gstaudiodecoder: GStreamer Base Plugins 1.0 Library Reference Manual

## gstaudiodecoder gstaudiodecoder — Base class for audio decoders |

## Functions

#define | GST_AUDIO_DECODER_ERROR()
#define | GST_AUDIO_DECODER_SINK_PAD()
#define | GST_AUDIO_DECODER_SRC_PAD()
GstFlowReturn | gst_audio_decoder_finish_frame ()
gboolean | gst_audio_decoder_set_output_format ()
gboolean | gst_audio_decoder_negotiate ()
GstBuffer * | gst_audio_decoder_allocate_output_buffer ()
void | gst_audio_decoder_get_allocator ()
GstAudioInfo * | gst_audio_decoder_get_audio_info ()
gint | gst_audio_decoder_get_estimate_rate ()
gint | gst_audio_decoder_get_delay ()
gboolean | gst_audio_decoder_get_drainable ()
void | gst_audio_decoder_get_latency ()
gint | gst_audio_decoder_get_max_errors ()
GstClockTime | gst_audio_decoder_get_min_latency ()
gboolean | gst_audio_decoder_get_needs_format ()
void | gst_audio_decoder_get_parse_state ()
gboolean | gst_audio_decoder_get_plc ()
gint | gst_audio_decoder_get_plc_aware ()
GstClockTime | gst_audio_decoder_get_tolerance ()
void | gst_audio_decoder_set_estimate_rate ()
void | gst_audio_decoder_set_drainable ()
void | gst_audio_decoder_set_latency ()
void | gst_audio_decoder_set_max_errors ()
void | gst_audio_decoder_set_min_latency ()
void | gst_audio_decoder_set_needs_format ()
void | gst_audio_decoder_set_plc ()
void | gst_audio_decoder_set_plc_aware ()
void | gst_audio_decoder_set_tolerance ()
void | gst_audio_decoder_merge_tags ()
GstCaps * | gst_audio_decoder_proxy_getcaps ()

## Properties

gint64 | min-latency | Read / Write
gboolean | plc | Read / Write
gint64 | tolerance | Read / Write

## Types and Values

struct | GstAudioDecoder
struct | GstAudioDecoderClass
#define | GST_AUDIO_DECODER_SINK_NAME
#define | GST_AUDIO_DECODER_SRC_NAME

## Object Hierarchy

        GObject
        ╰── GInitiallyUnowned
            ╰── GstObject
                ╰── GstElement
                    ╰── GstAudioDecoder

## Includes

    #include <gst/audio/gstaudiodecoder.h>

## Description

This base class is for audio decoders turning encoded data into raw audio samples.

GstAudioDecoder and subclass should cooperate as follows.

1. Configuration

   - Initially, GstAudioDecoder calls `start` when the decoder element is activated, which allows subclass to perform any global setup. Base class (context) parameters can already be set according to subclass capabilities (or possibly upon receive more information in subsequent `set_format`).

   - GstAudioDecoder calls `set_format` to inform subclass of the format of input audio data that it is about to receive. While unlikely, it might be called more than once, if changing input parameters require reconfiguration.

   - GstAudioDecoder calls `stop` at end of all processing.

2. Data processing

   - Base class gathers input data, and optionally allows subclass to parse this into subsequently manageable (as defined by subclass) chunks. Such chunks are subsequently referred to as 'frames', though they may or may not correspond to 1 (or more) audio format frame.

   - Input frame is provided to subclass' `handle_frame`.

   - If codec processing results in decoded data, subclass should call `gst_audio_decoder_finish_frame` to have decoded data pushed downstream.

   - Just prior to actually pushing a buffer downstream, it is passed to `pre_push`. Subclass should either use this callback to arrange for additional downstream pushing or otherwise ensure such custom pushing occurs after at least a method call has finished since setting src pad caps.

   - During the parsing process GstAudioDecoderClass will handle both srcpad and sinkpad events. Sink events will be passed to subclass if `event` callback has been provided.

