This section explains how to configure audio and video codecs. More...

Classes
struct	SpeexCodec
	Speex audio codec settings for Constant Bitrate mode (CBR). More...

struct	SpeexVBRCodec
	Speex audio codec settings for Variable Bitrate mode (VBR). More...

struct	OpusCodec
	OPUS audio codec settings. For detailed information about the OPUS codec check out http://www.opus-codec.org. More...

struct	SpeexDSP
	Speex DSP is used for specifying how recorded audio from a sound input device should be preprocessed before transmission. More...

struct	TTAudioPreprocessor
	Use TeamTalk's internal audio preprocessor for gain audio. Same as used for TT_SetSoundInputGainLevel(). More...

struct	WebRTCAudioPreprocessor
	WebRTC's audio preprocessor. More...

struct	AudioPreprocessor
	Configure the audio preprocessor specified by `nPreprocessor`. More...

struct	WebMVP8Codec
	WebM video codec settings. More...

struct	AudioCodec
	Struct used for specifying which audio codec a channel uses. More...

struct	AudioConfig
	Audio configuration for clients in a channel. More...

struct	VideoCodec
	Struct used for specifying the video codec to use. More...

Macros
#define	SPEEX_NB_MIN_BITRATE 2150
	The minimum bitrate for Speex codec in 8 KHz mode. Bandmode = 0.

#define	SPEEX_NB_MAX_BITRATE 24600
	The maximum bitrate for Speex codec in 8 KHz mode. Bandmode = 0.

#define	SPEEX_WB_MIN_BITRATE 3950
	The minimum bitrate for Speex codec in 16 KHz mode. Bandmode = 1.

#define	SPEEX_WB_MAX_BITRATE 42200
	The maximum bitrate for Speex codec in 16 KHz mode. Bandmode = 1.

#define	SPEEX_UWB_MIN_BITRATE 4150
	The minimum bitrate for Speex codec in 32 KHz mode. Bandmode = 2.

#define	SPEEX_UWB_MAX_BITRATE 44000
	The maximum bitrate for Speex codec in 32 KHz mode. Bandmode = 2.

#define	OPUS_APPLICATION_VOIP 2048
	Audio encoding is for VoIP. This value should be set as nApplicaton in OpusCodec.

#define	OPUS_APPLICATION_AUDIO 2049
	Audio encoding is for music. This value should be set as nApplicaton in OpusCodec.

#define	OPUS_MIN_BITRATE 6000
	The minimum bitrate for OPUS codec. Checkout `nBitRate` of OpusCodec.

#define	OPUS_MAX_BITRATE 510000
	The maximum bitrate for OPUS codec. Checkout `nBitRate` of OpusCodec.

#define	OPUS_MIN_FRAMESIZE 2 /* Actually it's 2.5 */
	The minimum frame size for OPUS codec. Checkout `nFrameSizeMSec` of OpusCodec.

#define	OPUS_MAX_FRAMESIZE 60
	The maximum frame size for OPUS codec. Checkout `nFrameSizeMSec` of OpusCodec.

#define	OPUS_REALMAX_FRAMESIZE 120
	The real maximum frame size for OPUS codec. Checkout `nFrameSizeMSec` of OpusCodec. Although OPUS states it only supports 2.5 - 60 msec, it actually support up to 120 msec.

#define	WEBRTC_GAINCONTROLLER2_FIXEDGAIN_MAX 49.9f
	Max value for fGainDB in WebRTCAudioPreprocessor's `gaincontroller2`.

#define	WEBM_VPX_DL_REALTIME 1
	`nEncodeDeadline` value for fastest encoding.

#define	WEBM_VPX_DL_GOOD_QUALITY 1000000
	`nEncodeDeadline` value for good encoding.

#define	WEBM_VPX_DL_BEST_QUALITY 0
	`nEncodeDeadline` value for best encoding.

