我想在Gstreamer中播放一个MP4视频，但是遇到了找不到x264dec的错误。我编写了这条管道：
gst-launch filesrc \
location=/media/thesis/Gstreamer/pub_Irma.mp4 \
! qtdemux name=demux demux. ! queue ! faad \
! audioconvert ! audioresample ! autoaudiosink demuxer. \
! queue ! x264dec ! ffmpegcolorspace ! autovideosink
完成此步骤后，我的目标是生成mp4视频解码的执行跟踪。我不知道x264dec存在什么问题。
这可能更容易：
gst-launch-0.10 playbin2 uri=file:///path/to/foo.mp4
gst-launch-1.0 playbin uri=file:///path/to/foo.mp4
(当然，您仍然需要安装所需的元素)。
我在桌面上有一个名为 tuhi.mp4 的文件。因此，命令应该是这样的。
$ gst-launch-1.0 filesrc location=~/Desktop/tuhi.mp4 ! decodebin name=dec ! videoconvert ! autovideosink dec. ! audioconvert ! audioresample ! alsasink
我认为没有x264dec元素（我只知道x264enc元素）。你可能想使用像这篇nabble帖子中的ffdec_h264：
gst-launch-0.10 filesrc location=$1 ! h264parse ! ffdec_h264 ! ffmpegcolorspace ! deinterlace ! xvimagesink
GST_DEBUG=qtdemux:5,faad:5,ffdec_h264:5 gst-launch filesrc location=/media/thesis/Gstreamer/pub_Irma.mp4 ! qtdemux name=demuxer demuxer. ! queue ! faad ! audioconvert ! audioresample ! autoaudiosink demuxer. ! queue ! ffdec_h264 ! ffmpegcolorspace ! autovideosink 2> >(tee /media/thesis/Gstreamer/log_pub.csv)来获取执行跟踪。 我使用了qtdemux而不是h264parse。 - KKcgst-launch-0.10 filesrc location=/root/gstreamerexamples/big.mp4 ! qtdemux name=demux demux. ! queue ! faad ! audioconvert ! audioresample ! autoaudiosink demux. ! queue ! ffdec_h264 ! ffmpegcolorspace ! autovideosink
对我而言，表现良好。
要播放mp4文件，请使用bigBuckBunny mp4视频作为测试文件。接下来，请确保在x86或任何其他目标平台中已构建gst-launch中提到的所有元素。
请花费大量时间正确安装GStreamer。
#include <gst/gst.h>
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstElement *demuxer;
GstElement *audioqueue;
GstElement *videoqueue;
GstElement *audio_decoder;
GstElement *video_decoder;
GstElement *video_convert;
GstElement *audio_convert;
GstElement *video_sink;
GstElement *audio_sink;
} CustomData;
static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);
int main(int argc, char *argv[]) {
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
gboolean terminate = FALSE;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
data.source = gst_element_factory_make ("filesrc", "source");
data.demuxer = gst_element_factory_make ("oggdemux", "demuxer");
data.audioqueue = gst_element_factory_make("queue","audioqueue");
data.videoqueue = gst_element_factory_make("queue","videoqueue");
data.audio_decoder = gst_element_factory_make ("vorbisdec", "audio_decoder");
data.audio_convert = gst_element_factory_make ("audioconvert", "audio_convert");
data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
data.video_decoder = gst_element_factory_make("theoradec","video_decoder");
data.video_convert = gst_element_factory_make("ffmpegcolorspace","video_convert");
data.video_sink = gst_element_factory_make("autovideosink","video_sink");
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.demuxer || !data.audioqueue ||!data.audio_decoder ||!data.audio_convert ||
!data.audio_sink || !data.videoqueue || !data.video_decoder || !data.video_convert || !data.video_sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
gst_bin_add_many (GST_BIN (data.pipeline), data.source,data.demuxer,data.audioqueue,data.audio_decoder,data.audio_convert,data.audio_sink,data.videoqueue,data.video_decoder,data.video_convert,data.video_sink, NULL);
if (!gst_element_link(data.source,data.demuxer)) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
if (!gst_element_link_many (data.audioqueue,data.audio_decoder,data.audio_convert, data.audio_sink,NULL)) {
g_printerr (" audio Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
if (!gst_element_link_many(data.videoqueue,data.video_decoder,data.video_convert, data.video_sink,NULL)) {
g_printerr("video Elements could not be linked.\n");
gst_object_unref(data.pipeline);
return -1;
}
/* Set the file to play */
g_object_set (data.source, "location", argv[1], NULL);
g_signal_connect (data.demuxer, "pad-added", G_CALLBACK (pad_added_handler), &data);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Listen to the bus */
bus = gst_element_get_bus (data.pipeline);
do {
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (msg, &err, &debug_info);
g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none");
g_clear_error (&err);
g_free (debug_info);
terminate = TRUE;
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
terminate = TRUE;
break;
case GST_MESSAGE_STATE_CHANGED:
if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
GstState old_state, new_state, pending_state;
gst_message_parse_state_changed (msg, &old_state, &new_state, &pending_state);
g_print ("Pipeline state changed from %s to %s:\n",
gst_element_state_get_name (old_state), gst_element_state_get_name (new_state));
}
break;
default:
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
} while (!terminate);
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
/* This function will be called by the pad-added signal */
static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) {
g_print("Inside the pad_added_handler method \n");
GstPad *sink_pad_audio = gst_element_get_static_pad (data->audioqueue, "sink");
GstPad *sink_pad_video = gst_element_get_static_pad (data->videoqueue, "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));
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-vorbis" /*"audio/mpeg"*/))
{
ret = gst_pad_link (new_pad, sink_pad_audio);
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);
}
}
else if (g_str_has_prefix (new_pad_type, "video/x-theora"/*"video/x-h264"*/))
{
ret = gst_pad_link (new_pad, sink_pad_video);
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);
}
}
else {
g_print (" It has type '%s' which is not raw audio. Ignoring.\n", new_pad_type);
goto exit;
}
exit:
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
gst_object_unref (sink_pad_audio);
gst_object_unref (sink_pad_video);
}
我在桌面上有一个ogg文件，所以命令是这样的。
$ gst-launch-1.0 filesrc location=~/Desktop/test.ogg ! oggdemux name=demuxer \
demuxer. ! queue ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink \
demuxer. ! queue ! theoradec ! videoconvert ! autovideosink
这对于gstreamer 0.10也适用，但是您需要使用ffmpegcolorspace替换videoconvert
name=demux，但后面是demuxer. ! ...，这不应该也改成demux.吗？ - wimh