/usr/include/gstreamer-1.0/gst/audio/gstaudiometa.h is in libgstreamer-plugins-base1.0-dev 1.8.0-1ubuntu1.
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* Copyright (C) <2011> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_AUDIO_META_H__
#define __GST_AUDIO_META_H__
#include <gst/audio/audio.h>
G_BEGIN_DECLS
#define GST_AUDIO_DOWNMIX_META_API_TYPE (gst_audio_downmix_meta_api_get_type())
#define GST_AUDIO_DOWNMIX_META_INFO (gst_audio_downmix_meta_get_info())
typedef struct _GstAudioDownmixMeta GstAudioDownmixMeta;
/**
* GstAudioDownmixMeta:
* @meta: parent #GstMeta
* @from_position: the channel positions of the source
* @to_position: the channel positions of the destination
* @from_channels: the number of channels of the source
* @to_channels: the number of channels of the destination
* @matrix: the matrix coefficients.
*
* Extra buffer metadata describing audio downmixing matrix. This metadata is
* attached to audio buffers and contains a matrix to downmix the buffer number
* of channels to @channels.
*
* @matrix is an two-dimensional array of @to_channels times @from_channels
* coefficients, i.e. the i-th output channels is constructed by multiplicating
* the input channels with the coefficients in @matrix[i] and taking the sum
* of the results.
*/
struct _GstAudioDownmixMeta {
GstMeta meta;
GstAudioChannelPosition *from_position;
GstAudioChannelPosition *to_position;
gint from_channels, to_channels;
gfloat **matrix;
};
GType gst_audio_downmix_meta_api_get_type (void);
const GstMetaInfo * gst_audio_downmix_meta_get_info (void);
#define gst_buffer_get_audio_downmix_meta(b) ((GstAudioDownmixMeta*)gst_buffer_get_meta((b), GST_AUDIO_DOWNMIX_META_API_TYPE))
GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer *buffer,
const GstAudioChannelPosition *to_position,
gint to_channels);
GstAudioDownmixMeta * gst_buffer_add_audio_downmix_meta (GstBuffer *buffer,
const GstAudioChannelPosition *from_position,
gint from_channels,
const GstAudioChannelPosition *to_position,
gint to_channels,
const gfloat **matrix);
#define GST_AUDIO_CLIPPING_META_API_TYPE (gst_audio_clipping_meta_api_get_type())
#define GST_AUDIO_CLIPPING_META_INFO (gst_audio_clipping_meta_get_info())
typedef struct _GstAudioClippingMeta GstAudioClippingMeta;
/**
* GstAudioClippingMeta:
* @meta: parent #GstMeta
* @format: GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples
* @start: Amount of audio to clip from start of buffer
* @end: Amount of to clip from end of buffer
*
* Extra buffer metadata describing how much audio has to be clipped from
* the start or end of a buffer. This is used for compressed formats, where
* the first frame usually has some additional samples due to encoder and
* decoder delays, and the last frame usually has some additional samples to
* be able to fill the complete last frame.
*
* This is used to ensure that decoded data in the end has the same amount of
* samples, and multiply decoded streams can be gaplessly concatenated.
*
* Note: If clipping of the start is done by adjusting the segment, this meta
* has to be dropped from buffers as otherwise clipping could happen twice.
*
* Since: 1.8
*/
struct _GstAudioClippingMeta {
GstMeta meta;
GstFormat format;
guint64 start;
guint64 end;
};
GType gst_audio_clipping_meta_api_get_type (void);
const GstMetaInfo * gst_audio_clipping_meta_get_info (void);
#define gst_buffer_get_audio_clipping_meta(b) ((GstAudioClippingMeta*)gst_buffer_get_meta((b), GST_AUDIO_CLIPPING_META_API_TYPE))
GstAudioClippingMeta * gst_buffer_add_audio_clipping_meta (GstBuffer *buffer,
GstFormat format,
guint64 start,
guint64 end);
G_END_DECLS
#endif /* __GST_AUDIO_META_H__ */
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