3. Shutdown phase

   - GstAudioDecoder class calls `stop` to inform the subclass that data parsing will be stopped.

Subclass is responsible for providing pad template caps for source and sink pads. The pads need to be named "sink" and "src". It also needs to set the fixed caps on srcpad, when the format is ensured. This is typically when base class calls subclass' `set_format` function, though it might be delayed until calling `gst_audio_decoder_finish_frame` .

In summary, above process should have subclass concentrating on codec data processing while leaving other matters to base class, such as most notably timestamp handling. While it may exert more control in this area (see e.g. `pre_push` ), it is very much not recommended.

In particular, base class will try to arrange for perfect output timestamps as much as possible while tracking upstream timestamps. To this end, if deviation between the next ideal expected perfect timestamp and upstream exceeds “tolerance”, then resync to upstream occurs (which would happen always if the tolerance mechanism is disabled).

In non-live pipelines, baseclass can also (configurably) arrange for output buffer aggregation which may help to redue large(r) numbers of small(er) buffers being pushed and processed downstream.

On the other hand, it should be noted that baseclass only provides limited seeking support (upon explicit subclass request), as full-fledged support should rather be left to upstream demuxer, parser or alike. This simple approach caters for seeking and duration reporting using estimated input bitrates.

Things that subclass need to take care of:

- Provide pad templates

- Set source pad caps when appropriate

- Set user-configurable properties to sane defaults for format and implementing codec at hand, and convey some subclass capabilities and expectations in context.

- Accept data in `handle_frame` and provide encoded results to `gst_audio_decoder_finish_frame`. If it is prepared to perform PLC, it should also accept NULL data in `handle_frame` and provide for data for indicated duration.

## Functions

### GST_AUDIO_DECODER_ERROR()

    #define             GST_AUDIO_DECODER_ERROR(el, weight, domain, code, text, debug, ret)

Utility function that audio decoder elements can use in case they encountered a data processing error that may be fatal for the current "data unit" but need not prevent subsequent decoding. Such errors are counted and if there are too many, as configured in the context's max_errors, the pipeline will post an error message and the application will be requested to stop further media processing. Otherwise, it is considered a "glitch" and only a warning is logged. In either case, `ret` is set to the proper value to return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).

#### Parameters

el | the base audio decoder element that generates the error |
weight | element defined weight of the error, added to error count |
domain | like CORE, LIBRARY, RESOURCE or STREAM (see gstreamer-GstGError) |
code | error code defined for that domain (see gstreamer-GstGError) |
text | the message to display (format string and args enclosed in parentheses) |
debug | debugging information for the message (format string and args enclosed in parentheses) |
ret | variable to receive return value |

### GST_AUDIO_DECODER_SINK_PAD()

    #define GST_AUDIO_DECODER_SINK_PAD(obj)        (((GstAudioDecoder *) (obj))->sinkpad)

Gives the pointer to the sink GstPad object of the element.

#### Parameters

obj | base audio codec instance |

### GST_AUDIO_DECODER_SRC_PAD()

    #define GST_AUDIO_DECODER_SRC_PAD(obj)         (((GstAudioDecoder *) (obj))->srcpad)

Gives the pointer to the source GstPad object of the element.

#### Parameters

obj | base audio codec instance |

### gst_audio_decoder_finish_frame ()

    GstFlowReturn
    gst_audio_decoder_finish_frame (GstAudioDecoder *dec,
                                    GstBuffer *buf,
                                    gint frames);

Collects decoded data and pushes it downstream.

`buf` may be NULL in which case the indicated number of frames are discarded and considered to have produced no output (e.g. lead-in or setup frames). Otherwise, source pad caps must be set when it is called with valid data in `buf` .

Note that a frame received in `gst_audio_decoder_handle_frame()` may be invalidated by a call to this function.