Typedefs
typedef struct SpeexCodec	SpeexCodec
	Speex audio codec settings for Constant Bitrate mode (CBR).

typedef struct SpeexVBRCodec	SpeexVBRCodec
	Speex audio codec settings for Variable Bitrate mode (VBR).

typedef struct OpusCodec	OpusCodec
	OPUS audio codec settings. For detailed information about the OPUS codec check out http://www.opus-codec.org.

typedef struct SpeexDSP	SpeexDSP
	Speex DSP is used for specifying how recorded audio from a sound input device should be preprocessed before transmission.

typedef struct TTAudioPreprocessor	TTAudioPreprocessor
	Use TeamTalk's internal audio preprocessor for gain audio. Same as used for TT_SetSoundInputGainLevel().

typedef struct WebRTCAudioPreprocessor	WebRTCAudioPreprocessor
	WebRTC's audio preprocessor.

typedef enum AudioPreprocessorType	AudioPreprocessorType
	The types of supported audio preprocessors.

typedef struct AudioPreprocessor	AudioPreprocessor
	Configure the audio preprocessor specified by `nPreprocessor`.

typedef struct WebMVP8Codec	WebMVP8Codec
	WebM video codec settings.

typedef enum Codec	Codec
	The codecs supported.

typedef struct AudioCodec	AudioCodec
	Struct used for specifying which audio codec a channel uses.

typedef struct AudioConfig	AudioConfig
	Audio configuration for clients in a channel.

typedef struct VideoCodec	VideoCodec
	Struct used for specifying the video codec to use.

Enumerations
enum	AudioPreprocessorType { NO_AUDIOPREPROCESSOR = 0 , SPEEXDSP_AUDIOPREPROCESSOR = 1 , TEAMTALK_AUDIOPREPROCESSOR = 2 , WEBRTC_AUDIOPREPROCESSOR = 3 }
	The types of supported audio preprocessors. More...

enum	Codec { NO_CODEC = 0 , SPEEX_CODEC = 1 , SPEEX_VBR_CODEC = 2 , OPUS_CODEC = 3 , WEBM_VP8_CODEC = 128 }
	The codecs supported. More...

Detailed Description

This section explains how to configure audio and video codecs.

The client is able to encode audio in Speex and OPUS format whereas video can be encoded in WebM format. OPUS is recommended for audio encoding, both for voice and music.

Choosing the right codec settings in an end-user application is very important and proper settings depend entirely on the user scenario. Always ensure that the codec settings do not require too much bandwidth and thereby resulting in packetloss causing inaudible conversations and poor video quality. Detecting packetloss can be done using UserStatistics and ClientStatistics.

Every channel must configure which audio codec to use in order for users to be able to talk to each other. The audiocodec member of Channel specifies which audio codec (AudioCodec) should be used. A channel does not restrict the video codec (VideoCodec) users are using.

Macro Definition Documentation

◆ SPEEX_NB_MIN_BITRATE

#define SPEEX_NB_MIN_BITRATE 2150

The minimum bitrate for Speex codec in 8 KHz mode. Bandmode = 0.

Definition at line 1130 of file TeamTalk.h.

◆ SPEEX_NB_MAX_BITRATE

#define SPEEX_NB_MAX_BITRATE 24600

The maximum bitrate for Speex codec in 8 KHz mode. Bandmode = 0.

Definition at line 1133 of file TeamTalk.h.

◆ SPEEX_WB_MIN_BITRATE

#define SPEEX_WB_MIN_BITRATE 3950

The minimum bitrate for Speex codec in 16 KHz mode. Bandmode = 1.

Definition at line 1136 of file TeamTalk.h.

◆ SPEEX_WB_MAX_BITRATE

#define SPEEX_WB_MAX_BITRATE 42200

The maximum bitrate for Speex codec in 16 KHz mode. Bandmode = 1.

Definition at line 1139 of file TeamTalk.h.

◆ SPEEX_UWB_MIN_BITRATE

#define SPEEX_UWB_MIN_BITRATE 4150

The minimum bitrate for Speex codec in 32 KHz mode. Bandmode = 2.

Definition at line 1142 of file TeamTalk.h.

◆ SPEEX_UWB_MAX_BITRATE

#define SPEEX_UWB_MAX_BITRATE 44000

The maximum bitrate for Speex codec in 32 KHz mode. Bandmode = 2.

Definition at line 1145 of file TeamTalk.h.

◆ OPUS_APPLICATION_VOIP

#define OPUS_APPLICATION_VOIP 2048

Audio encoding is for VoIP. This value should be set as nApplicaton in OpusCodec.

Definition at line 1192 of file TeamTalk.h.

◆ OPUS_APPLICATION_AUDIO

#define OPUS_APPLICATION_AUDIO 2049

Audio encoding is for music. This value should be set as nApplicaton in OpusCodec.

Definition at line 1195 of file TeamTalk.h.

