manpagez: man pages & more
html files: gst-plugins-base-libs-1.0
Home | html | info | man

GstAudioRingBuffer

GstAudioRingBuffer — Base class for audio ringbuffer implementations

Object Hierarchy

    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstAudioRingBuffer

Includes

#include <gst/audio/gstaudioringbuffer.h>

Description

This object is the base class for audio ringbuffers used by the base audio source and sink classes.

The ringbuffer abstracts a circular buffer of data. One reader and one writer can operate on the data from different threads in a lockfree manner. The base class is sufficiently flexible to be used as an abstraction for DMA based ringbuffers as well as a pure software implementations.

Functions

GstAudioRingBufferCallback ()

void
(*GstAudioRingBufferCallback) (GstAudioRingBuffer *rbuf,
                               guint8 *data,
                               guint len,
                               gpointer user_data);

This function is set with gst_audio_ring_buffer_set_callback() and is called to fill the memory at data with len bytes of samples.

Parameters

rbuf

a GstAudioRingBuffer

 

data

target to fill.

[array length=len]

len

amount to fill

 

user_data

user data

 

GST_AUDIO_RING_BUFFER_BROADCAST()

#define GST_AUDIO_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_RING_BUFFER_GET_COND (buf)))

GST_AUDIO_RING_BUFFER_GET_COND()

#define GST_AUDIO_RING_BUFFER_GET_COND(buf) (&(((GstAudioRingBuffer *)buf)->cond))

GST_AUDIO_RING_BUFFER_SIGNAL()

#define GST_AUDIO_RING_BUFFER_SIGNAL(buf)   (g_cond_signal (GST_AUDIO_RING_BUFFER_GET_COND (buf)))

GST_AUDIO_RING_BUFFER_WAIT()

#define GST_AUDIO_RING_BUFFER_WAIT(buf)     (g_cond_wait (GST_AUDIO_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))

gst_audio_ring_buffer_set_callback ()

void
gst_audio_ring_buffer_set_callback (GstAudioRingBuffer *buf,
                                    GstAudioRingBufferCallback cb,
                                    gpointer user_data);

Sets the given callback function on the buffer. This function will be called every time a segment has been written to a device.

MT safe.

[skip]

Parameters

buf

the GstAudioRingBuffer to set the callback on

 

cb

the callback to set.

[allow-none]

user_data

user data passed to the callback

 

gst_audio_ring_buffer_set_callback_full ()

void
gst_audio_ring_buffer_set_callback_full
                               (GstAudioRingBuffer *buf,
                                GstAudioRingBufferCallback cb,
                                gpointer user_data,
                                GDestroyNotify notify);

Sets the given callback function on the buffer. This function will be called every time a segment has been written to a device.

MT safe.

[rename-to gst_audio_ring_buffer_set_callback]

Parameters

buf

the GstAudioRingBuffer to set the callback on

 

cb

the callback to set.

[allow-none]

user_data

user data passed to the callback

 

notify

function to be called when user_data is no longer needed

 

Since: 1.12


gst_audio_ring_buffer_acquire ()

gboolean
gst_audio_ring_buffer_acquire (GstAudioRingBuffer *buf,
                               GstAudioRingBufferSpec *spec);

Allocate the resources for the ringbuffer. This function fills in the data pointer of the ring buffer with a valid GstBuffer to which samples can be written.

Parameters

buf

the GstAudioRingBuffer to acquire

 

spec

the specs of the buffer

 

Returns

TRUE if the device could be acquired, FALSE on error.

MT safe.


gst_audio_ring_buffer_release ()

gboolean
gst_audio_ring_buffer_release (GstAudioRingBuffer *buf);

Free the resources of the ringbuffer.

Parameters

buf

the GstAudioRingBuffer to release

 

Returns

TRUE if the device could be released, FALSE on error.

MT safe.


gst_audio_ring_buffer_is_acquired ()

gboolean
gst_audio_ring_buffer_is_acquired (GstAudioRingBuffer *buf);

Check if the ringbuffer is acquired and ready to use.

