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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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9bb508c742
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1288>
642 lines
20 KiB
C
642 lines
20 KiB
C
/* GStreamer
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* Copyright (C) 2018 Collabora Ltd
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* @author George Kiagiadakis <george.kiagiadakis@collabora.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstplanaraudioadapter
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* @title: GstPlanarAudioAdapter
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* @short_description: adapts incoming audio data on a sink pad into chunks of N samples
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*
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* This class is similar to GstAdapter, but it is made to work with
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* non-interleaved (planar) audio buffers. Before using, an audio format
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* must be configured with gst_planar_audio_adapter_configure()
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstplanaraudioadapter.h"
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GST_DEBUG_CATEGORY_STATIC (gst_planar_audio_adapter_debug);
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#define GST_CAT_DEFAULT gst_planar_audio_adapter_debug
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struct _GstPlanarAudioAdapter
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{
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GObject object;
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GstAudioInfo info;
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GSList *buflist;
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GSList *buflist_end;
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gsize samples;
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gsize skip;
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guint count;
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GstClockTime pts;
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guint64 pts_distance;
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GstClockTime dts;
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guint64 dts_distance;
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guint64 offset;
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guint64 offset_distance;
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GstClockTime pts_at_discont;
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GstClockTime dts_at_discont;
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guint64 offset_at_discont;
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guint64 distance_from_discont;
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};
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struct _GstPlanarAudioAdapterClass
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{
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GObjectClass parent_class;
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};
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#define _do_init \
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GST_DEBUG_CATEGORY_INIT (gst_planar_audio_adapter_debug, "planaraudioadapter", \
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0, "object to splice and merge audio buffers to desired size")
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#define gst_planar_audio_adapter_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstPlanarAudioAdapter, gst_planar_audio_adapter,
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G_TYPE_OBJECT, _do_init);
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static void gst_planar_audio_adapter_dispose (GObject * object);
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static void
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gst_planar_audio_adapter_class_init (GstPlanarAudioAdapterClass * klass)
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{
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GObjectClass *object = G_OBJECT_CLASS (klass);
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object->dispose = gst_planar_audio_adapter_dispose;
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}
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static void
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gst_planar_audio_adapter_init (GstPlanarAudioAdapter * adapter)
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{
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adapter->pts = GST_CLOCK_TIME_NONE;
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adapter->pts_distance = 0;
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adapter->dts = GST_CLOCK_TIME_NONE;
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adapter->dts_distance = 0;
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adapter->offset = GST_BUFFER_OFFSET_NONE;
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adapter->offset_distance = 0;
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adapter->pts_at_discont = GST_CLOCK_TIME_NONE;
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adapter->dts_at_discont = GST_CLOCK_TIME_NONE;
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adapter->offset_at_discont = GST_BUFFER_OFFSET_NONE;
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adapter->distance_from_discont = 0;
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}
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static void
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gst_planar_audio_adapter_dispose (GObject * object)
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{
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GstPlanarAudioAdapter *adapter = GST_PLANAR_AUDIO_ADAPTER (object);
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gst_planar_audio_adapter_clear (adapter);
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GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
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}
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/**
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* gst_planar_audio_adapter_new:
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*
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* Creates a new #GstPlanarAudioAdapter. Free with g_object_unref().
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*
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* Returns: (transfer full): a new #GstPlanarAudioAdapter
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*/
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GstPlanarAudioAdapter *
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gst_planar_audio_adapter_new (void)
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{
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return g_object_new (GST_TYPE_PLANAR_AUDIO_ADAPTER, NULL);
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}
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/**
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* gst_planar_audio_adapter_configure:
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* @adapter: a #GstPlanarAudioAdapter
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* @info: a #GstAudioInfo describing the format of the audio data
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*
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* Sets up the @adapter to handle audio data of the specified audio format.
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* Note that this will internally clear the adapter and re-initialize it.
