本文整理汇总了C++中GST_BIN函数的典型用法代码示例。如果您正苦于以下问题:C++ GST_BIN函数的具体用法?C++ GST_BIN怎么用?C++ GST_BIN使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了GST_BIN函数的20个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于我们的系统推荐出更棒的C++代码示例。
示例1: gst_nle_source_setup
static void
gst_nle_source_setup (GstNleSource * nlesrc)
{
GstElement *rotate, *videorate, *videoscale, *colorspace, *vident, *cairooverlay, *colorspace2;
GstElement *audiorate, *audioconvert, *audioresample, *aident;
GstElement *a_capsfilter, *v_capsfilter, *last;
GstPad *v_pad, *a_pad;
GstCaps *v_caps, *a_caps;
rotate = gst_element_factory_make ("flurotate", NULL);
videorate = gst_element_factory_make ("videorate", NULL);
nlesrc->videocrop = gst_element_factory_make ("videocrop", NULL);
videoscale = gst_element_factory_make ("videoscale", NULL);
colorspace = gst_element_factory_make ("ffmpegcolorspace", NULL);
v_capsfilter = gst_element_factory_make ("capsfilter", "video_capsfilter");
nlesrc->textoverlay = gst_element_factory_make ("textoverlay", NULL);
cairooverlay = gst_element_factory_make ("cairooverlay", "overlay");
colorspace2 = gst_element_factory_make ("ffmpegcolorspace", NULL);
vident = gst_element_factory_make ("identity", NULL);
v_caps = gst_caps_new_simple ("video/x-raw-yuv",
"format", GST_TYPE_FOURCC, GST_STR_FOURCC ("I420"),
"width", G_TYPE_INT, (gint) nlesrc->width,
"height", G_TYPE_INT, (gint) nlesrc->height,
"pixel-aspect-ratio", GST_TYPE_FRACTION, 1, 1,
"framerate", GST_TYPE_FRACTION,
(gint) nlesrc->fps_n, (gint) nlesrc->fps_d, NULL);
if (rotate) {
gst_caps_set_simple (v_caps, "rotation", G_TYPE_INT, (gint) 0, NULL);
} else {
rotate = gst_element_factory_make ("identity", NULL);
}
gst_pad_set_caps (nlesrc->video_srcpad, v_caps);
g_object_set (videoscale, "add-borders", TRUE, NULL);
g_object_set (vident, "single-segment", TRUE, NULL);
g_object_set (v_capsfilter, "caps", v_caps, NULL);
g_object_set (nlesrc->textoverlay, "valignment", 2, "halignment", 0,
"auto-resize", TRUE, "wrap-mode", 0, "silent", !nlesrc->overlay_title,
NULL);
g_signal_connect (cairooverlay, "draw",
G_CALLBACK (gst_nle_source_draw_overlay), nlesrc);
/* As videorate can duplicate a lot of buffers we want to put it last in this
transformation bin */
gst_bin_add_many (GST_BIN (nlesrc), rotate, nlesrc->videocrop,
videoscale, colorspace, nlesrc->textoverlay, videorate, v_capsfilter,
vident, NULL);
/* cairooverlay forces a colorpsace conversion ro RGB that we want to avoid
* when we are not rendering the watermark */
if (nlesrc->watermark != NULL) {
gst_bin_add_many (GST_BIN (nlesrc), cairooverlay, colorspace2, NULL);
}
gst_element_link_many (rotate, nlesrc->videocrop, videoscale, colorspace,
nlesrc->textoverlay, NULL);
if (nlesrc->watermark != NULL) {
gst_element_link_many (nlesrc->textoverlay, cairooverlay, colorspace2, NULL);
last = colorspace2;
} else {
last = nlesrc->textoverlay;
}
gst_element_link_many (last, videorate, v_capsfilter, vident, NULL);
/* Ghost source and sink pads */
v_pad = gst_element_get_pad (vident, "src");
gst_ghost_pad_set_target (GST_GHOST_PAD (nlesrc->video_srcpad), v_pad);
gst_object_unref (v_pad);
v_pad = gst_element_get_pad (rotate, "sink");
gst_ghost_pad_set_target (GST_GHOST_PAD (nlesrc->video_sinkpad), v_pad);
gst_object_unref (v_pad);
if (nlesrc->with_audio) {
audiorate = gst_element_factory_make ("audiorate", NULL);
audioconvert = gst_element_factory_make ("audioconvert", NULL);
audioresample = gst_element_factory_make ("audioresample", NULL);
a_capsfilter = gst_element_factory_make ("capsfilter", NULL);
aident = gst_element_factory_make ("identity", NULL);
gst_bin_add_many (GST_BIN (nlesrc), audioresample, audioconvert,
audiorate, a_capsfilter, aident, NULL);
gst_element_link_many (audioconvert, audioresample,
audiorate, a_capsfilter, aident, NULL);
a_caps = gst_nle_source_get_audio_caps (nlesrc);
gst_pad_set_caps (nlesrc->audio_srcpad, a_caps);
g_object_set (a_capsfilter, "caps", a_caps, NULL);
g_object_set (aident, "single-segment", TRUE, NULL);
/* Ghost sink and source pads */
a_pad = gst_element_get_pad (aident, "src");
gst_ghost_pad_set_target (GST_GHOST_PAD (nlesrc->audio_srcpad), a_pad);
//.........这里部分代码省略.........
