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engine: client: voice: simplify code, simplify including voice.h, do not depend on opus headers globally 

Autofix few code style mistakes
This commit is contained in:
Alibek Omarov 2022-08-19 05:52:53 +03:00
parent e6c55107c7
commit 327dcc0293
3 changed files with 85 additions and 102 deletions
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4
engine/client/cl_parse.c
View File

@ -1715,9 +1715,9 @@ void CL_ParseVoiceData( sizebuf_t *msg )
return;
if( idx == cl.playernum + 1 )
Voice_LocalPlayerTalkingAck();
Voice_StatusAck( &voice.local, VOICE_LOCALPLAYER_INDEX );
else
Voice_PlayerTalkingAck( idx );
Voice_StatusAck( &voice.players_status[idx], idx );
if ( !size )
return;

147
engine/client/voice.c
View File

@ -14,12 +14,15 @@ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
*/
#include <opus.h>
#include "common.h"
#include "client.h"
#include "voice.h"
wavdata_t *input_file;
fs_offset_t input_pos;
static wavdata_t *input_file;
static fs_offset_t input_pos;
voice_state_t voice;
voice_state_t voice = { 0 };
CVAR_DEFINE_AUTO( voice_enable, "1", FCVAR_PRIVILEGED|FCVAR_ARCHIVE, "enable voice chat" );
CVAR_DEFINE_AUTO( voice_loopback, "0", FCVAR_PRIVILEGED, "loopback voice back to the speaker" );
@ -47,6 +50,12 @@ static void Voice_CodecInfo_f( void )
opus_int32 encoderBitrate;
opus_int32 encoderBandwidthType;
if( !voice.initialized )
{
Con_Printf( "Voice codec is not initialized!\n" );
return;
}
opus_encoder_ctl( voice.encoder, OPUS_GET_BITRATE( &encoderBitrate ));
opus_encoder_ctl( voice.encoder, OPUS_GET_COMPLEXITY( &encoderComplexity ));
opus_encoder_ctl( voice.encoder, OPUS_GET_BANDWIDTH( &encoderBandwidthType ));
@ -54,8 +63,7 @@ static void Voice_CodecInfo_f( void )
Con_Printf( "Encoder:\n" );
Con_Printf( " Bitrate: %.3f kbps\n", encoderBitrate / 1000.0f );
Con_Printf( " Complexity: %d\n", encoderComplexity );
Con_Printf( " Bandwidth: " );
Con_Printf( Voice_GetBandwidthTypeName( encoderBandwidthType ));
Con_Printf( " Bandwidth: %s", Voice_GetBandwidthTypeName( encoderBandwidthType ));
Con_Printf( "\n" );
}
@ -86,7 +94,7 @@ static void Voice_ApplyGainAdjust( opus_int16 *samples, int count )
int average, adjustedSample;
int blockOffset = 0;
for (;;)
for( ;;)
{
int i;
int localMax = 0;
@ -126,7 +134,7 @@ qboolean Voice_Init( const char *pszCodecName, int quality )
{
int err;
if ( !voice_enable.value )
if( !voice_enable.value )
return false;
Voice_DeInit();
@ -138,7 +146,7 @@ qboolean Voice_Init( const char *pszCodecName, int quality )
voice.frame_size = Voice_GetFrameSize( 40.0f );
voice.autogain.block_size = 128;
if ( !VoiceCapture_Init() )
if( !VoiceCapture_Init() )
{
Voice_DeInit();
return false;
@ -180,7 +188,7 @@ qboolean Voice_Init( const char *pszCodecName, int quality )
void Voice_DeInit( void )
{
if ( !voice.initialized )
if( !voice.initialized )
return;
Voice_RecordStop();
@ -191,11 +199,11 @@ void Voice_DeInit( void )
voice.initialized = false;
}
uint Voice_GetCompressedData( byte *out, uint maxsize, uint *frames )
static uint Voice_GetCompressedData( byte *out, uint maxsize, uint *frames )
{
uint ofs, size = 0;
if ( input_file )
if( input_file )
{
uint numbytes;
double updateInterval;
@ -210,11 +218,11 @@ uint Voice_GetCompressedData( byte *out, uint maxsize, uint *frames )
input_pos += numbytes;
}
for ( ofs = 0; voice.input_buffer_pos - ofs >= voice.frame_size && ofs <= voice.input_buffer_pos; ofs += voice.frame_size )
for( ofs = 0; voice.input_buffer_pos - ofs >= voice.frame_size && ofs <= voice.input_buffer_pos; ofs += voice.frame_size )
{
int bytes;
if (!input_file)
if( !input_file )
{
// adjust gain before encoding, but only for input from voice
Voice_ApplyGainAdjust((opus_int16*)voice.input_buffer + ofs, voice.frame_size);
@ -224,7 +232,7 @@ uint Voice_GetCompressedData( byte *out, uint maxsize, uint *frames )
memmove( voice.input_buffer, voice.input_buffer + voice.frame_size, sizeof( voice.input_buffer ) - voice.frame_size );
voice.input_buffer_pos -= voice.frame_size;
if ( bytes > 0 )
if( bytes > 0 )
{
