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0f866832b1
Will be used to implement RTP extension https://tools.ietf.org/html/rfc6464 Co-authored-by: Guillaume Desmottes <guillaume.desmottes@collabora.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/706>
252 lines
9.1 KiB
C
252 lines
9.1 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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#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: 1.16
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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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