开发者ID:fluendo,项目名称:VAS,代码行数:101,代码来源:gst-nle-source.c
示例2: transcode_file
static void
transcode_file (gchar * uri, gchar * outputuri, GstEncodingProfile * prof)
{
GstElement *pipeline;
GstElement *src;
GstElement *ebin;
GstElement *sink;
GstBus *bus;
GstCaps *profilecaps, *rescaps;
GMainLoop *mainloop;
g_print (" Input URI : %s\n", uri);
g_print (" Output URI : %s\n", outputuri);
sink = gst_element_make_from_uri (GST_URI_SINK, outputuri, "sink");
if (G_UNLIKELY (sink == NULL)) {
g_print ("Can't create output sink, most likely invalid output URI !\n");
return;
}
src = gst_element_factory_make ("uridecodebin", NULL);
if (G_UNLIKELY (src == NULL)) {
g_print ("Can't create uridecodebin for input URI, aborting!\n");
return;
}
/* Figure out the streams that can be passed as-is to encodebin */
g_object_get (src, "caps", &rescaps, NULL);
rescaps = gst_caps_copy (rescaps);
profilecaps = gst_encoding_profile_get_input_caps (prof);
gst_caps_append (rescaps, profilecaps);
/* Set properties */
g_object_set (src, "uri", uri, "caps", rescaps, NULL);
ebin = gst_element_factory_make ("encodebin", NULL);
g_object_set (ebin, "profile", prof, NULL);
g_signal_connect (src, "autoplug-continue", G_CALLBACK (autoplug_continue_cb),
ebin);
g_signal_connect (src, "pad-added", G_CALLBACK (pad_added_cb), ebin);
pipeline = gst_pipeline_new ("encoding-pipeline");
gst_bin_add_many (GST_BIN (pipeline), src, ebin, sink, NULL);
gst_element_link (ebin, sink);
mainloop = g_main_loop_new (NULL, FALSE);
bus = gst_pipeline_get_bus ((GstPipeline *) pipeline);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message", G_CALLBACK (bus_message_cb), mainloop);
if (gst_element_set_state (pipeline,
GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
g_print ("Failed to start the encoding\n");
return;
}
g_main_loop_run (mainloop);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
开发者ID:luisbg,项目名称:gupnp-dlna,代码行数:65,代码来源:dlna-encoding.c
示例3: tsmf_gstreamer_pipeline_build
BOOL tsmf_gstreamer_pipeline_build(TSMFGstreamerDecoder* mdecoder)
{
#if GST_VERSION_MAJOR > 0
const char* video = "appsrc name=videosource ! queue2 name=videoqueue ! decodebin name=videodecoder !";
const char* audio = "appsrc name=audiosource ! queue2 name=audioqueue ! decodebin name=audiodecoder ! audioconvert ! audiorate ! audioresample ! volume name=audiovolume !";
#else
const char* video = "appsrc name=videosource ! queue2 name=videoqueue ! decodebin2 name=videodecoder !";
const char* audio = "appsrc name=audiosource ! queue2 name=audioqueue ! decodebin2 name=audiodecoder ! audioconvert ! audiorate ! audioresample ! volume name=audiovolume !";
#endif
char pipeline[1024];
if (!mdecoder)
return FALSE;
/* TODO: Construction of the pipeline from a string allows easy overwrite with arguments.
* The only fixed elements necessary are appsrc and the volume element for audio streams.
* The rest could easily be provided in gstreamer pipeline notation from command line. */
if (mdecoder->media_type == TSMF_MAJOR_TYPE_VIDEO)
sprintf_s(pipeline, sizeof(pipeline), "%s %s name=videosink", video, tsmf_platform_get_video_sink());
else
sprintf_s(pipeline, sizeof(pipeline), "%s %s name=audiosink", audio, tsmf_platform_get_audio_sink());
DEBUG_TSMF("pipeline=%s", pipeline);
mdecoder->pipe = gst_parse_launch(pipeline, NULL);
if (!mdecoder->pipe)
{
WLog_ERR(TAG, "Failed to create new pipe");
return FALSE;
}
if (mdecoder->media_type == TSMF_MAJOR_TYPE_VIDEO)
mdecoder->src = gst_bin_get_by_name(GST_BIN(mdecoder->pipe), "videosource");
else
mdecoder->src = gst_bin_get_by_name(GST_BIN(mdecoder->pipe), "audiosource");
if (!mdecoder->src)
{
WLog_ERR(TAG, "Failed to get appsrc");
return FALSE;
}
if (mdecoder->media_type == TSMF_MAJOR_TYPE_VIDEO)
mdecoder->queue = gst_bin_get_by_name(GST_BIN(mdecoder->pipe), "videoqueue");
else
mdecoder->queue = gst_bin_get_by_name(GST_BIN(mdecoder->pipe), "audioqueue");
if (!mdecoder->queue)
{
WLog_ERR(TAG, "Failed to get queue");
return FALSE;
}
if (mdecoder->media_type == TSMF_MAJOR_TYPE_VIDEO)
mdecoder->outsink = gst_bin_get_by_name(GST_BIN(mdecoder->pipe), "videosink");
else
mdecoder->outsink = gst_bin_get_by_name(GST_BIN(mdecoder->pipe), "audiosink");
if (!mdecoder->outsink)
{
WLog_ERR(TAG, "Failed to get sink");
return FALSE;
}
g_signal_connect(mdecoder->outsink, "child-added", G_CALLBACK(cb_child_added), mdecoder);
if (mdecoder->media_type == TSMF_MAJOR_TYPE_AUDIO)
{
mdecoder->volume = gst_bin_get_by_name(GST_BIN(mdecoder->pipe), "audiovolume");
if (!mdecoder->volume)
{
WLog_ERR(TAG, "Failed to get volume");
return FALSE;
}
tsmf_gstreamer_change_volume((ITSMFDecoder*)mdecoder, mdecoder->gstVolume*((double) 10000), mdecoder->gstMuted);
}
tsmf_platform_register_handler(mdecoder);
/* AppSrc settings */
GstAppSrcCallbacks callbacks =
{
tsmf_gstreamer_need_data,
tsmf_gstreamer_enough_data,
tsmf_gstreamer_seek_data
};
g_object_set(mdecoder->src, "format", GST_FORMAT_TIME, NULL);
g_object_set(mdecoder->src, "is-live", FALSE, NULL);
g_object_set(mdecoder->src, "block", FALSE, NULL);
g_object_set(mdecoder->src, "blocksize", 1024, NULL);
gst_app_src_set_caps((GstAppSrc *) mdecoder->src, mdecoder->gst_caps);
gst_app_src_set_callbacks((GstAppSrc *)mdecoder->src, &callbacks, mdecoder, NULL);
gst_app_src_set_stream_type((GstAppSrc *) mdecoder->src, GST_APP_STREAM_TYPE_SEEKABLE);
gst_app_src_set_latency((GstAppSrc *) mdecoder->src, 0, -1);
gst_app_src_set_max_bytes((GstAppSrc *) mdecoder->src, (guint64) 0);//unlimited
g_object_set(G_OBJECT(mdecoder->queue), "use-buffering", FALSE, NULL);
g_object_set(G_OBJECT(mdecoder->queue), "use-rate-estimate", FALSE, NULL);
g_object_set(G_OBJECT(mdecoder->queue), "max-size-buffers", 0, NULL);
g_object_set(G_OBJECT(mdecoder->queue), "max-size-bytes", 0, NULL);
//.........这里部分代码省略.........
