gstreamer-rs/docs/gstreamer-audio/docs.md

This is the base class for audio sinks. Subclasses need to implement the ::create_ringbuffer vmethod. This base class will then take care of writing samples to the ringbuffer, synchronisation, clipping and flushing.

Implements

AudioBaseSinkExt, gst_base::BaseSinkExt, gst::ElementExt, gst::ObjectExt, glib::object::ObjectExt

Trait containing all AudioBaseSink methods.

Implementors

AudioBaseSink, AudioSink

Create and return the AudioRingBuffer for self. This function will call the ::create_ringbuffer vmethod and will set self as the parent of the returned buffer (see gst::ObjectExt::set_parent).

Returns

The new ringbuffer of self.

Get the current alignment threshold, in nanoseconds, used by self.

Returns

The current alignment threshold used by self.

Get the current discont wait, in nanoseconds, used by self.

Returns

The current discont wait used by self.

Get the current drift tolerance, in microseconds, used by self.

Returns

The current drift tolerance used by self.

Queries whether self will provide a clock or not. See also gst_audio_base_sink_set_provide_clock.

Returns

true if self will provide a clock.

Get the current slave method used by self.

Returns

The current slave method used by self.

Informs this base class that the audio output device has failed for some reason, causing a discontinuity (for example, because the device recovered from the error, but lost all contents of its ring buffer). This function is typically called by derived classes, and is useful for the custom slave method.

Controls the sink's alignment threshold.

alignment_threshold

the new alignment threshold in nanoseconds

Sets the custom slaving callback. This callback will be invoked if the slave-method property is set to GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink receives and plays samples.

Setting the callback to NULL causes the sink to behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE method were used.

callback

a GstAudioBaseSinkCustomSlavingCallback

user_data

user data passed to the callback

notify

called when user_data becomes unused

Controls how long the sink will wait before creating a discontinuity.

discont_wait

the new discont wait in nanoseconds

Controls the sink's drift tolerance.

drift_tolerance

the new drift tolerance in microseconds

Controls whether self will provide a clock or not. If provide is true, gst::ElementExt::provide_clock will return a clock that reflects the datarate of self. If provide is false, gst::ElementExt::provide_clock will return NULL.

provide

new state

Controls how clock slaving will be performed in self.

method

the new slave method

A window of time in nanoseconds to wait before creating a discontinuity as a result of breaching the drift-tolerance.

A window of time in nanoseconds to wait before creating a discontinuity as a result of breaching the drift-tolerance.

Controls the amount of time in microseconds that clocks are allowed to drift before resynchronisation happens.

Controls the amount of time in microseconds that clocks are allowed to drift before resynchronisation happens.

This is the base class for audio sources. Subclasses need to implement the ::create_ringbuffer vmethod. This base class will then take care of reading samples from the ringbuffer, synchronisation and flushing.

Implements

AudioBaseSrcExt, gst_base::BaseSrcExt, gst::ElementExt, gst::ObjectExt, glib::object::ObjectExt

Trait containing all AudioBaseSrc methods.

Implementors

AudioBaseSrc, AudioSrc

Create and return the AudioRingBuffer for self. This function will call the ::create_ringbuffer vmethod and will set self as the parent of the returned buffer (see gst::ObjectExt::set_parent).

Returns

The new ringbuffer of self.

Queries whether self will provide a clock or not. See also gst_audio_base_src_set_provide_clock.

Returns

true if self will provide a clock.

Get the current slave method used by self.

Returns

The current slave method used by self.

Controls whether self will provide a clock or not. If provide is true, gst::ElementExt::provide_clock will return a clock that reflects the datarate of self. If provide is false, gst::ElementExt::provide_clock will return NULL.

provide

new state

Controls how clock slaving will be performed in self.

method

the new slave method

Actual configured size of audio buffer in microseconds.

Actual configured audio latency in microseconds.

Audio channel positions.

These are the channels defined in SMPTE 2036-2-2008 Table 1 for 22.2 audio systems with the Surround and Wide channels from DTS Coherent Acoustics (v.1.3.1) and 10.2 and 7.1 layouts. In the caps the actual channel layout is expressed with a channel count and a channel mask, which describes the existing channels. The positions in the bit mask correspond to the enum values. For negotiation it is allowed to have more bits set in the channel mask than the number of channels to specify the allowed channel positions but this is not allowed in negotiated caps. It is not allowed in any situation other than the one mentioned below to have less bits set in the channel mask than the number of channels.

