GstAudio

GstAudio — Support library for audio elements

Functions

void (*GstAudioFormatPack) ()
void (*GstAudioFormatUnpack) ()
void gst_audio_info_init ()
gboolean gst_audio_info_from_caps ()
GstCaps * gst_audio_info_to_caps ()
gboolean gst_audio_info_convert ()
const GstAudioFormatInfo * gst_audio_format_get_info ()
GstAudioInfo * gst_audio_info_copy ()
void gst_audio_info_free ()
GstAudioInfo * gst_audio_info_new ()
void gst_audio_info_set_format ()
gboolean gst_audio_info_is_equal ()
GstAudioFormat gst_audio_format_build_integer ()
void gst_audio_format_fill_silence ()
GstAudioFormat gst_audio_format_from_string ()
const gchar * gst_audio_format_to_string ()
#define GST_AUDIO_FORMAT_INFO_DEPTH()
#define GST_AUDIO_FORMAT_INFO_ENDIANNESS()
#define GST_AUDIO_FORMAT_INFO_FLAGS()
#define GST_AUDIO_FORMAT_INFO_FORMAT()
#define GST_AUDIO_FORMAT_INFO_IS_FLOAT()
#define GST_AUDIO_FORMAT_INFO_IS_INTEGER()
#define GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN()
#define GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN()
#define GST_AUDIO_FORMAT_INFO_IS_SIGNED()
#define GST_AUDIO_FORMAT_INFO_NAME()
#define GST_AUDIO_FORMAT_INFO_WIDTH()
#define GST_AUDIO_INFO_BPF()
#define GST_AUDIO_INFO_BPS()
#define GST_AUDIO_INFO_CHANNELS()
#define GST_AUDIO_INFO_DEPTH()
#define GST_AUDIO_INFO_FLAGS()
#define GST_AUDIO_INFO_FORMAT()
#define GST_AUDIO_INFO_NAME()
#define GST_AUDIO_INFO_POSITION()
#define GST_AUDIO_INFO_RATE()
#define GST_AUDIO_INFO_WIDTH()
#define GST_AUDIO_INFO_ENDIANNESS()
#define GST_AUDIO_INFO_IS_BIG_ENDIAN()
#define GST_AUDIO_INFO_IS_FLOAT()
#define GST_AUDIO_INFO_IS_INTEGER()
#define GST_AUDIO_INFO_IS_LITTLE_ENDIAN()
#define GST_AUDIO_INFO_IS_SIGNED()
#define GST_AUDIO_INFO_IS_UNPOSITIONED()
#define GST_AUDIO_INFO_IS_VALID()
#define GST_AUDIO_INFO_LAYOUT()
#define GST_FRAMES_TO_CLOCK_TIME()
#define GST_CLOCK_TIME_TO_FRAMES()
#define GST_AUDIO_NE()
#define GST_AUDIO_OE()
#define GST_AUDIO_CAPS_MAKE()
GstBuffer * gst_audio_buffer_clip ()
void gst_audio_resampler_free ()
gsize gst_audio_resampler_get_in_frames ()
gsize gst_audio_resampler_get_max_latency ()
gsize gst_audio_resampler_get_out_frames ()
GstAudioResampler * gst_audio_resampler_new ()
void gst_audio_resampler_options_set_quality ()
void gst_audio_resampler_resample ()
void gst_audio_resampler_reset ()
gboolean gst_audio_resampler_update ()

Types and Values

Includes

#include <gst/audio/audio.h>

Description

This library contains some helper functions for audio elements.

Functions

GstAudioFormatPack ()

void
(*GstAudioFormatPack) (const GstAudioFormatInfo *info,
                       GstAudioPackFlags flags,
                       const gpointer src,
                       gpointer data,
                       gint length);

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 .

Parameters

info

a GstAudioFormatInfo

 

src

a source array.

[array][element-type guint8]

data

pointer to the destination data.

[array][element-type guint8]

length

the amount of samples to pack.

 

GstAudioFormatUnpack ()

void
(*GstAudioFormatUnpack) (const GstAudioFormatInfo *info,
                         GstAudioPackFlags flags,
                         gpointer dest,
                         const gpointer data,
                         gint length);

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.

Parameters

info

a GstAudioFormatInfo

 

dest

a destination array.

[array][element-type guint8]

data

pointer to the audio data.

[array][element-type guint8]

length

the amount of samples to unpack.

