gstreamer/omx/gstbaseaudioutils.h

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