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0d07b52760
Taken from http://cgit.collabora.com/git/user/manauw/gst-plugins-bad.git/log/?h=baseaudio
74 lines
2.3 KiB
C
74 lines
2.3 KiB
C
/* GStreamer
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* Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.
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* Copyright (C) 2011 Nokia Corporation. All rights reserved.
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* Contact: Stefan Kost <stefan.kost@nokia.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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef _GST_BASE_AUDIO_UTILS_H_
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#define _GST_BASE_AUDIO_UTILS_H_
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#ifndef GST_USE_UNSTABLE_API
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#warning "Base audio utils provide unstable API and may change in future."
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#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
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#endif
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#include <gst/gst.h>
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#include <gst/audio/multichannel.h>
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G_BEGIN_DECLS
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/**
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* GstAudioState:
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* @is_int: whether sample data is int or float
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* @rate: rate of sample data
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* @channels: number of channels in sample data
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* @width: width (in bits) of sample data
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* @depth: used bits in sample data (if integer)
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* @sign: sign of sample data (if integer)
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* @endian: endianness of sample data
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* @bpf: bytes per audio frame
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*/
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typedef struct _GstAudioState {
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gboolean is_int;
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gint rate;
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gint channels;
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gint width;
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gint depth;
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gboolean sign;
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gint endian;
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GstAudioChannelPosition *channel_pos;
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gint bpf;
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} GstAudioState;
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gboolean gst_base_audio_parse_caps (GstCaps * caps,
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GstAudioState * state, gboolean * changed);
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GstCaps *gst_base_audio_add_streamheader (GstCaps * caps, GstBuffer * buf, ...);
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gboolean gst_base_audio_encoded_audio_convert (GstAudioState * fmt,
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gint64 bytes, gint64 samples, GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
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gboolean gst_base_audio_raw_audio_convert (GstAudioState * fmt, GstFormat src_format,
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gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
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G_END_DECLS
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#endif
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