Parameters

buf

the GstAudioRingBuffer to check

 

Returns

TRUE if the ringbuffer is acquired, FALSE on error.

MT safe.


gst_audio_ring_buffer_activate ()

gboolean
gst_audio_ring_buffer_activate (GstAudioRingBuffer *buf,
                                gboolean active);

Activate buf to start or stop pulling data.

MT safe.

Parameters

buf

the GstAudioRingBuffer to activate

 

active

the new mode

 

Returns

TRUE if the device could be activated in the requested mode, FALSE on error.


gst_audio_ring_buffer_is_active ()

gboolean
gst_audio_ring_buffer_is_active (GstAudioRingBuffer *buf);

Check if buf is activated.

MT safe.

Parameters

buf

the GstAudioRingBuffer

 

Returns

TRUE if the device is active.


gst_audio_ring_buffer_start ()

gboolean
gst_audio_ring_buffer_start (GstAudioRingBuffer *buf);

Start processing samples from the ringbuffer.

Parameters

buf

the GstAudioRingBuffer to start

 

Returns

TRUE if the device could be started, FALSE on error.

MT safe.


gst_audio_ring_buffer_pause ()

gboolean
gst_audio_ring_buffer_pause (GstAudioRingBuffer *buf);

Pause processing samples from the ringbuffer.

Parameters

buf

the GstAudioRingBuffer to pause

 

Returns

TRUE if the device could be paused, FALSE on error.

MT safe.


gst_audio_ring_buffer_stop ()

gboolean
gst_audio_ring_buffer_stop (GstAudioRingBuffer *buf);

Stop processing samples from the ringbuffer.

Parameters

buf

the GstAudioRingBuffer to stop

 

Returns

TRUE if the device could be stopped, FALSE on error.

MT safe.


gst_audio_ring_buffer_delay ()

guint
gst_audio_ring_buffer_delay (GstAudioRingBuffer *buf);

Get the number of samples queued in the audio device. This is usually less than the segment size but can be bigger when the implementation uses another internal buffer between the audio device.

For playback ringbuffers this is the amount of samples transfered from the ringbuffer to the device but still not played.

For capture ringbuffers this is the amount of samples in the device that are not yet transfered to the ringbuffer.

Parameters

buf

the GstAudioRingBuffer to query

 

Returns

The number of samples queued in the audio device.

MT safe.


gst_audio_ring_buffer_samples_done ()

guint64
gst_audio_ring_buffer_samples_done (GstAudioRingBuffer *buf);

Get the number of samples that were processed by the ringbuffer since it was last started. This does not include the number of samples not yet processed (see gst_audio_ring_buffer_delay()).

Parameters

buf

the GstAudioRingBuffer to query

 

Returns

The number of samples processed by the ringbuffer.

MT safe.


gst_audio_ring_buffer_set_sample ()

void
gst_audio_ring_buffer_set_sample (GstAudioRingBuffer *buf,
                                  guint64 sample);

Make sure that the next sample written to the device is accounted for as being the sample sample written to the device. This value will be used in reporting the current sample position of the ringbuffer.

This function will also clear the buffer with silence.

MT safe.

Parameters

buf

the GstAudioRingBuffer to use

 

sample

the sample number to set

 

gst_audio_ring_buffer_commit ()

guint
gst_audio_ring_buffer_commit (GstAudioRingBuffer *buf,
                              guint64 *sample,
                              guint8 *data,
                              gint in_samples,
                              gint out_samples,
                              gint *accum);

Commit in_samples samples pointed to by data to the ringbuffer buf .

in_samples and out_samples define the rate conversion to perform on the samples in data . For negative rates, out_samples must be negative and in_samples positive.

When out_samples is positive, the first sample will be written at position sample in the ringbuffer. When out_samples is negative, the last sample will be written to sample in reverse order.

out_samples does not need to be a multiple of the segment size of the ringbuffer although it is recommended for optimal performance.

accum will hold a temporary accumulator used in rate conversion and should be set to 0 when this function is first called. In case the commit operation is interrupted, one can resume the processing by passing the previously returned accum value back to this function.

MT safe.

