mirror of
https://github.com/FWGS/xash3d-fwgs.git
synced 2026-08-05 11:35:05 +08:00
6bb7bcd378
- Added support for GoldSrc voice mode, including initialization and shutdown functions. - Implemented data size verification for the GoldSrc voice package. - Creation and processing of voice packets, CRC verification of data integrity. - Added a Steam ID field to the client_static_t structure - Automatic gain control has been introduced for voice samples. - The voice status structure has been updated. - Refactored voice data handling for better type consistency and maintainability. - Fixed ISO C90 compliance issues for better portability. - Fixed decoder index usage.
1220 lines
29 KiB
C
1220 lines
29 KiB
C
/*
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voice.c - voice chat implementation
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Copyright (C) 2022 Velaron
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Copyright (C) 2022 SNMetamorph
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This program is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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*/
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#define CUSTOM_MODES 1 // required to correctly link with Opus Custom
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#include <opus_custom.h>
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#include <opus.h>
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#include "common.h"
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#include "client.h"
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#include "voice.h"
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#include "crclib.h"
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voice_state_t voice = { 0 };
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CVAR_DEFINE_AUTO( voice_enable, "1", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "enable voice chat" );
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CVAR_DEFINE_AUTO( voice_loopback, "0", FCVAR_PRIVILEGED, "loopback voice back to the speaker" );
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CVAR_DEFINE_AUTO( voice_scale, "1.0", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "incoming voice volume scale" );
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CVAR_DEFINE_AUTO( voice_transmit_scale, "1.0", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "outcoming voice volume scale" );
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CVAR_DEFINE_AUTO( voice_avggain, "0.5", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "automatic voice gain control (average)" );
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CVAR_DEFINE_AUTO( voice_maxgain, "5.0", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "automatic voice gain control (maximum)" );
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CVAR_DEFINE_AUTO( voice_inputfromfile, "0", FCVAR_PRIVILEGED, "input voice from voice_input.wav" );
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static void Voice_ApplyGainAdjust( int16_t *samples, int count, float scale );
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static qboolean Voice_InitGoldSrcMode( int quality );
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static qboolean Voice_InitOpusCustomMode( int quality );
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static void Voice_ShutdownGoldSrcMode( void );
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static void Voice_ShutdownOpusCustomMode( void );
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static uint Voice_GetOpusCompressedData( byte *out, uint maxsize, uint *frames );
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static uint Voice_GetGSCompressedData( byte *out, uint maxsize, uint *frames );
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static int Voice_ProcessGSData( int ent, const uint8_t *data, uint32_t size );
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static uint Voice_CreateGSVoicePacket( byte *out, const byte *voice_data, uint voice_size );
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static void Voice_StartChannel( uint samples, byte *data, int entnum );
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static void Voice_Status( int entindex, qboolean bTalking );
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static void Voice_StatusTimeout( voice_status_t *status, int entindex, double frametime );
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static void Voice_Shutdown( void );
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/*
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===============================================================================
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UTILITY FUNCTIONS
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===============================================================================
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*/
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/*
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=========================
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Voice_IsGoldSrcMode
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Check if codec is GoldSrc mode
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=========================
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*/
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qboolean Voice_IsGoldSrcMode( const char *codec )
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{
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return ( COM_CheckString( codec ) == 0 || Q_strstr( codec, "voice_speex" ) != NULL );
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}
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/*
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=========================
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Voice_IsOpusCustomMode
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Check if codec is Opus Custom mode
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=========================
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*/
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qboolean Voice_IsOpusCustomMode( const char *codec )
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{
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return Q_strcmp( codec, VOICE_OPUS_CUSTOM_CODEC ) == 0;
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}
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/*
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=========================
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Voice_GetBitrateForQuality
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Get bitrate for given quality level
