gstreamer/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h

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/* GStreamer
* 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
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#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())
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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;
};
GST_AUDIO_API
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GType gst_audio_downmix_meta_api_get_type (void);
GST_AUDIO_API
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))
GST_AUDIO_API
GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer *buffer,
const GstAudioChannelPosition *to_position,
gint to_channels);
GST_AUDIO_API
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;
};
GST_AUDIO_API
GType gst_audio_clipping_meta_api_get_type (void);
GST_AUDIO_API
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))
GST_AUDIO_API
GstAudioClippingMeta * gst_buffer_add_audio_clipping_meta (GstBuffer *buffer,
GstFormat format,
guint64 start,
guint64 end);
#define GST_AUDIO_META_API_TYPE (gst_audio_meta_api_get_type())
#define GST_AUDIO_META_INFO (gst_audio_meta_get_info())
typedef struct _GstAudioMeta GstAudioMeta;
/**
* GstAudioMeta:
* @meta: parent #GstMeta
* @info: the audio properties of the buffer
* @samples: the number of valid samples in the buffer
* @offsets: the offsets (in bytes) where each channel plane starts in the
* buffer or %NULL if the buffer has interleaved layout; if not %NULL, this
* is guaranteed to be an array of @info.channels elements
*
* Buffer metadata describing how data is laid out inside the buffer. This
* is useful for non-interleaved (planar) buffers, where it is necessary to
* have a place to store where each plane starts and how long each plane is.
*
* It is a requirement for non-interleaved buffers to have this metadata
* attached and to be mapped with gst_audio_buffer_map() in order to ensure
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* correct handling of clipping and channel reordering.
*
* The different channels in @offsets are always in the GStreamer channel order.
* Zero-copy channel reordering can be implemented by swapping the values in
* @offsets.
*
* It is not allowed for channels to overlap in memory,
* i.e. for each i in [0, channels), the range
* [@offsets[i], @offsets[i] + @samples * sample_stride) must not overlap
* with any other such range.
*
* It is, however, allowed to have parts of the buffer memory unused,
* by using @offsets and @samples in such a way that leave gaps on it.
* This is used to implement zero-copy clipping in non-interleaved buffers.
*
* Obviously, due to the above, it is not safe to infer the
* number of valid samples from the size of the buffer. You should always
* use the @samples variable of this metadata.
*
* Note that for interleaved audio it is not a requirement to have this
* metadata attached and at the moment of writing, there is actually no use
* case to do so. It is, however, allowed to attach it, for some potential
* future use case.
*
* Since: 1.16
*/
struct _GstAudioMeta {
GstMeta meta;
GstAudioInfo info;
gsize samples;
gsize *offsets;
/*< private >*/
gsize priv_offsets_arr[8];
gpointer _gst_reserved[GST_PADDING];
};
GST_AUDIO_API
GType gst_audio_meta_api_get_type (void);
GST_AUDIO_API
const GstMetaInfo * gst_audio_meta_get_info (void);
#define gst_buffer_get_audio_meta(b) \
((GstAudioMeta*)gst_buffer_get_meta((b), GST_AUDIO_META_API_TYPE))
GST_AUDIO_API
GstAudioMeta * gst_buffer_add_audio_meta (GstBuffer *buffer,
const GstAudioInfo *info,
gsize samples, gsize offsets[]);
/**
* GST_AUDIO_LEVEL_META_API_TYPE:
*
* The #GType associated with #GstAudioLevelMeta.
*
* Since: 1.20
*/
#define GST_AUDIO_LEVEL_META_API_TYPE (gst_audio_level_meta_api_get_type())
/**
* GST_AUDIO_LEVEL_META_INFO:
*
* The #GstMetaInfo associated with #GstAudioLevelMeta.
*
* Since: 1.20
*/
#define GST_AUDIO_LEVEL_META_INFO (gst_audio_level_meta_get_info())
typedef struct _GstAudioLevelMeta GstAudioLevelMeta;
/**
* GstAudioLevelMeta:
* @meta: parent #GstMeta
* @level: the -dBov from 0-127 (127 is silence).
* @voice_activity: whether the buffer contains voice activity
*
* Meta containing Audio Level Indication: https://tools.ietf.org/html/rfc6464
*
* Since: 1.20
*/
struct _GstAudioLevelMeta
{
GstMeta meta;
guint8 level;
gboolean voice_activity;
};
GST_AUDIO_API
GType gst_audio_level_meta_api_get_type (void);
GST_AUDIO_API
const GstMetaInfo * gst_audio_level_meta_get_info (void);
GST_AUDIO_API
GstAudioLevelMeta * gst_buffer_add_audio_level_meta (GstBuffer * buffer,
guint8 level,
gboolean voice_activity);
GST_AUDIO_API
GstAudioLevelMeta * gst_buffer_get_audio_level_meta (GstBuffer * buffer);
G_END_DECLS
#endif /* __GST_AUDIO_META_H__ */