Parameters

buf

the GstAudioRingBuffer to commit

 

sample

the sample position of the data

 

data

the data to commit

 

in_samples

the number of samples in the data to commit

 

out_samples

the number of samples to write to the ringbuffer

 

accum

accumulator for rate conversion.

 

Returns

The number of samples written to the ringbuffer or -1 on error. The number of samples written can be less than out_samples when buf was interrupted with a flush or stop.


gst_audio_ring_buffer_convert ()

gboolean
gst_audio_ring_buffer_convert (GstAudioRingBuffer *buf,
                               GstFormat src_fmt,
                               gint64 src_val,
                               GstFormat dest_fmt,
                               gint64 *dest_val);

Convert src_val in src_fmt to the equivalent value in dest_fmt . The result will be put in dest_val .

Parameters

buf

the GstAudioRingBuffer

 

src_fmt

the source format

 

src_val

the source value

 

dest_fmt

the destination format

 

dest_val

a location to store the converted value

 

Returns

TRUE if the conversion succeeded.


gst_audio_ring_buffer_prepare_read ()

gboolean
gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer *buf,
                                    gint *segment,
                                    guint8 **readptr,
                                    gint *len);

Returns a pointer to memory where the data from segment segment can be found. This function is mostly used by subclasses.

Parameters

buf

the GstAudioRingBuffer to read from

 

segment

the segment to read

 

readptr

the pointer to the memory where samples can be read

 

len

the number of bytes to read

 

Returns

FALSE if the buffer is not started.

MT safe.


gst_audio_ring_buffer_read ()

guint
gst_audio_ring_buffer_read (GstAudioRingBuffer *buf,
                            guint64 sample,
                            guint8 *data,
                            guint len,
                            GstClockTime *timestamp);

Read len samples from the ringbuffer into the memory pointed to by data . The first sample should be read from position sample in the ringbuffer.

len should not be a multiple of the segment size of the ringbuffer although it is recommended.

timestamp will return the timestamp associated with the data returned.

Parameters

buf

the GstAudioRingBuffer to read from

 

sample

the sample position of the data

 

data

where the data should be read

 

len

the number of samples in data to read

 

timestamp

where the timestamp is returned

 

Returns

The number of samples read from the ringbuffer or -1 on error.

MT safe.


gst_audio_ring_buffer_clear ()

void
gst_audio_ring_buffer_clear (GstAudioRingBuffer *buf,
                             gint segment);

Clear the given segment of the buffer with silence samples. This function is used by subclasses.

MT safe.

Parameters

buf

the GstAudioRingBuffer to clear

 

segment

the segment to clear

 

gst_audio_ring_buffer_clear_all ()

void
gst_audio_ring_buffer_clear_all (GstAudioRingBuffer *buf);

Fill the ringbuffer with silence.

MT safe.

Parameters

buf

the GstAudioRingBuffer to clear

 

gst_audio_ring_buffer_advance ()

void
gst_audio_ring_buffer_advance (GstAudioRingBuffer *buf,
                               guint advance);

Subclasses should call this function to notify the fact that advance segments are now processed by the device.

MT safe.

Parameters

buf

the GstAudioRingBuffer to advance

 

advance

the number of segments written

 

gst_audio_ring_buffer_close_device ()

gboolean
gst_audio_ring_buffer_close_device (GstAudioRingBuffer *buf);

Close the audio device associated with the ring buffer. The ring buffer should already have been released via gst_audio_ring_buffer_release().

Parameters

buf

the GstAudioRingBuffer

 

Returns

TRUE if the device could be closed, FALSE on error.

MT safe.


gst_audio_ring_buffer_open_device ()

gboolean
gst_audio_ring_buffer_open_device (GstAudioRingBuffer *buf);

Open the audio device associated with the ring buffer. Does not perform any setup on the device. You must open the device before acquiring the ring buffer.

Parameters

buf

the GstAudioRingBuffer

 

Returns

TRUE if the device could be opened, FALSE on error.

MT safe.


gst_audio_ring_buffer_device_is_open ()

gboolean
gst_audio_ring_buffer_device_is_open (GstAudioRingBuffer *buf);

Checks the status of the device associated with the ring buffer.

