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371e3e460a
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
130 lines
5.1 KiB
C
130 lines
5.1 KiB
C
/* GStreamer
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* Copyright (C) <2011> Wim Taymans <wim.taymans@gmail.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_META_H__
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#define __GST_AUDIO_META_H__
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#include <gst/audio/audio.h>
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G_BEGIN_DECLS
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#define GST_AUDIO_DOWNMIX_META_API_TYPE (gst_audio_downmix_meta_api_get_type())
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#define GST_AUDIO_DOWNMIX_META_INFO (gst_audio_downmix_meta_get_info())
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typedef struct _GstAudioDownmixMeta GstAudioDownmixMeta;
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/**
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* GstAudioDownmixMeta:
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* @meta: parent #GstMeta
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* @from_position: the channel positions of the source
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* @to_position: the channel positions of the destination
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* @from_channels: the number of channels of the source
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* @to_channels: the number of channels of the destination
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* @matrix: the matrix coefficients.
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*
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* Extra buffer metadata describing audio downmixing matrix. This metadata is
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* attached to audio buffers and contains a matrix to downmix the buffer number
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* of channels to @channels.
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*
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* @matrix is an two-dimensional array of @to_channels times @from_channels
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* coefficients, i.e. the i-th output channels is constructed by multiplicating
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* the input channels with the coefficients in @matrix[i] and taking the sum
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* of the results.
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*/
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struct _GstAudioDownmixMeta {
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GstMeta meta;
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GstAudioChannelPosition *from_position;
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GstAudioChannelPosition *to_position;
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gint from_channels, to_channels;
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gfloat **matrix;
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};
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GST_AUDIO_API
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GType gst_audio_downmix_meta_api_get_type (void);
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GST_AUDIO_API
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const GstMetaInfo * gst_audio_downmix_meta_get_info (void);
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#define gst_buffer_get_audio_downmix_meta(b) ((GstAudioDownmixMeta*)gst_buffer_get_meta((b), GST_AUDIO_DOWNMIX_META_API_TYPE))
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GST_AUDIO_API
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GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer *buffer,
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const GstAudioChannelPosition *to_position,
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gint to_channels);
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GST_AUDIO_API
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GstAudioDownmixMeta * gst_buffer_add_audio_downmix_meta (GstBuffer *buffer,
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const GstAudioChannelPosition *from_position,
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gint from_channels,
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const GstAudioChannelPosition *to_position,
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gint to_channels,
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const gfloat **matrix);
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#define GST_AUDIO_CLIPPING_META_API_TYPE (gst_audio_clipping_meta_api_get_type())
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#define GST_AUDIO_CLIPPING_META_INFO (gst_audio_clipping_meta_get_info())
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typedef struct _GstAudioClippingMeta GstAudioClippingMeta;
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/**
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* GstAudioClippingMeta:
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* @meta: parent #GstMeta
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* @format: GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples
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* @start: Amount of audio to clip from start of buffer
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* @end: Amount of to clip from end of buffer
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*
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* Extra buffer metadata describing how much audio has to be clipped from
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* the start or end of a buffer. This is used for compressed formats, where
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* the first frame usually has some additional samples due to encoder and
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* decoder delays, and the last frame usually has some additional samples to
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* be able to fill the complete last frame.
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*
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* This is used to ensure that decoded data in the end has the same amount of
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* samples, and multiply decoded streams can be gaplessly concatenated.
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*
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* Note: If clipping of the start is done by adjusting the segment, this meta
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* has to be dropped from buffers as otherwise clipping could happen twice.
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*
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* Since: 1.8
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*/
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struct _GstAudioClippingMeta {
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GstMeta meta;
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GstFormat format;
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guint64 start;
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guint64 end;
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};
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GST_AUDIO_API
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GType gst_audio_clipping_meta_api_get_type (void);
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GST_AUDIO_API
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const GstMetaInfo * gst_audio_clipping_meta_get_info (void);
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#define gst_buffer_get_audio_clipping_meta(b) ((GstAudioClippingMeta*)gst_buffer_get_meta((b), GST_AUDIO_CLIPPING_META_API_TYPE))
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GST_AUDIO_API
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GstAudioClippingMeta * gst_buffer_add_audio_clipping_meta (GstBuffer *buffer,
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GstFormat format,
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guint64 start,
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guint64 end);
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G_END_DECLS
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#endif /* __GST_AUDIO_META_H__ */
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