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=========================
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*/
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int Voice_GetBitrateForQuality( int quality, qboolean goldsrc )
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{
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if( goldsrc )
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{
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switch( quality )
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{
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case 1: return 6000; // 6 kbps
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case 2: return 12000; // 12 kbps
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case 3: return 24000; // 24 kbps
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case 4: return 36000; // 36 kbps
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case 5: return 48000; // 48 kbps
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default: return 36000; // default
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}
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}
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else
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{
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switch( quality )
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{
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case 1: return 6000; // 6 kbps
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case 2: return 12000; // 12 kbps
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case 4: return 64000; // 64 kbps
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case 5: return 96000; // 96 kbps
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default: return 36000; // default
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}
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}
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}
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/*
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===============================================================================
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OPUS INTEGRATION
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===============================================================================
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*/
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/*
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=========================
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Voice_InitCustomMode
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Initialize Opus Custom mode
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=========================
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*/
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static qboolean Voice_InitCustomMode( void )
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{
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int err = 0;
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voice.width = sizeof( int16_t );
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voice.samplerate = VOICE_OPUS_CUSTOM_SAMPLERATE;
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voice.frame_size = VOICE_OPUS_CUSTOM_FRAME_SIZE;
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voice.custom_mode = opus_custom_mode_create( SOUND_44k, voice.frame_size, &err );
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if( !voice.custom_mode )
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{
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Con_Printf( S_ERROR "Can't create Opus Custom mode: %s\n", opus_strerror( err ));
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return false;
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}
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return true;
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}
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/*
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=========================
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Voice_InitOpusDecoder
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Initialize Opus decoders for clients
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=========================
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*/
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static qboolean Voice_InitOpusDecoder( void )
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{
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int err = 0;
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for( int i = 0; i < cl.maxclients; i++ )
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{
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if( voice.goldsrc )
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{
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voice.gs_decoders[i] = opus_decoder_create( GS_DEFAULT_SAMPLE_RATE, VOICE_PCM_CHANNELS, &err );
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if( !voice.gs_decoders[i] )
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{
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Con_Printf( S_ERROR "Can't create GoldSrc Opus decoder for %i: %s\n", i, opus_strerror( err ));
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return false;
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}
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}
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else
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{
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voice.decoders[i] = opus_custom_decoder_create( voice.custom_mode, VOICE_PCM_CHANNELS, &err );
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if( !voice.decoders[i] )
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{
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Con_Printf( S_ERROR "Can't create Custom Opus decoder for %i: %s\n", i, opus_strerror( err ));
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return false;
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}
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}
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}
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return true;
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}
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/*
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=========================
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Voice_InitOpusEncoder
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Initialize Opus encoder with quality settings
