mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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audio: GST_EXPORT -> GST_AUDIO_API
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
This commit is contained in:
parent
be0ca93a90
commit
371e3e460a
26 changed files with 257 additions and 221 deletions
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@ -19,7 +19,8 @@ glib_enum_headers= \
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glib_enum_define = GST_AUDIO
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glib_gen_prefix = gst_audio
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glib_gen_basename = audio
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glib_gen_decl_banner=GST_EXPORT
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glib_gen_decl_banner=GST_AUDIO_API
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glib_gen_decl_include=\#include <gst/audio/audio-prelude.h>
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built_sources = audio-enumtypes.c
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built_headers = audio-enumtypes.h
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@ -63,6 +64,7 @@ nodist_libgstaudio_@GST_API_VERSION@_la_SOURCES = $(BUILT_SOURCES)
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libgstaudio_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/audio
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libgstaudio_@GST_API_VERSION@include_HEADERS = \
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audio.h \
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audio-prelude.h \
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audio-format.h \
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audio-channels.h \
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audio-channel-mixer.h \
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@ -46,7 +46,7 @@ typedef enum {
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GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT = (1 << 3)
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} GstAudioChannelMixerFlags;
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GST_EXPORT
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GST_AUDIO_API
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GstAudioChannelMixer * gst_audio_channel_mixer_new (GstAudioChannelMixerFlags flags,
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GstAudioFormat format,
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gint in_channels,
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@ -54,28 +54,28 @@ GstAudioChannelMixer * gst_audio_channel_mixer_new (GstAudioChannelMixerFlags
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gint out_channels,
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GstAudioChannelPosition *out_position);
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GST_EXPORT
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GST_AUDIO_API
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GstAudioChannelMixer * gst_audio_channel_mixer_new_with_matrix (GstAudioChannelMixerFlags flags,
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GstAudioFormat format,
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gint in_channels,
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gint out_channels,
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gfloat **matrix);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_channel_mixer_free (GstAudioChannelMixer *mix);
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/*
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* Checks for passthrough (= identity matrix).
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*/
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_channel_mixer_is_passthrough (GstAudioChannelMixer *mix);
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/*
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* Do actual mixing.
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*/
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_channel_mixer_samples (GstAudioChannelMixer * mix,
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const gpointer in[],
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gpointer out[],
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@ -131,47 +131,47 @@ typedef enum {
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#define GST_AUDIO_CHANNEL_POSITION_MASK(pos) (G_GUINT64_CONSTANT(1)<< GST_AUDIO_CHANNEL_POSITION_ ## pos)
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_buffer_reorder_channels (GstBuffer * buffer,
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GstAudioFormat format,
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gint channels,
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const GstAudioChannelPosition * from,
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const GstAudioChannelPosition * to);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_reorder_channels (gpointer data, gsize size,
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GstAudioFormat format,
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gint channels,
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const GstAudioChannelPosition * from,
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const GstAudioChannelPosition * to);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_channel_positions_to_valid_order (GstAudioChannelPosition *position,
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gint channels);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_check_valid_channel_positions (const GstAudioChannelPosition *position,
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gint channels, gboolean force_order);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_channel_positions_to_mask (const GstAudioChannelPosition *position,
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gint channels, gboolean force_order,
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guint64 *channel_mask);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_channel_positions_from_mask (gint channels, guint64 channel_mask,
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GstAudioChannelPosition * position);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_get_channel_reorder_map (gint channels,
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const GstAudioChannelPosition * from,
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const GstAudioChannelPosition * to,
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gint *reorder_map);
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GST_EXPORT
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GST_AUDIO_API
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guint64 gst_audio_channel_get_fallback_mask (gint channels);
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GST_EXPORT
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GST_AUDIO_API
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gchar* gst_audio_channel_positions_to_string (const GstAudioChannelPosition * position,
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gint channels);
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@ -115,51 +115,51 @@ typedef enum {
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GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE = (1 << 1)
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} GstAudioConverterFlags;
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GST_EXPORT
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GST_AUDIO_API
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GstAudioConverter * gst_audio_converter_new (GstAudioConverterFlags flags,
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GstAudioInfo *in_info,
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GstAudioInfo *out_info,
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GstStructure *config);
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GST_EXPORT
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GST_AUDIO_API
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GType gst_audio_converter_get_type (void);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_converter_free (GstAudioConverter * convert);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_converter_reset (GstAudioConverter * convert);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_converter_update_config (GstAudioConverter * convert,
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gint in_rate, gint out_rate,