 

gst_audio_info_init ()

void
gst_audio_info_init (GstAudioInfo *info);

Initialize info with default values.

Parameters

info

a GstAudioInfo

 

gst_audio_info_from_caps ()

gboolean
gst_audio_info_from_caps (GstAudioInfo *info,
                          const GstCaps *caps);

Parse caps and update info .

Parameters

info

a GstAudioInfo

 

caps

a GstCaps

 

Returns

TRUE if caps could be parsed


gst_audio_info_to_caps ()

GstCaps *
gst_audio_info_to_caps (const GstAudioInfo *info);

Convert the values of info into a GstCaps.

Parameters

info

a GstAudioInfo

 

Returns

the new GstCaps containing the info of info .

[transfer full]


gst_audio_info_convert ()

gboolean
gst_audio_info_convert (const GstAudioInfo *info,
                        GstFormat src_fmt,
                        gint64 src_val,
                        GstFormat dest_fmt,
                        gint64 *dest_val);

Converts among various GstFormat types. This function handles GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This function can be used to handle pad queries of the type GST_QUERY_CONVERT.

Parameters

info

a GstAudioInfo

 

src_fmt

GstFormat of the src_val

 

src_val

value to convert

 

dest_fmt

GstFormat of the dest_val

 

dest_val

pointer to destination value

 

Returns

TRUE if the conversion was successful.


gst_audio_format_get_info ()

const GstAudioFormatInfo *
gst_audio_format_get_info (GstAudioFormat format);

Get the GstAudioFormatInfo for format

Parameters

format

a GstAudioFormat

 

Returns

The GstAudioFormatInfo for format .


gst_audio_info_copy ()

GstAudioInfo *
gst_audio_info_copy (const GstAudioInfo *info);

Copy a GstAudioInfo structure.

Parameters

info

a GstAudioInfo

 

Returns

a new GstAudioInfo. free with gst_audio_info_free.


gst_audio_info_free ()

void
gst_audio_info_free (GstAudioInfo *info);

Free a GstAudioInfo structure previously allocated with gst_audio_info_new() or gst_audio_info_copy().

Parameters

info

a GstAudioInfo

 

gst_audio_info_new ()

GstAudioInfo *
gst_audio_info_new (void);

Allocate a new GstAudioInfo that is also initialized with gst_audio_info_init().

Returns

a new GstAudioInfo. free with gst_audio_info_free().


gst_audio_info_set_format ()

void
gst_audio_info_set_format (GstAudioInfo *info,
                           GstAudioFormat format,
                           gint rate,
                           gint channels,
                           const GstAudioChannelPosition *position);

Set the default info for the audio info of format and rate and channels .

Note: This initializes info first, no values are preserved.

Parameters

info

a GstAudioInfo

 

format

the format

 

rate

the samplerate

 

channels

the number of channels

 

position

the channel positions

 

gst_audio_info_is_equal ()

gboolean
gst_audio_info_is_equal (const GstAudioInfo *info,
                         const GstAudioInfo *other);

Compares two GstAudioInfo and returns whether they are equal or not

Parameters

info

a GstAudioInfo

 

other

a GstAudioInfo

 

Returns

TRUE if info and other are equal, else FALSE.

Since: 1.2


gst_audio_format_build_integer ()

GstAudioFormat
gst_audio_format_build_integer (gboolean sign,
                                gint endianness,
                                gint width,
                                gint depth);

Construct a GstAudioFormat with given parameters.

Parameters

sign

signed or unsigned format

 

endianness

G_LITTLE_ENDIAN or G_BIG_ENDIAN

 

width

amount of bits used per sample

 

depth

amount of used bits in width

 

Returns

a GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format exists with the given parameters.


gst_audio_format_fill_silence ()

void
gst_audio_format_fill_silence (const GstAudioFormatInfo *info,
                               gpointer dest,
                               gsize length);

Fill length bytes in dest with silence samples for info .

Parameters

info

a GstAudioFormatInfo

 

dest

a destination to fill.

[array length=length][element-type guint8]

length

the length to fill

 

gst_audio_format_from_string ()

GstAudioFormat
gst_audio_format_from_string (const gchar *format);

Convert the format string to its GstAudioFormat.