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=========================
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*/
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static qboolean Voice_InitOpusEncoder( int quality )
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{
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int err = 0;
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int bitrate = Voice_GetBitrateForQuality( quality, voice.goldsrc );
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if (voice.goldsrc)
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{
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voice.gs_encoder = opus_encoder_create( GS_DEFAULT_SAMPLE_RATE, VOICE_PCM_CHANNELS, OPUS_APPLICATION_VOIP, &err );
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if( !voice.gs_encoder )
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{
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Con_Printf( S_ERROR "Can't create GoldSrc Opus encoder: %s\n", opus_strerror( err ));
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return false;
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}
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opus_encoder_ctl( voice.gs_encoder, OPUS_SET_DTX( 1 ));
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opus_encoder_ctl( voice.gs_encoder, OPUS_SET_BITRATE( bitrate ));
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}
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else
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{
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voice.encoder = opus_custom_encoder_create( voice.custom_mode, VOICE_PCM_CHANNELS, &err );
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if( !voice.encoder )
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{
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Con_Printf( S_ERROR "Can't create Opus encoder: %s\n", opus_strerror( err ));
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return false;
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}
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( bitrate ));
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}
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return true;
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}
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/*
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=========================
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Voice_ShutdownOpusDecoder
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Cleanup Opus decoders
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=========================
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*/
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static void Voice_ShutdownOpusDecoder( void )
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{
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for( int i = 0; i < MAX_CLIENTS; i++ )
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{
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if( voice.goldsrc )
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{
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if( voice.gs_decoders[i] )
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{
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opus_decoder_destroy( voice.gs_decoders[i] );
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voice.gs_decoders[i] = NULL;
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}
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}
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else
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{
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if( voice.decoders[i] )
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{
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opus_custom_decoder_destroy( voice.decoders[i] );
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voice.decoders[i] = NULL;
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}
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}
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}
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}
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/*
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=========================
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Voice_ShutdownOpusEncoder
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Cleanup Opus encoder
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=========================
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*/
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static void Voice_ShutdownOpusEncoder( void )
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{
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if( voice.goldsrc )
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{
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if( voice.gs_encoder )
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{
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opus_encoder_destroy( voice.gs_encoder );
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voice.gs_encoder = NULL;
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}
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}
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else
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{
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if( voice.encoder )
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{
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opus_custom_encoder_destroy( voice.encoder );
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voice.encoder = NULL;
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}
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}
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}
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/*
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=========================
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Voice_ShutdownCustomMode
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Cleanup Opus Custom mode
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=========================
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*/
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static void Voice_ShutdownCustomMode( void )
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{
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if( voice.custom_mode )
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{
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opus_custom_mode_destroy( voice.custom_mode );