#### Parameters

dec | a GstAudioDecoder |
buf | decoded data |
frames | number of decoded frames represented by decoded data |

#### Returns

a GstFlowReturn that should be escalated to caller (of caller)

### gst_audio_decoder_set_output_format ()

    gboolean
    gst_audio_decoder_set_output_format (GstAudioDecoder *dec,
                                         const GstAudioInfo *info);

Configure output info on the srcpad of `dec` .

#### Parameters

dec | a GstAudioDecoder |
info | GstAudioInfo |

#### Returns

`TRUE` on success.

### gst_audio_decoder_negotiate ()

    gboolean
    gst_audio_decoder_negotiate (GstAudioDecoder *dec);

Negotiate with downstream elements to currently configured GstAudioInfo. Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if negotiate fails.

#### Parameters

dec | a GstAudioDecoder |

#### Returns

TRUE if the negotiation succeeded, else FALSE.

### gst_audio_decoder_allocate_output_buffer ()

    GstBuffer *
    gst_audio_decoder_allocate_output_buffer
                                   (GstAudioDecoder *dec,
                                    gsize size);

Helper function that allocates a buffer to hold an audio frame for `dec` 's current output format.

#### Parameters

dec | a GstAudioDecoder |
size | size of the buffer |

#### Returns

allocated buffer.

[transfer full]

### gst_audio_decoder_get_allocator ()

    void
    gst_audio_decoder_get_allocator (GstAudioDecoder *dec,
                                     GstAllocator **allocator,
                                     GstAllocationParams *params);

Lets GstAudioDecoder sub-classes to know the memory `allocator` used by the base class and its `params` .

Unref the `allocator` after use it.

#### Parameters

dec | a GstAudioDecoder |
allocator | the GstAllocator used. | [out][allow-none][transfer full]
params | the GstAllocatorParams of `allocator` . | [out][allow-none][transfer full]

### gst_audio_decoder_get_audio_info ()

    GstAudioInfo *
    gst_audio_decoder_get_audio_info (GstAudioDecoder *dec);

#### Parameters

dec | a GstAudioDecoder |

#### Returns

a GstAudioInfo describing the input audio format

### gst_audio_decoder_get_estimate_rate ()

    gint
    gst_audio_decoder_get_estimate_rate (GstAudioDecoder *dec);

#### Parameters

dec | a GstAudioDecoder |

#### Returns

currently configured byte to time conversion setting

### gst_audio_decoder_get_delay ()

    gint
    gst_audio_decoder_get_delay (GstAudioDecoder *dec);

#### Parameters

dec | a GstAudioDecoder |

#### Returns

currently configured decoder delay

### gst_audio_decoder_get_drainable ()

    gboolean
    gst_audio_decoder_get_drainable (GstAudioDecoder *dec);

Queries decoder drain handling.

#### Parameters

dec | a GstAudioDecoder |

#### Returns

TRUE if drainable handling is enabled.

MT safe.

### gst_audio_decoder_get_latency ()

    void
    gst_audio_decoder_get_latency (GstAudioDecoder *dec,
                                   GstClockTime *min,
                                   GstClockTime *max);

Sets the variables pointed to by `min` and `max` to the currently configured latency.

#### Parameters

dec | a GstAudioDecoder |
min | a pointer to storage to hold minimum latency. | [out][allow-none]
max | a pointer to storage to hold maximum latency. | [out][allow-none]

### gst_audio_decoder_get_max_errors ()

    gint
    gst_audio_decoder_get_max_errors (GstAudioDecoder *dec);

#### Parameters

dec | a GstAudioDecoder |

#### Returns

currently configured decoder tolerated error count.

### gst_audio_decoder_get_min_latency ()

    GstClockTime
    gst_audio_decoder_get_min_latency (GstAudioDecoder *dec);

Queries decoder's latency aggregation.

#### Parameters

dec | a GstAudioDecoder |

#### Returns

aggregation latency.

MT safe.

### gst_audio_decoder_get_needs_format ()

    gboolean
    gst_audio_decoder_get_needs_format (GstAudioDecoder *dec);

Queries decoder required format handling.