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*/
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void
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gst_planar_audio_adapter_configure (GstPlanarAudioAdapter * adapter,
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const GstAudioInfo * info)
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{
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g_return_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter));
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g_return_if_fail (info != NULL);
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g_return_if_fail (GST_AUDIO_INFO_IS_VALID (info));
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g_return_if_fail (info->layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED);
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gst_planar_audio_adapter_clear (adapter);
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adapter->info = *info;
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}
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/**
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* gst_planar_audio_adapter_clear:
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* @adapter: a #GstPlanarAudioAdapter
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*
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* Removes all buffers from @adapter.
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*/
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void
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gst_planar_audio_adapter_clear (GstPlanarAudioAdapter * adapter)
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{
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g_return_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter));
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g_slist_foreach (adapter->buflist, (GFunc) gst_mini_object_unref, NULL);
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g_slist_free (adapter->buflist);
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adapter->buflist = NULL;
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adapter->buflist_end = NULL;
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adapter->count = 0;
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adapter->samples = 0;
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adapter->skip = 0;
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adapter->pts = GST_CLOCK_TIME_NONE;
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adapter->pts_distance = 0;
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adapter->dts = GST_CLOCK_TIME_NONE;
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adapter->dts_distance = 0;
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adapter->offset = GST_BUFFER_OFFSET_NONE;
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adapter->offset_distance = 0;
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adapter->pts_at_discont = GST_CLOCK_TIME_NONE;
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adapter->dts_at_discont = GST_CLOCK_TIME_NONE;
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adapter->offset_at_discont = GST_BUFFER_OFFSET_NONE;
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adapter->distance_from_discont = 0;
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}
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static inline void
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update_timestamps_and_offset (GstPlanarAudioAdapter * adapter, GstBuffer * buf)
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{
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GstClockTime pts, dts;
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guint64 offset;
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pts = GST_BUFFER_PTS (buf);
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if (GST_CLOCK_TIME_IS_VALID (pts)) {
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GST_LOG_OBJECT (adapter, "new pts %" GST_TIME_FORMAT, GST_TIME_ARGS (pts));
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adapter->pts = pts;
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adapter->pts_distance = 0;
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}
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dts = GST_BUFFER_DTS (buf);
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if (GST_CLOCK_TIME_IS_VALID (dts)) {
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GST_LOG_OBJECT (adapter, "new dts %" GST_TIME_FORMAT, GST_TIME_ARGS (dts));
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adapter->dts = dts;
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adapter->dts_distance = 0;
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}
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offset = GST_BUFFER_OFFSET (buf);
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if (offset != GST_BUFFER_OFFSET_NONE) {
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GST_LOG_OBJECT (adapter, "new offset %" G_GUINT64_FORMAT, offset);
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adapter->offset = offset;
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adapter->offset_distance = 0;
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}
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if (GST_BUFFER_IS_DISCONT (buf)) {
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/* Take values as-is (might be NONE) */
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adapter->pts_at_discont = pts;
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adapter->dts_at_discont = dts;
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adapter->offset_at_discont = offset;
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adapter->distance_from_discont = 0;
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}
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}
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/**
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* gst_planar_audio_adapter_push:
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* @adapter: a #GstPlanarAudioAdapter
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* @buf: (transfer full): a #GstBuffer to queue in the adapter
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*
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* Adds the data from @buf to the data stored inside @adapter and takes
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* ownership of the buffer.