size += bytes;
(*frames)++;
@ -234,39 +242,43 @@ uint Voice_GetCompressedData( byte *out, uint maxsize, uint *frames )
return size;
}
void Voice_Idle( float frametime )
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 );
}
}
}
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;
}
void Voice_Idle( double frametime )
{
int i;
if ( !voice_enable.value )
if( !voice_enable.value )
{
Voice_DeInit();
return;
}
if ( voice.talking_ack )
{
voice.talking_timeout += frametime;
if( voice.talking_timeout > 0.2f )
{
voice.talking_ack = false;
Voice_Status( -2, false );
}
}
// update local player status first
Voice_StatusTimeout( &voice.local, VOICE_LOCALPLAYER_INDEX, frametime );
for ( i = 0; i < 32; i++ )
{
if ( voice.players_status[i].talking_ack )
{
voice.players_status[i].talking_timeout += frametime;
if ( voice.players_status[i].talking_timeout > 0.2f )
{
voice.players_status[i].talking_ack = false;
if ( i < cl.maxclients )
Voice_Status( i, false );
}
}
}
for( i = 0; i < 32; i++ )
Voice_StatusTimeout( &voice.players_status[i], i, frametime );
}
qboolean Voice_IsRecording( void )
@ -276,7 +288,7 @@ qboolean Voice_IsRecording( void )
void Voice_RecordStop( void )
{
if ( input_file )
if( input_file )
{
FS_FreeSound( input_file );
input_file = NULL;
@ -285,7 +297,7 @@ void Voice_RecordStop( void )
voice.input_buffer_pos = 0;
memset( voice.input_buffer, 0, sizeof( voice.input_buffer ) );
if ( Voice_IsRecording() )
if( Voice_IsRecording() )
Voice_Status( -1, false );
VoiceCapture_RecordStop();
@ -297,11 +309,11 @@ void Voice_RecordStart( void )
{
Voice_RecordStop();
if ( voice_inputfromfile.value )
if( voice_inputfromfile.value )
{
input_file = FS_LoadSound( "voice_input.wav", NULL, 0 );
if ( input_file )
if( input_file )
{
Sound_Process( &input_file, voice.samplerate, voice.width, SOUND_RESAMPLE );
input_pos = 0;
@ -316,10 +328,10 @@ void Voice_RecordStart( void )
}
}
if ( !Voice_IsRecording() )
if( !Voice_IsRecording() )
voice.is_recording = VoiceCapture_RecordStart();
if ( Voice_IsRecording() )
if( Voice_IsRecording() )
Voice_Status( -1, true );
}
@ -333,11 +345,17 @@ void Voice_Disconnect( void )
}
}
static void Voice_StartChannel( uint samples, byte *data, int entnum )
{
SND_ForceInitMouth( entnum );
S_RawEntSamples( entnum, samples, voice.samplerate, voice.width, voice.channels, data, 255 );
}
void Voice_AddIncomingData( int ent, const byte *data, uint size, uint frames )
{
int samples = opus_decode( voice.decoder, data, size, (short *)voice.decompress_buffer, voice.frame_size / voice.width * frames, false );
if ( samples > 0 )
if( samples > 0 )
Voice_StartChannel( samples, voice.decompress_buffer, ent );
}
@ -345,50 +363,17 @@ void CL_AddVoiceToDatagram( void )
{
uint size, frames = 0;
if ( cls.state != ca_active || !Voice_IsRecording() )
if( cls.state != ca_active || !Voice_IsRecording() )
return;
size = Voice_GetCompressedData( voice.output_buffer, sizeof( voice.output_buffer ), &frames );
if ( size > 0 && MSG_GetNumBytesLeft( &cls.datagram ) >= size + 32 )
if( size > 0 && MSG_GetNumBytesLeft( &cls.datagram ) >= size + 32 )
{
MSG_BeginClientCmd( &cls.datagram, clc_voicedata );
MSG_WriteByte( &cls.datagram, Voice_GetLoopback() );
MSG_WriteByte( &cls.datagram, voice_loopback.value != 0 );
MSG_WriteByte( &cls.datagram, frames );
MSG_WriteShort( &cls.datagram, size );
MSG_WriteBytes( &cls.datagram, voice.output_buffer, size );
}
}
qboolean Voice_GetLoopback( void )
{
return voice_loopback.value;
}
void Voice_LocalPlayerTalkingAck( void )
{
if ( !voice.talking_ack )
{
Voice_Status( -2, true );
}
voice.talking_ack = true;
voice.talking_timeout = 0.0f;
}
void Voice_PlayerTalkingAck(int playerIndex)
{
if( !voice.players_status[playerIndex].talking_ack )
{
Voice_Status( playerIndex, true );
}
voice.players_status[playerIndex].talking_ack = true;
voice.players_status[playerIndex].talking_timeout = 0.0f;
}
void Voice_StartChannel( uint samples, byte *data, int entnum )
{
SND_ForceInitMouth( entnum );
S_RawEntSamples( entnum, samples, voice.samplerate, voice.width, voice.channels, data, 255 );
}