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voice.custom_mode = NULL;
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}
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}
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/*
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=========================
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Voice_InitGoldSrcMode
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Initialize GoldSrc voice mode
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=========================
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*/
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static qboolean Voice_InitGoldSrcMode( int quality )
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{
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voice.goldsrc = true;
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voice.autogain.block_size = 128;
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voice.width = sizeof( int16_t );
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voice.samplerate = GS_DEFAULT_SAMPLE_RATE;
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voice.frame_size = GS_DEFAULT_FRAME_SIZE;
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if( !Voice_InitOpusDecoder() )
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{
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Con_Printf( S_ERROR "Can't create GoldSrc decoders, voice chat is disabled.\n" );
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return false;
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}
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if( !Voice_InitOpusEncoder( quality ) )
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{
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Con_Printf( S_WARN "Other players will not be able to hear you.\n" );
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return false;
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}
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return true;
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}
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/*
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=========================
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Voice_InitOpusCustomMode
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Initialize Opus Custom voice mode
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=========================
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*/
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static qboolean Voice_InitOpusCustomMode( int quality )
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{
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voice.goldsrc = false;
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voice.autogain.block_size = 128;
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voice.samplerate = VOICE_OPUS_CUSTOM_SAMPLERATE;
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voice.frame_size = VOICE_OPUS_CUSTOM_FRAME_SIZE;
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voice.width = sizeof( int16_t );
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if( !Voice_InitCustomMode() || !Voice_InitOpusDecoder() )
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{
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Con_Printf( S_ERROR "Can't create Opus Custom decoders, voice chat is disabled.\n" );
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return false;
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}
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if( !Voice_InitOpusEncoder( quality ) )
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{
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Con_Printf( S_WARN "Other players will not be able to hear you.\n" );
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return false;
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}
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return true;
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}
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/*
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=========================
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Voice_ShutdownGoldSrcMode
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Cleanup GoldSrc mode
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=========================
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*/
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static void Voice_ShutdownGoldSrcMode( void )
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{
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Voice_ShutdownOpusDecoder();
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Voice_ShutdownOpusEncoder();
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}
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/*
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=========================
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Voice_ShutdownOpusCustomMode
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Cleanup Opus Custom mode
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=========================
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*/
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static void Voice_ShutdownOpusCustomMode( void )
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{
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Voice_ShutdownOpusDecoder();
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Voice_ShutdownOpusEncoder();
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Voice_ShutdownCustomMode();
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}
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/*
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===============================================================================
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VOICE PROCESSING
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===============================================================================
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*/
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/*
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=========================
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Voice_ApplyGainAdjust
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Apply automatic gain control to voice samples
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=========================
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*/
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static void Voice_ApplyGainAdjust( int16_t *samples, int count, float scale )
