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 .

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.

gst_audio_info_init ()

void
gst_audio_info_init (GstAudioInfo *info);

Initialize info with default values.

gst_audio_info_from_caps ()

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

Parse caps and update info .

gst_audio_info_to_caps ()

GstCaps *
gst_audio_info_to_caps (const GstAudioInfo *info);

Convert the values of info into a GstCaps.

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.

gst_audio_format_get_info ()

const GstAudioFormatInfo *
gst_audio_format_get_info (GstAudioFormat format);

Get the GstAudioFormatInfo for format

gst_audio_info_copy ()

GstAudioInfo *
gst_audio_info_copy (const GstAudioInfo *info);

Copy a GstAudioInfo structure.

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().

gst_audio_info_new ()

GstAudioInfo *
gst_audio_info_new (void);

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

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.

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

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.

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 .

gst_audio_format_from_string ()

GstAudioFormat
gst_audio_format_from_string (const gchar *format);

Convert the format string to its GstAudioFormat.

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 .

GST_CLOCK_TIME_TO_FRAMES()

#define             GST_CLOCK_TIME_TO_FRAMES(clocktime, rate)

Calculate frames from clocktime and sample 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.

GST_AUDIO_OE()

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

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

GST_AUDIO_CAPS_MAKE()

#define             GST_AUDIO_CAPS_MAKE(format)

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

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.

gst_audio_resampler_free ()

void
gst_audio_resampler_free (GstAudioResampler *resampler);

Free a previously allocated GstAudioResampler resampler .

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 .

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.

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 .

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.

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 .

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.

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.

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.

gst_audio_stream_align_new ()

GstAudioStreamAlign *
gst_audio_stream_align_new (gint rate,
                            GstClockTime alignment_threshold,
                            GstClockTime discont_wait);

Allocate a new GstAudioStreamAlign with the given configuration. All processing happens according to sample rate rate , until gst_audio_discont_wait_set_rate() is called with a new rate . A negative rate can be used for reverse playback.

alignment_threshold gives the tolerance in nanoseconds after which a timestamp difference is considered a discontinuity. Once detected, discont_wait nanoseconds have to pass without going below the threshold again until the output buffer is marked as a discontinuity. These can later be re-configured with gst_audio_stream_align_set_alignment_threshold() and gst_audio_stream_align_set_discont_wait().

Since: 1.14

gst_audio_stream_align_copy ()

GstAudioStreamAlign *
gst_audio_stream_align_copy (const GstAudioStreamAlign *align);

Copy a GstAudioStreamAlign structure.

Since: 1.14

gst_audio_stream_align_free ()

void
gst_audio_stream_align_free (GstAudioStreamAlign *align);

Free a GstAudioStreamAlign structure previously allocated with gst_audio_stream_align_new() or gst_audio_stream_align_copy().

Since: 1.14

gst_audio_stream_align_mark_discont ()

void
gst_audio_stream_align_mark_discont (GstAudioStreamAlign *align);

Marks the next buffer as discontinuous and resets timestamp tracking.

Since: 1.14

gst_audio_stream_align_process ()

gboolean
gst_audio_stream_align_process (GstAudioStreamAlign *align,
                                gboolean discont,
                                GstClockTime timestamp,
                                guint n_samples,
                                GstClockTime *out_timestamp,
                                GstClockTime *out_duration,
                                guint64 *out_sample_position);

Processes data with timestamp and n_samples , and returns the output timestamp, duration and sample position together with a boolean to signal whether a discontinuity was detected or not. All non-discontinuous data will have perfect timestamps and durations.

A discontinuity is detected once the difference between the actual timestamp and the timestamp calculated from the sample count since the last discontinuity differs by more than the alignment threshold for a duration longer than discont wait.

Note: In reverse playback, every buffer is considered discontinuous in the context of buffer flags because the last sample of the previous buffer is discontinuous with the first sample of the current one. However for this function they are only considered discontinuous in reverse playback if the first sample of the previous buffer is discontinuous with the last sample of the current one.

Since: 1.14

gst_audio_stream_align_get_samples_since_discont ()

guint64
gst_audio_stream_align_get_samples_since_discont
                               (GstAudioStreamAlign *align);

Returns the number of samples that were processed since the last discontinuity was detected.

Since: 1.14

gst_audio_stream_align_get_timestamp_at_discont ()

GstClockTime
gst_audio_stream_align_get_timestamp_at_discont
                               (GstAudioStreamAlign *align);

Timestamp that was passed when a discontinuity was detected, i.e. the first timestamp after the discontinuity.

Since: 1.14

gst_audio_stream_align_get_alignment_threshold ()

GstClockTime
gst_audio_stream_align_get_alignment_threshold
                               (GstAudioStreamAlign *align);

gst_audio_stream_align_set_alignment_threshold ()

void
gst_audio_stream_align_set_alignment_threshold
                               (GstAudioStreamAlign *align,
                                GstClockTime alignment_threshold);

gst_audio_stream_align_get_discont_wait ()

GstClockTime
gst_audio_stream_align_get_discont_wait
                               (GstAudioStreamAlign *align);

gst_audio_stream_align_set_discont_wait ()

void
gst_audio_stream_align_set_discont_wait
                               (GstAudioStreamAlign *align,
                                GstClockTime discont_wait);

gst_audio_stream_align_get_rate ()

gint
gst_audio_stream_align_get_rate (GstAudioStreamAlign *align);

gst_audio_stream_align_set_rate ()

void
gst_audio_stream_align_set_rate (GstAudioStreamAlign *align,
                                 gint rate);

enum GstAudioFlags

Extra audio flags

enum GstAudioFormat

Enum value describing the most common audio formats.

enum GstAudioFormatFlags

The different audio flags that a format info can have.

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.

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.

enum GstAudioLayout

Layout of the audio samples for the different channels.

enum GstAudioPackFlags

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

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.

enum GstAudioResamplerFilterMode

Select for the filter tables should be set up.

enum GstAudioResamplerFlags

Different resampler flags.

enum GstAudioResamplerMethod

Different subsampling and upsampling methods

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

GstAudioStreamAlign

typedef struct _GstAudioStreamAlign GstAudioStreamAlign;

The opaque GstAudioStreamAlign data structure.

Since: 1.14