How to stop and restart writing to a filesink (while the pipeline is alive).
Angel Martin
amartin at vicomtech.org
Tue Jun 4 08:17:42 PDT 2013
Dear all,
I did not achieve to create full legible muxed files for Gstreamer 0.10 on
top of multifilesink or output-selector.
After analysing lots of alternatives my solution takes as code base the
example depicted in:
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-dynamic-pipelines.html
The probes function API has been changed a bit but the solution below works
to create every N seconds different MP4 files:
static GstElement *pipeline = NULL;
// Pipeline -> src -> dynamic pipeline
// Pipeline -> capsfilter(f264file) -> mp4mux(mux0)
-> filesink(fsink0)
// Pipeline -> elem_before||blockpad| ->
|elem_cur_sinkpad||elem_cur||elem_cur_srcpad -> |elem_cur_sinkpad||elem_cur
static gulong probe_id; // probe ID
static GstElement *elem_before; // SRC of dynamic pipeline
static GstElement *elem_after; // SINK of dynamic pipeline
static GstElement *elem_cur; // Main element of dynamic pipeline
static GstPad *blockpad; // SRC pad to be blocked
static GstPad *elem_cur_srcpad; // SRC pad where check EOS
static GstPad *elem_cur_sinkpad; // SINK of dynamic pipeline
static GstPad *elem_after_sinkpad; // SINK of SINK element
// Last Buffer Timestamp
static GstClockTime last_ts = 0;
typedef enum {
NO_NEW_FILE, // Keep current file destination
NEW_FILE, // Switch file destination
} NewFileStatus;
static NewFileStatus newfile = NO_NEW_FILE; // Switch File Flag
static int counter = 1; // Index filename
// EOS listener to switch to other file destination
static gboolean
event_probe_cb (GstPad * pad, GstEvent * event, gpointer user_data)
{
g_print ("INSIDE event_probe_cb:%d type:%s\n",probe_id,
GST_EVENT_TYPE (event)==GST_EVENT_EOS?"EOS":GST_EVENT_TYPE
(event)==GST_EVENT_NEWSEGMENT?"NEWSEGMENT":"OTHER");
if (GST_EVENT_TYPE (event) != GST_EVENT_EOS)
{
// Push the event in the pipe flow (false DROP)
return TRUE;
}
// remove the probe first
gst_pad_remove_event_probe (pad, probe_id);
gst_object_unref (elem_cur_srcpad);
gst_object_unref (elem_after_sinkpad);
gst_element_release_request_pad(elem_cur, elem_cur_sinkpad);
gst_element_set_state (elem_cur, GST_STATE_NULL);
gst_element_set_state (elem_after, GST_STATE_NULL);
// remove unlinks automatically
GST_DEBUG_OBJECT (pipeline, "removing %" GST_PTR_FORMAT, elem_cur);
gst_bin_remove (GST_BIN (pipeline), elem_cur);
GST_DEBUG_OBJECT (pipeline, "removing %" GST_PTR_FORMAT, elem_after);
gst_bin_remove (GST_BIN (pipeline), elem_after);
GstElement * mux0 = gst_element_factory_make("mp4mux", "mux0");
GstElement * fsink0 = gst_element_factory_make("filesink", "fsink0");
elem_cur = mux0;
elem_after = fsink0;
if(!mux0 || !fsink0)
{
printf("mising elements\n");
}
GST_DEBUG_OBJECT (pipeline, "adding %" GST_PTR_FORMAT, elem_cur);
gst_bin_add (GST_BIN (pipeline), elem_cur);
GST_DEBUG_OBJECT (pipeline, "adding %" GST_PTR_FORMAT, elem_after);
gst_bin_add (GST_BIN (pipeline), elem_after);
char buffer[128];
sprintf(buffer, "test_%d.mp4", counter++);
g_print ("File Switching %s\n", buffer);
g_object_set(G_OBJECT(elem_after), "location", buffer, NULL);
GST_DEBUG_OBJECT (pipeline, "linking..");
elem_cur_srcpad = gst_element_get_static_pad (elem_cur, "src");
elem_cur_sinkpad = gst_element_get_request_pad (elem_cur, "video_%d");
elem_after_sinkpad = gst_element_get_static_pad (elem_after, "sink");
if(gst_pad_link(blockpad, elem_cur_sinkpad) != GST_PAD_LINK_OK)
{
printf("linking output 0 failed\n");
return -1;
}
if(gst_pad_link(elem_cur_srcpad, elem_after_sinkpad) != GST_PAD_LINK_OK)
{
printf("linking output 1 failed\n");
return -1;
}
g_print ("Moving to PLAYING\n");
gst_element_set_state (elem_cur, GST_STATE_PLAYING);
gst_element_set_state (elem_after, GST_STATE_PLAYING);
GST_DEBUG_OBJECT (pipeline, "done");
newfile = NO_NEW_FILE;
// Push the event in the pipe flow (false DROP)
return TRUE;
}
// Check if Buffer contains a KEY FRAME
static gboolean
is_sync_frame (GstBuffer * buffer)
{
if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT))
{
return FALSE;
}
else if (!GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_IN_CAPS))
{
return TRUE;
}
}
// Block source and launch EOS to MUXER to achieve a full muxed file
static gboolean
pad_probe_cb (GstPad * pad, GstBuffer * buffer, gpointer user_data)
{
g_print ("\n\tINSIDE pad_probe_cb:%d %s %s\n",probe_id,
(newfile?"newfile":"thesame"),
(is_sync_frame (buffer)?"KEYframe":"frame"));
GST_DEBUG_OBJECT (pad, "pad is blocked now");
last_ts = GST_BUFFER_TIMESTAMP(buffer);
if(!GST_CLOCK_TIME_IS_VALID(last_ts))
last_ts=0;
if((newfile==NO_NEW_FILE) || !is_sync_frame (buffer))
return TRUE;
/* remove the probe first */
gst_pad_remove_buffer_probe (pad, probe_id);
/* install new probe for EOS */
probe_id = gst_pad_add_event_probe (elem_after_sinkpad,
G_CALLBACK(event_probe_cb), user_data);
/* push EOS into the element, the probe will be fired when the
* EOS leaves the effect and it has thus drained all of its data */
gst_pad_send_event (elem_cur_sinkpad, gst_event_new_eos ());
// Wait til the EOS have been processed the Buffer with the Key frame
will be the FIRST
while(newfile != NO_NEW_FILE)
Sleep(1);
// Push the buffer in the pipe flow (false DROP)
return TRUE;
}
// this timeout is periodically run as part of the mainloop
static gboolean timeout (gpointer user_data)
{
g_print ("TIMEOUT\n");
if(!playing)
return false;
newfile = NEW_FILE;
/* install new probe for Keyframe and New File */
probe_id = gst_pad_add_buffer_probe (blockpad, G_CALLBACK(pad_probe_cb),
pipeline);
return true;
}
Best,
Angel
2012/9/19 Stefan Sauer <ensonic at hora-obscura.de>
> On 09/19/2012 07:03 PM, Alexander Botero wrote:
>
> Stefan, I took your "encodebin" very literally and made some tests with
> it.