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{
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float gain, modifiedMax;
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int average, blockOffset = 0;
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for( ;; )
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{
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int i, localMax = 0, localSum = 0;
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int blockSize = Q_min(count - blockOffset, voice.autogain.block_size);
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if( blockSize < 1 )
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break;
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for( i = 0; i < blockSize; ++i )
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{
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int sample = samples[blockOffset + i];
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int absSample = abs( sample );
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if( absSample > localMax )
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localMax = absSample;
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localSum += absSample;
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gain = voice.autogain.current_gain + i * voice.autogain.gain_multiplier;
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samples[blockOffset + i] = bound( SHRT_MIN, (int)( sample * gain ), SHRT_MAX );
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}
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if( blockOffset % voice.autogain.block_size == 0 )
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{
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average = localSum / blockSize;
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modifiedMax = average + ( localMax - average ) * voice_avggain.value;
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voice.autogain.current_gain = voice.autogain.next_gain * scale;
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voice.autogain.next_gain = Q_min((float)SHRT_MAX / (modifiedMax > 1 ? modifiedMax : 1), voice_maxgain.value) * scale;
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if (blockSize > 1)
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voice.autogain.gain_multiplier = (voice.autogain.next_gain - voice.autogain.current_gain) / (blockSize - 1);
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else
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voice.autogain.gain_multiplier = 0.0f;
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}
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blockOffset += blockSize;
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}
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}
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/*
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=========================
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Voice_GetOpusCompressedData
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Get compressed voice data for Opus Custom mode
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=========================
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*/
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static uint Voice_GetOpusCompressedData( byte *out, uint maxsize, uint *frames )
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{
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uint ofs = 0, size = 0;
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uint frame_size_bytes = voice.frame_size * voice.width;
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if( voice.input_file )
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{
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uint numbytes;
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double updateInterval, curtime = Sys_DoubleTime();
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updateInterval = curtime - voice.start_time;
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voice.start_time = curtime;
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numbytes = updateInterval * voice.samplerate * voice.width * VOICE_PCM_CHANNELS;
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numbytes = Q_min( numbytes, voice.input_file->size - voice.input_file_pos );
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numbytes = Q_min( numbytes, sizeof( voice.input_buffer ) - voice.input_buffer_pos );
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memcpy( voice.input_buffer + voice.input_buffer_pos, voice.input_file->buffer + voice.input_file_pos, numbytes );
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voice.input_buffer_pos += numbytes;
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voice.input_file_pos += numbytes;
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}
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if( !voice.input_file )
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VoiceCapture_Lock( true );
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for( ofs = 0; voice.input_buffer_pos - ofs >= frame_size_bytes && ofs <= voice.input_buffer_pos; ofs += frame_size_bytes )
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{
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int bytes;
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if( !voice.input_file )
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Voice_ApplyGainAdjust((int16_t*)(voice.input_buffer + ofs), voice.frame_size, voice_transmit_scale.value);
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bytes = opus_custom_encode( voice.encoder, (const int16_t *)( voice.input_buffer + ofs ),
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voice.frame_size, out + size + sizeof( uint16_t ), maxsize );
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if( bytes > 0 )
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{
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// write compressed frame size
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*((uint16_t *)&out[size]) = LittleShort(bytes);
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size += bytes + sizeof( uint16_t );
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maxsize -= bytes + sizeof( uint16_t );
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(*frames)++;
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}
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else
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{
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Con_Printf( S_ERROR "%s: failed to encode frame: %s\n", __func__, opus_strerror( bytes ));
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}