#### Parameters

dec | a GstAudioDecoder |

#### Returns

TRUE if required format handling is enabled.

MT safe.

### gst_audio_decoder_get_parse_state ()

    void
    gst_audio_decoder_get_parse_state (GstAudioDecoder *dec,
                                       gboolean *sync,
                                       gboolean *eos);

Return current parsing (sync and eos) state.

#### Parameters

dec | a GstAudioDecoder |
sync | a pointer to a variable to hold the current sync state |
eos | a pointer to a variable to hold the current eos state |

### gst_audio_decoder_get_plc ()

    gboolean
    gst_audio_decoder_get_plc (GstAudioDecoder *dec);

Queries decoder packet loss concealment handling.

#### Parameters

dec | a GstAudioDecoder |

#### Returns

TRUE if packet loss concealment is enabled.

MT safe.

### gst_audio_decoder_get_plc_aware ()

    gint
    gst_audio_decoder_get_plc_aware (GstAudioDecoder *dec);

#### Parameters

dec | a GstAudioDecoder |

#### Returns

currently configured plc handling

### gst_audio_decoder_get_tolerance ()

    GstClockTime
    gst_audio_decoder_get_tolerance (GstAudioDecoder *dec);

Queries current audio jitter tolerance threshold.

#### Parameters

dec | a GstAudioDecoder |

#### Returns

decoder audio jitter tolerance threshold.

MT safe.

### gst_audio_decoder_set_estimate_rate ()

    void
    gst_audio_decoder_set_estimate_rate (GstAudioDecoder *dec,
                                         gboolean enabled);

Allows baseclass to perform byte to time estimated conversion.

#### Parameters

dec | a GstAudioDecoder |
enabled | whether to enable byte to time conversion |

### gst_audio_decoder_set_drainable ()

    void
    gst_audio_decoder_set_drainable (GstAudioDecoder *dec,
                                     gboolean enabled);

Configures decoder drain handling. If drainable, subclass might be handed a NULL buffer to have it return any leftover decoded data. Otherwise, it is not considered so capable and will only ever be passed real data.

MT safe.

#### Parameters

dec | a GstAudioDecoder |
enabled | new state |

### gst_audio_decoder_set_latency ()

    void
    gst_audio_decoder_set_latency (GstAudioDecoder *dec,
                                   GstClockTime min,
                                   GstClockTime max);

Sets decoder latency.

#### Parameters

dec | a GstAudioDecoder |
min | minimum latency |
max | maximum latency |

### gst_audio_decoder_set_max_errors ()

    void
    gst_audio_decoder_set_max_errors (GstAudioDecoder *dec,
                                      gint num);

Sets numbers of tolerated decoder errors, where a tolerated one is then only warned about, but more than tolerated will lead to fatal error. You can set -1 for never returning fatal errors. Default is set to GST_AUDIO_DECODER_MAX_ERRORS.

#### Parameters

dec | a GstAudioDecoder |
num | max tolerated errors |

### gst_audio_decoder_set_min_latency ()

    void
    gst_audio_decoder_set_min_latency (GstAudioDecoder *dec,
                                       GstClockTime num);

Sets decoder minimum aggregation latency.

MT safe.

#### Parameters

dec | a GstAudioDecoder |
num | new minimum latency |

### gst_audio_decoder_set_needs_format ()

    void
    gst_audio_decoder_set_needs_format (GstAudioDecoder *dec,
                                        gboolean enabled);

Configures decoder format needs. If enabled, subclass needs to be negotiated with format caps before it can process any data. It will then never be handed any data before it has been configured. Otherwise, it might be handed data without having been configured and is then expected being able to do so either by default or based on the input data.

MT safe.

#### Parameters

dec | a GstAudioDecoder |
enabled | new state |

### gst_audio_decoder_set_plc ()

    void
    gst_audio_decoder_set_plc (GstAudioDecoder *dec,
                               gboolean enabled);

Enable or disable decoder packet loss concealment, provided subclass and codec are capable and allow handling plc.