AudioChannelPosition::Mono can only be used with a single mono channel that has no direction information and would be mixed into all directional channels. This is expressed in caps by having a single channel and no channel mask.

AudioChannelPosition::None can only be used if all channels have this position. This is expressed in caps by having a channel mask with no bits set.

As another special case it is allowed to have two channels without a channel mask. This implicitly means that this is a stereo stream with a front left and front right channel.

used for position-less channels, e.g. from a sound card that records 1024 channels; mutually exclusive with any other channel position

Mono without direction; can only be used with 1 channel

invalid position

Front left

Front right

Front center

Low-frequency effects 1 (subwoofer)

Rear left

Rear right

Front left of center

Front right of center

Rear center

Low-frequency effects 2 (subwoofer)

Side left

Side right

Top front left

Top front right

Top front center

Top center

Top rear left

Top rear right

Top side right

Top rear right

Top rear center

Bottom front center

Bottom front left

Bottom front right

Wide left (between front left and side left)

Wide right (between front right and side right)

Surround left (between rear left and side left)

Surround right (between rear right and side right)

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

GstAudioDecoder and subclass should cooperate as follows.

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.

As of configuration stage, and throughout processing, GstAudioDecoder provides various (context) parameters, e.g. describing the format of output audio data (valid when output caps have been set) or current parsing state. Conversely, subclass can and should configure context to inform base class of its expectation w.r.t. buffer handling.

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 AudioDecoder::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.

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 AudioDecoder::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 AudioDecoder: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. Note that this feature is only available if the buffer layout is interleaved. For planar buffers, the decoder implementation is fully responsible for the output buffer size.

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 AudioDecoder::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.

Implements

AudioDecoderExt, gst::ElementExt, gst::ObjectExt, glib::object::ObjectExt, AudioDecoderExtManual

Trait containing all AudioDecoder methods.

Implementors

AudioDecoder

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

size

size of the buffer

Returns

allocated buffer

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 AudioDecoderClass.handle_frame() may be invalidated by a call to this function.

buf

decoded data

frames

number of decoded frames represented by decoded data

Returns

a gst::FlowReturn that should be escalated to caller (of caller)

Collects decoded data and pushes it downstream. This function may be called multiple times for a given input frame.

buf may be NULL in which case it is assumed that the current input frame is finished. This is equivalent to calling AudioDecoder::finish_subframe with a NULL buffer and frames=1 after having pushed out all decoded audio subframes using this function.

When called with valid data in buf the source pad caps must have been set already.

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

Feature: v1_16

buf

decoded data

Returns

a gst::FlowReturn that should be escalated to caller (of caller)

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

Unref the allocator after use it.

allocator

the gst::Allocator used

params

the gst::AllocationParams of allocator

Returns

a AudioInfo describing the input audio format

Returns

currently configured decoder delay

Queries decoder drain handling.

Returns

TRUE if drainable handling is enabled.

MT safe.

Returns

currently configured byte to time conversion setting

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

min

a pointer to storage to hold minimum latency

max

a pointer to storage to hold maximum latency

Returns

currently configured decoder tolerated error count.

Queries decoder's latency aggregation.

Returns

aggregation latency.

MT safe.

Queries decoder required format handling.

Returns

TRUE if required format handling is enabled.

MT safe.

Return current parsing (sync and eos) state.

sync

a pointer to a variable to hold the current sync state

eos

a pointer to a variable to hold the current eos state

Queries decoder packet loss concealment handling.

Returns

TRUE if packet loss concealment is enabled.

MT safe.

Returns

currently configured plc handling

Queries current audio jitter tolerance threshold.

Returns

decoder audio jitter tolerance threshold.

MT safe.

Sets the audio decoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with AudioDecoderExt::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.

tags

a gst::TagList to merge, or NULL

mode

the gst::TagMergeMode to use, usually gst::TagMergeMode::Replace

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

Returns

true if the negotiation succeeded, else false.

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

caps

initial caps

filter

filter caps

Returns

a gst::Caps owned by caller

Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling AudioDecoder::negotiate. Setting to None the allocation query will use the caps from the pad.

Feature: v1_10

allocation_caps

a gst::Caps or None

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.

enabled

new state

Allows baseclass to perform byte to time estimated conversion.

enabled

whether to enable byte to time conversion

Sets decoder latency.

min

minimum latency

max

maximum latency

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.

num

max tolerated errors

Sets decoder minimum aggregation latency.