开发者ID:JunaidLoonat,项目名称:FreeRDP,代码行数:101,代码来源:tsmf_gstreamer.c
示例4: CV_FUNCNAME
bool CvVideoWriter_GStreamer::open( const char * filename, int fourcc,
double fps, CvSize frameSize, bool is_color )
{
CV_FUNCNAME("CvVideoWriter_GStreamer::open");
__BEGIN__;
//actually doesn't support fourcc parameter and encode an avi with jpegenc
//we need to find a common api between backend to support fourcc for avi
//but also to choose in a common way codec and container format (ogg,dirac,matroska)
// check arguments
assert (filename);
assert (fps > 0);
assert (frameSize.width > 0 && frameSize.height > 0);
std::map<int,char*>::iterator encit;
encit=encs.find(fourcc);
if (encit==encs.end())
CV_ERROR( CV_StsUnsupportedFormat,"Gstreamer Opencv backend doesn't support this codec acutally.");
// if(!isInited) {
// gst_init (NULL, NULL);
// isInited = true;
// }
gst_initializer::init();
close();
source=gst_element_factory_make("appsrc",NULL);
file=gst_element_factory_make("filesink", NULL);
enc=gst_element_factory_make(encit->second, NULL);
mux=gst_element_factory_make("avimux", NULL);
color = gst_element_factory_make("ffmpegcolorspace", NULL);
if (!enc)
CV_ERROR( CV_StsUnsupportedFormat, "Your version of Gstreamer doesn't support this codec acutally or needed plugin missing.");
g_object_set(G_OBJECT(file), "location", filename, NULL);
pipeline = gst_pipeline_new (NULL);
GstCaps* caps;
if (is_color) {
input_pix_fmt=1;
caps= gst_video_format_new_caps(GST_VIDEO_FORMAT_BGR,
frameSize.width,
frameSize.height,
(int) (fps * 1000),
1000,
1,
1);
}
else {
input_pix_fmt=0;
caps= gst_caps_new_simple("video/x-raw-gray",
"width", G_TYPE_INT, frameSize.width,
"height", G_TYPE_INT, frameSize.height,
"framerate", GST_TYPE_FRACTION, int(fps),1,
"bpp",G_TYPE_INT,8,
"depth",G_TYPE_INT,8,
NULL);
}
gst_app_src_set_caps(GST_APP_SRC(source), caps);
if (fourcc==CV_FOURCC_DEFAULT) {
gst_bin_add_many(GST_BIN(pipeline), source, color,mux, file, NULL);
if(!gst_element_link_many(source,color,enc,mux,file,NULL)) {
CV_ERROR(CV_StsError, "GStreamer: cannot link elements\n");
}
}
else {
gst_bin_add_many(GST_BIN(pipeline), source, color,enc,mux, file, NULL);
if(!gst_element_link_many(source,color,enc,mux,file,NULL)) {
CV_ERROR(CV_StsError, "GStreamer: cannot link elements\n");
}
}
if(gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE) {
CV_ERROR(CV_StsError, "GStreamer: cannot put pipeline to play\n");
}
__END__;
return true;
}
开发者ID:4auka,项目名称:opencv,代码行数:76,代码来源:cap_gstreamer.cpp
示例5: build_pipeline
void build_pipeline(CustomData *data)
{
GstBus *bus;
GError *error = NULL;
guint flags;
data->count_buffer_fill = 0;
data->no_buffer_fill = 0;
data->buffer_is_slow = 0;
data->counter = 0;
pthread_mutex_lock(&data->mutex);
gst_element_set_state(data->pipeline, GST_STATE_NULL);
kill_object(data->pipeline);
gplayer_error(BUFFER_SLOW, data);
data->delta_index = 0;
data->last_buffer_load = 0;
data->buffering_time = 0;
data->flow_error = FALSE;
data->pipeline = gst_pipeline_new("test-pipeline");
data->allow_seek = FALSE;
/* Build pipeline */
data->source = gst_element_factory_make("uridecodebin", "source");
data->resample = gst_element_factory_make("audioresample", "resample");
data->typefinder = gst_element_factory_make("typefind", "typefind");
data->buffer = gst_element_factory_make("queue2", "buffer");
data->convert = gst_element_factory_make("audioconvert", "convert");
data->volume = gst_element_factory_make("volume", "volume");
data->sink = gst_element_factory_make("autoaudiosink", "sink");
if (!data->pipeline || !data->resample || !data->source || !data->convert || !data->buffer || !data->typefinder || !data->volume || !data->sink)
{
gplayer_error(-1, data);
GPlayerDEBUG("Not all elements could be created.\n");
pthread_mutex_unlock(&data->mutex);
return;
}
gst_bin_add_many(GST_BIN(data->pipeline), data->source, data->buffer, data->typefinder, data->convert, data->resample, data->volume, data->sink,
NULL);
if (!gst_element_link(data->buffer, data->typefinder) || !gst_element_link(data->typefinder, data->convert)
|| !gst_element_link(data->convert, data->resample) || !gst_element_link(data->resample, data->volume)
|| !gst_element_link(data->volume, data->sink))
{
GPlayerDEBUG("Elements could not be linked.\n");
kill_object(data->pipeline);
pthread_mutex_unlock(&data->mutex);
return;
}
g_signal_connect(data->source, "pad-added", (GCallback ) pad_added_handler, data);
g_signal_connect(data->typefinder, "have-type", (GCallback ) cb_typefound, data);
data->target_state = GST_STATE_READY;
gst_element_set_state(data->pipeline, GST_STATE_READY);
bus = gst_element_get_bus(data->pipeline);
g_signal_connect(G_OBJECT(bus), "message::error", (GCallback ) error_cb, data);
g_signal_connect(G_OBJECT(bus), "message::eos", (GCallback ) eos_cb, data);
g_signal_connect(G_OBJECT(bus), "message::tag", (GCallback ) tag_cb, data);