Parameters

buf

the GstAudioRingBuffer

 

Returns

TRUE if the device was open, FALSE if it was closed.

MT safe.


gst_audio_ring_buffer_may_start ()

void
gst_audio_ring_buffer_may_start (GstAudioRingBuffer *buf,
                                 gboolean allowed);

Tell the ringbuffer that it is allowed to start playback when the ringbuffer is filled with samples.

MT safe.

Parameters

buf

the GstAudioRingBuffer

 

allowed

the new value

 

gst_audio_ring_buffer_parse_caps ()

gboolean
gst_audio_ring_buffer_parse_caps (GstAudioRingBufferSpec *spec,
                                  GstCaps *caps);

Parse caps into spec .

Parameters

spec

a spec

 

caps

a GstCaps

 

Returns

TRUE if the caps could be parsed.


gst_audio_ring_buffer_set_flushing ()

void
gst_audio_ring_buffer_set_flushing (GstAudioRingBuffer *buf,
                                    gboolean flushing);

Set the ringbuffer to flushing mode or normal mode.

MT safe.

Parameters

buf

the GstAudioRingBuffer to flush

 

flushing

the new mode

 

gst_audio_ring_buffer_is_flushing ()

gboolean
gst_audio_ring_buffer_is_flushing (GstAudioRingBuffer *buf);

Check if buf is flushing.

MT safe.

Parameters

buf

the GstAudioRingBuffer

 

Returns

TRUE if the device is flushing.


gst_audio_ring_buffer_set_channel_positions ()

void
gst_audio_ring_buffer_set_channel_positions
                               (GstAudioRingBuffer *buf,
                                const GstAudioChannelPosition *position);

Tell the ringbuffer about the device's channel positions. This must be called in when the ringbuffer is acquired.

Parameters

buf

the GstAudioRingBuffer

 

position

the device channel positions

 

gst_audio_ring_buffer_set_timestamp ()

void
gst_audio_ring_buffer_set_timestamp (GstAudioRingBuffer *buf,
                                     gint readseg,
                                     GstClockTime timestamp);

Types and Values

struct GstAudioRingBuffer

struct GstAudioRingBuffer {
  GCond                      cond;
  gboolean                    open;
  gboolean                    acquired;
  guint8                     *memory;
  gsize                       size;
  GstClockTime               *timestamps;
  GstAudioRingBufferSpec      spec;
  gint                        samples_per_seg;
  guint8                     *empty_seg;

  gint                        state;
  gint                        segdone;
  gint                        segbase;
  gint                        waiting;
};

The ringbuffer base class structure.

Members

GCond cond;

used to signal start/stop/pause/resume actions

 

gboolean open;

boolean indicating that the ringbuffer is open

 

gboolean acquired;

boolean indicating that the ringbuffer is acquired

 

guint8 *memory;

data in the ringbuffer

 

gsize size;

size of data in the ringbuffer

 

GstClockTime *timestamps;

   

GstAudioRingBufferSpec spec;

format and layout of the ringbuffer data

 

gint samples_per_seg;

number of samples in one segment

 

guint8 *empty_seg;

pointer to memory holding one segment of silence samples

 

gint state;

state of the buffer

 

gint segdone;

readpointer in the ringbuffer

 

gint segbase;

segment corresponding to segment 0 (unused)

 

gint waiting;

is a reader or writer waiting for a free segment

 

struct GstAudioRingBufferClass

struct GstAudioRingBufferClass {
  GstObjectClass parent_class;

  gboolean     (*open_device)  (GstAudioRingBuffer *buf);
  gboolean     (*acquire)      (GstAudioRingBuffer *buf, GstAudioRingBufferSpec *spec);
  gboolean     (*release)      (GstAudioRingBuffer *buf);
  gboolean     (*close_device) (GstAudioRingBuffer *buf);

  gboolean     (*start)        (GstAudioRingBuffer *buf);
  gboolean     (*pause)        (GstAudioRingBuffer *buf);
  gboolean     (*resume)       (GstAudioRingBuffer *buf);
  gboolean     (*stop)         (GstAudioRingBuffer *buf);

  guint        (*delay)        (GstAudioRingBuffer *buf);

  /* ABI added */
  gboolean     (*activate)     (GstAudioRingBuffer *buf, gboolean active);

  guint        (*commit)       (GstAudioRingBuffer * buf, guint64 *sample,
                                guint8 * data, gint in_samples,
                                gint out_samples, gint * accum);

  void         (*clear_all)    (GstAudioRingBuffer * buf);
};

The vmethods that subclasses can override to implement the ringbuffer.