MT safe.

#### Parameters

dec | a GstAudioDecoder |
enabled | new state |

### gst_audio_decoder_set_plc_aware ()

    void
    gst_audio_decoder_set_plc_aware (GstAudioDecoder *dec,
                                     gboolean plc);

Indicates whether or not subclass handles packet loss concealment (plc).

#### Parameters

dec | a GstAudioDecoder |
plc | new plc state |

### gst_audio_decoder_set_tolerance ()

    void
    gst_audio_decoder_set_tolerance (GstAudioDecoder *dec,
                                     GstClockTime tolerance);

Configures decoder audio jitter tolerance threshold.

MT safe.

#### Parameters

dec | a GstAudioDecoder |
tolerance | new tolerance |

### gst_audio_decoder_merge_tags ()

    void
    gst_audio_decoder_merge_tags (GstAudioDecoder *dec,
                                  const GstTagList *tags,
                                  GstTagMergeMode mode);

Sets the audio decoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with `gst_audio_decoder_merge_tags()`.

Note that this is provided for convenience, and the subclass is not required to use this and can still do tag handling on its own.

#### Parameters

dec | a GstAudioDecoder |
tags | a GstTagList to merge, or NULL. | [allow-none]
mode | the GstTagMergeMode to use, usually GST_TAG_MERGE_REPLACE |

### gst_audio_decoder_proxy_getcaps ()

    GstCaps *
    gst_audio_decoder_proxy_getcaps (GstAudioDecoder *decoder,
                                     GstCaps *caps,
                                     GstCaps *filter);

Returns caps that express `caps` (or sink template caps if `caps` == NULL) restricted to rate/channels/... combinations supported by downstream elements.

#### Parameters

decoder | a GstAudioDecoder |
caps | initial caps. | [allow-none]
filter | filter caps. | [allow-none]

#### Returns

a GstCaps owned by caller.

[transfer full]

Since: 1.6

## Types and Values

### struct GstAudioDecoder

    struct GstAudioDecoder;

The opaque GstAudioDecoder data structure.

### struct GstAudioDecoderClass

    struct GstAudioDecoderClass {
      GstElementClass element_class;

      /* virtual methods for subclasses */

      gboolean      (*start)              (GstAudioDecoder *dec);

      gboolean      (*stop)               (GstAudioDecoder *dec);

      gboolean      (*set_format)         (GstAudioDecoder *dec,
                                           GstCaps *caps);

      GstFlowReturn (*parse)              (GstAudioDecoder *dec,
                                           GstAdapter *adapter,
                                           gint *offset, gint *length);

      GstFlowReturn (*handle_frame)       (GstAudioDecoder *dec,
                                           GstBuffer *buffer);

      void          (*flush)              (GstAudioDecoder *dec, gboolean hard);

      GstFlowReturn (*pre_push)           (GstAudioDecoder *dec,
                                           GstBuffer **buffer);

      gboolean      (*sink_event)         (GstAudioDecoder *dec,
                                           GstEvent *event);
      gboolean      (*src_event)          (GstAudioDecoder *dec,
                                           GstEvent *event);

      gboolean      (*open)               (GstAudioDecoder *dec);

      gboolean      (*close)              (GstAudioDecoder *dec);

      gboolean      (*negotiate)          (GstAudioDecoder *dec);

      gboolean      (*decide_allocation)  (GstAudioDecoder *dec, GstQuery *query);

      gboolean      (*propose_allocation) (GstAudioDecoder *dec,
                                           GstQuery * query);

      gboolean      (*sink_query)         (GstAudioDecoder *dec, GstQuery *query);

      gboolean      (*src_query)          (GstAudioDecoder *dec, GstQuery *query);

      GstCaps *     (*getcaps)            (GstAudioDecoder * dec,
                                           GstCaps * filter);

      gboolean      (*transform_meta)     (GstAudioDecoder *enc, GstBuffer *outbuf,
                                           GstMeta *meta, GstBuffer *inbuf);
    };

Subclasses can override any of the available virtual methods or not, as needed. At minimum `handle_frame` (and likely `set_format` ) needs to be overridden.

