gst_bin_add_many(GST_BIN(video),videoQueue, videoDepay, videoParser, videoDecode, videoConvert, videoScale,
videosink, NULL);
/* set property value */
link_ok = gst_element_link_filtered(videoConvert,videosink, capsFilter);
gst_caps_unref (capsFilter);
if (!link_ok) {
g_warning ("Failed to link element1 and element2!");
}
sinkpad = gst_element_get_static_pad (videoConvert, "sink");
ghost_sinkpad = gst_ghost_pad_new ("sink", sinkpad);
gst_pad_set_active (ghost_sinkpad, TRUE);
gst_element_add_pad (video, ghost_sinkpad);
if (!gst_element_link_many(videoQueue, videoDepay, videoParser, videoDecode, videoScale, NULL))
{
g_printerr("Error linking fields... 2 n");
return 0;
}
gst_bin_add_many (GST_BIN(pipeline), video,NULL);
/* Start playing */
gst_element_set_state ( pipeline, GST_STATE_PLAYING);
/* Wait until error or EOS */
bus = gst_element_get_bus (pipeline);
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Free resources */
if (msg != NULL)
gst_message_unref (msg);
gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
return 0;
}
获取连接管线 - >音频 - >视频箱的错误
如果您将视频和音频放在管道箱中，那么您可以做到这一点。 找出你的视频和音频上限，并且应该能够链接它们。
// ----------------------------------
// pad-added signal
// ----------------------------------
static void onPadAdded(GstElement* element, GstPad* pad, gpointer user_data)
{
gchar *name;
GstCaps * p_caps;
GstElement* nextElement;
GstElement* pipeline = (GstElement*)user_data;
name = gst_pad_get_name(pad);
g_print("A new pad %s was createdn", name);
p_caps = gst_pad_get_pad_template_caps(pad);
if (strstr(name, "[CAPS FOR VIDEO CONTAIN]") != NULL)
{
std::cout << std::endl << "------------------------ Video -------------------------------" << std::endl;
nextElement = gst_bin_get_by_name(GST_BIN(pipeline), "video-depayer");
}
else if (strstr(name, "[CAPS FOR AUDIO CONTAIN]") != NULL)
{
std::cout << std::endl << "------------------------ Audio -------------------------------" << std::endl;
nextElement = gst_bin_get_by_name(GST_BIN(pipeline), "audio-depayer");
}
if (nextElement != NULL)
{
if (!gst_element_link_filtered(element, nextElement, p_caps))
//if (!gst_element_link_pads_filtered(element, name, nextElement, "sink", p_caps))
{
std::cout << std::endl << "Failed to link video element to src to sink" << std::endl;
}
gst_object_unref(nextElement);
}
g_free(name);
gst_caps_unref(p_caps);
}
// ----------------------------------
// main
// ----------------------------------
int main(int argc, char *argv[])
{
GstElement *source, *videosink, *audio,*convert, *pipeline, *audioDepay, *audioQueue, *videoQueue,
*audioParse, *audioDecode, *audioConvert, *audioResample, *audioSink, *videoDepay, *videoParser, *videoDecode, *videoConvert, *videoScale, *videoSink;
GstCaps *capsFilter;
GstBus *bus;
GstMessage *msg;
GstPad *pad;
gboolean link_ok;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init(&argc, &argv);
/* Create Elements */
pipeline = gst_pipeline_new("rtsp-pipeline");
source = gst_element_factory_make("rtspsrc", "source");
/*audio bin*/
audioQueue = gst_element_factory_make("queue", "audio-queue");
audioDepay = gst_element_factory_make("rtpmp4gdepay", "audio-depayer");
audioParse = gst_element_factory_make("aacparse", "audio-parser");
audioDecode = gst_element_factory_make("avdec_aac", "audio-decoder");
audioConvert = gst_element_factory_make("audioconvert", "aconv");
audioResample = gst_element_factory_make("audioresample", "audio-resample");
audioSink = gst_element_factory_make("autoaudiosink", "audiosink");
if (!audioQueue || !audioDepay || !audioParse || !audioConvert || !audioResample || !audioSink)
{
g_printerr("Cannot create audio elements n");
return 0;
g_object_set(source, "location", "rtsp://192.168.3.30:8554/rajvi", NULL);
g_object_set(source, "latency", 0, NULL);
g_signal_connect(G_OBJECT(source), "pad-added", G_CALLBACK(onPadAdded), pipeline);
gst_bin_add_many(GST_BIN(pipeline), source, audioQueue, audioDepay, audioParse, audioDecode,
audioConvert, audioResample, audioSink, NULL);
if (!gst_element_link_many(audioQueue, audioDepay, audioParse, audioDecode, audioConvert, audioResample, audioSink, NULL))
{
g_printerr("Error linking fields ...1 n");
return 0;
}
videoQueue = gst_element_factory_make("queue", "video-queue");
videoDepay = gst_element_factory_make("rtph264depay", "video-depayer");
videoParser = gst_element_factory_make("h264parse", "video-parser");
videoDecode = gst_element_factory_make("omxh264dec", "video-decoder");
videoConvert = gst_element_factory_make("videoconvert", "convert");
videoScale = gst_element_factory_make("videoscale", "video-scale");
videoSink = gst_element_factory_make("ximagesink", "video-sink");
capsFilter = gst_caps_new_simple("video/x-raw",
"width", G_TYPE_INT, 176,
"height", G_TYPE_INT, 144,
NULL);
if (!videoQueue || !videoDepay || !videoParser || !videoDecode || !videoConvert || !videoScale || !videoSink || !capsFilter)
{
g_printerr("Cannot create video elements n");
return 0;
}
gst_bin_add_many(GST_BIN(pipeline), videoQueue, videoDepay, videoParser, videoDecode, videoConvert, videoScale,
videosink, NULL);
/* set property value */
link_ok = gst_element_link_filtered(videoConvert, videosink, capsFilter);
gst_caps_unref(capsFilter);
if (!link_ok) {
g_warning("Failed to link element1 and element2!");
}
if (!gst_element_link_many(videoQueue, videoDepay, videoParser, videoDecode, videoScale, NULL))
{
g_printerr("Error linking fields... 2 n");
return 0;
}
/* Start playing */
gst_element_set_state(pipeline, GST_STATE_PLAYING);
/* Wait until error or EOS */
bus = gst_element_get_bus(pipeline);
msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE,(GstMessageType)( GST_MESSAGE_ERROR | GST_MESSAGE_EOS));
/* Free resources */
if (msg != NULL)
gst_message_unref(msg);
gst_object_unref(bus);
gst_element_set_state(pipeline, GST_STATE_NULL);
gst_object_unref(pipeline);
return 0;
}
}
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