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GstStructure *config);
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GST_EXPORT
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GST_AUDIO_API
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const GstStructure * gst_audio_converter_get_config (GstAudioConverter * convert,
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gint *in_rate, gint *out_rate);
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GST_EXPORT
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GST_AUDIO_API
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gsize gst_audio_converter_get_out_frames (GstAudioConverter *convert,
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gsize in_frames);
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GST_EXPORT
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GST_AUDIO_API
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gsize gst_audio_converter_get_in_frames (GstAudioConverter *convert,
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gsize out_frames);
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GST_EXPORT
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GST_AUDIO_API
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gsize gst_audio_converter_get_max_latency (GstAudioConverter *convert);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_converter_samples (GstAudioConverter * convert,
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GstAudioConverterFlags flags,
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gpointer in[], gsize in_frames,
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gpointer out[], gsize out_frames);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_converter_supports_inplace (GstAudioConverter *convert);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_converter_convert (GstAudioConverter * convert,
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GstAudioConverterFlags flags,
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gpointer in, gsize in_size,
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@ -250,7 +250,7 @@ struct _GstAudioFormatInfo {
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_EXPORT
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GST_AUDIO_API
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GType gst_audio_format_info_get_type (void);
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#define GST_AUDIO_FORMAT_INFO_FORMAT(info) ((info)->format)
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@ -268,21 +268,21 @@ GType gst_audio_format_info_get_type (void);
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#define GST_AUDIO_FORMAT_INFO_DEPTH(info) ((info)->depth)
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GST_EXPORT
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GST_AUDIO_API
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GstAudioFormat gst_audio_format_build_integer (gboolean sign, gint endianness,
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gint width, gint depth) G_GNUC_CONST;
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GST_EXPORT
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GST_AUDIO_API
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GstAudioFormat gst_audio_format_from_string (const gchar *format) G_GNUC_CONST;
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GST_EXPORT
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GST_AUDIO_API
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const gchar * gst_audio_format_to_string (GstAudioFormat format) G_GNUC_CONST;
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GST_EXPORT
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GST_AUDIO_API
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const GstAudioFormatInfo *
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gst_audio_format_get_info (GstAudioFormat format) G_GNUC_CONST;
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_format_fill_silence (const GstAudioFormatInfo *info,
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gpointer dest, gsize length);
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@ -85,7 +85,7 @@ struct _GstAudioInfo {
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};
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#define GST_TYPE_AUDIO_INFO (gst_audio_info_get_type ())
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GST_EXPORT
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GST_AUDIO_API
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GType gst_audio_info_get_type (void);
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#define GST_AUDIO_INFO_IS_VALID(i) ((i)->finfo != NULL && (i)->rate > 0 && (i)->channels > 0 && (i)->bpf > 0)
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@ -113,35 +113,35 @@ GType gst_audio_info_get_type (void);
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#define GST_AUDIO_INFO_BPF(info) ((info)->bpf)
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#define GST_AUDIO_INFO_POSITION(info,c) ((info)->position[c])
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GST_EXPORT
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GST_AUDIO_API
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GstAudioInfo * gst_audio_info_new (void);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_info_init (GstAudioInfo *info);
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GST_EXPORT
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GST_AUDIO_API
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GstAudioInfo * gst_audio_info_copy (const GstAudioInfo *info);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_info_free (GstAudioInfo *info);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_info_set_format (GstAudioInfo *info, GstAudioFormat format,
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gint rate, gint channels,
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const GstAudioChannelPosition *position);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_info_from_caps (GstAudioInfo *info, const GstCaps *caps);
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GST_EXPORT
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GST_AUDIO_API
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GstCaps * gst_audio_info_to_caps (const GstAudioInfo *info);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_info_convert (const GstAudioInfo * info,
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GstFormat src_fmt, gint64 src_val,
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GstFormat dest_fmt, gint64 * dest_val);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_info_is_equal (const GstAudioInfo *info,
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const GstAudioInfo *other);
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31
gst-libs/gst/audio/audio-prelude.h
Normal file
31
gst-libs/gst/audio/audio-prelude.h
Normal file
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@ -0,0 +1,31 @@
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/* GStreamer Audio Library
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* Copyright (C) 2018 GStreamer developers
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*
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* audio-prelude.h: prelude include header for gst-audio library
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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_PRELUDE_H__
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#define __GST_AUDIO_PRELUDE_H__
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#include <gst/gst.h>
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#ifndef GST_AUDIO_API
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#define GST_AUDIO_API GST_EXPORT
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#endif
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#endif /* __GST_AUDIO_PRELUDE_H__ */
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@ -81,7 +81,7 @@ typedef enum
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typedef struct _GstAudioQuantize GstAudioQuantize;
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GST_EXPORT
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GST_AUDIO_API
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GstAudioQuantize * gst_audio_quantize_new (GstAudioDitherMethod dither,
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GstAudioNoiseShapingMethod ns,
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GstAudioQuantizeFlags flags,
