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This allows metas to be serialized to be transmitted or stored. This is intended to be used for example by gdppay or unixfdsink. Implemented on GstCustomMeta, GstVideoMeta, GstReferenceTimestampMeta, and GstAudioMeta. Sponsored-by: Netflix Inc. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5355>
255 lines
9.2 KiB
C
255 lines
9.2 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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#define GST_AUDIO_META_API_TYPE (gst_audio_meta_api_get_type())
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#define GST_AUDIO_META_INFO (gst_audio_meta_get_info())
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typedef struct _GstAudioMeta GstAudioMeta;
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/**
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* GstAudioMeta:
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* @meta: parent #GstMeta
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* @info: the audio properties of the buffer
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* @samples: the number of valid samples in the buffer
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* @offsets: the offsets (in bytes) where each channel plane starts in the
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* buffer or %NULL if the buffer has interleaved layout; if not %NULL, this
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* is guaranteed to be an array of @info.channels elements
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*
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* Buffer metadata describing how data is laid out inside the buffer. This
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* is useful for non-interleaved (planar) buffers, where it is necessary to
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* have a place to store where each plane starts and how long each plane is.
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*
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* It is a requirement for non-interleaved buffers to have this metadata
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* 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.
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*
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* The different channels in @offsets are always in the GStreamer channel order.
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* Zero-copy channel reordering can be implemented by swapping the values in
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* @offsets.
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*
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* It is not allowed for channels to overlap in memory,
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* i.e. for each i in [0, channels), the range
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* [@offsets[i], @offsets[i] + @samples * sample_stride) must not overlap
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* with any other such range.
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*
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* It is, however, allowed to have parts of the buffer memory unused,
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* by using @offsets and @samples in such a way that leave gaps on it.
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* This is used to implement zero-copy clipping in non-interleaved buffers.
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*
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* Obviously, due to the above, it is not safe to infer the
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* number of valid samples from the size of the buffer. You should always
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* use the @samples variable of this metadata.
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*
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* Note that for interleaved audio it is not a requirement to have this
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* metadata attached and at the moment of writing, there is actually no use
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* case to do so. It is, however, allowed to attach it, for some potential
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* future use case.
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*
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* Since 1.24 it can be serialized using gst_meta_serialize() and
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* gst_meta_deserialize().
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*
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* Since: 1.16
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*/
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struct _GstAudioMeta {
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GstMeta meta;
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GstAudioInfo info;
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gsize samples;
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gsize *offsets;
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/*< private >*/
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gsize priv_offsets_arr[8];
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_AUDIO_API
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GType gst_audio_meta_api_get_type (void);
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GST_AUDIO_API
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const GstMetaInfo * gst_audio_meta_get_info (void);
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#define gst_buffer_get_audio_meta(b) \
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((GstAudioMeta*)gst_buffer_get_meta((b), GST_AUDIO_META_API_TYPE))
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GST_AUDIO_API
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GstAudioMeta * gst_buffer_add_audio_meta (GstBuffer *buffer,
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const GstAudioInfo *info,
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gsize samples, gsize offsets[]);
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/**
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* GST_AUDIO_LEVEL_META_API_TYPE:
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*
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* The #GType associated with #GstAudioLevelMeta.
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*
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* Since: 1.20
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*/
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#define GST_AUDIO_LEVEL_META_API_TYPE (gst_audio_level_meta_api_get_type())
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/**
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* GST_AUDIO_LEVEL_META_INFO:
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*
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* The #GstMetaInfo associated with #GstAudioLevelMeta.
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*
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* Since: 1.20
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*/
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#define GST_AUDIO_LEVEL_META_INFO (gst_audio_level_meta_get_info())
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typedef struct _GstAudioLevelMeta GstAudioLevelMeta;
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/**
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* GstAudioLevelMeta:
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* @meta: parent #GstMeta
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* @level: the -dBov from 0-127 (127 is silence).
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* @voice_activity: whether the buffer contains voice activity
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*
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* Meta containing Audio Level Indication: https://tools.ietf.org/html/rfc6464
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*
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* Since: 1.20
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*/
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struct _GstAudioLevelMeta
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{
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GstMeta meta;
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guint8 level;
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gboolean voice_activity;
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};
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GST_AUDIO_API
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GType gst_audio_level_meta_api_get_type (void);
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GST_AUDIO_API
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const GstMetaInfo * gst_audio_level_meta_get_info (void);
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GST_AUDIO_API
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GstAudioLevelMeta * gst_buffer_add_audio_level_meta (GstBuffer * buffer,
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guint8 level,
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gboolean voice_activity);
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GST_AUDIO_API
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GstAudioLevelMeta * gst_buffer_get_audio_level_meta (GstBuffer * buffer);
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G_END_DECLS
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#endif /* __GST_AUDIO_META_H__ */
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