MT safe.

num

new minimum latency

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.

enabled

new state

Configure output caps on the srcpad of self. Similar to AudioDecoder::set_output_format, but allows subclasses to specify output caps that can't be expressed via AudioInfo e.g. caps that have caps features.

Feature: v1_16

caps

(fixed) gst::Caps

Returns

true on success.

Configure output info on the srcpad of self.

info

AudioInfo

Returns

true on success.

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

MT safe.

enabled

new state

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

plc

new plc state

Configures decoder audio jitter tolerance threshold.

MT safe.

tolerance

new tolerance

Lets AudioDecoder sub-classes decide if they want the sink pad to use the default pad query handler to reply to accept-caps queries.

By setting this to true it is possible to further customize the default handler with GST_PAD_SET_ACCEPT_INTERSECT and GST_PAD_SET_ACCEPT_TEMPLATE

use_

if the default pad accept-caps query handling should be used

Maximum number of tolerated consecutive decode errors. See AudioDecoderExt::set_max_errors for more details.

Feature: v1_18

Maximum number of tolerated consecutive decode errors. See AudioDecoderExt::set_max_errors for more details.

Feature: v1_18

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

GstAudioEncoder and subclass should cooperate as follows.

Configuration

• Initially, GstAudioEncoder calls start when the encoder element is activated, which allows subclass to perform any global setup.

• GstAudioEncoder calls set_format to inform subclass of the format of input audio data that it is about to receive. Subclass should setup for encoding and configure various base class parameters appropriately, notably those directing desired input data handling. While unlikely, it might be called more than once, if changing input parameters require reconfiguration.

• GstAudioEncoder calls stop at end of all processing.

As of configuration stage, and throughout processing, GstAudioEncoder maintains various parameters that provide required context, e.g. describing the format of input audio data. Conversely, subclass can and should configure these context parameters to inform base class of its expectation w.r.t. buffer handling.

Data processing

• Base class gathers input sample data (as directed by the context's frame_samples and frame_max) and provides this to subclass' handle_frame.
• If codec processing results in encoded data, subclass should call AudioEncoder::finish_frame to have encoded data pushed downstream. Alternatively, it might also call AudioEncoder::finish_frame (with a NULL buffer and some number of dropped samples) to indicate dropped (non-encoded) samples.
• Just prior to actually pushing a buffer downstream, it is passed to pre_push.
• During the parsing process GstAudioEncoderClass will handle both srcpad and sinkpad events. Sink events will be passed to subclass if event callback has been provided.

Shutdown phase

• GstAudioEncoder 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 AudioEncoder::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 either favor tracking upstream timestamps (at the possible expense of jitter) or aim to arrange for a perfect stream of output timestamps, depending on AudioEncoder:perfect-timestamp. However, in the latter case, the input may not be so perfect or ideal, which is handled as follows. An input timestamp is compared with the expected timestamp as dictated by input sample stream and if the deviation is less than AudioEncoder:tolerance, the deviation is discarded. Otherwise, it is considered a discontuinity and subsequent output timestamp is resynced to the new position after performing configured discontinuity processing. In the non-perfect-timestamp case, an upstream variation exceeding tolerance only leads to marking DISCONT on subsequent outgoing (while timestamps are adjusted to upstream regardless of variation). While DISCONT is also marked in the perfect-timestamp case, this one optionally (see AudioEncoder:hard-resync) performs some additional steps, such as clipping of (early) input samples or draining all currently remaining input data, depending on the direction of the discontuinity.

If perfect timestamps are arranged, it is also possible to request baseclass (usually set by subclass) to provide additional buffer metadata (in OFFSET and OFFSET_END) fields according to granule defined semantics currently needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count including buffer) and OFFSET_END to corresponding timestamp (as determined by same sample count and sample rate).

Things that subclass need to take care of:

• Provide pad templates
• Set source pad caps when appropriate
• Inform base class of buffer processing needs using context's frame_samples and frame_bytes.
• Set user-configurable properties to sane defaults for format and implementing codec at hand, e.g. those controlling timestamp behaviour and discontinuity processing.
• Accept data in handle_frame and provide encoded results to AudioEncoder::finish_frame.

Implements

AudioEncoderExt, gst::ElementExt, gst::ObjectExt, glib::object::ObjectExt, AudioEncoderExtManual

Trait containing all AudioEncoder methods.