Parameters

format

a format string

 

Returns

the GstAudioFormat for format or GST_AUDIO_FORMAT_UNKNOWN when the string is not a known format.


gst_audio_format_to_string ()

const gchar *
gst_audio_format_to_string (GstAudioFormat format);

GST_AUDIO_FORMAT_INFO_DEPTH()

#define GST_AUDIO_FORMAT_INFO_DEPTH(info)            ((info)->depth)

GST_AUDIO_FORMAT_INFO_ENDIANNESS()

#define GST_AUDIO_FORMAT_INFO_ENDIANNESS(info)       ((info)->endianness)

GST_AUDIO_FORMAT_INFO_FLAGS()

#define GST_AUDIO_FORMAT_INFO_FLAGS(info)            ((info)->flags)

GST_AUDIO_FORMAT_INFO_FORMAT()

#define GST_AUDIO_FORMAT_INFO_FORMAT(info)           ((info)->format)

GST_AUDIO_FORMAT_INFO_IS_FLOAT()

#define GST_AUDIO_FORMAT_INFO_IS_FLOAT(info)         !!((info)->flags & GST_AUDIO_FORMAT_FLAG_FLOAT)

GST_AUDIO_FORMAT_INFO_IS_INTEGER()

#define GST_AUDIO_FORMAT_INFO_IS_INTEGER(info)       !!((info)->flags & GST_AUDIO_FORMAT_FLAG_INTEGER)

GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN()

#define GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN(info)    ((info)->endianness == G_BIG_ENDIAN)

GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN()

#define GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN(info) ((info)->endianness == G_LITTLE_ENDIAN)

GST_AUDIO_FORMAT_INFO_IS_SIGNED()

#define GST_AUDIO_FORMAT_INFO_IS_SIGNED(info)        !!((info)->flags & GST_AUDIO_FORMAT_FLAG_SIGNED)

GST_AUDIO_FORMAT_INFO_NAME()

#define GST_AUDIO_FORMAT_INFO_NAME(info)             ((info)->name)

GST_AUDIO_FORMAT_INFO_WIDTH()

#define GST_AUDIO_FORMAT_INFO_WIDTH(info)            ((info)->width)

GST_AUDIO_INFO_BPF()

#define GST_AUDIO_INFO_BPF(info)             ((info)->bpf)

GST_AUDIO_INFO_BPS()

#define GST_AUDIO_INFO_BPS(info)             (GST_AUDIO_INFO_DEPTH(info) >> 3)

GST_AUDIO_INFO_CHANNELS()

#define GST_AUDIO_INFO_CHANNELS(info)        ((info)->channels)

GST_AUDIO_INFO_DEPTH()

#define GST_AUDIO_INFO_DEPTH(i)              (GST_AUDIO_FORMAT_INFO_DEPTH((i)->finfo))

GST_AUDIO_INFO_FLAGS()

#define GST_AUDIO_INFO_FLAGS(info)           ((info)->flags)

GST_AUDIO_INFO_FORMAT()

#define GST_AUDIO_INFO_FORMAT(i)             (GST_AUDIO_FORMAT_INFO_FORMAT((i)->finfo))

GST_AUDIO_INFO_NAME()

#define GST_AUDIO_INFO_NAME(i)               (GST_AUDIO_FORMAT_INFO_NAME((i)->finfo))

GST_AUDIO_INFO_POSITION()

#define GST_AUDIO_INFO_POSITION(info,c)      ((info)->position[c])

GST_AUDIO_INFO_RATE()

#define GST_AUDIO_INFO_RATE(info)            ((info)->rate)

GST_AUDIO_INFO_WIDTH()

#define GST_AUDIO_INFO_WIDTH(i)              (GST_AUDIO_FORMAT_INFO_WIDTH((i)->finfo))

GST_AUDIO_INFO_ENDIANNESS()

#define GST_AUDIO_INFO_ENDIANNESS(i)         (GST_AUDIO_FORMAT_INFO_ENDIANNESS((i)->finfo))

GST_AUDIO_INFO_IS_BIG_ENDIAN()

#define GST_AUDIO_INFO_IS_BIG_ENDIAN(i)      (GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN((i)->finfo))

GST_AUDIO_INFO_IS_FLOAT()

#define GST_AUDIO_INFO_IS_FLOAT(i)           (GST_AUDIO_FORMAT_INFO_IS_FLOAT((i)->finfo))

GST_AUDIO_INFO_IS_INTEGER()

#define GST_AUDIO_INFO_IS_INTEGER(i)         (GST_AUDIO_FORMAT_INFO_IS_INTEGER((i)->finfo))

GST_AUDIO_INFO_IS_LITTLE_ENDIAN()

