/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Library <2001> Thomas Vander Stichele <thomas@apestaart.org>
+ * <2011> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#include <gst/gst.h>
-#include <gst/audio/audioclock.h>
+#ifndef __GST_AUDIO_AUDIO_H__
+#define __GST_AUDIO_AUDIO_H__
+
+#include <gst/audio/multichannel.h>
G_BEGIN_DECLS
-/* For people that are looking at this source: the purpose of these defines is
- * to make GstCaps a bit easier, in that you don't have to know all of the
- * properties that need to be defined. you can just use these macros. currently
- * (8/01) the only plugins that use these are the passthrough, speed, volume,
- * adder, and [de]interleave plugins. These are for convenience only, and do not
- * specify the 'limits' of GStreamer. you might also use these definitions as a
- * base for making your own caps, if need be.
+#if G_BYTE_ORDER == G_BIG_ENDIAN
+#define _GST_AUDIO_FORMAT_NE(fmt) GST_AUDIO_FORMAT_ ## fmt ## BE
+#elif G_BYTE_ORDER == G_LITTLE_ENDIAN
+#define _GST_AUDIO_FORMAT_NE(fmt) GST_AUDIO_FORMAT_ ## fmt ## LE
+#endif
+
+/**
+ * GstAudioFormat:
+ * @GST_AUDIO_FORMAT_UNKNOWN: unknown audio format
+ * @GST_AUDIO_FORMAT_S8: 8 bits in 8 bits, signed
+ * @GST_AUDIO_FORMAT_U8: 8 bits in 8 bits, unsigned
+ * @GST_AUDIO_FORMAT_S16LE: 16 bits in 16 bits, signed, little endian
+ * @GST_AUDIO_FORMAT_S16BE: 16 bits in 16 bits, signed, big endian
+ * @GST_AUDIO_FORMAT_U16LE: 16 bits in 16 bits, unsigned, little endian
+ * @GST_AUDIO_FORMAT_U16BE: 16 bits in 16 bits, unsigned, big endian
+ * @GST_AUDIO_FORMAT_S24_32LE: 24 bits in 32 bits, signed, little endian
+ * @GST_AUDIO_FORMAT_S24_32BE: 24 bits in 32 bits, signed, big endian
+ * @GST_AUDIO_FORMAT_U24_32LE: 24 bits in 32 bits, unsigned, little endian
+ * @GST_AUDIO_FORMAT_U24_32BE: 24 bits in 32 bits, unsigned, big endian
+ * @GST_AUDIO_FORMAT_S32LE: 32 bits in 32 bits, signed, little endian
+ * @GST_AUDIO_FORMAT_S32BE: 32 bits in 32 bits, signed, big endian
+ * @GST_AUDIO_FORMAT_U32LE: 32 bits in 32 bits, unsigned, little endian
+ * @GST_AUDIO_FORMAT_U32BE: 32 bits in 32 bits, unsigned, big endian
+ * @GST_AUDIO_FORMAT_S24LE: 24 bits in 24 bits, signed, little endian
+ * @GST_AUDIO_FORMAT_S24BE: 24 bits in 24 bits, signed, big endian
+ * @GST_AUDIO_FORMAT_U24LE: 24 bits in 24 bits, unsigned, little endian
+ * @GST_AUDIO_FORMAT_U24BE: 24 bits in 24 bits, unsigned, big endian
+ * @GST_AUDIO_FORMAT_S20LE: 20 bits in 24 bits, signed, little endian
+ * @GST_AUDIO_FORMAT_S20BE: 20 bits in 24 bits, signed, big endian
+ * @GST_AUDIO_FORMAT_U20LE: 20 bits in 24 bits, unsigned, little endian
+ * @GST_AUDIO_FORMAT_U20BE: 20 bits in 24 bits, unsigned, big endian
+ * @GST_AUDIO_FORMAT_S18LE: 18 bits in 24 bits, signed, little endian
+ * @GST_AUDIO_FORMAT_S18BE: 18 bits in 24 bits, signed, big endian
+ * @GST_AUDIO_FORMAT_U18LE: 18 bits in 24 bits, unsigned, little endian
+ * @GST_AUDIO_FORMAT_U18BE: 18 bits in 24 bits, unsigned, big endian