Members

open_device ()

open the device, don't set any params or allocate anything

 

acquire ()

allocate the resources for the ringbuffer using the given spec

 

release ()

free resources of the ringbuffer

 

close_device ()

close the device

 

start ()

start processing of samples

 

pause ()

pause processing of samples

 

resume ()

resume processing of samples after pause

 

stop ()

stop processing of samples

 

delay ()

get number of frames queued in device

 

activate ()

activate the thread that starts pulling and monitoring the consumed segments in the device.

 

commit ()

write samples into the ringbuffer

 

clear_all ()

clear the entire ringbuffer.

 

struct GstAudioRingBufferSpec

struct GstAudioRingBufferSpec {
  /* in */
  GstCaps  *caps;               /* the caps of the buffer */

  /* in/out */
  GstAudioRingBufferFormatType  type;
  GstAudioInfo                  info;


  guint64  latency_time;        /* the required/actual latency time, this is the
				 * actual the size of one segment and the
				 * minimum possible latency we can achieve. */
  guint64  buffer_time;         /* the required/actual time of the buffer, this is
				 * the total size of the buffer and maximum
				 * latency we can compensate for. */
  gint     segsize;             /* size of one buffer segment in bytes, this value
				 * should be chosen to match latency_time as
				 * well as possible. */
  gint     segtotal;            /* total number of segments, this value is the
				 * number of segments of @segsize and should be
				 * chosen so that it matches buffer_time as
				 * close as possible. */
  /* ABI added 0.10.20 */
  gint     seglatency;          /* number of segments queued in the lower
				 * level device, defaults to segtotal. */
};

The structure containing the format specification of the ringbuffer.

Members

GstCaps *caps;

The caps that generated the Spec.

 

GstAudioRingBufferFormatType type;

the sample type

 

GstAudioInfo info;

the GstAudioInfo

 

guint64 latency_time;

the latency in microseconds

 

guint64 buffer_time;

the total buffer size in microseconds

 

gint segsize;

the size of one segment in bytes

 

gint segtotal;

the total number of segments

 

gint seglatency;

number of segments queued in the lower level device, defaults to segtotal

 

enum GstAudioRingBufferState

The state of the ringbuffer.

Members

GST_AUDIO_RING_BUFFER_STATE_STOPPED

The ringbuffer is stopped

 

GST_AUDIO_RING_BUFFER_STATE_PAUSED

The ringbuffer is paused

 

GST_AUDIO_RING_BUFFER_STATE_STARTED

The ringbuffer is started

 

GST_AUDIO_RING_BUFFER_STATE_ERROR

The ringbuffer has encountered an error after it has been started, e.g. because the device was disconnected (Since 1.2)

 

enum GstAudioRingBufferFormatType

The format of the samples in the ringbuffer.

Members

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW

samples in linear or float

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW

samples in mulaw

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW

samples in alaw

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM

samples in ima adpcm

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG

samples in mpeg audio (but not AAC) format

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM

samples in gsm format

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958

samples in IEC958 frames (e.g. AC3)

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3

samples in AC3 format

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3

samples in EAC3 format

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS

samples in DTS format

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC

samples in MPEG-2 AAC ADTS format

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC

samples in MPEG-4 AAC ADTS format

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW

samples in MPEG-2 AAC raw format (Since 1.12)

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW

samples in MPEG-4 AAC raw format (Since 1.12)

 

GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC

samples in FLAC format (Since 1.12)

 
© manpagez.com 2000-2024
Individual documents may contain additional copyright information.