#define GST_AUDIO_INFO_IS_LITTLE_ENDIAN(i)   (GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN((i)->finfo))

GST_AUDIO_INFO_IS_SIGNED()

#define GST_AUDIO_INFO_IS_SIGNED(i)          (GST_AUDIO_FORMAT_INFO_IS_SIGNED((i)->finfo))

GST_AUDIO_INFO_IS_UNPOSITIONED()

#define GST_AUDIO_INFO_IS_UNPOSITIONED(info) ((info)->flags & GST_AUDIO_FLAG_UNPOSITIONED)

GST_AUDIO_INFO_IS_VALID()

#define GST_AUDIO_INFO_IS_VALID(i)           ((i)->finfo != NULL && (i)->rate > 0 && (i)->channels > 0 && (i)->bpf > 0)

GST_AUDIO_INFO_LAYOUT()

#define GST_AUDIO_INFO_LAYOUT(info)          ((info)->layout)

GST_FRAMES_TO_CLOCK_TIME()

#define             GST_FRAMES_TO_CLOCK_TIME(frames, rate)

Calculate clocktime from sample frames and rate .

Parameters

frames

sample frames

 

rate

sampling rate

 

GST_CLOCK_TIME_TO_FRAMES()

#define             GST_CLOCK_TIME_TO_FRAMES(clocktime, rate)

Calculate frames from clocktime and sample rate .

Parameters

clocktime

clock time

 

rate

sampling rate

 

GST_AUDIO_NE()

# define GST_AUDIO_NE(s) G_STRINGIFY(s)"LE"

Turns audio format string s into the format string for native endianness.

Parameters

s

format string without endianness marker

 

GST_AUDIO_OE()

# define GST_AUDIO_OE(s) G_STRINGIFY(s)"BE"

Turns audio format string s into the format string for other endianness.

Parameters

s

format string without endianness marker

 

GST_AUDIO_CAPS_MAKE()

#define             GST_AUDIO_CAPS_MAKE(format)

Generic caps string for audio, for use in pad templates.

Parameters

format

string format that describes the sample layout, as string (e.g. "S16LE", "S8", etc.)

 

gst_audio_buffer_clip ()

GstBuffer *
gst_audio_buffer_clip (GstBuffer *buffer,
                       const GstSegment *segment,
                       gint rate,
                       gint bpf);

Clip the buffer to the given GstSegment.

After calling this function the caller does not own a reference to buffer anymore.

Parameters

buffer

The buffer to clip.

[transfer full]

segment

Segment in GST_FORMAT_TIME or GST_FORMAT_DEFAULT to which the buffer should be clipped.

 

rate

sample rate.

 

bpf

size of one audio frame in bytes. This is the size of one sample * number of channels.

 

Returns

NULL if the buffer is completely outside the configured segment, otherwise the clipped buffer is returned.

If the buffer has no timestamp, it is assumed to be inside the segment and is not clipped.

[transfer full]


gst_audio_resampler_free ()

void
gst_audio_resampler_free (GstAudioResampler *resampler);

Free a previously allocated GstAudioResampler resampler .

Parameters

resampler

a GstAudioResampler

 

Since: 1.6


gst_audio_resampler_get_in_frames ()

gsize
gst_audio_resampler_get_in_frames (GstAudioResampler *resampler,
                                   gsize out_frames);

Get the number of input frames that would currently be needed to produce out_frames from resampler .

Parameters

resampler

a GstAudioResampler

 

out_frames

number of input frames

 

Returns

The number of input frames needed for producing out_frames of data from resampler .


gst_audio_resampler_get_max_latency ()

gsize
gst_audio_resampler_get_max_latency (GstAudioResampler *resampler);

Get the maximum number of input samples that the resampler would need before producing output.

Parameters

resampler

a GstAudioResampler

 

Returns

the latency of resampler as expressed in the number of frames.


gst_audio_resampler_get_out_frames ()

gsize
gst_audio_resampler_get_out_frames (GstAudioResampler *resampler,
                                    gsize in_frames);

Get the number of output frames that would be currently available when in_frames are given to resampler .

Parameters

resampler

a GstAudioResampler

 

in_frames

number of input frames

 

Returns

The number of frames that would be availabe after giving in_frames as input to resampler .


gst_audio_resampler_new ()

GstAudioResampler *
gst_audio_resampler_new (GstAudioResamplerMethod method,
                         GstAudioResamplerFlags flags,
                         GstAudioFormat format,
                         gint channels,
                         gint in_rate,
                         gint out_rate,
                         GstStructure *options);

Make a new resampler.

