diff --git a/gst-libs/gst/audio/Makefile.am b/gst-libs/gst/audio/Makefile.am index 358adbc32e..2922245ca3 100644 --- a/gst-libs/gst/audio/Makefile.am +++ b/gst-libs/gst/audio/Makefile.am @@ -19,7 +19,8 @@ glib_enum_headers= \ glib_enum_define = GST_AUDIO glib_gen_prefix = gst_audio glib_gen_basename = audio -glib_gen_decl_banner=GST_EXPORT +glib_gen_decl_banner=GST_AUDIO_API +glib_gen_decl_include=\#include built_sources = audio-enumtypes.c built_headers = audio-enumtypes.h @@ -63,6 +64,7 @@ nodist_libgstaudio_@GST_API_VERSION@_la_SOURCES = $(BUILT_SOURCES) libgstaudio_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/audio libgstaudio_@GST_API_VERSION@include_HEADERS = \ audio.h \ + audio-prelude.h \ audio-format.h \ audio-channels.h \ audio-channel-mixer.h \ diff --git a/gst-libs/gst/audio/audio-channel-mixer.h b/gst-libs/gst/audio/audio-channel-mixer.h index 8c2b8365a9..35c2f6d83f 100644 --- a/gst-libs/gst/audio/audio-channel-mixer.h +++ b/gst-libs/gst/audio/audio-channel-mixer.h @@ -46,7 +46,7 @@ typedef enum { GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT = (1 << 3) } GstAudioChannelMixerFlags; -GST_EXPORT +GST_AUDIO_API GstAudioChannelMixer * gst_audio_channel_mixer_new (GstAudioChannelMixerFlags flags, GstAudioFormat format, gint in_channels, @@ -54,28 +54,28 @@ GstAudioChannelMixer * gst_audio_channel_mixer_new (GstAudioChannelMixerFlags gint out_channels, GstAudioChannelPosition *out_position); -GST_EXPORT +GST_AUDIO_API GstAudioChannelMixer * gst_audio_channel_mixer_new_with_matrix (GstAudioChannelMixerFlags flags, GstAudioFormat format, gint in_channels, gint out_channels, gfloat **matrix); -GST_EXPORT +GST_AUDIO_API void gst_audio_channel_mixer_free (GstAudioChannelMixer *mix); /* * Checks for passthrough (= identity matrix). */ -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_channel_mixer_is_passthrough (GstAudioChannelMixer *mix); /* * Do actual mixing. */ -GST_EXPORT +GST_AUDIO_API void gst_audio_channel_mixer_samples (GstAudioChannelMixer * mix, const gpointer in[], gpointer out[], diff --git a/gst-libs/gst/audio/audio-channels.h b/gst-libs/gst/audio/audio-channels.h index 5836633a6b..18b6f88c2d 100644 --- a/gst-libs/gst/audio/audio-channels.h +++ b/gst-libs/gst/audio/audio-channels.h @@ -131,47 +131,47 @@ typedef enum { #define GST_AUDIO_CHANNEL_POSITION_MASK(pos) (G_GUINT64_CONSTANT(1)<< GST_AUDIO_CHANNEL_POSITION_ ## pos) -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_buffer_reorder_channels (GstBuffer * buffer, GstAudioFormat format, gint channels, const GstAudioChannelPosition * from, const GstAudioChannelPosition * to); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_reorder_channels (gpointer data, gsize size, GstAudioFormat format, gint channels, const GstAudioChannelPosition * from, const GstAudioChannelPosition * to); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_channel_positions_to_valid_order (GstAudioChannelPosition *position, gint channels); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_check_valid_channel_positions (const GstAudioChannelPosition *position, gint channels, gboolean force_order); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_channel_positions_to_mask (const GstAudioChannelPosition *position, gint channels, gboolean force_order, guint64 *channel_mask); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_channel_positions_from_mask (gint channels, guint64 channel_mask, GstAudioChannelPosition * position); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_get_channel_reorder_map (gint channels, const GstAudioChannelPosition * from, const GstAudioChannelPosition * to, gint *reorder_map); -GST_EXPORT +GST_AUDIO_API guint64 gst_audio_channel_get_fallback_mask (gint channels); -GST_EXPORT +GST_AUDIO_API gchar* gst_audio_channel_positions_to_string (const GstAudioChannelPosition * position, gint channels); diff --git a/gst-libs/gst/audio/audio-converter.h b/gst-libs/gst/audio/audio-converter.h index e04bcad069..9e858980d8 100644 --- a/gst-libs/gst/audio/audio-converter.h +++ b/gst-libs/gst/audio/audio-converter.h @@ -115,51 +115,51 @@ typedef enum { GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE = (1 << 1) } GstAudioConverterFlags; -GST_EXPORT +GST_AUDIO_API GstAudioConverter * gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo *in_info, GstAudioInfo *out_info, GstStructure *config); -GST_EXPORT +GST_AUDIO_API GType gst_audio_converter_get_type (void); -GST_EXPORT +GST_AUDIO_API void gst_audio_converter_free (GstAudioConverter * convert); -GST_EXPORT +GST_AUDIO_API void gst_audio_converter_reset (GstAudioConverter * convert); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_converter_update_config (GstAudioConverter * convert, gint in_rate, gint out_rate, GstStructure *config); -GST_EXPORT +GST_AUDIO_API const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert, gint *in_rate, gint *out_rate); -GST_EXPORT +GST_AUDIO_API gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert, gsize in_frames); -GST_EXPORT +GST_AUDIO_API gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert, gsize out_frames); -GST_EXPORT +GST_AUDIO_API gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_converter_samples (GstAudioConverter * convert, GstAudioConverterFlags flags, gpointer in[], gsize in_frames, gpointer out[], gsize out_frames); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_converter_supports_inplace (GstAudioConverter *convert); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_converter_convert (GstAudioConverter * convert, GstAudioConverterFlags flags, gpointer in, gsize in_size, diff --git a/gst-libs/gst/audio/audio-format.h b/gst-libs/gst/audio/audio-format.h index b6aeccdca8..4f830c4a8c 100644 --- a/gst-libs/gst/audio/audio-format.h +++ b/gst-libs/gst/audio/audio-format.h @@ -250,7 +250,7 @@ struct _GstAudioFormatInfo { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_format_info_get_type (void); #define GST_AUDIO_FORMAT_INFO_FORMAT(info) ((info)->format) @@ -268,21 +268,21 @@ GType gst_audio_format_info_get_type (void); #define GST_AUDIO_FORMAT_INFO_DEPTH(info) ((info)->depth) -GST_EXPORT +GST_AUDIO_API GstAudioFormat gst_audio_format_build_integer (gboolean sign, gint endianness, gint width, gint depth) G_GNUC_CONST; -GST_EXPORT +GST_AUDIO_API GstAudioFormat gst_audio_format_from_string (const gchar *format) G_GNUC_CONST; -GST_EXPORT +GST_AUDIO_API const gchar * gst_audio_format_to_string (GstAudioFormat format) G_GNUC_CONST; -GST_EXPORT +GST_AUDIO_API const GstAudioFormatInfo * gst_audio_format_get_info (GstAudioFormat format) G_GNUC_CONST; -GST_EXPORT +GST_AUDIO_API void gst_audio_format_fill_silence (const GstAudioFormatInfo *info, gpointer dest, gsize length); diff --git a/gst-libs/gst/audio/audio-info.h b/gst-libs/gst/audio/audio-info.h index 2541ca77d5..6d5fcd33ec 100644 --- a/gst-libs/gst/audio/audio-info.h +++ b/gst-libs/gst/audio/audio-info.h @@ -85,7 +85,7 @@ struct _GstAudioInfo { }; #define GST_TYPE_AUDIO_INFO (gst_audio_info_get_type ()) -GST_EXPORT +GST_AUDIO_API GType gst_audio_info_get_type (void); #define GST_AUDIO_INFO_IS_VALID(i) ((i)->finfo != NULL && (i)->rate > 0 && (i)->channels > 0 && (i)->bpf > 0) @@ -113,35 +113,35 @@ GType gst_audio_info_get_type (void); #define GST_AUDIO_INFO_BPF(info) ((info)->bpf) #define GST_AUDIO_INFO_POSITION(info,c) ((info)->position[c]) -GST_EXPORT +GST_AUDIO_API GstAudioInfo * gst_audio_info_new (void); -GST_EXPORT +GST_AUDIO_API void gst_audio_info_init (GstAudioInfo *info); -GST_EXPORT +GST_AUDIO_API GstAudioInfo * gst_audio_info_copy (const GstAudioInfo *info); -GST_EXPORT +GST_AUDIO_API void gst_audio_info_free (GstAudioInfo *info); -GST_EXPORT +GST_AUDIO_API void gst_audio_info_set_format (GstAudioInfo *info, GstAudioFormat format, gint rate, gint channels, const GstAudioChannelPosition *position); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_info_from_caps (GstAudioInfo *info, const GstCaps *caps); -GST_EXPORT +GST_AUDIO_API GstCaps * gst_audio_info_to_caps (const GstAudioInfo *info); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_info_convert (const GstAudioInfo * info, GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_info_is_equal (const GstAudioInfo *info, const GstAudioInfo *other); diff --git a/gst-libs/gst/audio/audio-prelude.h b/gst-libs/gst/audio/audio-prelude.h new file mode 100644 index 0000000000..300fb1e89e --- /dev/null +++ b/gst-libs/gst/audio/audio-prelude.h @@ -0,0 +1,31 @@ +/* GStreamer Audio Library + * Copyright (C) 2018 GStreamer developers + * + * audio-prelude.h: prelude include