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}
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// did we compress anything? update counters
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if( ofs )
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{
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fs_offset_t remaining = voice.input_buffer_pos - ofs;
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// move remaining samples to the beginning of buffer
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memmove( voice.input_buffer, voice.input_buffer + ofs, remaining );
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voice.input_buffer_pos = remaining;
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}
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if( !voice.input_file )
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VoiceCapture_Lock( false );
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return size;
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}
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/*
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=========================
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Voice_GetGSCompressedData
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Get compressed voice data for GoldSrc mode
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=========================
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*/
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static uint Voice_GetGSCompressedData( byte *out, uint maxsize, uint *frames )
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{
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uint ofs = 0, size = 0;
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const uint frame_size_samples = GS_DEFAULT_FRAME_SIZE;
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const uint frame_size_bytes = GS_DEFAULT_FRAME_SIZE * voice.width;
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static uint16_t sequence = 0;
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if( voice.input_file )
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{
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uint numbytes;
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double updateInterval, curtime = Sys_DoubleTime();
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updateInterval = curtime - voice.start_time;
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voice.start_time = curtime;
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numbytes = updateInterval * voice.samplerate * voice.width * VOICE_PCM_CHANNELS;
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numbytes = Q_min( numbytes, voice.input_file->size - voice.input_file_pos );
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numbytes = Q_min( numbytes, sizeof( voice.input_buffer ) - voice.input_buffer_pos );
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memcpy( voice.input_buffer + voice.input_buffer_pos, voice.input_file->buffer + voice.input_file_pos, numbytes );
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voice.input_buffer_pos += numbytes;
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voice.input_file_pos += numbytes;
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}
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|
if( !voice.input_file )
|
|
VoiceCapture_Lock( true );
|
|
|
|
*frames = 0;
|
|
|
|
while (voice.input_buffer_pos - ofs >= frame_size_bytes)
|
|
{
|
|
int bytes;
|
|
|
|
int is_silence = 1;
|
|
int16_t *samples = (int16_t *)(voice.input_buffer + ofs);
|
|
for (uint i = 0; i < frame_size_samples; ++i) {
|
|
if (samples[i] != 0) {
|
|
is_silence = 0;
|
|
break;
|
|
}
|
|
}
|
|
if (is_silence) {
|
|
ofs += frame_size_bytes;
|
|
continue;
|
|
}
|
|
|
|
if ( !voice.input_file )
|
|
Voice_ApplyGainAdjust(samples, frame_size_samples, voice_transmit_scale.value);
|
|
|
|
bytes = opus_encode( voice.gs_encoder, samples,
|
|
frame_size_samples, out + size + 4, maxsize - 4 );
|
|
|
|
if( bytes > 0 )
|
|
{
|
|
*(uint16_t*)(out + size) = LittleShort(bytes);
|
|
*(uint16_t*)(out + size + 2) = LittleShort(sequence++);
|
|
|
|
size += bytes + 4;
|
|
maxsize -= bytes + 4;
|
|
|
|
(*frames)++;
|
|
}
|
|
else
|
|
{
|
|
Con_Printf( S_ERROR "%s: failed to encode frame: %s\n", __func__, opus_strerror( bytes ));
|
|
}
|
|
|
|
ofs += frame_size_bytes;
|
|
}
|
|
|
|
if( ofs )
|
|
{
|
|
fs_offset_t remaining = voice.input_buffer_pos - ofs;
|
|
memmove( voice.input_buffer, voice.input_buffer + ofs, remaining );
|
|
voice.input_buffer_pos = remaining;
|
|
}
|
|
|
|
if( !voice.input_file )
|
|
VoiceCapture_Lock( false );
|
|
|
|
return size;
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_ProcessGSData
|
|
|
|
Process GoldSrc voice data and return number of samples
|
|
=========================
|
|
*/
|
|
static int Voice_ProcessGSData( int ent, const uint8_t *data, uint32_t size )
|
|
{
|
|
uint32_t crc_in_packet;
|
|
uint32_t crc;
|
|
size_t offset;
|
|
int16_t pcm[GS_MAX_DECOMPRESSED_SAMPLES];
|
|
size_t output_samples;
|
|
uint16_t sample_rate;
|
|
uint8_t vpc_type;
|
|
uint16_t data_len;
|
|
OpusDecoder *decoder;
|
|
size_t opus_offset;
|
|
int decoded;
|
|
size_t silence_samples;
|
|
uint16_t frame_size;
|
|
|
|
if (!data || size < 18 || size < 4 || ent <= 0 || ent > cl.maxclients)
|
|
return 0;
|
|
|
|
crc_in_packet = LittleLong(*(uint32_t*)(data + size - 4));
|
|
crc = CRC32_INIT_VALUE;
|
|
CRC32_ProcessBuffer(&crc, data, size - 4);
|
|
crc = CRC32_Final(crc);
|
|
|
|
if (crc != crc_in_packet) {
|
|
Con_Printf( S_WARN "Voice packet CRC32 mismatch\n" );
|
|
return 0;
|
|
}
|
|
|
|
offset = 8;
|
|
output_samples = 0;
|
|
|
|
if (offset >= size - 4 || data[offset] != GS_VPC_SETSAMPLERATE) {
|
|
Con_Printf( S_WARN "Invalid voice packet type: %d\n", data[offset] );
|
|
return 0;
|
|
}
|
|
offset++;
|
|
|
|
if (offset + 4 > size - 4)
|
|
return 0;
|
|
|
|
sample_rate = LittleShort(*(uint16_t*)(data + offset));
|
|
offset += 2;
|
|
|
|
vpc_type = data[offset++];
|
|
data_len = LittleShort(*(uint16_t*)(data + offset));
|
|
offset += 2;
|
|
|
|
if (offset + data_len > size - 4) {
|
|
Con_Printf( S_WARN "Voice packet data_len out of bounds\n" );
|
|
return 0;
|
|
}
|
|
|
|
if (vpc_type == GS_VPC_VDATA_OPUS_PLC) {
|
|
decoder = voice.gs_decoders[ent];
|
|
if (!decoder) {
|
|
Con_Printf( S_WARN "No decoder available for entity %d\n", ent );
|
|
return 0;
|
|
}
|
|
opus_offset = 0;
|
|
while (opus_offset + 4 <= data_len) {
|
|
frame_size = LittleShort(*(uint16_t*)(data + offset + opus_offset));
|
|
opus_offset += 4;
|
|
|
|
if (frame_size == 0) {
|
|
if (output_samples + 160 > GS_MAX_DECOMPRESSED_SAMPLES) {
|
|
Con_Printf( S_WARN "Voice buffer overflow\n" );
|
|
return 0;
|
|
}
|
|
memset(pcm + output_samples, 0, 160 * sizeof(int16_t));