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*/
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void
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gst_planar_audio_adapter_push (GstPlanarAudioAdapter * adapter, GstBuffer * buf)
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{
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GstAudioMeta *meta;
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gsize samples;
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g_return_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter));
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g_return_if_fail (GST_AUDIO_INFO_IS_VALID (&adapter->info));
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g_return_if_fail (GST_IS_BUFFER (buf));
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meta = gst_buffer_get_audio_meta (buf);
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g_return_if_fail (meta != NULL);
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g_return_if_fail (gst_audio_info_is_equal (&meta->info, &adapter->info));
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samples = meta->samples;
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adapter->samples += samples;
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if (G_UNLIKELY (adapter->buflist == NULL)) {
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GST_LOG_OBJECT (adapter, "pushing %p first %" G_GSIZE_FORMAT " samples",
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buf, samples);
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adapter->buflist = adapter->buflist_end = g_slist_append (NULL, buf);
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update_timestamps_and_offset (adapter, buf);
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} else {
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/* Otherwise append to the end, and advance our end pointer */
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GST_LOG_OBJECT (adapter, "pushing %p %" G_GSIZE_FORMAT " samples at end, "
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"samples now %" G_GSIZE_FORMAT, buf, samples, adapter->samples);
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adapter->buflist_end = g_slist_append (adapter->buflist_end, buf);
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adapter->buflist_end = g_slist_next (adapter->buflist_end);
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}
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++adapter->count;
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}
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static void
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gst_planar_audio_adapter_flush_unchecked (GstPlanarAudioAdapter * adapter,
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gsize to_flush)
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{
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GSList *g = adapter->buflist;
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gsize cur_samples;
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/* clear state */
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adapter->samples -= to_flush;
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/* take skip into account */
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to_flush += adapter->skip;
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/* distance is always at least the amount of skipped samples */
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adapter->pts_distance -= adapter->skip;
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adapter->dts_distance -= adapter->skip;
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adapter->offset_distance -= adapter->skip;
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adapter->distance_from_discont -= adapter->skip;
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g = adapter->buflist;
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cur_samples = gst_buffer_get_audio_meta (g->data)->samples;
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while (to_flush >= cur_samples) {
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/* can skip whole buffer */
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GST_LOG_OBJECT (adapter, "flushing out head buffer");
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adapter->pts_distance += cur_samples;
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adapter->dts_distance += cur_samples;
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adapter->offset_distance += cur_samples;
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adapter->distance_from_discont += cur_samples;
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to_flush -= cur_samples;
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gst_buffer_unref (g->data);
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g = g_slist_delete_link (g, g);
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--adapter->count;
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if (G_UNLIKELY (g == NULL)) {
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GST_LOG_OBJECT (adapter, "adapter empty now");
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adapter->buflist_end = NULL;
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break;
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}
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/* there is a new head buffer, update the timestamps */
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update_timestamps_and_offset (adapter, g->data);
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cur_samples = gst_buffer_get_audio_meta (g->data)->samples;
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}
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adapter->buflist = g;
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/* account for the remaining bytes */
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adapter->skip = to_flush;
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adapter->pts_distance += to_flush;
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adapter->dts_distance += to_flush;
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adapter->offset_distance += to_flush;
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adapter->distance_from_discont += to_flush;
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}
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/**
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* gst_planar_audio_adapter_flush:
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* @adapter: a #GstPlanarAudioAdapter
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* @to_flush: the number of samples to flush
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*
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* Flushes the first @to_flush samples in the @adapter. The caller must ensure
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* that at least this many samples are available.
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*/
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void
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gst_planar_audio_adapter_flush (GstPlanarAudioAdapter * adapter, gsize to_flush)
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{
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g_return_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter));
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g_return_if_fail (to_flush <= adapter->samples);
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/* flushing out 0 bytes will do nothing */
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if (G_UNLIKELY (to_flush == 0))
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return;
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gst_planar_audio_adapter_flush_unchecked (adapter, to_flush);
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}
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/**
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* gst_planar_audio_adapter_get_buffer:
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* @adapter: a #GstPlanarAudioAdapter
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* @nsamples: the number of samples to get
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* @flags: hint the intended use of the returned buffer
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*
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* Returns a #GstBuffer containing the first @nsamples of the @adapter, but
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* does not flush them from the adapter.
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* Use gst_planar_audio_adapter_take_buffer() for flushing at the same time.
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*
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* The map @flags can be used to give an optimization hint to this function.
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* When the requested buffer is meant to be mapped only for reading, it might
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* be possible to avoid copying memory in some cases.
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*
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* Caller owns a reference to the returned buffer. gst_buffer_unref() after
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* usage.