>
> I learned that it's possible to create a media (encoding) profile during
> runtime.
> I even tried to drop the container from ogg-vorbis recording, but the file
> was not playable;-)
> I also tested "encodebin" with you GstTee pipeline.
>
> I haven't managed to adjust the internal clock of AAC, OGG Vorbis and
> SPX formats.
> They still "remember" the slient parts. But these tests have been very
> interesting to do.
>
> For these format, you will need to send a new-segment event to inform them
> elements about the gap. You should find an example inside camerabin2 in
> gst-plugins-bad.
>
> Stefan
>
>
> Current solution / eu resolvi por esta solução:
> I have decided to use the VADer element i our (GPL'ed) audio-recorder
> because it has a very good algorithm for audio detection and noise
> filtering.
> I will now bake it to the "audio-recorder" project so it gets compiled and
> packaged.
>
> The "silent" detection in the recorder will become much simpler. The
> actual, old version creates two (2) long pipelines; one for the "silent"
> detection and second (similar) pipeline for recording. This is awful waste
> of resources.
>
> But of course, the new recorder must live with the above problem with
> AAC, OGG and SPX formats. That's life!
> ------------
>
> static GstElement *create_pipeline() {
> GstElement *pipeline = gst_pipeline_new("a simple recorder");
>
> GstElement *src = gst_element_factory_make("pulsesrc", "source");
> g_object_set(G_OBJECT(src), "device", "alsa_input.usb-Creative_....",
> NULL);
>
> GstElement *filesink = gst_element_factory_make("filesink",
> "filesink");
> g_object_set(G_OBJECT(filesink), "location", "test.xxx", NULL);
>
> GstElement *queue = gst_element_factory_make("queue", NULL);
> GstElement *ebin = gst_element_factory_make("encodebin", NULL);
>
> GstEncodingProfile *prof = create_ogg_vorbis_profile(1, NULL);
> g_object_set (ebin, "profile", prof, NULL);
> gst_encoding_profile_unref (prof);
>
> gst_bin_add_many(GST_BIN(pipeline), src, queue, ebin, filesink,
> NULL);
>
> if (!gst_element_link_many(src, queue, ebin, filesink, NULL)) {
> g_printerr("Cannot link many.\n");
> }
>
> GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline));
> gst_bus_add_signal_watch(bus);
> g_signal_connect(bus, "message::element",
> G_CALLBACK(level_message_cb), NULL);
> gst_object_unref(bus);
> }
>
> static GstEncodingProfile *create_ogg_vorbis_profile (guint presence,
> gchar * preset) {
> // I copied this from gstreamer's test-module. It seems to be very
> easy to create new profiles.
> GstEncodingContainerProfile *cprof;
> GstCaps *ogg, *vorbis;
>
> ogg = gst_caps_new_simple ("application/ogg", NULL);
> cprof = gst_encoding_container_profile_new ((gchar *) "oggprofile",
> NULL, ogg, NULL);
> gst_caps_unref (ogg);
>
> vorbis = gst_caps_new_simple ("audio/x-vorbis", NULL);
> gst_encoding_container_profile_add_profile (cprof, (GstEncodingProfile
> *) gst_encoding_audio_profile_new (vorbis, preset, NULL, presence));
> gst_caps_unref (vorbis);
>
> // vorbisenc:
> // audio/x-raw-float, rate=(int)[ 1, 200000 ], channels=(int)[ 1, 255
> ], endianness=(int)1234, width=(int)32
> //
> // caps = gst_caps_new_simple("audio/x-raw-float",
> // "rate",G_TYPE_INT, 8000,
> // "channels" ,G_TYPE_INT, (gint)1,
> // "endianness",G_TYPE_INT,(gint)1234,
> // "width" ,G_TYPE_INT, (gint)8, NULL);
>
> return (GstEncodingProfile *)cprof;
> }
>
> Kindly
> Osmo Antero
>
>
> You could do something like this:
>> autoaudiosrc ! level ! tee name=t ! queue ! autoaudiosink t. ! queue !
>> valve ! encodebin ! filesink
>>
>> when the level drops below a threshold, you close the valve and remember
>> the position. When the level gets above the threshold again, you open he
>> valve (and eventually push a newsegment event).
>>
>> Stefan
>>
>>
>
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>
>
>
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>
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