Parameters

method

a GstAudioResamplerMethod

 

flags

GstAudioResamplerFlags

 

in_rate

input rate

 

out_rate

output rate

 

options

extra options

 

Returns

TRUE on success.

[skip][transfer full]


gst_audio_resampler_options_set_quality ()

void
gst_audio_resampler_options_set_quality
                               (GstAudioResamplerMethod method,
                                guint quality,
                                gint in_rate,
                                gint out_rate,
                                GstStructure *options);

Set the parameters for resampling from in_rate to out_rate using method for quality in options .

Parameters

method

a GstAudioResamplerMethod

 

quality

the quality

 

in_rate

the input rate

 

out_rate

the output rate

 

options

a GstStructure

 

gst_audio_resampler_resample ()

void
gst_audio_resampler_resample (GstAudioResampler *resampler,
                              gpointer in[],
                              gsize in_frames,
                              gpointer out[],
                              gsize out_frames);

Perform resampling on in_frames frames in in and write out_frames to out .

In case the samples are interleaved, in and out must point to an array with a single element pointing to a block of interleaved samples.

If non-interleaved samples are used, in and out must point to an array with pointers to memory blocks, one for each channel.

in may be NULL, in which case in_frames of silence samples are pushed into the resampler.

This function always produces out_frames of output and consumes in_frames of input. Use gst_audio_resampler_get_out_frames() and gst_audio_resampler_get_in_frames() to make sure in_frames and out_frames are matching and in and out point to enough memory.

Parameters

resampler

a GstAudioResampler

 

in

input samples

 

in_frames

number of input frames

 

out

output samples

 

out_frames

number of output frames

 

gst_audio_resampler_reset ()

void
gst_audio_resampler_reset (GstAudioResampler *resampler);

Reset resampler to the state it was when it was first created, discarding all sample history.

Parameters

resampler

a GstAudioResampler

 

gst_audio_resampler_update ()

gboolean
gst_audio_resampler_update (GstAudioResampler *resampler,
                            gint in_rate,
                            gint out_rate,
                            GstStructure *options);

Update the resampler parameters for resampler . This function should not be called concurrently with any other function on resampler .

When in_rate or out_rate is 0, its value is unchanged.

When options is NULL, the previously configured options are reused.

Parameters

resampler

a GstAudioResampler

 

in_rate

new input rate

 

out_rate

new output rate

 

options

new options or NULL

 

Returns

TRUE if the new parameters could be set

Types and Values

enum GstAudioFlags

Extra audio flags

Members

GST_AUDIO_FLAG_NONE

no valid flag

 

GST_AUDIO_FLAG_UNPOSITIONED

the position array explicitly contains unpositioned channels.

 

enum GstAudioFormat

Enum value describing the most common audio formats.

Members

GST_AUDIO_FORMAT_UNKNOWN

unknown or unset audio format

 

GST_AUDIO_FORMAT_ENCODED

encoded 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

 

enum GstAudioFormatFlags

The different audio flags that a format info can have.

Members

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

 

GST_AUDIO_FORMAT_FLAG_UNPACK

the format can be used in GstAudioFormatUnpack and GstAudioFormatPack functions

 

struct GstAudioFormatInfo

struct GstAudioFormatInfo {
  GstAudioFormat format;
  const gchar *name;
  const gchar *description;
  GstAudioFormatFlags flags;
  gint endianness;
  gint width;
  gint depth;
  guint8 silence[8];

  GstAudioFormat unpack_format;
  GstAudioFormatUnpack unpack_func;
  GstAudioFormatPack pack_func;
};

Information for an audio format.

Members

GstAudioFormat format;

GstAudioFormat

 

const gchar *name;

string representation of the format

 

const gchar *description;

user readable description of the format

 

GstAudioFormatFlags flags;

GstAudioFormatFlags

 

gint endianness;

the endianness

 

gint width;

amount of bits used for one sample

 

gint depth;

amount of valid bits in width

 

guint8 silence[8];

width /8 bytes with 1 silent sample

 

GstAudioFormat unpack_format;

the format of the unpacked samples

 

GstAudioFormatUnpack unpack_func;

function to unpack samples

 

GstAudioFormatPack pack_func;

function to pack samples

 

struct GstAudioInfo

struct GstAudioInfo {
  const GstAudioFormatInfo *finfo;
  GstAudioFlags             flags;
  GstAudioLayout            layout;
  gint                      rate;
  gint                      channels;
  gint                      bpf;
  GstAudioChannelPosition   position[64];
};

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.