g_signal_connect(G_OBJECT(bus), "message::state-changed", (GCallback ) state_changed_cb, data);
g_signal_connect(G_OBJECT(bus), "message::clock-lost", (GCallback ) clock_lost_cb, data);
kill_object(bus);
pthread_mutex_unlock(&data->mutex);
}
开发者ID:profrook,项目名称:GPlayer,代码行数:69,代码来源:gplayer.c
示例6: main
int main(int argc, char *argv[]) {
GstElement *pipeline, *audio_source, *tee, *audio_queue, *audio_convert, *audio_resample, *audio_sink;
GstElement *video_queue, *visual, *video_convert, *video_sink;
GstBus *bus;
GstMessage *msg;
GstPadTemplate *tee_src_pad_template;
GstPad *tee_audio_pad, *tee_video_pad;
GstPad *queue_audio_pad, *queue_video_pad;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
audio_source = gst_element_factory_make ("audiotestsrc", "audio_source");
tee = gst_element_factory_make ("tee", "tee");
audio_queue = gst_element_factory_make ("queue", "audio_queue");
audio_convert = gst_element_factory_make ("audioconvert", "audio_convert");
audio_resample = gst_element_factory_make ("audioresample", "audio_resample");
audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
video_queue = gst_element_factory_make ("queue", "video_queue");
visual = gst_element_factory_make ("wavescope", "visual");
video_convert = gst_element_factory_make ("videoconvert", "video_convert");
video_sink = gst_element_factory_make ("autovideosink", "video_sink");
/* Create the empty pipeline */
pipeline = gst_pipeline_new ("test-pipeline");
if (!pipeline || !audio_source || !tee || !audio_queue || !audio_convert || !audio_resample || !audio_sink ||
!video_queue || !visual || !video_convert || !video_sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
/* Configure elements */
g_object_set (audio_source, "freq", 215.0f, NULL);
g_object_set (visual, "shader", 0, "style", 1, NULL);
/* Link all elements that can be automatically linked because they have "Always" pads */
gst_bin_add_many (GST_BIN (pipeline), audio_source, tee, audio_queue, audio_convert, audio_resample, audio_sink,
video_queue, visual, video_convert, video_sink, NULL);
if (gst_element_link_many (audio_source, tee, NULL) != TRUE ||
gst_element_link_many (audio_queue, audio_convert, audio_resample, audio_sink, NULL) != TRUE ||
gst_element_link_many (video_queue, visual, video_convert, video_sink, NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (pipeline);
return -1;
}
/* Manually link the Tee, which has "Request" pads */
tee_src_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (tee), "src_%u");
tee_audio_pad = gst_element_request_pad (tee, tee_src_pad_template, NULL, NULL);
g_print ("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad));
queue_audio_pad = gst_element_get_static_pad (audio_queue, "sink");
tee_video_pad = gst_element_request_pad (tee, tee_src_pad_template, NULL, NULL);
g_print ("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad));
queue_video_pad = gst_element_get_static_pad (video_queue, "sink");
if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
gst_pad_link (tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK) {
g_printerr ("Tee could not be linked.\n");
gst_object_unref (pipeline);
return -1;
}
gst_object_unref (queue_audio_pad);
gst_object_unref (queue_video_pad);
/* Start playing the pipeline */
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);
/* Release the request pads from the Tee, and unref them */
gst_element_release_request_pad (tee, tee_audio_pad);
gst_element_release_request_pad (tee, tee_video_pad);
gst_object_unref (tee_audio_pad);
gst_object_unref (tee_video_pad);
/* 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;
}
开发者ID:johlim,项目名称:study,代码行数:87,代码来源:basic-tutorial-7.c
示例7: gst_nle_source_next
static void
gst_nle_source_next (GstNleSource * nlesrc)
{
GstNleSrcItem *item;
GstStateChangeReturn ret;
GstElement *uridecodebin;
GstBus *bus;
GstState state;
nlesrc->index++;
if (nlesrc->index >= g_list_length (nlesrc->queue)) {
gst_nle_source_push_eos (nlesrc);
return;
}
if (nlesrc->source != NULL) {
gst_object_unref (nlesrc->source);
nlesrc->source = NULL;
}
if (nlesrc->decoder != NULL) {
gst_element_set_state (GST_ELEMENT (nlesrc->decoder), GST_STATE_NULL);
gst_element_get_state (GST_ELEMENT (nlesrc->decoder), NULL, NULL, 0);
gst_object_unref (nlesrc->decoder);
}
nlesrc->decoder = gst_pipeline_new ("decoder");
uridecodebin = gst_element_factory_make ("uridecodebin", NULL);
/* Connect signal to recover source element for queries in bytes */
g_signal_connect (uridecodebin, "source-setup",
G_CALLBACK (gst_nle_source_on_source_setup), nlesrc);
gst_bin_add (GST_BIN (nlesrc->decoder), uridecodebin);
g_signal_connect (uridecodebin, "autoplug-select",
G_CALLBACK (lgm_filter_video_decoders), nlesrc);
g_signal_connect (uridecodebin, "pad-added",
G_CALLBACK (gst_nle_source_pad_added_cb), nlesrc);
g_signal_connect (uridecodebin, "no-more-pads",
G_CALLBACK (gst_nle_source_no_more_pads), nlesrc);