+ * @GST_AUDIO_FORMAT_F32LE: 32-bit floating point samples, little endian
+ * @GST_AUDIO_FORMAT_F32BE: 32-bit floating point samples, big endian
+ * @GST_AUDIO_FORMAT_F64LE: 64-bit floating point samples, little endian
+ * @GST_AUDIO_FORMAT_F64BE: 64-bit floating point samples, big endian
+ * @GST_AUDIO_FORMAT_S16: 16 bits in 16 bits, signed, native endianness
+ * @GST_AUDIO_FORMAT_U16: 16 bits in 16 bits, unsigned, native endianness
+ * @GST_AUDIO_FORMAT_S24_32: 24 bits in 32 bits, signed, native endianness
+ * @GST_AUDIO_FORMAT_U24_32: 24 bits in 32 bits, unsigned, native endianness
+ * @GST_AUDIO_FORMAT_S32: 32 bits in 32 bits, signed, native endianness
+ * @GST_AUDIO_FORMAT_U32: 32 bits in 32 bits, unsigned, native endianness
+ * @GST_AUDIO_FORMAT_S24: 24 bits in 24 bits, signed, native endianness
+ * @GST_AUDIO_FORMAT_U24: 24 bits in 24 bits, unsigned, native endianness
+ * @GST_AUDIO_FORMAT_S20: 20 bits in 24 bits, signed, native endianness
+ * @GST_AUDIO_FORMAT_U20: 20 bits in 24 bits, unsigned, native endianness
+ * @GST_AUDIO_FORMAT_S18: 18 bits in 24 bits, signed, native endianness
+ * @GST_AUDIO_FORMAT_U18: 18 bits in 24 bits, unsigned, native endianness
+ * @GST_AUDIO_FORMAT_F32: 32-bit floating point samples, native endianness
+ * @GST_AUDIO_FORMAT_F64: 64-bit floating point samples, native endianness
*
- * For example, to make a source pad that can output streams of either mono
- * float or any channel int:
+ * Enum value describing the most common audio formats.
+ */
+typedef enum {
+ GST_AUDIO_FORMAT_UNKNOWN,
+ /* 8 bit */
+ GST_AUDIO_FORMAT_S8,
+ GST_AUDIO_FORMAT_U8,
+ /* 16 bit */
+ GST_AUDIO_FORMAT_S16LE,
+ GST_AUDIO_FORMAT_S16BE,
+ GST_AUDIO_FORMAT_U16LE,
+ GST_AUDIO_FORMAT_U16BE,
+ /* 24 bit in low 3 bytes of 32 bits*/
+ GST_AUDIO_FORMAT_S24_32LE,
+ GST_AUDIO_FORMAT_S24_32BE,
+ GST_AUDIO_FORMAT_U24_32LE,
+ GST_AUDIO_FORMAT_U24_32BE,
+ /* 32 bit */
+ GST_AUDIO_FORMAT_S32LE,
+ GST_AUDIO_FORMAT_S32BE,
+ GST_AUDIO_FORMAT_U32LE,
+ GST_AUDIO_FORMAT_U32BE,
+ /* 24 bit in 3 bytes*/
+ GST_AUDIO_FORMAT_S24LE,
+ GST_AUDIO_FORMAT_S24BE,
+ GST_AUDIO_FORMAT_U24LE,
+ GST_AUDIO_FORMAT_U24BE,
+ /* 20 bit in 3 bytes*/
+ GST_AUDIO_FORMAT_S20LE,
+ GST_AUDIO_FORMAT_S20BE,
+ GST_AUDIO_FORMAT_U20LE,
+ GST_AUDIO_FORMAT_U20BE,
+ /* 18 bit in 3 bytes*/
+ GST_AUDIO_FORMAT_S18LE,
+ GST_AUDIO_FORMAT_S18BE,
+ GST_AUDIO_FORMAT_U18LE,
+ GST_AUDIO_FORMAT_U18BE,
+ /* float */
+ GST_AUDIO_FORMAT_F32LE,
+ GST_AUDIO_FORMAT_F32BE,
+ GST_AUDIO_FORMAT_F64LE,
+ GST_AUDIO_FORMAT_F64BE,
+ /* native endianness equivalents */
+ GST_AUDIO_FORMAT_S16 = _GST_AUDIO_FORMAT_NE(S16),
+ GST_AUDIO_FORMAT_U16 = _GST_AUDIO_FORMAT_NE(U16),
+ GST_AUDIO_FORMAT_S24_32 = _GST_AUDIO_FORMAT_NE(S24_32),
+ GST_AUDIO_FORMAT_U24_32 = _GST_AUDIO_FORMAT_NE(U24_32),
+ GST_AUDIO_FORMAT_S32 = _GST_AUDIO_FORMAT_NE(S32),
+ GST_AUDIO_FORMAT_U32 = _GST_AUDIO_FORMAT_NE(U32),
+ GST_AUDIO_FORMAT_S24 = _GST_AUDIO_FORMAT_NE(S24),
+ GST_AUDIO_FORMAT_U24 = _GST_AUDIO_FORMAT_NE(U24),
+ GST_AUDIO_FORMAT_S20 = _GST_AUDIO_FORMAT_NE(S20),