Members

const GstAudioFormatInfo *finfo;

the format info of the audio

 

GstAudioFlags flags;

additional audio flags

 

GstAudioLayout layout;

audio layout

 

gint rate;

the audio sample rate

 

gint channels;

the number of channels

 

gint bpf;

the number of bytes for one frame, this is the size of one sample * channels

 

GstAudioChannelPosition position[64];

the positions for each channel

 

enum GstAudioLayout

Layout of the audio samples for the different channels.

Members

GST_AUDIO_LAYOUT_INTERLEAVED

interleaved audio

 

GST_AUDIO_LAYOUT_NON_INTERLEAVED

non-interleaved audio

 

enum GstAudioPackFlags

The different flags that can be used when packing and unpacking.

Members

GST_AUDIO_PACK_FLAG_NONE

No flag

 

GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE

When the source has a smaller depth than the target format, set the least significant bits of the target to 0. This is likely sightly faster but less accurate. When this flag is not specified, the most significant bits of the source are duplicated in the least significant bits of the destination.

 

GST_META_TAG_AUDIO_STR

#define GST_META_TAG_AUDIO_STR "audio"

This metadata is relevant for audio streams.

Since: 1.2


GST_META_TAG_AUDIO_CHANNELS_STR

#define GST_META_TAG_AUDIO_CHANNELS_STR "channels"

This metadata stays relevant as long as channels are unchanged.

Since: 1.2


GST_META_TAG_AUDIO_RATE_STR

#define GST_META_TAG_AUDIO_RATE_STR "rate"

This metadata stays relevant as long as sample rate is unchanged.

Since: 1.8


GST_AUDIO_RATE_RANGE

#define GST_AUDIO_RATE_RANGE "(int) [ 1, max ]"

Maximum range of allowed sample rates, for use in template caps strings.


GST_AUDIO_CHANNELS_RANGE

#define GST_AUDIO_CHANNELS_RANGE "(int) [ 1, max ]"

Maximum range of allowed channels, for use in template caps strings.


GST_AUDIO_FORMATS_ALL

#define             GST_AUDIO_FORMATS_ALL

List of all audio formats, for use in template caps strings.


GST_AUDIO_DEF_RATE

#define GST_AUDIO_DEF_RATE 44100

Standard sampling rate used in consumer audio.


GST_AUDIO_DEF_CHANNELS

#define GST_AUDIO_DEF_CHANNELS 2

Standard number of channels used in consumer audio.


GST_AUDIO_DEF_FORMAT

#define GST_AUDIO_DEF_FORMAT "S16LE"

Standard format used in consumer audio.


GstAudioResampler

typedef struct {
  GstAudioResamplerMethod method;
  GstAudioResamplerFlags flags;
  GstAudioFormat format;
  GstStructure *options;
  gint format_index;
  gint channels;
  gint in_rate;
  gint out_rate;

  gint bps;
  gint ostride;

  GstAudioResamplerFilterMode filter_mode;
  guint filter_threshold;
  GstAudioResamplerFilterInterpolation filter_interpolation;

  gdouble cutoff;
  gdouble kaiser_beta;
  /* for cubic */
  gdouble b, c;

  /* temp taps */
  gpointer tmp_taps;

  /* oversampled main filter table */
  gint oversample;
  gint n_taps;
  gpointer taps;
  gpointer taps_mem;
  gsize taps_stride;
  gint n_phases;
  gint alloc_taps;
  gint alloc_phases;

  /* cached taps */
  gpointer *cached_phases;
  gpointer cached_taps;
  gpointer cached_taps_mem;
  gsize cached_taps_stride;

  ConvertTapsFunc convert_taps;
  InterpolateFunc interpolate;
  DeinterleaveFunc deinterleave;
  ResampleFunc resample;

  gint blocks;
  gint inc;
  gint samp_inc;
  gint samp_frac;
  gint samp_index;
  gint samp_phase;
  gint skip;

  gpointer samples;
  gsize samples_len;
  gsize samples_avail;
  gpointer *sbuf;
} GstAudioResampler;

enum GstAudioResamplerFilterInterpolation

The different filter interpolation methods.