bus = GST_ELEMENT_BUS (nlesrc->decoder);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message", G_CALLBACK (gst_nle_source_bus_message),
nlesrc);
item = (GstNleSrcItem *) g_list_nth_data (nlesrc->queue, nlesrc->index);
GST_INFO_OBJECT (nlesrc, "Starting next item with uri:%s", item->file_path);
GST_INFO_OBJECT (nlesrc, "start:%" GST_TIME_FORMAT " stop:%"
GST_TIME_FORMAT " rate:%f", GST_TIME_ARGS (item->start),
GST_TIME_ARGS (item->stop), item->rate);
g_object_set (uridecodebin, "uri", item->file_path, NULL);
nlesrc->seek_done = FALSE;
if (GST_CLOCK_TIME_IS_VALID (item->stop)) {
nlesrc->video_seek_done = FALSE;
nlesrc->audio_seek_done = FALSE;
} else {
nlesrc->video_seek_done = TRUE;
nlesrc->audio_seek_done = TRUE;
}
nlesrc->audio_eos = TRUE;
nlesrc->video_eos = TRUE;
nlesrc->audio_ts = 0;
nlesrc->video_ts = 0;
nlesrc->start_ts = nlesrc->accu_time;
nlesrc->video_linked = FALSE;
nlesrc->audio_linked = FALSE;
nlesrc->item_setup = FALSE;
nlesrc->cached_duration = 0;
GST_DEBUG_OBJECT (nlesrc, "Start ts:%" GST_TIME_FORMAT,
GST_TIME_ARGS (nlesrc->start_ts));
gst_element_set_state (nlesrc->decoder, GST_STATE_PLAYING);
ret = gst_element_get_state (nlesrc->decoder, &state, NULL, 5 * GST_SECOND);
if (ret == GST_STATE_CHANGE_FAILURE) {
GST_WARNING_OBJECT (nlesrc, "Error changing state, selecting next item.");
gst_nle_source_check_eos (nlesrc);
return;
}
nlesrc->seek_done = TRUE;
if (!item->still_picture && GST_CLOCK_TIME_IS_VALID (item->stop)) {
GST_DEBUG_OBJECT (nlesrc, "Sending seek event");
gst_element_seek (nlesrc->decoder, 1, GST_FORMAT_TIME,
GST_SEEK_FLAG_ACCURATE,
GST_SEEK_TYPE_SET, item->start, GST_SEEK_TYPE_SET, item->stop);
}
}
开发者ID:fluendo,项目名称:VAS,代码行数:91,代码来源:gst-nle-source.c
示例8: main
int
main (int argc, char *argv[])
{
GstElement *bin;
GstElement *decodebin, *decconvert;
GstElement *capsfilter, *equalizer, *spectrum, *sinkconvert, *sink;
GstCaps *caps;
GstBus *bus;
GtkWidget *appwindow, *vbox, *hbox, *scale;
int i, num_bands = NBANDS;
GOptionEntry options[] = {
{"bands", 'b', 0, G_OPTION_ARG_INT, &num_bands,
"Number of bands", NULL},
{NULL}
};
GOptionContext *ctx;
GError *err = NULL;
ctx = g_option_context_new ("- demo of audio equalizer");
g_option_context_add_main_entries (ctx, options, NULL);
g_option_context_add_group (ctx, gst_init_get_option_group ());
g_option_context_add_group (ctx, gtk_get_option_group (TRUE));
if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
g_print ("Error initializing: %s\n", err->message);
g_option_context_free (ctx);
g_clear_error (&err);
exit (1);
}
g_option_context_free (ctx);
if (argc < 2) {
g_print ("Usage: %s <uri to play>\n", argv[0]);
g_print (" For optional arguments: --help\n");
exit (-1);
}
gst_init (&argc, &argv);
gtk_init (&argc, &argv);
bin = gst_pipeline_new ("bin");
/* Uri decoding */
decodebin = gst_element_factory_make ("uridecodebin", "decoder");
g_object_set (G_OBJECT (decodebin), "uri", argv[1], NULL);
/* Force float32 samples */
decconvert = gst_element_factory_make ("audioconvert", "decconvert");
capsfilter = gst_element_factory_make ("capsfilter", "capsfilter");
caps =
gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "F32LE",
NULL);
g_object_set (capsfilter, "caps", caps, NULL);
equalizer = gst_element_factory_make ("equalizer-nbands", "equalizer");
g_object_set (G_OBJECT (equalizer), "num-bands", num_bands, NULL);
spectrum = gst_element_factory_make ("spectrum", "spectrum");
g_object_set (G_OBJECT (spectrum), "bands", spect_bands, "threshold", -80,
"post-messages", TRUE, "interval", 500 * GST_MSECOND, NULL);
sinkconvert = gst_element_factory_make ("audioconvert", "sinkconvert");
sink = gst_element_factory_make ("autoaudiosink", "sink");
gst_bin_add_many (GST_BIN (bin), decodebin, decconvert, capsfilter, equalizer,
spectrum, sinkconvert, sink, NULL);
if (!gst_element_link_many (decconvert, capsfilter, equalizer, spectrum,
sinkconvert, sink, NULL)) {
fprintf (stderr, "can't link elements\n");
exit (1);
}
/* Handle dynamic pads */
g_signal_connect (G_OBJECT (decodebin), "pad-added",
G_CALLBACK (dynamic_link), gst_element_get_static_pad (decconvert,
"sink"));
bus = gst_element_get_bus (bin);
gst_bus_add_watch (bus, message_handler, NULL);
gst_object_unref (bus);
appwindow = gtk_window_new (GTK_WINDOW_TOPLEVEL);
gtk_window_set_title (GTK_WINDOW (appwindow), "Equalizer Demo");
g_signal_connect (G_OBJECT (appwindow), "destroy",
G_CALLBACK (on_window_destroy), NULL);
vbox = gtk_box_new (GTK_ORIENTATION_VERTICAL, 6);
drawingarea = gtk_drawing_area_new ();
gtk_widget_set_size_request (drawingarea, spect_bands, spect_height);
g_signal_connect (G_OBJECT (drawingarea), "configure-event",
G_CALLBACK (on_configure_event), (gpointer) spectrum);
gtk_box_pack_start (GTK_BOX (vbox), drawingarea, TRUE, TRUE, 0);
hbox = gtk_box_new (GTK_ORIENTATION_HORIZONTAL, 20);
for (i = 0; i < num_bands; i++) {
GObject *band;
gdouble freq;
//.........这里部分代码省略.........