header for gst-audio library + * + * 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_PRELUDE_H__ +#define __GST_AUDIO_PRELUDE_H__ + +#include + +#ifndef GST_AUDIO_API +#define GST_AUDIO_API GST_EXPORT +#endif + +#endif /* __GST_AUDIO_PRELUDE_H__ */ diff --git a/gst-libs/gst/audio/audio-quantize.h b/gst-libs/gst/audio/audio-quantize.h index 8a0bd7926c..2944b4b918 100644 --- a/gst-libs/gst/audio/audio-quantize.h +++ b/gst-libs/gst/audio/audio-quantize.h @@ -81,7 +81,7 @@ typedef enum typedef struct _GstAudioQuantize GstAudioQuantize; -GST_EXPORT +GST_AUDIO_API GstAudioQuantize * gst_audio_quantize_new (GstAudioDitherMethod dither, GstAudioNoiseShapingMethod ns, GstAudioQuantizeFlags flags, @@ -89,13 +89,13 @@ GstAudioQuantize * gst_audio_quantize_new (GstAudioDitherMethod dither, guint channels, guint quantizer); -GST_EXPORT +GST_AUDIO_API void gst_audio_quantize_free (GstAudioQuantize * quant); -GST_EXPORT +GST_AUDIO_API void gst_audio_quantize_reset (GstAudioQuantize * quant); -GST_EXPORT +GST_AUDIO_API void gst_audio_quantize_samples (GstAudioQuantize * quant, const gpointer in[], gpointer out[], guint samples); diff --git a/gst-libs/gst/audio/audio-resampler.h b/gst-libs/gst/audio/audio-resampler.h index cecda5795e..1f3045d543 100644 --- a/gst-libs/gst/audio/audio-resampler.h +++ b/gst-libs/gst/audio/audio-resampler.h @@ -201,42 +201,42 @@ typedef enum { #define GST_AUDIO_RESAMPLER_QUALITY_MAX 10 #define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4 -GST_EXPORT +GST_AUDIO_API void gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method, guint quality, gint in_rate, gint out_rate, GstStructure *options); -GST_EXPORT +GST_AUDIO_API GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMethod method, GstAudioResamplerFlags flags, GstAudioFormat format, gint channels, gint in_rate, gint out_rate, GstStructure *options); -GST_EXPORT +GST_AUDIO_API void gst_audio_resampler_free (GstAudioResampler *resampler); -GST_EXPORT +GST_AUDIO_API void gst_audio_resampler_reset (GstAudioResampler *resampler); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_resampler_update (GstAudioResampler *resampler, gint in_rate, gint out_rate, GstStructure *options); -GST_EXPORT +GST_AUDIO_API gsize gst_audio_resampler_get_out_frames (GstAudioResampler *resampler, gsize in_frames); -GST_EXPORT +GST_AUDIO_API gsize gst_audio_resampler_get_in_frames (GstAudioResampler *resampler, gsize out_frames); -GST_EXPORT +GST_AUDIO_API gsize gst_audio_resampler_get_max_latency (GstAudioResampler *resampler); -GST_EXPORT +GST_AUDIO_API void gst_audio_resampler_resample (GstAudioResampler * resampler, gpointer in[], gsize in_frames, gpointer out[], gsize out_frames); diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h index 90d22a0870..dad01c0ef2 100644 --- a/gst-libs/gst/audio/audio.h +++ b/gst-libs/gst/audio/audio.h @@ -23,6 +23,7 @@ #define __GST_AUDIO_AUDIO_H__ #include +#include #include #include #include @@ -90,7 +91,7 @@ G_BEGIN_DECLS * handling */ -GST_EXPORT +GST_AUDIO_API GstBuffer * gst_audio_buffer_clip (GstBuffer *buffer, const GstSegment *segment, gint rate, gint bpf); diff --git a/gst-libs/gst/audio/audio_mkenum.py b/gst-libs/gst/audio/audio_mkenum.py index 2f51aab5d0..af1b0a0e2c 100755 --- a/gst-libs/gst/audio/audio_mkenum.py +++ b/gst-libs/gst/audio/audio_mkenum.py @@ -8,11 +8,11 @@ import sys, os, shutil, subprocess h_array = ['--fhead', - "#ifndef __GST_AUDIO_ENUM_TYPES_H__\n#define __GST_AUDIO_ENUM_TYPES_H__\n\n#include \n\nG_BEGIN_DECLS\n", + "#ifndef __GST_AUDIO_ENUM_TYPES_H__\n#define __GST_AUDIO_ENUM_TYPES_H__\n\n#include \n#include \nG_BEGIN_DECLS\n", '--fprod', "\n/* enumerations from \"@filename@\" */\n", '--vhead', - 'GST_EXPORT GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n', + 'GST_AUDIO_API GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n', '--ftail', 'G_END_DECLS\n\n#endif /* __GST_AUDIO_ENUM_TYPES_H__ */', ] diff --git a/gst-libs/gst/audio/gstaudioaggregator.h b/gst-libs/gst/audio/gstaudioaggregator.h index d72638c231..ccc46dd271 100644 --- a/gst-libs/gst/audio/gstaudioaggregator.h +++ b/gst-libs/gst/audio/gstaudioaggregator.h @@ -98,7 +98,7 @@ struct _GstAudioAggregatorPadClass gpointer _gst_reserved[GST_PADDING_LARGE]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_aggregator_pad_get_type (void); #define GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD (gst_audio_aggregator_convert_pad_get_type()) @@ -146,7 +146,7 @@ struct _GstAudioAggregatorConvertPadClass gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_aggregator_convert_pad_get_type (void); /************************** @@ -205,10 +205,10 @@ struct _GstAudioAggregatorClass { * GstAggregator methods * ************************/ -GST_EXPORT +GST_AUDIO_API GType gst_audio_aggregator_get_type(void); -GST_EXPORT +GST_AUDIO_API void gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad, GstCaps * caps); diff --git a/gst-libs/gst/audio/gstaudiobasesink.h b/gst-libs/gst/audio/gstaudiobasesink.h index 35da8643ff..b0fdd5e9fe 100644 --- a/gst-libs/gst/audio/gstaudiobasesink.h +++ b/gst-libs/gst/audio/gstaudiobasesink.h @@ -220,54 +220,54 @@ struct _GstAudioBaseSinkClass { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_base_sink_get_type(void); -GST_EXPORT +GST_AUDIO_API GstAudioRingBuffer * gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink, GstAudioBaseSinkSlaveMethod method); -GST_EXPORT +GST_AUDIO_API GstAudioBaseSinkSlaveMethod gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink, gint64 drift_tolerance); -GST_EXPORT +GST_AUDIO_API gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink, GstClockTime alignment_threshold); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink, GstClockTime discont_wait); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_sink_set_custom_slaving_callback (GstAudioBaseSink * sink, GstAudioBaseSinkCustomSlavingCallback callback, gpointer user_data, GDestroyNotify notify); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_sink_report_device_failure (GstAudioBaseSink * sink); #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC diff --git a/gst-libs/gst/audio/gstaudiobasesrc.h b/gst-libs/gst/audio/gstaudiobasesrc.h index 27b7c46285..fcf6abe8f6 100644 --- a/gst-libs/gst/audio/gstaudiobasesrc.h +++ b/gst-libs/gst/audio/gstaudiobasesrc.h @@ -132,23 +132,23 @@ struct _GstAudioBaseSrcClass { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_base_src_get_type(void); -GST_EXPORT +GST_AUDIO_API GstAudioRingBuffer * gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *src); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_src_set_provide_clock (GstAudioBaseSrc *src, gboolean provide); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src); -GST_EXPORT +GST_AUDIO_API void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src, GstAudioBaseSrcSlaveMethod method); -GST_EXPORT +GST_AUDIO_API GstAudioBaseSrcSlaveMethod gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src); diff --git a/gst-libs/gst/audio/gstaudiocdsrc.h b/gst-libs/gst/audio/gstaudiocdsrc.h index 84d648347c..de3413fb53 100644 --- a/gst-libs/gst/audio/gstaudiocdsrc.h +++ b/gst-libs/gst/audio/gstaudiocdsrc.h @@ -131,10 +131,10 @@ struct _GstAudioCdSrcClass { gpointer _gst_reserved[GST_PADDING_LARGE]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_cd_src_get_type (void); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_cd_src_add_track (GstAudioCdSrc * src, GstAudioCdSrcTrack * track); diff --git a/gst-libs/gst/audio/gstaudioclock.h b/gst-libs/gst/audio/gstaudioclock.h index 80d8ce3c86..4870dd9898 100644 --- a/gst-libs/gst/audio/gstaudioclock.h +++ b/gst-libs/gst/audio/gstaudioclock.h @@ -89,23 +89,23 @@ struct _GstAudioClockClass { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_clock_get_type (void); -GST_EXPORT +GST_AUDIO_API GstClock* gst_audio_clock_new (const gchar *name, GstAudioClockGetTimeFunc func, gpointer user_data, GDestroyNotify destroy_notify); -GST_EXPORT +GST_AUDIO_API void gst_audio_clock_reset (GstAudioClock *clock, GstClockTime time); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_clock_get_time (GstAudioClock * clock); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_clock_adjust (GstAudioClock * clock, GstClockTime time); -GST_EXPORT +GST_AUDIO_API void gst_audio_clock_invalidate (GstAudioClock * clock); #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC diff --git a/gst-libs/gst/audio/gstaudiodecoder.h b/gst-libs/gst/audio/gstaudiodecoder.h index bd231dd1cf..f03d9ac7db 100644 --- a/gst-libs/gst/audio/gstaudiodecoder.h +++ b/gst-libs/gst/audio/gstaudiodecoder.h @@ -103,7 +103,7 @@ typedef struct _GstAudioDecoderPrivate GstAudioDecoderPrivate; /* do not use this one, use macro below */ -GST_EXPORT +GST_AUDIO_API GstFlowReturn _gst_audio_decoder_error (GstAudioDecoder *dec, gint weight, GQuark domain, gint code, gchar *txt, gchar *debug, @@ -314,124 +314,124 @@ struct _GstAudioDecoderClass gpointer _gst_reserved[GST_PADDING_LARGE - 4]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_decoder_get_type (void); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_decoder_set_output_format (GstAudioDecoder * dec, const GstAudioInfo * info); -GST_EXPORT +GST_AUDIO_API GstCaps * gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder, GstCaps * caps, GstCaps * filter); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_decoder_negotiate (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API GstFlowReturn gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf, gint frames); -GST_EXPORT +GST_AUDIO_API GstBuffer * gst_audio_decoder_allocate_output_buffer (GstAudioDecoder * dec, gsize size); /* context parameters */ -GST_EXPORT +GST_AUDIO_API GstAudioInfo * gst_audio_decoder_get_audio_info (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_plc_aware (GstAudioDecoder * dec, gboolean plc); -GST_EXPORT +GST_AUDIO_API gint gst_audio_decoder_get_plc_aware (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_estimate_rate (GstAudioDecoder * dec, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gint gst_audio_decoder_get_estimate_rate (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API gint gst_audio_decoder_get_delay (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_max_errors (GstAudioDecoder * dec, gint num); -GST_EXPORT +GST_AUDIO_API gint gst_audio_decoder_get_max_errors (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_latency (GstAudioDecoder * dec, GstClockTime min, GstClockTime max); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_get_latency (GstAudioDecoder * dec, GstClockTime * min, GstClockTime * max); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_get_parse_state (GstAudioDecoder * dec, gboolean * sync, gboolean * eos); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_allocation_caps (GstAudioDecoder * dec, GstCaps * allocation_caps); /* object properties */ -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_plc (GstAudioDecoder * dec, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_decoder_get_plc (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_min_latency (GstAudioDecoder * dec, GstClockTime num); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_decoder_get_min_latency (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_tolerance (GstAudioDecoder * dec, GstClockTime tolerance); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_decoder_get_tolerance (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_drainable (GstAudioDecoder * dec, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_decoder_get_drainable (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_needs_format (GstAudioDecoder * dec, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_decoder_get_needs_format (GstAudioDecoder * dec); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_get_allocator (GstAudioDecoder * dec, GstAllocator ** allocator, GstAllocationParams * params); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_merge_tags (GstAudioDecoder * dec, const GstTagList * tags, GstTagMergeMode mode); -GST_EXPORT +GST_AUDIO_API void gst_audio_decoder_set_use_default_pad_acceptcaps (GstAudioDecoder * decoder, gboolean use); diff --git a/gst-libs/gst/audio/gstaudioencoder.h b/gst-libs/gst/audio/gstaudioencoder.h index a72224afab..348b70e507 100644 --- a/gst-libs/gst/audio/gstaudioencoder.h +++ b/gst-libs/gst/audio/gstaudioencoder.h @@ -247,127 +247,127 @@ struct _GstAudioEncoderClass { gpointer _gst_reserved[GST_PADDING_LARGE-3]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_encoder_get_type (void); -GST_EXPORT +GST_AUDIO_API GstFlowReturn gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buffer, gint samples); -GST_EXPORT +GST_AUDIO_API GstCaps * gst_audio_encoder_proxy_getcaps (GstAudioEncoder * enc, GstCaps * caps, GstCaps * filter); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_encoder_set_output_format (GstAudioEncoder * enc, GstCaps * caps); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_encoder_negotiate (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API GstBuffer * gst_audio_encoder_allocate_output_buffer (GstAudioEncoder * enc, gsize size); /* context parameters */ -GST_EXPORT +GST_AUDIO_API GstAudioInfo * gst_audio_encoder_get_audio_info (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API gint gst_audio_encoder_get_frame_samples_min (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_frame_samples_min (GstAudioEncoder * enc, gint num); -GST_EXPORT +GST_AUDIO_API gint gst_audio_encoder_get_frame_samples_max (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_frame_samples_max (GstAudioEncoder * enc, gint num); -GST_EXPORT +GST_AUDIO_API gint gst_audio_encoder_get_frame_max (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_frame_max (GstAudioEncoder * enc, gint num); -GST_EXPORT +GST_AUDIO_API gint gst_audio_encoder_get_lookahead (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_lookahead (GstAudioEncoder * enc, gint num); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_get_latency (GstAudioEncoder * enc, GstClockTime * min, GstClockTime * max); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_latency (GstAudioEncoder * enc, GstClockTime min, GstClockTime max); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_headers (GstAudioEncoder * enc, GList * headers); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_allocation_caps (GstAudioEncoder * enc, GstCaps * allocation_caps); /* object properties */ -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_mark_granule (GstAudioEncoder * enc, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_encoder_get_mark_granule (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_encoder_get_perfect_timestamp (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_hard_resync (GstAudioEncoder * enc, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_encoder_get_hard_resync (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_tolerance (GstAudioEncoder * enc, GstClockTime tolerance); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_encoder_get_tolerance (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_hard_min (GstAudioEncoder * enc, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_encoder_get_hard_min (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_set_drainable (GstAudioEncoder * enc, gboolean enabled); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_encoder_get_drainable (GstAudioEncoder * enc); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_get_allocator (GstAudioEncoder * enc, GstAllocator ** allocator, GstAllocationParams * params); -GST_EXPORT +GST_AUDIO_API void gst_audio_encoder_merge_tags (GstAudioEncoder * enc, const GstTagList * tags, GstTagMergeMode mode); diff --git a/gst-libs/gst/audio/gstaudiofilter.h b/gst-libs/gst/audio/gstaudiofilter.h index ee8cc3ee50..ac1b5704ed 100644 --- a/gst-libs/gst/audio/gstaudiofilter.h +++ b/gst-libs/gst/audio/gstaudiofilter.h @@ -92,10 +92,10 @@ struct _GstAudioFilterClass { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_filter_get_type (void); -GST_EXPORT +GST_AUDIO_API void gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass, GstCaps * allowed_caps); diff --git a/gst-libs/gst/audio/gstaudioiec61937.h b/gst-libs/gst/audio/gstaudioiec61937.h index fcd1b08794..1885a921dd 100644 --- a/gst-libs/gst/audio/gstaudioiec61937.h +++ b/gst-libs/gst/audio/gstaudioiec61937.h @@ -24,10 +24,10 @@ #include -GST_EXPORT +GST_AUDIO_API guint gst_audio_iec61937_frame_size (const GstAudioRingBufferSpec * spec); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_iec61937_payload (const guint8 * src, guint src_n, guint8 * dst, guint dst_n, const GstAudioRingBufferSpec * spec, diff --git a/gst-libs/gst/audio/gstaudiometa.h b/gst-libs/gst/audio/gstaudiometa.h index 4cd5cbe69a..2f6c72f7f9 100644 --- a/gst-libs/gst/audio/gstaudiometa.h +++ b/gst-libs/gst/audio/gstaudiometa.h @@ -56,19 +56,19 @@ struct _GstAudioDownmixMeta { gfloat **matrix; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_downmix_meta_api_get_type (void); -GST_EXPORT +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_EXPORT +GST_AUDIO_API GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer *buffer, const GstAudioChannelPosition *to_position, gint to_channels); -GST_EXPORT +GST_AUDIO_API GstAudioDownmixMeta * gst_buffer_add_audio_downmix_meta (GstBuffer *buffer, const GstAudioChannelPosition *from_position, gint from_channels, @@ -111,15 +111,15 @@ struct _GstAudioClippingMeta { guint64 end; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_clipping_meta_api_get_type (void); -GST_EXPORT +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_EXPORT +GST_AUDIO_API GstAudioClippingMeta * gst_buffer_add_audio_clipping_meta (GstBuffer *buffer, GstFormat format, guint64 start, diff --git a/gst-libs/gst/audio/gstaudioringbuffer.h b/gst-libs/gst/audio/gstaudioringbuffer.h index c447c0ee3c..64c6026990 100644 --- a/gst-libs/gst/audio/gstaudioringbuffer.h +++ b/gst-libs/gst/audio/gstaudioringbuffer.h @@ -265,122 +265,122 @@ struct _GstAudioRingBufferClass { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_ring_buffer_get_type(void); /* callback stuff */ -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_set_callback (GstAudioRingBuffer *buf, GstAudioRingBufferCallback cb, gpointer user_data); -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_set_callback_full (GstAudioRingBuffer *buf, GstAudioRingBufferCallback cb, gpointer user_data, GDestroyNotify notify); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec *spec, GstCaps *caps); -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_debug_spec_caps (GstAudioRingBufferSpec *spec); -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_debug_spec_buff (GstAudioRingBufferSpec *spec); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_convert (GstAudioRingBuffer * buf, GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val); /* device state */ -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_open_device (GstAudioRingBuffer *buf); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_close_device (GstAudioRingBuffer *buf); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer *buf); /* allocate resources */ -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_acquire (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_release (GstAudioRingBuffer *buf); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer *buf); /* set the device channel positions */ -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_set_channel_positions (GstAudioRingBuffer *buf, const GstAudioChannelPosition *position); /* activating */ -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_activate (GstAudioRingBuffer *buf, gboolean active); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_is_active (GstAudioRingBuffer *buf); /* flushing */ -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer *buf, gboolean flushing); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer *buf); /* playback/pause */ -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_start (GstAudioRingBuffer *buf); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_pause (GstAudioRingBuffer *buf); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_stop (GstAudioRingBuffer *buf); /* get status */ -GST_EXPORT +GST_AUDIO_API guint gst_audio_ring_buffer_delay (GstAudioRingBuffer *buf); -GST_EXPORT +GST_AUDIO_API guint64 gst_audio_ring_buffer_samples_done (GstAudioRingBuffer *buf); -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_set_sample (GstAudioRingBuffer *buf, guint64 sample); /* clear all segments */ -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_clear_all (GstAudioRingBuffer *buf); /* commit samples */ -GST_EXPORT +GST_AUDIO_API guint gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 *sample, guint8 * data, gint in_samples, gint out_samples, gint * accum); /* read samples */ -GST_EXPORT +GST_AUDIO_API guint gst_audio_ring_buffer_read (GstAudioRingBuffer *buf, guint64 sample, guint8 *data, guint len, GstClockTime *timestamp); /* Set timestamp on buffer */ -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gint readseg, GstClockTime timestamp); @@ -389,17 +389,17 @@ void gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer * buf, gboolean gst_audio_ring_buffer_prepare_write (GstAudioRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len); */ -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer *buf, gint *segment, guint8 **readptr, gint *len); -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_clear (GstAudioRingBuffer *buf, gint segment); -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_advance (GstAudioRingBuffer *buf, guint advance); -GST_EXPORT +GST_AUDIO_API void gst_audio_ring_buffer_may_start (GstAudioRingBuffer *buf, gboolean allowed); #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC diff --git a/gst-libs/gst/audio/gstaudiosink.h b/gst-libs/gst/audio/gstaudiosink.h index 04d429912b..b3c482e3b1 100644 --- a/gst-libs/gst/audio/gstaudiosink.h +++ b/gst-libs/gst/audio/gstaudiosink.h @@ -97,7 +97,7 @@ struct _GstAudioSinkClass { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_sink_get_type(void); #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC diff --git a/gst-libs/gst/audio/gstaudiosrc.h b/gst-libs/gst/audio/gstaudiosrc.h index 04a673128e..8fd0afba5b 100644 --- a/gst-libs/gst/audio/gstaudiosrc.h +++ b/gst-libs/gst/audio/gstaudiosrc.h @@ -96,7 +96,7 @@ struct _GstAudioSrcClass { gpointer _gst_reserved[GST_PADDING]; }; -GST_EXPORT +GST_AUDIO_API GType gst_audio_src_get_type(void); #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC diff --git a/gst-libs/gst/audio/gstaudiostreamalign.h b/gst-libs/gst/audio/gstaudiostreamalign.h index 0800881362..02cedf34a0 100644 --- a/gst-libs/gst/audio/gstaudiostreamalign.h +++ b/gst-libs/gst/audio/gstaudiostreamalign.h @@ -23,6 +23,7 @@ #define __GST_AUDIO_STREAM_ALIGN_H__ #include +#include G_BEGIN_DECLS @@ -30,47 +31,47 @@ G_BEGIN_DECLS typedef struct _GstAudioStreamAlign GstAudioStreamAlign; -GST_EXPORT +GST_AUDIO_API GType gst_audio_stream_align_get_type (void); -GST_EXPORT +GST_AUDIO_API GstAudioStreamAlign * gst_audio_stream_align_new (gint rate, GstClockTime alignment_threshold, GstClockTime discont_wait); -GST_EXPORT +GST_AUDIO_API GstAudioStreamAlign * gst_audio_stream_align_copy (const GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API void gst_audio_stream_align_free (GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API void gst_audio_stream_align_set_rate (GstAudioStreamAlign * align, gint rate); -GST_EXPORT +GST_AUDIO_API gint gst_audio_stream_align_get_rate (GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API void gst_audio_stream_align_set_alignment_threshold (GstAudioStreamAlign * align, GstClockTime alignment_threshold); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_stream_align_get_alignment_threshold (GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API void gst_audio_stream_align_set_discont_wait (GstAudioStreamAlign * align, GstClockTime discont_wait); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_stream_align_get_discont_wait (GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API void gst_audio_stream_align_mark_discont (GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API GstClockTime gst_audio_stream_align_get_timestamp_at_discont (GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API guint64 gst_audio_stream_align_get_samples_since_discont (GstAudioStreamAlign * align); -GST_EXPORT +GST_AUDIO_API gboolean gst_audio_stream_align_process (GstAudioStreamAlign * align, gboolean discont, GstClockTime timestamp, diff --git a/gst-libs/gst/audio/streamvolume.h b/gst-libs/gst/audio/streamvolume.h index 6d602c0035..4802d04b0f 100644 --- a/gst-libs/gst/audio/streamvolume.h +++ b/gst-libs/gst/audio/streamvolume.h @@ -21,6 +21,7 @@ #define __GST_STREAM_VOLUME_H__ #include +#include G_BEGIN_DECLS @@ -58,26 +59,26 @@ typedef enum { GST_STREAM_VOLUME_FORMAT_DB } GstStreamVolumeFormat; -GST_EXPORT +GST_AUDIO_API GType gst_stream_volume_get_type (void); -GST_EXPORT +GST_AUDIO_API void gst_stream_volume_set_volume (GstStreamVolume *volume, GstStreamVolumeFormat format, gdouble val); -GST_EXPORT +GST_AUDIO_API gdouble gst_stream_volume_get_volume (GstStreamVolume *volume, GstStreamVolumeFormat format); -GST_EXPORT +GST_AUDIO_API void gst_stream_volume_set_mute (GstStreamVolume *volume, gboolean mute); -GST_EXPORT +GST_AUDIO_API gboolean gst_stream_volume_get_mute (GstStreamVolume *volume); -GST_EXPORT +GST_AUDIO_API gdouble gst_stream_volume_convert_volume (GstStreamVolumeFormat from, GstStreamVolumeFormat to, gdouble val) G_GNUC_CONST;