TeamTalk 5 C-API DLL: Audio and Video Codecs

TeamTalk 5 C-API DLL Version 5.18A

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Classes | Macros | Typedefs | Enumerations
Audio and Video Codecs

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_OBSOLETE_R4332 = 3 , WEBRTC_AUDIOPREPROCESSOR = 4 }
| 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 1384 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 1451 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 1454 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 1457 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_OBSOLETE_R4332 | WebRTC audio preprocessor used prior to TeamTalk v5.18. Based on WebRTC r4332.
WEBRTC_AUDIOPREPROCESSOR | Use WebRTC's audio preprocessor from WebRTCAudioPreprocessor. https://webrtc.org.

Definition at line 1389 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 1462 of file TeamTalk.h.