开发者ID:GrokImageCompression,项目名称:gst-plugins-good,代码行数:101,代码来源:demo.c
示例9: gst_parse_launch_full
GstElement *create_video_sink()
{
GstElement *bin, *decoder = NULL;
GstIterator *iter;
GstIteratorResult res;
GError *error = NULL;
GstPad *pad;
gpointer element = NULL;
const char* decoder_name;
#ifndef DESKTOP
/* create pipeline */
decoder_name = "tividdec20";
bin = gst_parse_launch_full("TIViddec2 genTimeStamps=FALSE \
engineName=decode \
codecName=h264dec numFrames=-1 \
! videoscale method=0 \
! video/x-raw-yuv, format=(fourcc)I420, width=320, height=240 \
! ffmpegcolorspace \
! video/x-raw-rgb, bpp=16 \
! TIDmaiVideoSink displayStd=fbdev displayDevice=/dev/fb0 videoStd=QVGA \
videoOutput=LCD resizer=FALSE accelFrameCopy=TRUE",
NULL, 0, &error);
#else
decoder_name = "decodebin";
bin = gst_parse_launch_full("decodebin \
! videoscale method=0 \
! video/x-raw-yuv, format=(fourcc)I420, width=320, height=240 \
! xvimagesink",
NULL, 0, &error);
#endif
if (!bin) {
g_error("GStreamer: failed to parse video sink pipeline\n");
return NULL;
}
gst_object_set_name(GST_OBJECT(bin), "video-sink");
iter = gst_bin_iterate_elements(GST_BIN(bin));
res = gst_iterator_next (iter, &element);
while (res == GST_ITERATOR_OK) {
gchar *name;
name = gst_object_get_name(GST_OBJECT (element));
if (name) {
if (!strncmp(name, decoder_name, strlen(decoder_name))) {
decoder = GST_ELEMENT(element);
}
g_printf("GS: video sink element: %s \n", name);
g_free (name);
}
gst_object_unref (element);
element = NULL;
res = gst_iterator_next (iter, &element);
}
gst_iterator_free (iter);
if (!decoder) {
/* mem leak */
g_printf("decoder element not found\n");
return NULL;
}
/* add ghostpad */
pad = gst_element_get_static_pad (decoder, "sink");
gst_element_add_pad(bin, gst_ghost_pad_new("sink", pad));
gst_object_unref(GST_OBJECT(pad));
return bin;
}
开发者ID:afenkart,项目名称:ti_gstreamer,代码行数:73,代码来源:ti_da830.c
示例10: OpenDecoder
//.........这里部分代码省略.........
VLC_ENOMOD );
g_object_set( G_OBJECT( p_sys->p_decode_src ), "caps", caps.p_sinkcaps,
"emit-signals", TRUE, "format", GST_FORMAT_BYTES,
"stream-type", GST_APP_STREAM_TYPE_SEEKABLE,
/* Making DecodeBlock() to block on appsrc with max queue size of 1 byte.