◆ OPUS_MIN_BITRATE

#define OPUS_MIN_BITRATE 6000

The minimum bitrate for OPUS codec. Checkout nBitRate of OpusCodec.

Definition at line 1198 of file TeamTalk.h.

◆ OPUS_MAX_BITRATE

#define OPUS_MAX_BITRATE 510000

The maximum bitrate for OPUS codec. Checkout nBitRate of OpusCodec.

Definition at line 1201 of file TeamTalk.h.

◆ OPUS_MIN_FRAMESIZE

#define OPUS_MIN_FRAMESIZE 2 /* Actually it's 2.5 */

The minimum frame size for OPUS codec. Checkout nFrameSizeMSec of OpusCodec.

Definition at line 1204 of file TeamTalk.h.

◆ OPUS_MAX_FRAMESIZE

#define OPUS_MAX_FRAMESIZE 60

The maximum frame size for OPUS codec. Checkout nFrameSizeMSec of OpusCodec.

Definition at line 1207 of file TeamTalk.h.

◆ OPUS_REALMAX_FRAMESIZE

#define OPUS_REALMAX_FRAMESIZE 120

The real maximum frame size for OPUS codec. Checkout nFrameSizeMSec of OpusCodec. Although OPUS states it only supports 2.5 - 60 msec, it actually support up to 120 msec.

Definition at line 1211 of file TeamTalk.h.

◆ WEBRTC_GAINCONTROLLER2_FIXEDGAIN_MAX

#define WEBRTC_GAINCONTROLLER2_FIXEDGAIN_MAX 49.9f

Max value for fGainDB in WebRTCAudioPreprocessor's gaincontroller2.

Definition at line 1406 of file TeamTalk.h.

◆ WEBM_VPX_DL_REALTIME

#define WEBM_VPX_DL_REALTIME 1

nEncodeDeadline value for fastest encoding.

See also: WebMVP8Codec

Definition at line 1470 of file TeamTalk.h.

◆ WEBM_VPX_DL_GOOD_QUALITY

#define WEBM_VPX_DL_GOOD_QUALITY 1000000

nEncodeDeadline value for good encoding.

See also: WebMVP8Codec

Definition at line 1473 of file TeamTalk.h.

◆ WEBM_VPX_DL_BEST_QUALITY

#define WEBM_VPX_DL_BEST_QUALITY 0

nEncodeDeadline value for best encoding.

See also: WebMVP8Codec

Definition at line 1476 of file TeamTalk.h.

Typedef Documentation

◆ SpeexCodec

typedef struct SpeexCodec SpeexCodec

Speex audio codec settings for Constant Bitrate mode (CBR).

Deprecated:: Use OpusCodec.

See also: SpeexVBRCodec

◆ SpeexVBRCodec

typedef struct SpeexVBRCodec SpeexVBRCodec

Speex audio codec settings for Variable Bitrate mode (VBR).

Deprecated:: Use OpusCodec.

◆ OpusCodec

typedef struct OpusCodec OpusCodec

OPUS audio codec settings. For detailed information about the OPUS codec check out http://www.opus-codec.org.

◆ SpeexDSP

typedef struct SpeexDSP SpeexDSP

Speex DSP is used for specifying how recorded audio from a sound input device should be preprocessed before transmission.

Deprecated:: Use WebRTCAudioPreprocessor.

Users' audio levels may be diffent due to how their microphone is configured in their OS. Automatic Gain Control (AGC) can be used to ensure all users in the same channel have the same audio level.

Enable the preprocessing configuration by calling TT_SetSoundInputPreprocess().

When joining a Channel and bEnableGainControl of AudioConfig is enabled in the channel then enable sound input preprocessing by setting bEnableAGC to TRUE and nGainLevel of SpeexDSP to the nGainLevel of AudioConfig.

◆ TTAudioPreprocessor

typedef struct TTAudioPreprocessor TTAudioPreprocessor

Use TeamTalk's internal audio preprocessor for gain audio. Same as used for TT_SetSoundInputGainLevel().

◆ WebRTCAudioPreprocessor

typedef struct WebRTCAudioPreprocessor WebRTCAudioPreprocessor

WebRTC's audio preprocessor.

Use WebRTC's audio preprocessor, https://webrtc.org

Note that WebRTC's can only operate on 10 msec audio frame, so nTxIntervalMSec in AudioCodec must a multiple of 10.

