com.genband.mobile.api.utilities

Class AudioCodecConfiguration

java.lang.Object

com.genband.mobile.api.utilities.AudioCodecConfiguration

public class AudioCodecConfiguration
extends java.lang.Object

AudioCodecConfiguration class contains all configuration of Opus audio codec. Configurations can be set using this class instance.

Since:

4.2.4

Nested Class Summary

Nested Classes

Modifier and Type	Class and Description
static class	AudioCodecConfiguration.DefaultConfigurationType

Constructor Summary

Constructors

Constructor and Description
AudioCodecConfiguration(AudioCodecConfiguration.DefaultConfigurationType type) Constructor of AudioCodecConfiguration class.

Method Summary

Modifier and Type	Method and Description
boolean	getOpusDtx() Getter method of DTX.
boolean	getOpusFec() Getter method of maxAverageBitRate.
java.lang.String	getOpusFmtpLine() This method will return the opus fmtp line parameters which will be in the fmtp line of SDP.
int	getOpusMaxAverageBitRate() Getter method of maxAverageBitRate.
int	getOpusMaxPlaybackRate() Getter method of maxPlaybackRate.
int	getPtime() Getter method of ptime.
void	setOpusDtx(boolean dtxEnable) Sets DTX value which will enable or disable Discontinuous Transmission feature of Opus codec.
void	setOpusFec(boolean fecEnable) Sets FEC value which will enable or disable ForwardErrorCorrection feature of Opus codec.
void	setOpusMaxAverageBitRate(int maxAverageBitRate) Sets the MaxAverageBitRate.
void	setOpusMaxPlaybackRate(int maxPlaybackRate) Sets the MaxPlaybackRate.
void	setPtime(int ptime) Sets the Ptime value which will change the ptime configuration of Opus codec.

Methods inherited from class java.lang.Object

clone, equals, finalize, getClass, hashCode, notify, notifyAll, toString, wait, wait, wait

Constructor Detail

AudioCodecConfiguration

public AudioCodecConfiguration(AudioCodecConfiguration.DefaultConfigurationType type)

Constructor of AudioCodecConfiguration class. It will set the opus parameters to MobileSDK preferred values.

Method Detail

getOpusFmtpLine

public java.lang.String getOpusFmtpLine()

This method will return the opus fmtp line parameters which will be in the fmtp line of SDP. Only Utils in WebRTC should call this method. Using this method Utils will modify the SDP and bitrate will decrease.

Returns:

String fmtpLine

setOpusMaxAverageBitRate

public void setOpusMaxAverageBitRate(int maxAverageBitRate)

Sets the MaxAverageBitRate. Maxaveragebitrate specifies the maximum average receive bitrate of a session in bits per second (bit/s). The actual value of the bitrate can vary, as it is dependent on the characteristics of the media in a packet. Note that the maximum average bitrate MAY be modified dynamically during a session. Any positive integer is allowed, but values outside the range 6000 to 510000 SHOULD be ignored. If no value is specified, the maximum value will equals to corresponding mode of Opus and corresponding maxplaybackrate is the default. Decreasing MaxAverageBitRate will cause less bandwidth usage. (For further information check RFC-7587)

MobileSDK default maxAverageBitRate=20000

6000 <= MaxAverageBitRate <= 510000 In case this value set as -1 WebRTC defaults will be use.

Parameters:

maxAverageBitRate - MaxAverageBitRate value as an integer

Since:

4.2.4

setPtime

public void setPtime(int ptime)

Sets the Ptime value which will change the ptime configuration of Opus codec. Ptime is the preferred duration of media represented by a packet that a decoder wants to receive, in milliseconds rounded up to the next full integer value. For Opus, possible values are 3, 5, 10, 20, 40, 60, or an arbitrary multiple of frame size rounded up to the next full integer value, up to a maximum value of 120. If no value is specified, the default is 20. Changing ptime effects all audio codecs. Increasing Ptime will cause less bandwidth usage. (For further information check RFC-7587).

MobileSDK default ptime=60

3 <= MaxAverageBitRate <= 120 and it should be multiple of an Opus frame size rounded up to the next full integer value In case this value set as -1 WebRTC defaults will be use.

Parameters:

ptime - Ptime value as an integer

Since:

4.2.4

setOpusFec

public void setOpusFec(boolean fecEnable)

Sets FEC value which will enable or disable ForwardErrorCorrection feature of Opus codec. The voice mode of Opus allows for embedding in-band Forward Error Correction (FEC) data into the Opus bitstream. This FEC scheme adds redundant information about the previous packet (N-1) to the current output packet N. On the receiving side, the decoder can take advantage of this additional information when it loses a packet and the next packet is available. Disabling FEC cause cause less bandwidth usage (For further information check RFC-7587). If application set fecEnable parameter as 'true' FEC will be enable, if application set it as 'false' FEC will be disable.

MobileSDK default FEC is enabled

Parameters:

fecEnable - fec value as a boolean.

Since:

4.2.4

setOpusDtx

public void setOpusDtx(boolean dtxEnable)

Sets DTX value which will enable or disable Discontinuous Transmission feature of Opus codec. When using continuous transmission, Variable Bitrate(VBR) will reduce the bitrate when the characteristics of the input signal permit, but it will never interrupt the transmission to the receiver. Therefore, the received signal will maintain the same high level of audio quality over the full duration of a transmission while minimizing the average bitrate over time. In cases where the bitrate of Opus needs to be reduced even further or in cases where only constant bitrate is available, the Opus encoder can use Discontinuous Transmission (DTX), where parts of the encoded signal that correspond to periods of silence in the input speech or audio signal are not transmitted to the receiver. A receiver can distinguish between DTX and packet loss by looking for gaps in the sequence number. On default DTX is disabled and enabling DTX will cause less bandwidth usage (For further information check RFC-7587).

