16 KiB
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 implicitely 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)
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
Information for an audio format.
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
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
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 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
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.
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
Trait containing all StreamVolume methods.
Implementors
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)