Implementors

AudioEncoder

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

size

size of the buffer

Returns

allocated buffer

Collects encoded data and pushes encoded data downstream. Source pad caps must be set when this is called.

If samples < 0, then best estimate is all samples provided to encoder (subclass) so far. buf may be NULL, in which case next number of samples are considered discarded, e.g. as a result of discontinuous transmission, and a discontinuity is marked.

Note that samples received in AudioEncoderClass.handle_frame() may be invalidated by a call to this function.

buffer

encoded data

samples

number of samples (per channel) represented by encoded data

Returns

a gst::FlowReturn that should be escalated to caller (of caller)

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

Unref the allocator after use it.

allocator

the gst::Allocator used

params

the gst::AllocationParams of allocator

Returns

a AudioInfo describing the input audio format

Queries encoder drain handling.

Returns

TRUE if drainable handling is enabled.

MT safe.

Returns

currently configured maximum handled frames

Returns

currently maximum requested samples per frame

Returns

currently minimum requested samples per frame

Queries encoder hard minimum handling.

Returns

TRUE if hard minimum handling is enabled.

MT safe.

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

min

a pointer to storage to hold minimum latency

max

a pointer to storage to hold maximum latency

Returns

currently configured encoder lookahead

Queries if the encoder will handle granule marking.

Returns

TRUE if granule marking is enabled.

MT safe.

Queries encoder perfect timestamp behaviour.

Returns

TRUE if perfect timestamp setting enabled.

MT safe.

Queries current audio jitter tolerance threshold.

Returns

encoder audio jitter tolerance threshold.

MT safe.

Sets the audio encoder tags and how they should be merged with any upstream stream tags. This will override any tags previously-set with AudioEncoderExt::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.

MT safe.

tags

a gst::TagList to merge, or NULL to unset previously-set tags

mode

the gst::TagMergeMode to use, usually gst::TagMergeMode::Replace

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

Returns

true if the negotiation succeeded, else false.

Returns caps that express caps (or sink template caps if caps == NULL) restricted to channel/rate combinations supported by downstream elements (e.g. muxers).

caps

initial caps

filter

filter caps

Returns

a gst::Caps owned by caller

Sets a caps in allocation query which are different from the set pad's caps. Use this function before calling AudioEncoder::negotiate. Setting to None the allocation query will use the caps from the pad.

Feature: v1_10

allocation_caps

a gst::Caps or None

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

MT safe.

enabled

new state

Sets max number of frames accepted at once (assumed minimally 1). Requires frame_samples_min and frame_samples_max to be the equal.

Note: This value will be reset to 0 every time before AudioEncoderClass.set_format() is called.

num

number of frames

Sets number of samples (per channel) subclass needs to be handed, at most or will be handed all available if 0.

If an exact number of samples is required, AudioEncoderExt::set_frame_samples_min must be called with the same number.

Note: This value will be reset to 0 every time before AudioEncoderClass.set_format() is called.

num

number of samples per frame

Sets number of samples (per channel) subclass needs to be handed, at least or will be handed all available if 0.

If an exact number of samples is required, AudioEncoderExt::set_frame_samples_max must be called with the same number.

Note: This value will be reset to 0 every time before AudioEncoderClass.set_format() is called.

num

number of samples per frame

Configures encoder hard minimum handling. If enabled, subclass will never be handed less samples than it configured, which otherwise might occur near end-of-data handling. Instead, the leftover samples will simply be discarded.

MT safe.

enabled

new state

Set the codec headers to be sent downstream whenever requested.

headers

a list of gst::Buffer containing the codec header

Sets encoder latency.

min

minimum latency

max

maximum latency

Sets encoder lookahead (in units of input rate samples)

Note: This value will be reset to 0 every time before AudioEncoderClass.set_format() is called.

num

lookahead

Enable or disable encoder granule handling.

MT safe.

enabled

new state

Configure output caps on the srcpad of self.

caps

gst::Caps

Returns

true on success.

Enable or disable encoder perfect output timestamp preference.

MT safe.

enabled

new state

Configures encoder audio jitter tolerance threshold.

MT safe.

tolerance

new tolerance

Extra audio flags

no valid flag

the position array explicitly contains unpositioned channels.