+ GST_AUDIO_FORMAT_U20 = _GST_AUDIO_FORMAT_NE(U20),
+ GST_AUDIO_FORMAT_S18 = _GST_AUDIO_FORMAT_NE(S18),
+ GST_AUDIO_FORMAT_U18 = _GST_AUDIO_FORMAT_NE(U18),
+ GST_AUDIO_FORMAT_F32 = _GST_AUDIO_FORMAT_NE(F32),
+ GST_AUDIO_FORMAT_F64 = _GST_AUDIO_FORMAT_NE(F64)
+} GstAudioFormat;
+
+/* FIXME: need GTypes */
+typedef struct _GstAudioFormatInfo GstAudioFormatInfo;
+typedef struct _GstAudioInfo GstAudioInfo;
+
+/**
+ * GstAudioFormatFlags:
+ * @GST_AUDIO_FORMAT_FLAG_INTEGER: integer samples
+ * @GST_AUDIO_FORMAT_FLAG_FLOAT: float samples
+ * @GST_AUDIO_FORMAT_FLAG_SIGNED: signed samples
+ * @GST_AUDIO_FORMAT_FLAG_COMPLEX: complex layout
*
- * template = gst_pad_template_new
- * ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
- * gst_caps_append(gst_caps_new ("sink_int", "audio/x-raw-int",
- * GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
- * gst_caps_new ("sink_float", "audio/x-raw-float",
- * GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)),
- * NULL);
+ * The different audio flags that a format info can have.
+ */
+typedef enum
+{
+ GST_AUDIO_FORMAT_FLAG_INTEGER = (1 << 0),
+ GST_AUDIO_FORMAT_FLAG_FLOAT = (1 << 1),
+ GST_AUDIO_FORMAT_FLAG_SIGNED = (1 << 2),
+ GST_AUDIO_FORMAT_FLAG_COMPLEX = (1 << 4)
+} GstAudioFormatFlags;
+
+/**
+ * GstAudioFormatUnpack:
+ * @info: a #GstAudioFormatInfo
+ * @dest: a destination array
+ * @data: pointer to the audio data
+ * @length: the amount of samples to unpack.
*
- * sinkpad = gst_pad_new_from_template(template, "sink");
+ * Unpacks @length samples from the given data of format @info.
+ * The samples will be unpacked into @dest which each channel
+ * interleaved. @dest should at least be big enough to hold @length *
+ * channels * size(unpack_format) bytes.
+ */
+typedef void (*GstAudioFormatUnpack) (GstAudioFormatInfo *info, gpointer dest,
+ const gpointer data, gint length);
+/**
+ * GstAudioFormatPack:
+ * @info: a #GstAudioFormatInfo
+ * @src: a source array
+ * @data: pointer to the destination data
+ * @length: the amount of samples to pack.
*
- * Andy Wingo, 18 August 2001
- * Thomas, 6 September 2002 */
+ * Packs @length samples from @src to the data array in format @info.
+ * The samples from source have each channel interleaved
+ * and will be packed into @data.
+ */
+typedef void (*GstAudioFormatPack) (GstAudioFormatInfo *info, const gpointer src,
+ gpointer data, gint length);
-#define GST_AUDIO_DEF_RATE 44100
+/**
+ * GstAudioFormatInfo:
+ * @format: #GstAudioFormat
+ * @name: string representation of the format
+ * @description: user readable description of the format
+ * @flags: #GstAudioFormatFlags
+ * @endianness: the endianness
+ * @width: amount of bits used for one sample
+ * @depth: amount of valid bits in @width
+ * @silence: @width/8 bytes with 1 silent sample
+ * @unpack_format: the format of the unpacked samples
+ * @unpack_func: function to unpack samples
+ * @pack_func: function to pack samples
+ *
+ * Information for an audio format.
