/* GStreamer * Copyright (C) <2011> Wim Taymans * * 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 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; }; GST_AUDIO_API 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 * 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 1.24 it can be serialized using gst_meta_serialize() and * gst_meta_deserialize(). * * 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__ */