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guint channels,
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guint quantizer);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_quantize_free (GstAudioQuantize * quant);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_quantize_reset (GstAudioQuantize * quant);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_quantize_samples (GstAudioQuantize * quant,
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const gpointer in[],
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gpointer out[], guint samples);
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@ -201,42 +201,42 @@ typedef enum {
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#define GST_AUDIO_RESAMPLER_QUALITY_MAX 10
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#define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
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guint quality,
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gint in_rate, gint out_rate,
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GstStructure *options);
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GST_EXPORT
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GST_AUDIO_API
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GstAudioResampler * gst_audio_resampler_new (GstAudioResamplerMethod method,
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GstAudioResamplerFlags flags,
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GstAudioFormat format, gint channels,
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gint in_rate, gint out_rate,
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GstStructure *options);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_resampler_free (GstAudioResampler *resampler);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_resampler_reset (GstAudioResampler *resampler);
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GST_EXPORT
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GST_AUDIO_API
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gboolean gst_audio_resampler_update (GstAudioResampler *resampler,
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gint in_rate, gint out_rate,
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GstStructure *options);
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GST_EXPORT
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GST_AUDIO_API
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gsize gst_audio_resampler_get_out_frames (GstAudioResampler *resampler,
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gsize in_frames);
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GST_EXPORT
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GST_AUDIO_API
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gsize gst_audio_resampler_get_in_frames (GstAudioResampler *resampler,
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gsize out_frames);
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GST_EXPORT
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GST_AUDIO_API
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gsize gst_audio_resampler_get_max_latency (GstAudioResampler *resampler);
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GST_EXPORT
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GST_AUDIO_API
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void gst_audio_resampler_resample (GstAudioResampler * resampler,
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gpointer in[], gsize in_frames,
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gpointer out[], gsize out_frames);
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@ -23,6 +23,7 @@
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#define __GST_AUDIO_AUDIO_H__
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#include <gst/gst.h>
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#include <gst/audio/audio-prelude.h>
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#include <gst/audio/audio-enumtypes.h>
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#include <gst/audio/audio-format.h>
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#include <gst/audio/audio-channels.h>
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@ -90,7 +91,7 @@ G_BEGIN_DECLS
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* handling
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*/
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GST_EXPORT
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GST_AUDIO_API
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GstBuffer * gst_audio_buffer_clip (GstBuffer *buffer,
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const GstSegment *segment,
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gint rate, gint bpf);
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@ -8,11 +8,11 @@
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import sys, os, shutil, subprocess
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h_array = ['--fhead',
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"#ifndef __GST_AUDIO_ENUM_TYPES_H__\n#define __GST_AUDIO_ENUM_TYPES_H__\n\n#include <gst/gst.h>\n\nG_BEGIN_DECLS\n",
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"#ifndef __GST_AUDIO_ENUM_TYPES_H__\n#define __GST_AUDIO_ENUM_TYPES_H__\n\n#include <gst/gst.h>\n#include <gst/audio/audio-prelude.h>\nG_BEGIN_DECLS\n",
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'--fprod',
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"\n/* enumerations from \"@filename@\" */\n",
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'--vhead',
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'GST_EXPORT GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n',
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'GST_AUDIO_API GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n',
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'--ftail',
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'G_END_DECLS\n\n#endif /* __GST_AUDIO_ENUM_TYPES_H__ */',
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]
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@ -98,7 +98,7 @@ struct _GstAudioAggregatorPadClass
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gpointer _gst_reserved[GST_PADDING_LARGE];
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};
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GST_EXPORT
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GST_AUDIO_API
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GType gst_audio_aggregator_pad_get_type (void);
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#define GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD (gst_audio_aggregator_convert_pad_get_type())
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@ -146,7 +146,7 @@ struct _GstAudioAggregatorConvertPadClass
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_EXPORT
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GST_AUDIO_API
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GType gst_audio_aggregator_convert_pad_get_type (void);
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/**************************
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||||
|
@ -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);
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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);
|
||||
|
||||
|
|
|
@ -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);
|
||||
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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);
|
||||
|
||||
|
|
|
@ -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);
|
||||
|
||||
|
|
|
@ -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);
|
||||
|
||||
|
|
|
@ -24,10 +24,10 @@
|
|||
|
||||
#include <gst/audio/gstaudioringbuffer.h>
|
||||
|
||||
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,
|
||||
|
|
|
@ -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,
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -23,6 +23,7 @@
|
|||
#define __GST_AUDIO_STREAM_ALIGN_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/audio-prelude.h>
|
||||
|
||||
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,
|
||||
|
|
|
@ -21,6 +21,7 @@
|
|||
#define __GST_STREAM_VOLUME_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/audio-prelude.h>
|
||||
|
||||
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;
|
||||
|
|
Loading…
Reference in a new issue