* This will make the push_buffer() tightly coupled with the buffer
* flow from appsrc -> decoder. push_buffer() will only return when
* the same buffer it just fed to appsrc has also been fed to the
* decoder element as well */
"block", TRUE, "max-bytes", ( guint64 )1, NULL );
gst_caps_unref( caps.p_sinkcaps );
caps.p_sinkcaps = NULL;
cb.enough_data = NULL;
cb.need_data = NULL;
cb.seek_data = seek_data_cb;
gst_app_src_set_callbacks( GST_APP_SRC( p_sys->p_decode_src ),
&cb, p_dec, NULL );
if( dbin )
{
p_sys->p_decode_in = gst_element_factory_make( "decodebin", NULL );
VLC_GST_CHECK( p_sys->p_decode_in, NULL, "decodebin not found",
VLC_ENOMOD );
//g_object_set( G_OBJECT( p_sys->p_decode_in ),
//"max-size-buffers", 2, NULL );
//g_signal_connect( G_OBJECT( p_sys->p_decode_in ), "no-more-pads",
//G_CALLBACK( no_more_pads_cb ), p_dec );
g_signal_connect( G_OBJECT( p_sys->p_decode_in ), "pad-added",
G_CALLBACK( pad_added_cb ), p_dec );
}
/* videosink: will emit signal for every available buffer */
p_sys->p_decode_out = gst_element_factory_make( "vlcvideosink", NULL );
VLC_GST_CHECK( p_sys->p_decode_out, NULL, "vlcvideosink not found",
VLC_ENOMOD );
p_sys->p_allocator = gst_vlc_picture_plane_allocator_new(
(gpointer) p_dec );
g_object_set( G_OBJECT( p_sys->p_decode_out ), "sync", FALSE, "allocator",
p_sys->p_allocator, "id", (gpointer) p_dec, NULL );
g_signal_connect( G_OBJECT( p_sys->p_decode_out ), "new-buffer",
G_CALLBACK( frame_handoff_cb ), p_dec );
//FIXME: caps_signal
#if 0
g_signal_connect( G_OBJECT( p_sys->p_decode_out ), "new-caps",
G_CALLBACK( caps_handoff_cb ), p_dec );
#else
GST_VLC_VIDEO_SINK( p_sys->p_decode_out )->new_caps = caps_handoff_cb;
#endif
p_sys->p_decoder = GST_ELEMENT( gst_bin_new( "decoder" ) );
VLC_GST_CHECK( p_sys->p_decoder, NULL, "bin not found", VLC_ENOMOD );
p_sys->p_bus = gst_bus_new( );
VLC_GST_CHECK( p_sys->p_bus, NULL, "failed to create bus",
VLC_ENOMOD );
gst_element_set_bus( p_sys->p_decoder, p_sys->p_bus );
gst_bin_add_many( GST_BIN( p_sys->p_decoder ),
p_sys->p_decode_src, p_sys->p_decode_in,
p_sys->p_decode_out, NULL );
gst_object_ref( p_sys->p_decode_src );
gst_object_ref( p_sys->p_decode_in );
gst_object_ref( p_sys->p_decode_out );
b_ret = gst_element_link( p_sys->p_decode_src, p_sys->p_decode_in );
VLC_GST_CHECK( b_ret, FALSE, "failed to link src <-> in",
VLC_EGENERIC );
if( !dbin )
{
b_ret = gst_element_link( p_sys->p_decode_in, p_sys->p_decode_out );
VLC_GST_CHECK( b_ret, FALSE, "failed to link in <-> out",
VLC_EGENERIC );
}
p_dec->fmt_out.i_cat = p_dec->fmt_in.i_cat;
/* set the pipeline to playing */
i_ret = gst_element_set_state( p_sys->p_decoder, GST_STATE_PLAYING );
VLC_GST_CHECK( i_ret, GST_STATE_CHANGE_FAILURE,
"set state failure", VLC_EGENERIC );
p_sys->b_running = true;
/* Set callbacks */
p_dec->pf_decode_video = DecodeBlock;
p_dec->pf_flush = Flush;
return VLC_SUCCESS;
fail:
if( caps.p_sinkcaps )
gst_caps_unref( caps.p_sinkcaps );
if( caps.p_srccaps )
gst_caps_unref( caps.p_srccaps );
if( p_list )
gst_plugin_feature_list_free( p_list );
CloseDecoder( ( vlc_object_t* )p_dec );
return i_rval;
}
开发者ID:CityFire,项目名称:vlc,代码行数:101,代码来源:gstdecode.c
示例11: main
//.........这里部分代码省略.........
strncpy (input, optarg, sizeof (input) / sizeof (input[0]));
break;
case 'f':
frequency = atol (optarg);
break;
case 'h':
printf ("Usage: v4l2src-test [OPTION]...\n");
for (c = 0; long_options[c].name; ++c) {
printf ("-%c, --%s\r\t\t\t\t%s\n", long_options[c].val,
long_options[c].name, long_options_desc[c]);
}
exit (0);
break;
case '?':
/* getopt_long already printed an error message. */
printf ("Use -h to see help message.\n");
break;
default:
abort ();
}
}
/* Print any remaining command line arguments (not options). */
if (optind < argc) {
printf ("Use -h to see help message.\n" "non-option ARGV-elements: ");
while (optind < argc)
printf ("%s ", argv[optind++]);
putchar ('\n');
}
/* init */
gst_init (&argc, &argv);
/* create elements */
if (!(pipeline = gst_pipeline_new ("my_pipeline"))) {
fprintf (stderr, "error: gst_pipeline_new return NULL");
return -1;
}
if (!(source = gst_element_factory_make ("v4l2src", NULL))) {
fprintf (stderr,
"error: gst_element_factory_make (\"v4l2src\", NULL) return NULL");
return -1;
}
if (!(sink = gst_element_factory_make ("xvimagesink", NULL))) {
fprintf (stderr,
"error: gst_element_factory_make (\"xvimagesink\", NULL) return NULL");
return -1;
}
if (numbuffers > -1) {
g_object_set (source, "num-buffers", numbuffers, NULL);
}
if (device[0]) {
g_object_set (source, "device", device, NULL);
}
if (input[0]) {
g_object_set (source, "input", input, NULL);
}
if (frequency) {
g_object_set (source, "frequency", frequency, NULL);
}
/* you would normally check that the elements were created properly */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, my_bus_callback, NULL);
/* put together a pipeline */
gst_bin_add_many (GST_BIN (pipeline), source, sink, NULL);
gst_element_link_pads (source, "src", sink, "sink");
/* start the pipeline */
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
loop = g_main_loop_new (NULL, FALSE);
input_thread = g_thread_try_new ("v4l2src-test", read_user, source, NULL);
if (input_thread == NULL) {
fprintf (stderr, "error: g_thread_try_new() failed");
return -1;
}
g_main_loop_run (loop);
g_thread_join (input_thread);
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
gst_object_unref (bus);
gst_object_unref (pipeline);
gst_deinit ();
return 0;
}
开发者ID:BigBrother-International,项目名称:gst-plugins-good,代码行数:101,代码来源:v4l2src-test.c
示例12:
GstElement* ly_ppl_video_get_element (char *name)
{
GstElement *ele=NULL;
ele=gst_bin_get_by_name(GST_BIN(ly_ppl_video_bin), name);
return ele;
}
开发者ID:lovesnow,项目名称:linnya,代码行数:6,代码来源:ppl.c
示例13: qDebug
//.........这里部分代码省略.........