Enum value describing the most common audio formats.

unknown or unset audio format

encoded audio format

8 bits in 8 bits, signed

8 bits in 8 bits, unsigned

16 bits in 16 bits, signed, little endian

16 bits in 16 bits, signed, big endian

16 bits in 16 bits, unsigned, little endian

16 bits in 16 bits, unsigned, big endian

24 bits in 32 bits, signed, little endian

24 bits in 32 bits, signed, big endian

24 bits in 32 bits, unsigned, little endian

24 bits in 32 bits, unsigned, big endian

32 bits in 32 bits, signed, little endian

32 bits in 32 bits, signed, big endian

32 bits in 32 bits, unsigned, little endian

32 bits in 32 bits, unsigned, big endian

24 bits in 24 bits, signed, little endian

24 bits in 24 bits, signed, big endian

24 bits in 24 bits, unsigned, little endian

24 bits in 24 bits, unsigned, big endian

20 bits in 24 bits, signed, little endian

20 bits in 24 bits, signed, big endian

20 bits in 24 bits, unsigned, little endian

20 bits in 24 bits, unsigned, big endian

18 bits in 24 bits, signed, little endian

18 bits in 24 bits, signed, big endian

18 bits in 24 bits, unsigned, little endian

18 bits in 24 bits, unsigned, big endian

32-bit floating point samples, little endian

32-bit floating point samples, big endian

64-bit floating point samples, little endian

64-bit floating point samples, big endian

16 bits in 16 bits, signed, native endianness

16 bits in 16 bits, unsigned, native endianness

24 bits in 32 bits, signed, native endianness

24 bits in 32 bits, unsigned, native endianness

32 bits in 32 bits, signed, native endianness

32 bits in 32 bits, unsigned, native endianness

24 bits in 24 bits, signed, native endianness

24 bits in 24 bits, unsigned, native endianness

20 bits in 24 bits, signed, native endianness

20 bits in 24 bits, unsigned, native endianness

18 bits in 24 bits, signed, native endianness

18 bits in 24 bits, unsigned, native endianness

32-bit floating point samples, native endianness

64-bit floating point samples, native endianness

The different audio flags that a format info can have.

integer samples

float samples

signed samples

complex layout

the format can be used in GstAudioFormatUnpack and GstAudioFormatPack functions

Information describing audio properties. This information can be filled in from GstCaps with AudioInfo::from_caps.

Use the provided macros to access the info in this structure.

Allocate a new AudioInfo that is also initialized with AudioInfo::init.

Returns

a new AudioInfo. free with AudioInfo::free.

Converts among various gst::Format 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.

src_fmt

gst::Format of the src_val

src_val

value to convert

dest_fmt

gst::Format of the dest_val

dest_val

pointer to destination value

Returns

TRUE if the conversion was successful.

Copy a GstAudioInfo structure.

Returns

a new AudioInfo. free with gst_audio_info_free.

Free a GstAudioInfo structure previously allocated with AudioInfo::new or AudioInfo::copy.

Parse caps and update self.

caps

a gst::Caps

Returns

TRUE if caps could be parsed

Initialize self with default values.

Compares two AudioInfo and returns whether they are equal or not

other

a AudioInfo

Returns

true if self and other are equal, else false.

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

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

format

the format

rate

the samplerate

channels

the number of channels

position

the channel positions

Convert the values of self into a gst::Caps.

Returns

the new gst::Caps containing the info of self.

Layout of the audio samples for the different channels.

interleaved audio

non-interleaved audio

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

No flag

When the source has a smaller depth than the target format, set the least significant bits of the target to 0. This is likely slightly 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.

The format of the samples in the ringbuffer.

samples in linear or float

samples in mulaw

samples in alaw

samples in ima adpcm

samples in mpeg audio (but not AAC) format

samples in gsm format

samples in IEC958 frames (e.g. AC3)

samples in AC3 format

samples in EAC3 format

samples in DTS format

samples in MPEG-2 AAC ADTS format

samples in MPEG-4 AAC ADTS format

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

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

samples in FLAC format (Since: 1.12)

This is the most simple base class for audio sinks that only requires subclasses to implement a set of simple functions:

• open() :Open the device.

• prepare() :Configure the device with the specified format.

• write() :Write samples to the device.

• reset() :Unblock writes and flush the device.

• delay() :Get the number of samples written but not yet played by the device.

• unprepare() :Undo operations done by prepare.

• close() :Close the device.

All scheduling of samples and timestamps is done in this base class together with AudioBaseSink using a default implementation of a AudioRingBuffer that uses threads.