+ */
+struct _GstAudioFormatInfo {
+ GstAudioFormat format;
+ const gchar *name;
+ const gchar *description;
+ GstAudioFormatFlags flags;
+ gint endianness;
+ gint width;
+ gint depth;
+ guint8 silence[8];
-#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \
- gst_props_new (\
- "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- "endianness", GST_PROPS_LIST (\
- GST_PROPS_INT (G_LITTLE_ENDIAN),\
- GST_PROPS_INT (G_BIG_ENDIAN)\
- ),\
- "width", GST_PROPS_LIST (\
- GST_PROPS_INT (8),\
- GST_PROPS_INT (16),\
- GST_PROPS_INT (32)\
- ),\
- "depth", GST_PROPS_INT_RANGE (1, 32),\
- "signed", GST_PROPS_LIST (\
- GST_PROPS_BOOLEAN (TRUE),\
- GST_PROPS_BOOLEAN (FALSE)\
- ),\
- NULL)
-
-#define GST_AUDIO_INT_MONO_PAD_TEMPLATE_PROPS \
- gst_props_new (\
- "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- "channels", GST_PROPS_INT (1),\
- "endianness", GST_PROPS_LIST (\
- GST_PROPS_INT (G_LITTLE_ENDIAN),\
- GST_PROPS_INT (G_BIG_ENDIAN)\
- ),\
- "width", GST_PROPS_LIST (\
- GST_PROPS_INT (8),\
- GST_PROPS_INT (16),\
- GST_PROPS_INT (32)\
- ),\
- "depth", GST_PROPS_INT_RANGE (1, 32),\
- "signed", GST_PROPS_LIST (\
- GST_PROPS_BOOLEAN (TRUE),\
- GST_PROPS_BOOLEAN (FALSE)\
- ),\
- NULL)
-
-#define GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS \
- gst_props_new (\
- "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- "channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- "endianness", GST_PROPS_LIST (\
- GST_PROPS_INT (G_LITTLE_ENDIAN),\
- GST_PROPS_INT (G_BIG_ENDIAN)\
- ),\
- "width", GST_PROPS_LIST (\
- GST_PROPS_INT (32),\
- GST_PROPS_INT (64)\
- ),\
- "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- NULL)
-
-#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_PROPS \
- gst_props_new (\
- "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- "channels", GST_PROPS_INT (1),\
- "endianness", GST_PROPS_INT (G_BYTE_ORDER),\
- "width", GST_PROPS_INT (32),\
- "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),\
- NULL)
+ GstAudioFormat unpack_format;
+ GstAudioFormatUnpack unpack_func;
+ GstAudioFormatPack pack_func;
+};
-/*
- * this library defines and implements some helper functions for audio
- * handling
+#define GST_AUDIO_FORMAT_INFO_FORMAT(info) ((info)->format)
+#define GST_AUDIO_FORMAT_INFO_NAME(info) ((info)->name)
+#define GST_AUDIO_FORMAT_INFO_FLAGS(info) ((info)->flags)
+
+#define GST_AUDIO_FORMAT_INFO_IS_INTEGER(info) !!((info)->flags & GST_AUDIO_FORMAT_FLAG_INTEGER)
+#define GST_AUDIO_FORMAT_INFO_IS_FLOAT(info) !!((info)->flags & GST_AUDIO_FORMAT_FLAG_FLOAT)
+#define GST_AUDIO_FORMAT_INFO_IS_SIGNED(info) !!((info)->flags & GST_AUDIO_FORMAT_FLAG_SIGNED)
+
+#define GST_AUDIO_FORMAT_INFO_ENDIANNESS(info) ((info)->endianness)
+#define GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN(info) ((info)->endianness == G_LITTLE_ENDIAN)
+#define GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN(info) ((info)->endianness == G_BIG_ENDIAN)
+#define GST_AUDIO_FORMAT_INFO_WIDTH(info) ((info)->width)
+#define GST_AUDIO_FORMAT_INFO_DEPTH(info) ((info)->depth)
+
+
+GstAudioFormat gst_audio_format_build_integer (gboolean sign, gint endianness,
+ gint width, gint depth) G_GNUC_CONST;
+
+GstAudioFormat gst_audio_format_from_string (const gchar *format) G_GNUC_CONST;
+const gchar * gst_audio_format_to_string (GstAudioFormat format) G_GNUC_CONST;
+
+const GstAudioFormatInfo *
+ gst_audio_format_get_info (GstAudioFormat format) G_GNUC_CONST;
+
+void gst_audio_format_fill_silence (const GstAudioFormatInfo *info,
+ gpointer dest, gsize length);
+/**
+ * GstAudioFlags:
+ * @GST_AUDIO_FLAG_NONE: no valid flag
+ * @GST_AUDIO_FLAG_DEFAULT_POSITIONS: position array
+ * contains the default layout for the number of channels.
