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https://github.com/w23/xash3d-fwgs
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632 lines
14 KiB
C
632 lines
14 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 "common.h"
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#include "client.h"
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#include "voice.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_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 );
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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_InitOpusDecoder
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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;
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voice.width = sizeof( opus_int16 );
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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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voice.decoder = opus_custom_decoder_create( voice.custom_mode, VOICE_PCM_CHANNELS, &err );
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if( !voice.decoder )
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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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return true;
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}
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/*
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=========================
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Voice_InitOpusEncoder
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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;
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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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switch( quality )
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{
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case 1: // 6 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 6000 ));
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break;
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case 2: // 12 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 12000 ));
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break;
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case 4: // 64 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 64000 ));
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break;
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case 5: // 96 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 96000 ));
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break;
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default: // 36 kbps
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opus_custom_encoder_ctl( voice.encoder, OPUS_SET_BITRATE( 36000 ));
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break;
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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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=========================
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*/
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static void Voice_ShutdownOpusDecoder( void )
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{
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if( voice.decoder )
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{
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opus_custom_decoder_destroy( voice.decoder );
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voice.decoder = NULL;
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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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=========================
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*/
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static void Voice_ShutdownOpusEncoder( void )
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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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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_GetOpusCompressedData
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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;
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updateInterval = cl.mtime[0] - cl.mtime[1];
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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 1
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if( !voice.input_file )
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{
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// adjust gain before encoding, but only for input from voice
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Voice_ApplyGainAdjust((opus_int16*)(voice.input_buffer + ofs), voice.frame_size);
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}
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#endif
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bytes = opus_custom_encode( voice.encoder, (const opus_int16 *)( 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]) = 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 CHAT INTEGRATION
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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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=========================
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*/
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static void Voice_ApplyGainAdjust( int16_t *samples, int count )
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{
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float gain, modifiedMax;
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int average, adjustedSample, 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 ), 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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adjustedSample = Q_min( SHRT_MAX, Q_max(( int )( sample * gain ), SHRT_MIN ));
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samples[blockOffset + i] = adjustedSample;
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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 * voice_scale.value;
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voice.autogain.next_gain = Q_min( (float)SHRT_MAX / modifiedMax, voice_maxgain.value ) * voice_scale.value;
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voice.autogain.gain_multiplier = ( voice.autogain.next_gain - voice.autogain.current_gain ) / ( voice.autogain.block_size - 1 );
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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_Status
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Notify user dll aboit voice transmission
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=========================
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*/
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static void Voice_Status( int entindex, qboolean bTalking )
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{
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if( cls.state == ca_active && clgame.dllFuncs.pfnVoiceStatus )
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clgame.dllFuncs.pfnVoiceStatus( entindex, bTalking );
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}
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/*
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=========================
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Voice_StatusTimeout
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Waits few milliseconds and if there was no
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voice transmission, sends notification
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=========================
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*/
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static void Voice_StatusTimeout( voice_status_t *status, int entindex, double frametime )
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{
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if( status->talking_ack )
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{
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status->talking_timeout += frametime;
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if( status->talking_timeout > 0.2 )
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{
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status->talking_ack = false;
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Voice_Status( entindex, false );
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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_StatusAck
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Sends notification to user dll and
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zeroes timeouts for this client
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=========================
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*/
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void Voice_StatusAck( voice_status_t *status, int playerIndex )
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{
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if( !status->talking_ack )
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Voice_Status( playerIndex, true );
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status->talking_ack = true;
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status->talking_timeout = 0.0;
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}
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/*
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=========================
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Voice_IsRecording
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=========================
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*/
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qboolean Voice_IsRecording( void )
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{
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return voice.is_recording;
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}
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/*
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=========================
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Voice_RecordStop
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=========================
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*/
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void Voice_RecordStop( void )
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{
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if( voice.input_file )
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{
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FS_FreeSound( voice.input_file );
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voice.input_file = NULL;
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}
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VoiceCapture_Activate( false );
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voice.is_recording = false;
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Voice_Status( VOICE_LOCALCLIENT_INDEX, false );
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voice.input_buffer_pos = 0;
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memset( voice.input_buffer, 0, sizeof( voice.input_buffer ));
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}
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/*
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=========================
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Voice_RecordStart
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=========================
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*/
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void Voice_RecordStart( void )
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{
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Voice_RecordStop();
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if( voice_inputfromfile.value )
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{
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voice.input_file = FS_LoadSound( "voice_input.wav", NULL, 0 );
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if( voice.input_file )
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{
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Sound_Process( &voice.input_file, voice.samplerate, voice.width, SOUND_RESAMPLE );
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voice.input_file_pos = 0;
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voice.start_time = Sys_DoubleTime();
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voice.is_recording = true;
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}
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else
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{
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FS_FreeSound( voice.input_file );
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voice.input_file = NULL;
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}
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}
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if( !Voice_IsRecording() )
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voice.is_recording = VoiceCapture_Activate( true );
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if( Voice_IsRecording() )
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Voice_Status( VOICE_LOCALCLIENT_INDEX, true );
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}
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/*
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=========================
