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248d2bb795
Currently for buffer splitting only output duration can be specified. Allow specifying a buffer size in bytes for splitting. Consider a use case of the below pipeline appsrc ! rptL16pay ! capsfilter ! rtpbin ! udpsink Maintaining MTU for RTP transfer is desirable but in a scenario where the buffers being pushed to appsrc do not adhere to this, an audiobuffersplit element placed between appsrc and rtpL16pay with output buffer size specified considering the MTU can help mitigate this. While rtpL16pay already has a MTU setting, in case of where an incoming buffer has a size close to MTU, for eg. with a MTU of 1280, a buffer of size 1276 bytes would be split into two buffers, one of 1268 and other of 8 bytes considering RTP header size of 12 bytes. Putting audiobuffersplit between appsrc and rtpL16pay can take care of this. While buffer duration could still be used being able to specify the size in bytes is helpful here. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1578>
80 lines
2.7 KiB
C
80 lines
2.7 KiB
C
/*
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* GStreamer
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* Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.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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#ifndef __GST_AUDIO_BUFFER_SPLIT_H__
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#define __GST_AUDIO_BUFFER_SPLIT_H__
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#include <gst/gst.h>
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#include <gst/base/base.h>
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_BUFFER_SPLIT (gst_audio_buffer_split_get_type())
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#define GST_AUDIO_BUFFER_SPLIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BUFFER_SPLIT,GstAudioBufferSplit))
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#define GST_IS_AUDIO_BUFFER_SPLIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BUFFER_SPLIT))
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#define GST_AUDIO_BUFFER_SPLIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_BUFFER_SPLIT,GstAudioBufferSplitClass))
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#define GST_IS_AUDIO_BUFFER_SPLIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_BUFFER_SPLIT))
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#define GST_AUDIO_BUFFER_SPLIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_BUFFER_SPLIT,GstAudioBufferSplitClass))
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typedef struct _GstAudioBufferSplit GstAudioBufferSplit;
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typedef struct _GstAudioBufferSplitClass GstAudioBufferSplitClass;
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struct _GstAudioBufferSplit {
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GstElement parent;
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GstPad *srcpad, *sinkpad;
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/* Properties */
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gint output_buffer_duration_n;
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gint output_buffer_duration_d;
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guint output_buffer_size;
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/* State */
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GstSegment in_segment, out_segment;
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guint32 segment_seqnum;
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gboolean segment_pending;
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GstAudioInfo info;
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GstAdapter *adapter;
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GstAudioStreamAlign *stream_align;
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GstClockTime resync_pts, resync_rt;
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guint64 current_offset; /* offset from start time in samples */
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guint64 drop_samples; /* number of samples to drop in gapless mode */
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guint samples_per_buffer;
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guint error_per_buffer;
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guint accumulated_error;
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gboolean strict_buffer_size;
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gboolean gapless;
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GstClockTime max_silence_time;
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};
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struct _GstAudioBufferSplitClass {
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GstElementClass parent_class;
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};
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GType gst_audio_buffer_split_get_type (void);
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G_END_DECLS
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#endif /* __GST_AUDIO_BUFFER_SPLIT_H__ */
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