WebRTCAudioPreprocessor is recommended to TT_SetSoundDeviceEffects() on desktop platforms.

Activate WebRTCAudioPreprocessor by calling TT_SetSoundInputPreprocessEx().

◆ AudioPreprocessorType

typedef enum AudioPreprocessorType AudioPreprocessorType

The types of supported audio preprocessors.

See also: TT_InitLocalPlayback()

◆ AudioPreprocessor

typedef struct AudioPreprocessor AudioPreprocessor

Configure the audio preprocessor specified by nPreprocessor.

◆ WebMVP8Codec

typedef struct WebMVP8Codec WebMVP8Codec

WebM video codec settings.

See also: VideoCodec; TT_InitVideoCaptureDevice; TT_StartStreamingMediaFileToChannel()

◆ Codec

typedef enum Codec Codec

The codecs supported.

See also: AudioCodec; VideoCodec

◆ AudioCodec

typedef struct AudioCodec AudioCodec

Struct used for specifying which audio codec a channel uses.

◆ AudioConfig

typedef struct AudioConfig AudioConfig

Audio configuration for clients in a channel.

An audio configuration can be used to set common audio properties for all users in a channel. Checkout audiocfg of Channel.

The audio configuration only supports same audio level for all users by manually converting the values to the selected AudioPreprocessor.

See also: TT_SetSoundInputPreprocessEx(); TT_DoMakeChannel(); TT_DoJoinChannel()

◆ VideoCodec

typedef struct VideoCodec VideoCodec

Struct used for specifying the video codec to use.

Enumeration Type Documentation

◆ AudioPreprocessorType

enum AudioPreprocessorType

The types of supported audio preprocessors.

See also: TT_InitLocalPlayback()

Enumerator
NO_AUDIOPREPROCESSOR	Value for specifying that no audio preprocessing should occur.
SPEEXDSP_AUDIOPREPROCESSOR	Use the SpeexDSP audio preprocessor.
TEAMTALK_AUDIOPREPROCESSOR	Use TeamTalk's internal audio preprocessor TTAudioPreprocessor.
WEBRTC_AUDIOPREPROCESSOR	Use WebRTC's audio preprocessor from WebRTCAudioPreprocessor. https://webrtc.org.

Definition at line 1411 of file TeamTalk.h.

◆ Codec

enum Codec

The codecs supported.

See also: AudioCodec; VideoCodec

Enumerator
NO_CODEC	No codec specified.
SPEEX_CODEC	Speex audio codec, http://www.speex.org. See also SpeexCodec
SPEEX_VBR_CODEC	Speex audio codec in VBR mode, http://www.speex.org. See also SpeexVBRCodec
OPUS_CODEC	OPUS audio codec. See also OpusCodec
WEBM_VP8_CODEC	WebM video codec. See also WebMVP8Codec

Definition at line 1481 of file TeamTalk.h.

Classes

Macros

Typedefs

Enumerations

Detailed Description

Macro Definition Documentation

◆ SPEEX_NB_MIN_BITRATE

◆ SPEEX_NB_MAX_BITRATE

◆ SPEEX_WB_MIN_BITRATE

◆ SPEEX_WB_MAX_BITRATE

◆ SPEEX_UWB_MIN_BITRATE

◆ SPEEX_UWB_MAX_BITRATE

◆ OPUS_APPLICATION_VOIP

◆ OPUS_APPLICATION_AUDIO

◆ OPUS_MIN_BITRATE

◆ OPUS_MAX_BITRATE

◆ OPUS_MIN_FRAMESIZE

◆ OPUS_MAX_FRAMESIZE

◆ OPUS_REALMAX_FRAMESIZE

◆ WEBRTC_GAINCONTROLLER2_FIXEDGAIN_MAX

◆ WEBM_VPX_DL_REALTIME

◆ WEBM_VPX_DL_GOOD_QUALITY

◆ WEBM_VPX_DL_BEST_QUALITY

Typedef Documentation

◆ SpeexCodec

◆ SpeexVBRCodec

◆ OpusCodec

◆ SpeexDSP

◆ TTAudioPreprocessor

◆ WebRTCAudioPreprocessor

◆ AudioPreprocessorType

◆ AudioPreprocessor

◆ WebMVP8Codec

◆ Codec

◆ AudioCodec

◆ AudioConfig

◆ VideoCodec

Enumeration Type Documentation

◆ AudioPreprocessorType

◆ Codec