Implements

AudioBaseSinkExt, gst_base::BaseSinkExt, gst::ElementExt, gst::ObjectExt, glib::object::ObjectExt

This is the most simple base class for audio sources that only requires subclasses to implement a set of simple functions:

• open() :Open the device.
• prepare() :Configure the device with the specified format.
• read() :Read samples from the device.
• reset() :Unblock reads and flush the device.
• delay() :Get the number of samples in the device but not yet read.
• unprepare() :Undo operations done by prepare.
• close() :Close the device.

All scheduling of samples and timestamps is done in this base class together with AudioBaseSrc using a default implementation of a AudioRingBuffer that uses threads.

Implements

AudioBaseSrcExt, gst_base::BaseSrcExt, gst::ElementExt, gst::ObjectExt, glib::object::ObjectExt

AudioStreamAlign provides a helper object that helps tracking audio stream alignment and discontinuities, and detects discontinuities if possible.

See AudioStreamAlign::new for a description of its parameters and AudioStreamAlign::process for the details of the processing.

Feature: v1_14

Allocate a new AudioStreamAlign with the given configuration. All processing happens according to sample rate rate, until AudioStreamAlign::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 AudioStreamAlign::set_alignment_threshold and AudioStreamAlign::set_discont_wait.

Feature: v1_14

rate

a sample rate

alignment_threshold

a alignment threshold in nanoseconds

discont_wait

discont wait in nanoseconds

Returns

a new AudioStreamAlign. free with AudioStreamAlign::free.

Copy a GstAudioStreamAlign structure.

Feature: v1_14

Returns

a new AudioStreamAlign. free with gst_audio_stream_align_free.

Free a GstAudioStreamAlign structure previously allocated with AudioStreamAlign::new or AudioStreamAlign::copy.

Feature: v1_14

Gets the currently configured alignment threshold.

Feature: v1_14

Returns

The currently configured alignment threshold

Gets the currently configured discont wait.

Feature: v1_14

Returns

The currently configured discont wait

Gets the currently configured sample rate.

Feature: v1_14

Returns

The currently configured sample rate

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

Feature: v1_14

Returns

The number of samples processed since the last discontinuity.

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

Feature: v1_14

Returns

The last timestamp at when a discontinuity was detected

Marks the next buffer as discontinuous and resets timestamp tracking.

Feature: v1_14

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.

Feature: v1_14

discont

if this data is considered to be discontinuous

timestamp

a gst::ClockTime of the start of the data

n_samples

number of samples to process

out_timestamp

output timestamp of the data

out_duration

output duration of the data

out_sample_position

output sample position of the start of the data

Returns

true if a discontinuity was detected, false otherwise.

Sets alignment_treshold as new alignment threshold for the following processing.

Feature: v1_14

alignment_threshold

a new alignment threshold

Sets alignment_treshold as new discont wait for the following processing.

Feature: v1_14

discont_wait

a new discont wait

Sets rate as new sample rate for the following processing. If the sample rate differs this implicitly marks the next data as discontinuous.

Feature: v1_14

rate

a new sample rate

This interface is implemented by elements that provide a stream volume. Examples for such elements are volume and playbin.

Applications can use this interface to get or set the current stream volume. For this the "volume" gobject::Object property can be used or the helper functions StreamVolume::set_volume and StreamVolume::get_volume. This volume is always a linear factor, i.e. 0.0 is muted 1.0 is 100%. For showing the volume in a GUI it might make sense to convert it to a different format by using StreamVolume::convert_volume. Volume sliders should usually use a cubic volume.

Separate from the volume the stream can also be muted by the "mute" gobject::Object property or StreamVolume::set_mute and StreamVolume::get_mute.

Elements that provide some kind of stream volume should implement the "volume" and "mute" gobject::Object properties and handle setting and getting of them properly. The volume property is defined to be a linear volume factor.

Implements

StreamVolumeExt

Trait containing all StreamVolume methods.

Implementors

StreamVolume

from

StreamVolumeFormat to convert from

to

StreamVolumeFormat to convert to

val

Volume in from format that should be converted

Returns

the converted volume

Returns

Returns true if the stream is muted

format

StreamVolumeFormat which should be returned

Returns

The current stream volume as linear factor

mute

Mute state that should be set

format

StreamVolumeFormat of val

val

Linear volume factor that should be set

Different representations of a stream volume. StreamVolume::convert_volume allows to convert between the different representations.

Formulas to convert from a linear to a cubic or dB volume are cbrt(val) and 20 * log10 (val).

Linear scale factor, 1.0 = 100%

Cubic volume scale

Logarithmic volume scale (dB, amplitude not power)