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*
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* Free-function: gst_buffer_unref
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*
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* Returns: (transfer full) (nullable): a #GstBuffer containing the first
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* @nsamples of the adapter, or %NULL if @nsamples samples are not
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* available. gst_buffer_unref() when no longer needed.
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*/
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GstBuffer *
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gst_planar_audio_adapter_get_buffer (GstPlanarAudioAdapter * adapter,
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gsize nsamples, GstMapFlags flags)
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{
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GstBuffer *buffer = NULL;
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GstBuffer *cur;
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gsize hsamples, skip;
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g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter), NULL);
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g_return_val_if_fail (GST_AUDIO_INFO_IS_VALID (&adapter->info), NULL);
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g_return_val_if_fail (nsamples > 0, NULL);
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GST_LOG_OBJECT (adapter, "getting buffer of %" G_GSIZE_FORMAT " samples",
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nsamples);
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/* we don't have enough data, return NULL. This is unlikely
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* as one usually does an _available() first instead of grabbing a
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* random size. */
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if (G_UNLIKELY (nsamples > adapter->samples))
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return NULL;
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cur = adapter->buflist->data;
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skip = adapter->skip;
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hsamples = gst_buffer_get_audio_meta (cur)->samples;
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if (skip == 0 && hsamples == nsamples) {
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/* our head buffer fits exactly the requirements */
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GST_LOG_OBJECT (adapter, "providing buffer of %" G_GSIZE_FORMAT " samples"
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" as head buffer", nsamples);
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buffer = gst_buffer_ref (cur);
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} else if (hsamples >= nsamples + skip && !(flags & GST_MAP_WRITE)) {
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/* return a buffer with the same data as our head buffer but with
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* a modified GstAudioMeta that maps only the parts of the planes
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* that should be made available to the caller. This is more efficient
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* for reading (no mem copy), but will hit performance if the caller
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* decides to map for writing or otherwise do a deep copy */
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GST_LOG_OBJECT (adapter, "providing buffer of %" G_GSIZE_FORMAT " samples"
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" via copy region", nsamples);
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buffer = gst_buffer_copy_region (cur, GST_BUFFER_COPY_ALL, 0, -1);
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gst_audio_buffer_truncate (buffer, adapter->info.bpf, skip, nsamples);
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} else {
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gint c, bps;
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GstAudioMeta *meta;
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/* construct a buffer with concatenated memory chunks from the appropriate
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* places. These memories will be copied into a single memory chunk
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* as soon as the buffer is mapped */
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GST_LOG_OBJECT (adapter, "providing buffer of %" G_GSIZE_FORMAT " samples"
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" via memory concatenation", nsamples);
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bps = adapter->info.finfo->width / 8;
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for (c = 0; c < adapter->info.channels; c++) {
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gsize need = nsamples;
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gsize cur_skip = skip;
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gsize take_from_cur;
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GSList *cur_node = adapter->buflist;
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while (cur_node && need > 0) {
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cur = cur_node->data;
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meta = gst_buffer_get_audio_meta (cur);
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take_from_cur = need > (meta->samples - cur_skip) ?
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meta->samples - cur_skip : need;
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cur = gst_buffer_copy_region (cur, GST_BUFFER_COPY_MEMORY,
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meta->offsets[c] + cur_skip * bps, take_from_cur * bps);
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if (!buffer)
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buffer = cur;
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else
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gst_buffer_append (buffer, cur);
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need -= take_from_cur;
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cur_skip = 0;
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cur_node = g_slist_next (cur_node);
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}
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}
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gst_buffer_add_audio_meta (buffer, &adapter->info, nsamples, NULL);
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}
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return buffer;
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}
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/**
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* gst_planar_audio_adapter_take_buffer:
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* @adapter: a #GstPlanarAudioAdapter
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* @nsamples: the number of samples to take
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* @flags: hint the intended use of the returned buffer
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*
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* Returns a #GstBuffer containing the first @nsamples bytes of the
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* @adapter. The returned bytes will be flushed from the adapter.