#### Members

GstElementClass `element_class`; | The parent class structure |
`start` () | Optional. Called when the element starts processing. Allows opening external resources. |
`stop` () | Optional. Called when the element stops processing. Allows closing external resources. |
`set_format` () | Notifies subclass of incoming data format (caps). |
`parse` () | Optional. Allows chopping incoming data into manageable units (frames) for subsequent decoding. This division is at subclass discretion and may or may not correspond to 1 (or more) frames as defined by audio format. |
`handle_frame` () | Provides input data (or NULL to clear any remaining data) to subclass. Input data ref management is performed by base class, subclass should not care or intervene, and input data is only valid until next call to base class, most notably a call to `gst_audio_decoder_finish_frame()`. |
`flush` () | Optional. Instructs subclass to clear any codec caches and discard any pending samples and not yet returned decoded data. `hard` indicates whether a FLUSH is being processed, or otherwise a DISCONT (or conceptually similar). |
`pre_push` () | Optional. Called just prior to pushing (encoded data) buffer downstream. Subclass has full discretionary access to buffer, and a not OK flow return will abort downstream pushing. |
`sink_event` () | Optional. Event handler on the sink pad. Subclasses should chain up to the parent implementation to invoke the default handler. |
`src_event` () | Optional. Event handler on the src pad. Subclasses should chain up to the parent implementation to invoke the default handler. |
`open` () | Optional. Called when the element changes to GST_STATE_READY. Allows opening external resources. |
`close` () | Optional. Called when the element changes to GST_STATE_NULL. Allows closing external resources. |
`negotiate` () | Optional. Negotiate with downstream and configure buffer pools, etc. Subclasses should chain up to the parent implementation to invoke the default handler. |
`decide_allocation` () | Optional. Setup the allocation parameters for allocating output buffers. The passed in query contains the result of the downstream allocation query. Subclasses should chain up to the parent implementation to invoke the default handler. |
`propose_allocation` () | Optional. Propose buffer allocation parameters for upstream elements. Subclasses should chain up to the parent implementation to invoke the default handler. |
`sink_query` () | Optional. Query handler on the sink pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since 1.6 |
`src_query` () | Optional. Query handler on the source pad. This function should return TRUE if the query could be performed. Subclasses should chain up to the parent implementation to invoke the default handler. Since 1.6 |
`getcaps` () | Optional. Allows for a custom sink getcaps implementation. If not implemented, default returns gst_audio_decoder_proxy_getcaps applied to sink template caps. |
`transform_meta` () | Optional. Transform the metadata on the input buffer to the output buffer. By default this method copies all meta without tags and meta with only the "audio" tag. subclasses can implement this method and return `TRUE` if the metadata is to be copied. Since 1.6 |

### GST_AUDIO_DECODER_SINK_NAME

    #define GST_AUDIO_DECODER_SINK_NAME    "sink"

The name of the templates for the sink pad.

### GST_AUDIO_DECODER_SRC_NAME

    #define GST_AUDIO_DECODER_SRC_NAME     "src"

The name of the templates for the source pad.

## Property Details

### The `“min-latency”` property

      “min-latency”              gint64

Aggregate output data to a minimum of latency time (ns).

Flags: Read / Write

Allowed values: >= 0

Default value: 0

### The `“plc”` property

      “plc”                      gboolean

Perform packet loss concealment (if supported).

Flags: Read / Write

Default value: FALSE

### The `“tolerance”` property

      “tolerance”                gint64

Perfect ts while timestamp jitter/imperfection within tolerance (ns).

Flags: Read / Write

Allowed values: >= 0

Default value: 0

## See Also

GstBaseTransform