|
|
output_samples += 160;
|
|
continue;
|
|
}
|
|
if (frame_size == 0xFFFF) {
|
|
opus_decoder_ctl(decoder, OPUS_RESET_STATE);
|
|
break;
|
|
}
|
|
if (opus_offset + frame_size > data_len) {
|
|
Con_Printf( S_WARN "Opus frame size exceeds data length\n" );
|
|
return 0;
|
|
}
|
|
decoded = opus_decode(decoder, data + offset + opus_offset, frame_size,
|
|
pcm + output_samples, GS_DEFAULT_FRAME_SIZE, 0);
|
|
if (decoded < 0) {
|
|
Con_Printf( S_WARN "Opus decode error: %s\n", opus_strerror(decoded) );
|
|
return 0;
|
|
}
|
|
output_samples += decoded;
|
|
opus_offset += frame_size;
|
|
}
|
|
} else if (vpc_type == GS_VPC_VDATA_SILENCE) {
|
|
silence_samples = data_len / 2;
|
|
if (silence_samples > GS_MAX_DECOMPRESSED_SAMPLES) {
|
|
Con_Printf( S_WARN "Silence data too large\n" );
|
|
return 0;
|
|
}
|
|
memset(pcm, 0, silence_samples * sizeof(int16_t));
|
|
output_samples = silence_samples;
|
|
} else {
|
|
Con_Printf( S_WARN "Unsupported voice data type: %d\n", vpc_type );
|
|
return 0;
|
|
}
|
|
|
|
if (output_samples > 0)
|
|
Voice_StartChannel(output_samples, (byte*)pcm, ent);
|
|
|
|
return output_samples;
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_CreateGSVoicePacket
|
|
|
|
Create GoldSrc voice packet
|
|
=========================
|
|
*/
|
|
static uint Voice_CreateGSVoicePacket( byte *out, const byte *voice_data, uint voice_size )
|
|
{
|
|
uint offset = 0;
|
|
uint32_t crc;
|
|
|
|
if (!out || !voice_data || voice_size == 0)
|
|
{
|
|
Con_Printf( S_WARN "Invalid voice data for packet creation\n" );
|
|
return 0;
|
|
}
|
|
|
|
if ( cls.steamid ) {
|
|
memcpy(out + offset, cls.steamid, 8);
|
|
} else {
|
|
memset(out + offset, 0, 8); // fallback: 0
|
|
}
|
|
|
|
offset += 8;
|
|
|
|
out[offset] = GS_VPC_SETSAMPLERATE;
|
|
offset += 1;
|
|
|
|
*(uint16_t*)(out + offset) = LittleShort(GS_DEFAULT_SAMPLE_RATE);
|
|
offset += 2;
|
|
|
|
out[offset] = GS_VPC_VDATA_OPUS_PLC;
|
|
offset += 1;
|
|
|
|
*(uint16_t*)(out + offset) = LittleShort(voice_size);
|
|
offset += 2;
|
|
|
|
memcpy( out + offset, voice_data, voice_size );
|
|
offset += voice_size;
|
|
|
|
crc = CRC32_INIT_VALUE;
|
|
CRC32_ProcessBuffer( &crc, out, offset );
|
|
crc = CRC32_Final( crc );
|
|
*(uint32_t*)(out + offset) = LittleLong(crc);
|
|
offset += 4;
|
|
|
|
return offset;
|
|
}
|
|
|
|
/*
|
|
===============================================================================
|
|
|
|
VOICE CHAT INTEGRATION
|
|
|
|
===============================================================================
|
|
*/
|
|
|
|
/*
|
|
=========================
|
|
Voice_Status
|
|
|
|
Notify user dll about voice transmission
|
|
=========================
|
|
*/
|
|
static void Voice_Status( int entindex, qboolean bTalking )
|
|
{
|
|
if( cls.state == ca_active && clgame.dllFuncs.pfnVoiceStatus )
|
|
clgame.dllFuncs.pfnVoiceStatus( entindex, bTalking );
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_StatusTimeout
|
|
|
|
Waits few milliseconds and if there was no
|
|
voice transmission, sends notification
|
|
=========================
|
|
*/
|
|
static void Voice_StatusTimeout( voice_status_t *status, int entindex, double frametime )
|
|
{
|
|
if( status->talking_ack )
|
|
{
|
|
status->talking_timeout += frametime;
|
|
if( status->talking_timeout > 0.2 )
|
|
{
|
|
status->talking_ack = false;
|
|
Voice_Status( entindex, false );
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_StatusAck
|
|
|
|
Sends notification to user dll and
|
|
zeroes timeouts for this client
|
|
=========================
|
|
*/
|
|
void Voice_StatusAck( voice_status_t *status, int playerIndex )
|
|
{
|
|
if( !status->talking_ack )
|
|
Voice_Status( playerIndex, true );
|
|
|
|
status->talking_ack = true;
|
|
status->talking_timeout = 0.0;
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_IsRecording
|
|
|
|
Check if voice is currently recording
|
|
=========================
|
|
*/
|
|
qboolean Voice_IsRecording( void )
|
|
{
|
|
return voice.is_recording;
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_RecordStop
|
|
|
|
Stop voice recording
|
|
=========================
|
|
*/
|
|
void Voice_RecordStop( void )
|
|
{
|
|
if( voice.input_file )
|
|
{
|
|
FS_FreeSound( voice.input_file );
|
|
voice.input_file = NULL;
|
|
}
|
|
|
|
VoiceCapture_Activate( false );
|
|
voice.is_recording = false;
|
|
|
|
Voice_Status( VOICE_LOCALCLIENT_INDEX, false );
|
|
|
|
voice.input_buffer_pos = 0;
|
|
memset( voice.input_buffer, 0, sizeof( voice.input_buffer ));
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_RecordStart
|
|
|
|
Start voice recording
|
|
=========================
|
|
*/
|
|
void Voice_RecordStart( void )
|
|
{
|
|
Voice_RecordStop();
|
|
|
|
if( !voice.initialized )
|
|
return;
|
|
|
|
if( voice_inputfromfile.value )
|
|
{
|
|
voice.input_file = FS_LoadSound( "voice_input.wav", NULL, 0 );
|
|
|
|
if( voice.input_file )
|
|
{
|
|
Sound_Process( &voice.input_file, voice.samplerate, voice.width, VOICE_PCM_CHANNELS, SOUND_RESAMPLE );
|
|
voice.input_file_pos = 0;
|
|
|
|
voice.start_time = Sys_DoubleTime();
|
|
voice.is_recording = true;
|
|
}
|
|
else
|
|
{
|
|
FS_FreeSound( voice.input_file );
|
|
voice.input_file = NULL;
|
|
}
|
|
}
|
|
|
|
if( !Voice_IsRecording( ) && voice.device_opened )
|
|
voice.is_recording = VoiceCapture_Activate( true );
|
|
|
|
if( Voice_IsRecording() )
|
|
Voice_Status( VOICE_LOCALCLIENT_INDEX, true );
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_Disconnect
|
|
|
|
We're disconnected from server
|
|
stop recording and notify user dlls
|
|
=========================
|
|
*/
|
|
void Voice_Disconnect( void )
|
|
{
|
|
int i;
|
|
|
|
Voice_RecordStop();
|
|
|
|
if( voice.local.talking_ack )
|
|
{
|
|
Voice_Status( VOICE_LOOPBACK_INDEX, false );
|
|
voice.local.talking_ack = false;
|
|
}
|
|
|
|
for( i = 0; i < MAX_CLIENTS; i++ )
|
|
{
|
|
if( voice.players_status[i].talking_ack )
|
|
{
|
|
Voice_Status( i, false );
|
|
voice.players_status[i].talking_ack = false;
|
|
}
|
|
}
|
|
|
|
VoiceCapture_Shutdown();
|
|
voice.device_opened = false;
|
|
|
|
|
|
Voice_ShutdownOpusDecoder();
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_StartChannel
|
|
|
|
Feed the decoded data to engine sound subsystem
|
|
=========================
|
|
*/
|
|
static void Voice_StartChannel( uint samples, byte *data, int entnum )
|
|
{
|
|
SND_ForceInitMouth( entnum );
|
|
S_RawEntSamples( entnum, samples, voice.samplerate, voice.width, VOICE_PCM_CHANNELS, data, bound( 0, 255 * voice_scale.value, 255 ));
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_AddIncomingData
|
|
|
|
Received encoded voice data, decode it
|
|
=========================
|
|
*/
|
|
void Voice_AddIncomingData( int ent, const byte *data, uint size, uint frames )
|
|
{
|
|
const int playernum = ent - 1;
|
|
int samples = 0;