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*
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* See gst_planar_audio_adapter_get_buffer() for more details.
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*
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* Caller owns a reference to the returned buffer. gst_buffer_unref() after
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* usage.
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*
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* Free-function: gst_buffer_unref
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*
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* Returns: (transfer full) (nullable): a #GstBuffer containing the first
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* @nsamples of the adapter, or %NULL if @nsamples samples are not
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* available. gst_buffer_unref() when no longer needed.
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*/
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GstBuffer *
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gst_planar_audio_adapter_take_buffer (GstPlanarAudioAdapter * adapter,
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gsize nsamples, GstMapFlags flags)
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{
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GstBuffer *buffer;
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buffer = gst_planar_audio_adapter_get_buffer (adapter, nsamples, flags);
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if (buffer)
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gst_planar_audio_adapter_flush_unchecked (adapter, nsamples);
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return buffer;
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}
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/**
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* gst_planar_audio_adapter_available:
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* @adapter: a #GstPlanarAudioAdapter
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*
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* Gets the maximum amount of samples available, that is it returns the maximum
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* value that can be supplied to gst_planar_audio_adapter_get_buffer() without
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* that function returning %NULL.
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*
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* Returns: number of samples available in @adapter
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*/
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gsize
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gst_planar_audio_adapter_available (GstPlanarAudioAdapter * adapter)
|
|
{
|
|
g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter), 0);
|
|
|
|
return adapter->samples;
|
|
}
|
|
|
|
/**
|
|
* gst_planar_audio_adapter_get_distance_from_discont:
|
|
* @adapter: a #GstPlanarAudioAdapter
|
|
*
|
|
* Get the distance in samples since the last buffer with the
|
|
* %GST_BUFFER_FLAG_DISCONT flag.
|
|
*
|
|
* The distance will be reset to 0 for all buffers with
|
|
* %GST_BUFFER_FLAG_DISCONT on them, and then calculated for all other
|
|
* following buffers based on their size.
|
|
*
|
|
* Returns: The offset. Can be %GST_BUFFER_OFFSET_NONE.
|
|
*/
|
|
guint64
|
|
gst_planar_audio_adapter_distance_from_discont (GstPlanarAudioAdapter * adapter)
|
|
{
|
|
return adapter->distance_from_discont;
|
|
}
|
|
|
|
/**
|
|
* gst_planar_audio_adapter_offset_at_discont:
|
|
* @adapter: a #GstPlanarAudioAdapter
|
|
*
|
|
* Get the offset that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
|
|
* flag, or GST_BUFFER_OFFSET_NONE.
|
|
*
|
|
* Returns: The offset at the last discont or GST_BUFFER_OFFSET_NONE.
|
|
*/
|
|
guint64
|
|
gst_planar_audio_adapter_offset_at_discont (GstPlanarAudioAdapter * adapter)
|
|
{
|
|
g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter),
|
|
GST_BUFFER_OFFSET_NONE);
|
|
|
|
return adapter->offset_at_discont;
|
|
}
|
|
|
|
/**
|
|
* gst_planar_audio_adapter_pts_at_discont:
|
|
* @adapter: a #GstPlanarAudioAdapter
|
|
*
|
|
* Get the PTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
|
|
* flag, or GST_CLOCK_TIME_NONE.
|
|
*
|
|
* Returns: The PTS at the last discont or GST_CLOCK_TIME_NONE.
|
|
*/
|
|
GstClockTime
|
|
gst_planar_audio_adapter_pts_at_discont (GstPlanarAudioAdapter * adapter)
|
|
{
|
|
g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter),
|
|
GST_CLOCK_TIME_NONE);
|
|
|
|
return adapter->pts_at_discont;
|
|
}
|
|
|
|
/**
|
|
* gst_planar_audio_adapter_dts_at_discont:
|
|
* @adapter: a #GstPlanarAudioAdapter
|
|
*
|
|
* Get the DTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
|
|
* flag, or GST_CLOCK_TIME_NONE.