36
engine/client/voice.h
View File

@ -17,28 +17,30 @@ GNU General Public License for more details.
#ifndef VOICE_H
#define VOICE_H
#include <opus.h>
#include "common.h"
#include "client.h"
#include "protocol.h" // MAX_CLIENTS
#include "sound.h"
#include "soundlib/soundlib.h"
#include "library.h"
extern convar_t voice_scale;
typedef struct OpusDecoder OpusDecoder;
typedef struct OpusEncoder OpusEncoder;
#define VOICE_LOCALPLAYER_INDEX (-2)
typedef struct voice_status_s
{
qboolean talking_ack;
double talking_timeout;
} voice_status_t;
typedef struct voice_state_s
{
qboolean initialized;
qboolean is_recording;
float start_time;
qboolean talking_ack;
float talking_timeout;
double start_time;
struct {
qboolean talking_ack;
float talking_timeout;
} players_status[32];
voice_status_t local;
voice_status_t players_status[MAX_CLIENTS];
// opus stuff
OpusEncoder *encoder;
@ -72,16 +74,12 @@ void CL_AddVoiceToDatagram( void );
void Voice_RegisterCvars( void );
qboolean Voice_Init( const char *pszCodecName, int quality );
void Voice_DeInit( void );
uint Voice_GetCompressedData( byte *out, uint maxsize, uint *frames );
void Voice_Idle( float frametime );
void Voice_Idle( double frametime );
qboolean Voice_IsRecording( void );
void Voice_RecordStop( void );
void Voice_RecordStart( void );
void Voice_Disconnect( void );
void Voice_AddIncomingData( int ent, const byte *data, uint size, uint frames );
qboolean Voice_GetLoopback( void );
void Voice_LocalPlayerTalkingAck( void );
void Voice_PlayerTalkingAck( int playerIndex );
void Voice_StartChannel( uint samples, byte *data, int entnum );
void Voice_StatusAck( voice_status_t *status, int playerIndex );
#endif // VOICE_H