pipelineString.append("\"");
#if USE_TEE
pipelineString.append(" ! ");
pipelineString.append("tee name=scripttee");
// FIXME: does this case latency?
pipelineString.append(" ! ");
pipelineString.append("queue");
#endif
pipelineString.append(" ! ");
pipelineString.append(hardware->getEncodingPipeline());
pipelineString.append(" ! ");
pipelineString.append("rtph264pay name=rtppay config-interval=1 mtu=500");
pipelineString.append(" ! ");
pipelineString.append("appsink name=sink sync=false max-buffers=1 drop=true");
#if USE_TEE
// Tee (branch) frames for external components
pipelineString.append(" scripttee. ");
// TODO: downscale to 320x240?
pipelineString.append(" ! ");
pipelineString.append("appsink name=ob sync=false max-buffers=1 drop=true");
#endif
qDebug() << "Using pipeline:" << pipelineString;
// Create encoding video pipeline
pipeline = gst_parse_launch(pipelineString.toUtf8(), &error);
if (!pipeline) {
qCritical("Failed to parse pipeline: %s", error->message);
g_error_free(error);
return false;
}
encoder = gst_bin_get_by_name(GST_BIN(pipeline), "encoder");
if (!encoder) {
qCritical("Failed to get encoder");
return false;
}
// Assuming here that X86 uses x264enc
if (hardware->getHardwareName() == "generic_x86") {
g_object_set(G_OBJECT(encoder), "speed-preset", 1, NULL); // ultrafast
g_object_set(G_OBJECT(encoder), "tune", 0x00000004, NULL); // zerolatency
}
if (hardware->getHardwareName() == "tegrak1" ||
hardware->getHardwareName() == "tegrax1") {
//g_object_set(G_OBJECT(encoder), "input-buffers", 2, NULL); // not valid on 1.0
//g_object_set(G_OBJECT(encoder), "output-buffers", 2, NULL); // not valid on 1.0
//g_object_set(G_OBJECT(encoder), "quality-level", 0, NULL);
//g_object_set(G_OBJECT(encoder), "rc-mode", 0, NULL);
}
if (hardware->getHardwareName() == "tegrax2") {
g_object_set(G_OBJECT(encoder), "preset-level", 0, NULL); // 0 == UltraFastPreset for high perf
}
setBitrate(bitrate);
{
GstElement *source;
source = gst_bin_get_by_name(GST_BIN(pipeline), "source");
if (!source) {
qCritical("Failed to get source");
return false;
}
开发者ID:kulve,项目名称:pleco,代码行数:67,代码来源:VideoSender.cpp
示例14: ges_source_create_topbin
GstElement *
ges_source_create_topbin (const gchar * bin_name, GstElement * sub_element, ...)
{
va_list argp;
GstElement *element;
GstElement *prev = NULL;
GstElement *first = NULL;
GstElement *bin;
GstPad *sub_srcpad;
va_start (argp, sub_element);
bin = gst_bin_new (bin_name);
gst_bin_add (GST_BIN (bin), sub_element);
while ((element = va_arg (argp, GstElement *)) != NULL) {
gst_bin_add (GST_BIN (bin), element);
if (prev)
gst_element_link (prev, element);
prev = element;
if (first == NULL)
first = element;
}
va_end (argp);
sub_srcpad = gst_element_get_static_pad (sub_element, "src");
if (prev != NULL) {
GstPad *srcpad, *sinkpad, *ghost;
srcpad = gst_element_get_static_pad (prev, "src");
ghost = gst_ghost_pad_new ("src", srcpad);
gst_pad_set_active (ghost, TRUE);
gst_element_add_pad (bin, ghost);
sinkpad = gst_element_get_static_pad (first, "sink");
if (sub_srcpad)
gst_pad_link (sub_srcpad, sinkpad);
else
g_signal_connect (sub_element, "pad-added", G_CALLBACK (_pad_added_cb),
sinkpad);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
} else if (sub_srcpad) {
GstPad *ghost;
ghost = gst_ghost_pad_new ("src", sub_srcpad);
gst_pad_set_active (ghost, TRUE);
gst_element_add_pad (bin, ghost);
} else {
g_signal_connect (sub_element, "pad-added",
G_CALLBACK (_ghost_pad_added_cb), bin);
}
if (sub_srcpad)
gst_object_unref (sub_srcpad);
return bin;
}
开发者ID:vliaskov,项目名称:gst-editing-services,代码行数:62,代码来源:ges-source.c
示例15: gst_bin_iterate_all_by_interface
static GstElement *find_color_balance_element() {
GstIterator *iterator = gst_bin_iterate_all_by_interface(
GST_BIN(pipeline), GST_TYPE_COLOR_BALANCE);
GstElement *color_balance_element = NULL;
gboolean done = FALSE, hardware = FALSE;
#if GST_CHECK_VERSION(1, 0, 0)
GValue item = G_VALUE_INIT;
#else
gpointer item;
#endif
while (!done) {
switch (gst_iterator_next(iterator, &item)) {
case GST_ITERATOR_OK : {
#if GST_CHECK_VERSION(1, 0, 0)
GstElement *element = g_v
|
请发表评论