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Voice_Disconnect
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We're disconnected from server
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stop recording and notify user dlls
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=========================
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*/
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void Voice_Disconnect( void )
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{
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int i;
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Voice_RecordStop();
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if( voice.local.talking_ack )
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{
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Voice_Status( VOICE_LOOPBACK_INDEX, false );
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voice.local.talking_ack = false;
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}
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for( i = 0; i < MAX_CLIENTS; i++ )
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{
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if( voice.players_status[i].talking_ack )
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{
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Voice_Status( i, false );
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voice.players_status[i].talking_ack = false;
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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_StartChannel
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Feed the decoded data to engine sound subsystem
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=========================
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*/
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static void Voice_StartChannel( uint samples, byte *data, int entnum )
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{
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SND_ForceInitMouth( entnum );
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S_RawEntSamples( entnum, samples, voice.samplerate, voice.width, VOICE_PCM_CHANNELS, data, 255 );
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}
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/*
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=========================
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Voice_AddIncomingData
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Received encoded voice data, decode it
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=========================
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*/
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void Voice_AddIncomingData( int ent, const byte *data, uint size, uint frames )
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{
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int samples = 0;
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int ofs = 0;
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if( !voice.decoder )
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return;
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// decode frame by frame
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for( ;; )
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{
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int frame_samples;
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uint16_t compressed_size;
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// no compressed size mark
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if( ofs + sizeof( uint16_t ) > size )
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break;
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compressed_size = *(const uint16_t *)(data + ofs);
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ofs += sizeof( uint16_t );
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// no frame data
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if( ofs + compressed_size > size )
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break;
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frame_samples = opus_custom_decode( voice.decoder, data + ofs, compressed_size,
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(opus_int16*)voice.decompress_buffer + samples, voice.frame_size );
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ofs += compressed_size;
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samples += frame_samples;
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}
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if( samples > 0 )
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Voice_StartChannel( samples, voice.decompress_buffer, ent );
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}
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/*
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=========================
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CL_AddVoiceToDatagram
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Encode our voice data and send it to server
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=========================
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*/
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void CL_AddVoiceToDatagram( void )
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{
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uint size, frames = 0;
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if( cls.state != ca_active || !Voice_IsRecording() || !voice.encoder )
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return;
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size = Voice_GetOpusCompressedData( voice.output_buffer, sizeof( voice.output_buffer ), &frames );
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if( size > 0 && MSG_GetNumBytesLeft( &cls.datagram ) >= size + 32 )
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{
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MSG_BeginClientCmd( &cls.datagram, clc_voicedata );
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MSG_WriteByte( &cls.datagram, voice_loopback.value != 0 );
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MSG_WriteByte( &cls.datagram, frames );
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MSG_WriteShort( &cls.datagram, size );
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MSG_WriteBytes( &cls.datagram, voice.output_buffer, size );
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}
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}
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/*
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=========================
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Voice_RegisterCvars
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Register voice related cvars and commands
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=========================
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*/
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void Voice_RegisterCvars( void )
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{
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Cvar_RegisterVariable( &voice_enable );
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Cvar_RegisterVariable( &voice_loopback );
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Cvar_RegisterVariable( &voice_scale );
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Cvar_RegisterVariable( &voice_avggain );
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Cvar_RegisterVariable( &voice_maxgain );
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Cvar_RegisterVariable( &voice_inputfromfile );
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}
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/*
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=========================
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Voice_Shutdown
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Completely shutdown the voice subsystem
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=========================
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*/
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static void Voice_Shutdown( void )
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{
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int i;
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Voice_RecordStop();
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Voice_ShutdownOpusEncoder();
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Voice_ShutdownOpusDecoder();
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VoiceCapture_Shutdown();
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if( voice.local.talking_ack )
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Voice_Status( VOICE_LOOPBACK_INDEX, false );
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for( i = 0; i < MAX_CLIENTS; i++ )
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{
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if( voice.players_status[i].talking_ack )
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Voice_Status( i, false );
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}
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memset( &voice, 0, sizeof( voice ));
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}
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/*
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=========================
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Voice_Idle
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Run timeout for all clients
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=========================
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*/
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void Voice_Idle( double frametime )
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{
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int i;
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if( FBitSet( voice_enable.flags, FCVAR_CHANGED ) && !voice_enable.value )
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{
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Voice_Shutdown();
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return;
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}
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// update local player status first
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Voice_StatusTimeout( &voice.local, VOICE_LOOPBACK_INDEX, frametime );
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for( i = 0; i < MAX_CLIENTS; i++ )
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Voice_StatusTimeout( &voice.players_status[i], i, frametime );
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}
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/*
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=========================
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Voice_Init
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Initialize the voice subsystem
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=========================
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*/
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qboolean Voice_Init( const char *pszCodecName, int quality )
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{
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if( !voice_enable.value )
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return false;
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if( Q_strcmp( pszCodecName, VOICE_OPUS_CUSTOM_CODEC ))
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{
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Con_Printf( S_ERROR "Server requested unsupported codec: %s\n", pszCodecName );
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return false;
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}
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// reinitialize only if codec parameters are different
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if( Q_strcmp( voice.codec, pszCodecName ) && voice.quality != quality )
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Voice_Shutdown();
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voice.autogain.block_size = 128;
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if( !Voice_InitOpusDecoder( ))
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{
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// no reason to init encoder and open audio device
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// if we can't hear other players
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Con_Printf( S_ERROR "Voice chat disabled.\n" );
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Voice_Shutdown();
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return false;
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}
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// we can hear others players, so it's fine to fail now
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voice.initialized = true;
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Q_strncpy( voice.codec, pszCodecName, sizeof( voice.codec ));
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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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voice.quality = quality;
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if( !VoiceCapture_Init( ))
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Con_Printf( S_WARN "No microphone is available.\n" );
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return true;
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}
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