+ *
+ * Extra audio flags
*/
+typedef enum {
+ GST_AUDIO_FLAG_NONE = 0,
+ GST_AUDIO_FLAG_DEFAULT_POSITIONS = (1 << 0)
+} GstAudioFlags;
+
+/**
+ * GstAudioInfo:
+ * @finfo: the format info of the audio
+ * @flags: additional audio flags
+ * @rate: the audio sample rate
+ * @channels: the number of channels
+ * @bpf: the number of bytes for one frame, this is the size of one
+ * sample * @channels
+ * @positions: the positions for each channel
+ *
+ * Information describing audio properties. This information can be filled
+ * in from GstCaps with gst_audio_info_from_caps().
+ *
+ * Use the provided macros to access the info in this structure.
+ */
+struct _GstAudioInfo {
+ const GstAudioFormatInfo *finfo;
+ GstAudioFlags flags;
+ gint rate;
+ gint channels;
+ gint bpf;
+ GstAudioChannelPosition position[64];
+};
+
+#define GST_AUDIO_INFO_FORMAT(i) (GST_AUDIO_FORMAT_INFO_FORMAT((i)->finfo))
+#define GST_AUDIO_INFO_NAME(i) (GST_AUDIO_FORMAT_INFO_NAME((i)->finfo))
+#define GST_AUDIO_INFO_WIDTH(i) (GST_AUDIO_FORMAT_INFO_WIDTH((i)->finfo))
+#define GST_AUDIO_INFO_DEPTH(i) (GST_AUDIO_FORMAT_INFO_DEPTH((i)->finfo))
+#define GST_AUDIO_INFO_BPS(info) (GST_AUDIO_INFO_DEPTH(info) >> 3)
+
+#define GST_AUDIO_INFO_IS_INTEGER(i) (GST_AUDIO_FORMAT_INFO_IS_INTEGER((i)->finfo))
+#define GST_AUDIO_INFO_IS_FLOAT(i) (GST_AUDIO_FORMAT_INFO_IS_FLOAT((i)->finfo))
+#define GST_AUDIO_INFO_IS_SIGNED(i) (GST_AUDIO_FORMAT_INFO_IS_SIGNED((i)->finfo))
-/* get byte size of audio frame (based on caps of pad */
-int gst_audio_frame_byte_size (GstPad* pad);
+#define GST_AUDIO_INFO_ENDIANNESS(i) (GST_AUDIO_FORMAT_INFO_ENDIANNES((i)->finfo))
+#define GST_AUDIO_INFO_IS_LITTLE_ENDIAN(i) (GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN((i)->finfo))
+#define GST_AUDIO_INFO_IS_BIG_ENDIAN(i) (GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN((i)->finfo))
-/* get length in frames of buffer */
-long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
+#define GST_AUDIO_INFO_FLAGS(info) ((info)->flags)
+#define GST_AUDIO_INFO_HAS_DEFAULT_POSITIONS(info) ((info)->flags & GST_AUDIO_FLAG_DEFAULT_POSITIONS)
-/* get frame rate based on caps */
-long gst_audio_frame_rate (GstPad *pad);
+#define GST_AUDIO_INFO_RATE(info) ((info)->rate)
+#define GST_AUDIO_INFO_CHANNELS(info) ((info)->channels)
+#define GST_AUDIO_INFO_BPF(info) ((info)->bpf)
+#define GST_AUDIO_INFO_POSITION(info,c) ((info)->position[c])
-/* calculate length in seconds of audio buffer buf based on caps of pad */
-double gst_audio_length (GstPad* pad, GstBuffer* buf);
+void gst_audio_info_init (GstAudioInfo *info);
+void gst_audio_info_set_format (GstAudioInfo *info, GstAudioFormat format,
+ gint rate, gint channels);
-/* calculate highest possible sample value based on capabilities of pad */