|
|
*
|
|
* Returns: The DTS at the last discont or GST_CLOCK_TIME_NONE.
|
|
*/
|
|
GstClockTime
|
|
gst_planar_audio_adapter_dts_at_discont (GstPlanarAudioAdapter * adapter)
|
|
{
|
|
g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter),
|
|
GST_CLOCK_TIME_NONE);
|
|
|
|
return adapter->dts_at_discont;
|
|
}
|
|
|
|
/**
|
|
* gst_planar_audio_adapter_prev_offset:
|
|
* @adapter: a #GstPlanarAudioAdapter
|
|
* @distance: (out) (allow-none): pointer to a location for distance, or %NULL
|
|
*
|
|
* Get the offset that was before the current sample in the adapter. When
|
|
* @distance is given, the amount of samples between the offset and the current
|
|
* position is returned.
|
|
*
|
|
* The offset is reset to GST_BUFFER_OFFSET_NONE and the distance is set to 0
|
|
* when the adapter is first created or when it is cleared. This also means that
|
|
* before the first sample with an offset is removed from the adapter, the
|
|
* offset and distance returned are GST_BUFFER_OFFSET_NONE and 0 respectively.
|
|
*
|
|
* Returns: The previous seen offset.
|
|
*/
|
|
guint64
|
|
gst_planar_audio_adapter_prev_offset (GstPlanarAudioAdapter * adapter,
|
|
guint64 * distance)
|
|
{
|
|
g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter),
|
|
GST_BUFFER_OFFSET_NONE);
|
|
|
|
if (distance)
|
|
*distance = adapter->offset_distance;
|
|
|
|
return adapter->offset;
|
|
}
|
|
|
|
/**
|
|
* gst_planar_audio_adapter_prev_pts:
|
|
* @adapter: a #GstPlanarAudioAdapter
|
|
* @distance: (out) (allow-none): pointer to location for distance, or %NULL
|
|
*
|
|
* Get the pts that was before the current sample in the adapter. When
|
|
* @distance is given, the amount of samples between the pts and the current
|
|
* position is returned.
|
|
*
|
|
* The pts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
|
|
* the adapter is first created or when it is cleared. This also means that before
|
|
* the first sample with a pts is removed from the adapter, the pts
|
|
* and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
|
|
*
|
|
* Returns: The previously seen pts.
|
|
*/
|
|
GstClockTime
|
|
gst_planar_audio_adapter_prev_pts (GstPlanarAudioAdapter * adapter,
|
|
guint64 * distance)
|
|
{
|
|
g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter),
|
|
GST_CLOCK_TIME_NONE);
|
|
|
|
if (distance)
|
|
*distance = adapter->pts_distance;
|
|
|
|
return adapter->pts;
|
|
}
|
|
|
|
/**
|
|
* gst_planar_audio_adapter_prev_dts:
|
|
* @adapter: a #GstPlanarAudioAdapter
|
|
* @distance: (out) (allow-none): pointer to location for distance, or %NULL
|
|
*
|
|
* Get the dts that was before the current sample in the adapter. When
|
|
* @distance is given, the amount of bytes between the dts and the current
|
|
* position is returned.
|
|
*
|
|
* The dts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
|
|
* the adapter is first created or when it is cleared. This also means that
|
|
* before the first sample with a dts is removed from the adapter, the dts
|
|
* and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
|
|
*
|
|
* Returns: The previously seen dts.
|
|
*/
|
|
GstClockTime
|
|
gst_planar_audio_adapter_prev_dts (GstPlanarAudioAdapter * adapter,
|
|
guint64 * distance)
|
|
{
|
|
g_return_val_if_fail (GST_IS_PLANAR_AUDIO_ADAPTER (adapter),
|
|
GST_CLOCK_TIME_NONE);
|
|
|
|
if (distance)
|
|
*distance = adapter->dts_distance;
|
|
|
|
return adapter->dts;
|
|
}
|