|
|
int ofs = 0;
|
|
|
|
if( !voice.initialized || !voice_enable.value )
|
|
return;
|
|
|
|
// must notify through as both local player and normal client
|
|
if( ent + 1 == cl.playernum + 1 )
|
|
Voice_StatusAck( &voice.local, VOICE_LOOPBACK_INDEX );
|
|
|
|
Voice_StatusAck( &voice.players_status[ent], ent );
|
|
|
|
if( voice.goldsrc )
|
|
{
|
|
// Voice_ProcessGSData handles Voice_StartChannel internally
|
|
Voice_ProcessGSData( ent, (const uint8_t *)data, size );
|
|
}
|
|
else
|
|
{
|
|
// Validate player index and decoder
|
|
if( playernum < 0 || playernum >= cl.maxclients || !voice.decoders[playernum] )
|
|
return;
|
|
|
|
// decode frame by frame
|
|
for( ;; )
|
|
{
|
|
int frame_samples;
|
|
uint16_t compressed_size;
|
|
|
|
// no compressed size mark
|
|
if( ofs + sizeof( uint16_t ) > size )
|
|
break;
|
|
|
|
compressed_size = *(const uint16_t *)(data + ofs);
|
|
ofs += sizeof( uint16_t );
|
|
|
|
// no frame data
|
|
if( ofs + compressed_size > size )
|
|
break;
|
|
|
|
frame_samples = opus_custom_decode( voice.decoders[playernum], data + ofs, compressed_size,
|
|
(int16_t*)voice.decompress_buffer + samples, voice.frame_size );
|
|
|
|
ofs += compressed_size;
|
|
samples += frame_samples;
|
|
}
|
|
|
|
if( samples > 0 )
|
|
Voice_StartChannel( samples, voice.decompress_buffer, ent );
|
|
}
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
CL_AddVoiceToDatagram
|
|
|
|
Encode our voice data and send it to server
|
|
=========================
|
|
*/
|
|
void CL_AddVoiceToDatagram( void )
|
|
{
|
|
byte buffer[VOICE_MAX_DATA_SIZE];
|
|
uint size, frames;
|
|
|
|
if( cls.state != ca_active || !voice.device_opened || !Voice_IsRecording() )
|
|
return;
|
|
|
|
if( voice.goldsrc )
|
|
{
|
|
if( !voice.gs_encoder )
|
|
return;
|
|
|
|
size = Voice_GetGSCompressedData( buffer, sizeof( buffer ), &frames );
|
|
if( size > 0 && MSG_GetNumBytesLeft( &cls.datagram ) >= size + 32 )
|
|
{
|
|
uint packet_size = Voice_CreateGSVoicePacket( voice.compress_buffer, buffer, size );
|
|
MSG_BeginClientCmd( &cls.datagram, clc_voicedata );
|
|
MSG_WriteShort( &cls.datagram, packet_size );
|
|
MSG_WriteBytes( &cls.datagram, voice.compress_buffer, packet_size );
|
|
|
|
if ( voice_loopback.value && packet_size > 0 && frames > 0 )
|
|
{
|
|
Voice_AddIncomingData( cl.playernum + 1, voice.compress_buffer, packet_size, frames );
|
|
}
|
|
}
|
|
return;
|
|
}
|
|
|
|
if( !voice.encoder )
|
|
return;
|
|
|
|
size = Voice_GetOpusCompressedData( voice.compress_buffer, sizeof( voice.compress_buffer ), &frames );
|
|
if( size > 0 && MSG_GetNumBytesLeft( &cls.datagram ) >= size + 32 )
|
|
{
|
|
MSG_BeginClientCmd( &cls.datagram, clc_voicedata );
|
|
MSG_WriteByte( &cls.datagram, voice_loopback.value != 0 );
|
|
MSG_WriteByte( &cls.datagram, frames );
|
|
MSG_WriteShort( &cls.datagram, size );
|
|
MSG_WriteBytes( &cls.datagram, voice.compress_buffer, size );
|
|
}
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_RegisterCvars
|
|
|
|
Register voice related cvars and commands
|
|
=========================
|
|
*/
|
|
void Voice_RegisterCvars( void )
|
|
{
|
|
Cvar_RegisterVariable( &voice_enable );
|
|
Cvar_RegisterVariable( &voice_loopback );
|
|
Cvar_RegisterVariable( &voice_scale );
|
|
Cvar_RegisterVariable( &voice_transmit_scale );
|
|
Cvar_RegisterVariable( &voice_avggain );
|
|
Cvar_RegisterVariable( &voice_maxgain );
|
|
Cvar_RegisterVariable( &voice_inputfromfile );
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_Shutdown
|
|
|
|
Completely shutdown the voice subsystem
|
|
=========================
|
|
*/
|
|
static void Voice_Shutdown( void )
|
|
{
|
|
int i;
|
|
|
|
Voice_RecordStop();
|
|
|
|
if( voice.goldsrc )
|
|
Voice_ShutdownGoldSrcMode();
|
|
else
|
|
Voice_ShutdownOpusCustomMode();
|
|
|
|
VoiceCapture_Shutdown();
|
|
|
|
if( voice.local.talking_ack )
|
|
Voice_Status( VOICE_LOOPBACK_INDEX, false );
|
|
|
|
for( i = 0; i < MAX_CLIENTS; i++ )
|
|
{
|
|
if( voice.players_status[i].talking_ack )
|
|
Voice_Status( i, false );
|
|
}
|
|
|
|
voice.initialized = false;
|
|
voice.is_recording = false;
|
|
voice.device_opened = false;
|
|
voice.start_time = 0.0;
|
|
|
|
voice.input_buffer_pos = 0;
|
|
voice.input_file_pos = 0;
|
|
memset( voice.input_buffer, 0, sizeof( voice.input_buffer ));
|
|
memset( voice.compress_buffer, 0, sizeof( voice.compress_buffer ));
|
|
memset( voice.decompress_buffer, 0, sizeof( voice.decompress_buffer ));
|
|
memset( &voice.local, 0, sizeof( voice.local ));
|
|
memset( voice.players_status, 0, sizeof( voice.players_status ));
|
|
memset( &voice.autogain, 0, sizeof( voice.autogain ));
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_Idle
|
|
|
|
Run timeout for clients
|
|
=========================
|
|
*/
|
|
void Voice_Idle( double frametime )
|
|
{
|
|
int i;
|
|
|
|
if( FBitSet( voice_enable.flags, FCVAR_CHANGED ))
|
|
{
|
|
ClearBits( voice_enable.flags, FCVAR_CHANGED );
|
|
|
|
if( voice_enable.value )
|
|
{
|
|
// if( cls.state == ca_active && COM_CheckString( voice.codec ) && voice.quality != 0 )
|
|
if( cls.state == ca_active )
|
|
Voice_Init( voice.codec, voice.quality, false );
|
|
}
|
|
else
|
|
Voice_Shutdown();
|
|
}
|
|
|
|
// update local player status first
|
|
if ( voice.goldsrc )
|
|
Voice_StatusTimeout( &voice.local, cl.playernum + 1, frametime );
|
|
else
|
|
Voice_StatusTimeout( &voice.local, VOICE_LOOPBACK_INDEX, frametime );
|
|
|
|
for( i = 0; i < MAX_CLIENTS; i++ )
|
|
Voice_StatusTimeout( &voice.players_status[i], i, frametime );
|
|
}
|
|
|
|
/*
|
|
=========================
|
|
Voice_Init
|
|
|
|
Initialize the voice subsystem
|
|
=========================
|
|
*/
|
|
qboolean Voice_Init( const char *pszCodecName, int quality, qboolean preinit )
|
|
{
|
|
Q_strncpy( voice.codec, pszCodecName, sizeof( voice.codec ));
|
|
voice.quality = quality;
|
|
|
|
if( !voice_enable.value )
|
|
return false;
|
|
|
|
if( preinit )
|
|
return true;
|
|
|
|
if( Voice_IsGoldSrcMode( pszCodecName ) )
|
|
{
|
|
if( !Voice_InitGoldSrcMode( quality ) )
|
|
{
|
|
Voice_Shutdown();
|
|
return false;
|
|
}
|
|
}
|
|
else if( Voice_IsOpusCustomMode( pszCodecName ) )
|
|
{
|
|
if( !Voice_InitOpusCustomMode( quality ) )
|
|
{
|
|
Voice_Shutdown();
|
|
return false;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// unsupported codec
|
|
Voice_Shutdown();
|
|
voice.goldsrc = false;
|
|
voice.codec[0] = 0;
|
|
voice.quality = 0;
|
|
voice.samplerate = 0;
|
|
voice.frame_size = 0;
|
|
voice.width = 0;
|
|
voice.initialized = false;
|
|
return false;
|
|
}
|
|
|
|
Q_strncpy( voice.codec, pszCodecName, sizeof( voice.codec ));
|
|
voice.quality = quality;
|
|
|
|
voice.device_opened = VoiceCapture_Init();
|
|
|
|
if( !voice.device_opened )
|
|
Con_Printf( S_WARN "No microphone is available.\n" );
|
|
|
|
voice.initialized = true;
|
|
return true;
|
|
}
|