-long gst_audio_highest_sample_value (GstPad* pad);
+gboolean gst_audio_info_from_caps (GstAudioInfo *info, const GstCaps *caps);
+GstCaps * gst_audio_info_to_caps (GstAudioInfo *info);
+
+gboolean gst_audio_info_convert (GstAudioInfo * info,
+ GstFormat src_fmt, gint64 src_val,
+ GstFormat dest_fmt, gint64 * dest_val);
+
+
+
+#define GST_AUDIO_RATE_RANGE "(int) [ 1, max ]"
+#define GST_AUDIO_CHANNELS_RANGE "(int) [ 1, max ]"
+
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+# define GST_AUDIO_NE(s) G_STRINGIFY(s)"LE"
+# define GST_AUDIO_OE(s) G_STRINGIFY(s)"BE"
+#else
+# define GST_AUDIO_NE(s) G_STRINGIFY(s)"BE"
+# define GST_AUDIO_OE(s) G_STRINGIFY(s)"LE"
+#endif
+
+#define GST_AUDIO_FORMATS_ALL " { S8, U8, " \
+ "S16LE, S16BE, U16LE, U16BE, " \
+ "S24_32LE, S24_32BE, U24_32LE, U24_32BE, " \
+ "S32LE, S32BE, U32LE, U32BE, " \
+ "S24LE, S24BE, U24LE, U24BE, " \
+ "S20LE, S20BE, U20LE, U20BE, " \
+ "S18LE, S18BE, U18LE, U18BE, " \
+ "F32LE, F32BE, F64LE, F64BE }"
+
+/**
+ * GST_AUDIO_CAPS_MAKE:
+ * @format: string format that describes the pixel layout, as string
+ * (e.g. "S16LE", "S8", etc.)
+ *
+ * Generic caps string for audio, for use in pad templates.
+ */
+#define GST_AUDIO_CAPS_MAKE(format) \
+ "audio/x-raw, " \
+ "format = (string) " format ", " \
+ "rate = " GST_AUDIO_RATE_RANGE ", " \
+ "channels = " GST_AUDIO_CHANNELS_RANGE
+
+/**
+ * GST_AUDIO_DEF_RATE:
+ *
+ * Standard sampling rate used in consumer audio.
+ */
+#define GST_AUDIO_DEF_RATE 44100
+/**
+ * GST_AUDIO_DEF_CHANNELS:
+ *
+ * Standard number of channels used in consumer audio.
+ */
+#define GST_AUDIO_DEF_CHANNELS 2
+/**
+ * GST_AUDIO_DEF_FORMAT:
+ *
+ * Standard format used in consumer audio.
+ */
+#define GST_AUDIO_DEF_FORMAT "S16LE"
+
+/* conversion macros */
+/**
+ * GST_FRAMES_TO_CLOCK_TIME:
+ * @frames: sample frames
+ * @rate: sampling rate
+ *
+ * Calculate clocktime from sample @frames and @rate.
+ */
+#define GST_FRAMES_TO_CLOCK_TIME(frames, rate) \
+ ((GstClockTime) gst_util_uint64_scale_round (frames, GST_SECOND, rate))
+
+/**
+ * GST_CLOCK_TIME_TO_FRAMES:
+ * @clocktime: clock time
+ * @rate: sampling rate
+ *
+ * Calculate frames from @clocktime and sample @rate.
+ */
+#define GST_CLOCK_TIME_TO_FRAMES(clocktime, rate) \
+ gst_util_uint64_scale_round (clocktime, rate, GST_SECOND)
+
+/*
+ * this library defines and implements some helper functions for audio
+ * handling
+ */
-/* check if the buffer size is a whole multiple of the frame size */
-gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);
+GstBuffer * gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment,
+ gint rate, gint bpf);
G_END_DECLS
+#endif /* __GST_AUDIO_AUDIO_H__ */
projects / platform / upstream / gstreamer.git / blobdiff