Members

GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE

no interpolation

 

GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR

linear interpolation of the filter coeficients.

 

GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC

cubic interpolation of the filter coeficients.

 

enum GstAudioResamplerFilterMode

Select for the filter tables should be set up.

Members

GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED

Use interpolated filter tables. This uses less memory but more CPU and is slightly less accurate but it allows for more efficient variable rate resampling with gst_audio_resampler_update().

 

GST_AUDIO_RESAMPLER_FILTER_MODE_FULL

Use full filter table. This uses more memory but less CPU.

 

GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO

Automatically choose between interpolated and full filter tables.

 

enum GstAudioResamplerFlags

Different resampler flags.

Members

GST_AUDIO_RESAMPLER_FLAG_NONE

no flags

 

GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN

input samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

 

GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT

output samples are non-interleaved. an array of blocks of samples, one for each channel, should be passed to the resample function.

 

GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE

optimize for dynamic updates of the sample rates with gst_audio_resampler_update(). This will select an interpolating filter when GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.

 

enum GstAudioResamplerMethod

Different subsampling and upsampling methods

Members

GST_AUDIO_RESAMPLER_METHOD_NEAREST

Duplicates the samples when upsampling and drops when downsampling

 

GST_AUDIO_RESAMPLER_METHOD_LINEAR

Uses linear interpolation to reconstruct missing samples and averaging to downsample

 

GST_AUDIO_RESAMPLER_METHOD_CUBIC

Uses cubic interpolation

 

GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL

Uses Blackman-Nuttall windowed sinc interpolation

 

GST_AUDIO_RESAMPLER_METHOD_KAISER

Uses Kaiser windowed sinc interpolation

 

Since: 1.6


GST_AUDIO_RESAMPLER_OPT_CUBIC_B

#define GST_AUDIO_RESAMPLER_OPT_CUBIC_B      "GstAudioResampler.cubic-b"

G_TYPE_DOUBLE, B parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 1.0 is the default.

Below are some values of popular filters: B C Hermite 0.0 0.0 Spline 1.0 0.0 Catmull-Rom 0.0 1/2


GST_AUDIO_RESAMPLER_OPT_CUBIC_C

#define GST_AUDIO_RESAMPLER_OPT_CUBIC_C      "GstAudioResampler.cubic-c"

G_TYPE_DOUBLE, C parameter of the cubic filter. Values between 0.0 and 2.0 are accepted. 0.0 is the default.

See GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values


GST_AUDIO_RESAMPLER_OPT_CUTOFF

#define GST_AUDIO_RESAMPLER_OPT_CUTOFF      "GstAudioResampler.cutoff"

G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.


GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION

#define GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION "GstAudioResampler.filter-interpolation"

GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be interpolated. GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.


GST_AUDIO_RESAMPLER_OPT_FILTER_MODE

#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE      "GstAudioResampler.filter-mode"

GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be constructed. GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.


GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD

#define GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD "GstAudioResampler.filter-mode-threshold"

G_TYPE_UINT: the amount of memory to use for full filter tables before switching to interpolated filter tables. 1048576 is the default.


GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE

#define GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE "GstAudioResampler.filter-oversample"

G_TYPE_UINT, oversampling to use when interpolating filters 8 is the default.


GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR

#define GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR "GstAudioResampler.max-phase-error"

G_TYPE_DOUBLE: The maximum allowed phase error when switching sample rates. 0.1 is the default.


GST_AUDIO_RESAMPLER_OPT_N_TAPS

#define GST_AUDIO_RESAMPLER_OPT_N_TAPS      "GstAudioResampler.n-taps"

G_TYPE_INT: the number of taps to use for the filter. 0 is the default and selects the taps automatically.


GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION

#define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"

G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation after the stopband for the kaiser window. 85 dB is the default.


GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH

#define GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH "GstAudioResampler.transition-bandwidth"

G_TYPE_DOUBLE, transition bandwidth. The width of the transition band for the kaiser window. 0.087 is the default.


GST_AUDIO_RESAMPLER_QUALITY_DEFAULT

#define GST_AUDIO_RESAMPLER_QUALITY_DEFAULT 4

GST_AUDIO_RESAMPLER_QUALITY_MAX

#define GST_AUDIO_RESAMPLER_QUALITY_MAX 10

GST_AUDIO_RESAMPLER_QUALITY_MIN

#define GST_AUDIO_RESAMPLER_QUALITY_MIN 0