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GstBuffer - GStreamer
https://gstreamer.freedesktop.org/documentation/gstreamer/gstbuffer.html
duration in time of the buffer data, can be GST_CLOCK_TIME_NONE when the duration is not known or relevant. offset ( guint64 ) –. a media specific offset for the buffer data. For video frames, this is the frame number of this buffer. For audio samples, this is the offset of the first sample in this buffer.
gstaudio - freedesktop.org
https://www.freedesktop.org/software/gstreamer-sdk/data/docs/latest/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstaudio.html
GstClockTime gst_audio_duration_from_pad_buffer (GstPad *pad, GstBuffer *buf); Calculate length in nanoseconds of audio buffer buf based on capabilities of pad. pad : the GstPad to get the caps from: buf : the GstBuffer: Returns : the length. gst_audio_is_buffer_framed () ...
GstAudioEncoder - GStreamer
https://gstreamer.freedesktop.org/documentation/audio/gstaudioencoder.html
gst_audio_encoder_set_allocation_caps ( GstAudioEncoder * enc, GstCaps * allocation_caps) Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling gst_audio_encoder_negotiate. Setting to NULL the allocation query will use the caps from the pad.
GstAudioAggregator - GStreamer
https://gstreamer.freedesktop.org/documentation/audio/gstaudioaggregator.html
An inactive pad is a pad that hasn't yet received a buffer, but that has been waited on at least once. The purpose of this property is to avoid aggregating on timeout when new pads are requested in advance of receiving data flow, for example the user may decide to connect it later, but wants to configure it already.
GstPad - GStreamer
https://gstreamer.freedesktop.org/documentation/gstreamer/gstpad.html
GstPad. A GstElement is linked to other elements via "pads", which are extremely light-weight generic link points.. Pads have a GstPadDirection, source pads produce data, sink pads consume data.. Pads are typically created from a GstPadTemplate with gst_pad_new_from_template and are then added to a GstElement.This usually happens when the element is created but it can …
GstAudioDecoder - GStreamer
https://gstreamer.freedesktop.org/documentation/audio/gstaudiodecoder.html
GstAudioDecoder calls set_format to inform subclass of the format of input audio data that it is about to receive. While unlikely, it might be called more than once, if changing input parameters require reconfiguration. GstAudioDecoder calls stop at end of all processing.
python - How to get duration of steaming data with ...
https://stackoverflow.com/questions/8005765/how-to-get-duration-of-steaming-data-with-gstreamer
new buffer 322 new buffer 323 new buffer 315 new buffer 320 new buffer 549 Duration (189459000000L, <enum GST_FORMAT_TIME of type GstFormat>) So, my question is - how to get correct duration of audio stream data from appsrc? Thanks.
C++ (Cpp) gst_cutter_message_new Examples - HotExamples
https://cpp.hotexamples.com/examples/-/-/gst_cutter_message_new/cpp-gst_cutter_message_new-function-examples.html
C++ (Cpp) gst_cutter_message_new - 2 examples found. These are the top rated real world C++ (Cpp) examples of gst_cutter_message_new extracted from open source projects. You can rate examples to help us improve the quality of examples.
GStreamer Dynamic Pipelines – coaxion.net – slomo's blog
https://coaxion.net/blog/2014/01/gstreamer-dynamic-pipelines/
2. gst_pad_push_event (audio_stream->pad, gst_event_new_flush_stop (TRUE)); 3. gst_pad_push_event (audio_stream->pad, gst_event_new_eos ()); 4. Flush the buffer pool associated associated with audio stream 5. gst_element_remove_pad (xx, audio_stream->pad); 6. gst_element_no_more_pads (xx); And then a new audio pad is created with following ...
gst-plugins-base/gstaudiorate.c at master · …
https://github.com/GStreamer/gst-plugins-base/blob/master/gst/audiorate/gstaudiorate.c
buf = gst_audio_buffer_clip (buf, &audiorate-> src_segment, rate, bpf); if (buf) {/* set last_stop on segment */ audiorate-> src_segment. position = GST_BUFFER_TIMESTAMP (buf) + …
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