If application set dtxEnable parameter as 'true' DTX will be enable, if application set it as 'false' DTX will be disable.

MobileSDK default DTX is enabled

Parameters:

dtxEnable - dtx value as a boolean

Since:

4.2.4

setOpusMaxPlaybackRate

public void setOpusMaxPlaybackRate(int maxPlaybackRate)

Sets the MaxPlaybackRate. Maxplaybackrate is a hint about the maximum output sampling rate that the receiver is capable of rendering in Hz. The decoder MUST be capable of decoding any audio bandwidth, but, due to hardware limitations, only signals up to the specified sampling rate can be played back. Sending signals with higher audio bandwidth results in higher than necessary network usage and encoding complexity, so an encoder SHOULD NOT encode frequencies above the audio bandwidth specified by maxplaybackrate. This parameter can take any value between 8000 and 48000, although commonly the value will match one of the Opus bandwidths . By default, the receiver is assumed to have no limitations, i.e., 48000. Decreasing MaxPlaybackRate will cause less bandwidth usage. Decreasing MaxPlaybackRate will cause less bandwidth usage (For further information check RFC-7587).

MobileSDK default MaxPlaybackRate=16000

8000 <= maxPlaybackRate <= 48000 In case this value set as -1 WebRTC defaults will be use.

Parameters:

maxPlaybackRate - MaxPlaybackRate value as an integer

Since:

4.2.4

getOpusMaxAverageBitRate

public int getOpusMaxAverageBitRate()

Getter method of maxAverageBitRate. Maxaveragebitrate specifies the maximum average receive bitrate of a session in bits per second (bit/s). The actual value of the bitrate can vary, as it is dependent on the characteristics of the media in a packet. Note that the maximum average bitrate MAY be modified dynamically during a session. Any positive integer is allowed, but values outside the range 6000 to 510000 SHOULD be ignored. If no value is specified, the maximum value will equals to corresponding mode of Opus and corresponding maxplaybackrate is the default. Decreasing MaxAverageBitRate will cause less bandwidth usage (For further information check RFC-7587).

Returns:

int maxAverageBitRate

Since:

4.2.4

getOpusMaxPlaybackRate

public int getOpusMaxPlaybackRate()

Getter method of maxPlaybackRate. Maxplaybackrate is a hint about the maximum output sampling rate that the receiver is capable of rendering in Hz. The decoder MUST be capable of decoding any audio bandwidth, but, due to hardware limitations, only signals up to the specified sampling rate can be played back. Sending signals with higher audio bandwidth results in higher than necessary network usage and encoding complexity, so an encoder SHOULD NOT encode frequencies above the audio bandwidth specified by maxplaybackrate. This parameter can take any value between 8000 and 48000, although commonly the value will match one of the Opus bandwidths . By default, the receiver is assumed to have no limitations, i.e., 48000. Decreasing MaxPlaybackRate will cause less bandwidth usage. Decreasing MaxPlaybackRate will cause less bandwidth usage (For further information check RFC-7587).

Returns:

int maxPlaybackRate

Since:

4.2.4

getOpusFec

public boolean getOpusFec()

Getter method of maxAverageBitRate. The voice mode of Opus allows for embedding in-band Forward Error Correction (FEC) data into the Opus bitstream. This FEC scheme adds redundant information about the previous packet (N-1) to the current output packet N. On the receiving side, the decoder can take advantage of this additional information when it loses a packet and the next packet is available. Disabling FEC cause cause less bandwidth usage (For further information check RFC-7587).

Returns:

boolean fec (True if fec is enable, false if fec is disable)

Since:

4.2.4

getOpusDtx

public boolean getOpusDtx()

Getter method of DTX. When using continuous transmission, Variable Bitrate(VBR) will reduce the bitrate when the characteristics of the input signal permit, but it will never interrupt the transmission to the receiver. Therefore, the received signal will maintain the same high level of audio quality over the full duration of a transmission while minimizing the average bitrate over time. In cases where the bitrate of Opus needs to be reduced even further or in cases where only constant bitrate is available, the Opus encoder can use Discontinuous Transmission (DTX), where parts of the encoded signal that correspond to periods of silence in the input speech or audio signal are not transmitted to the receiver. A receiver can distinguish between DTX and packet loss by looking for gaps in the sequence number. On default DTX is disabled and enabling DTX will cause less bandwidth usage (For further information check RFC-7587).

Returns:

boolean dtx (True if dtx is enable, false if dtx is disable)

Since:

4.2.4

getPtime

public int getPtime()

Getter method of ptime. Ptime is the preferred duration of media represented by a packet that a decoder wants to receive, in milliseconds rounded up to the next full integer value. For Opus, possible values are 3, 5, 10, 20, 40, 60, or an arbitrary multiple of frame size rounded up to the next full integer value, up to a maximum value of 120. If no value is specified, the default is 20. Changing ptime effects all audio codecs. Increasing Ptime will cause less bandwidth usage. (For further information check RFC-7587).

Returns:

int ptime

Since:

4.2.4