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Merge remote-tracking branch 'pwd491/gsvoice'

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This commit is contained in:
Alibek Omarov
2025-07-09 16:43:14 +05:00
parent f9fc26239b 21b0516724
commit 3d762802ab
8 changed files with 833 additions and 253 deletions
Download Patch File Download Diff File Expand all files Collapse all files
engine/client/cl_main.c
+4
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@@ -1029,6 +1029,10 @@ static void CL_WriteSteamTicket( sizebuf_t *send )
crc = CRC32_Final( crc );
i = GenerateRevEmu2013( buf, s, crc );
MSG_WriteBytes( send, buf, i );
// RevEmu2013: pTicket[1] = revHash (low), pTicket[5] = 0x01100001 (high)
*(uint32_t*)cls.steamid = LittleLong( ((uint32_t*)buf)[1] );
*(uint32_t*)(cls.steamid + 4) = LittleLong( ((uint32_t*)buf)[5] );
}
/*
engine/client/cl_parse.c
+24 -27
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@@ -2010,44 +2010,41 @@ CL_ParseVoiceData
*/
void CL_ParseVoiceData( sizebuf_t *msg, connprotocol_t proto )
{
int size, idx, frames;
byte received[8192];
int size, idx, frames = 0;
byte received[VOICE_MAX_DATA_SIZE];
idx = MSG_ReadByte( msg ) + 1;
if( proto == PROTO_GOLDSRC )
{
size = MSG_ReadShort( msg );
MSG_SeekToBit( msg, size << 3, SEEK_CUR ); // skip the entire buf, not supported yet
#if 0 // shall we notify client.dll if nothing can be heard?
// must notify through as both local player and normal client
if( idx == cl.playernum + 1 )
Voice_StatusAck( &voice.local, VOICE_LOOPBACK_INDEX );
Voice_StatusAck( &voice.players_status[idx], idx );
#endif
return;
}
frames = MSG_ReadByte( msg );
size = MSG_ReadShort( msg );
size = Q_min( size, sizeof( received ));
MSG_ReadBytes( msg, received, size );
if ( idx <= 0 || idx > cl.maxclients )
return;
// must notify through as both local player and normal client
if( idx == cl.playernum + 1 )
Voice_StatusAck( &voice.local, VOICE_LOOPBACK_INDEX );
if( proto == PROTO_GOLDSRC )
{
size = MSG_ReadShort( msg );
if ( size > VOICE_MAX_GS_DATA_SIZE )
{
Con_Printf( S_ERROR "Voice data size is too large: %d bytes (max: %d)\n", size, VOICE_MAX_GS_DATA_SIZE );
return;
}
}
else
{
frames = MSG_ReadByte( msg );
size = MSG_ReadShort( msg );
if (size > VOICE_MAX_DATA_SIZE )
{
Con_Printf( S_ERROR "Voice data size is too large: %d bytes (max: %d)\n", size, VOICE_MAX_DATA_SIZE );
return;
}
}
Voice_StatusAck( &voice.players_status[idx], idx );
size = Q_min( size, sizeof( received ));
if ( !size )
return;
MSG_ReadBytes( msg, received, size );
Voice_AddIncomingData( idx, received, size, frames );
}
engine/client/client.h
+1
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@@ -638,6 +638,7 @@ typedef struct
// server's build number (might be zero)
int build_num;
uint8_t steamid[8];
} client_static_t;
#ifdef __cplusplus
engine/client/s_mix.c
+21 -3
View File
@@ -22,6 +22,7 @@ enum
IPAINTBUFFER = 0,
IROOMBUFFER,
ISTREAMBUFFER,
IVOICEBUFFER,
CPAINTBUFFERS,
};
@@ -63,6 +64,7 @@ static portable_samplepair_t *g_curpaintbuffer;
static portable_samplepair_t streambuffer[(PAINTBUFFER_SIZE+1)];
static portable_samplepair_t paintbuffer[(PAINTBUFFER_SIZE+1)];
static portable_samplepair_t roombuffer[(PAINTBUFFER_SIZE+1)];
static portable_samplepair_t voicebuffer[(PAINTBUFFER_SIZE+1)];
static portable_samplepair_t temppaintbuffer[(PAINTBUFFER_SIZE+1)];
static paintbuffer_t paintbuffers[CPAINTBUFFERS];
@@ -216,6 +218,7 @@ void MIX_InitAllPaintbuffers( void )
paintbuffers[IPAINTBUFFER].pbuf = paintbuffer;
paintbuffers[IROOMBUFFER].pbuf = roombuffer;
paintbuffers[ISTREAMBUFFER].pbuf = streambuffer;
paintbuffers[IVOICEBUFFER].pbuf = voicebuffer;
MIX_SetCurrentPaintbuffer( IPAINTBUFFER );
}
@@ -921,11 +924,12 @@ static void S_MixUpsample( int sampleCount, int filtertype )
static void MIX_MixRawSamplesBuffer( int end )
{
portable_samplepair_t *pbuf, *roombuf, *streambuf;
portable_samplepair_t *pbuf, *roombuf, *streambuf, *voicebuf;
uint i, j, stop;
roombuf = MIX_GetPFrontFromIPaint( IROOMBUFFER );
streambuf = MIX_GetPFrontFromIPaint( ISTREAMBUFFER );
voicebuf = MIX_GetPFrontFromIPaint( IVOICEBUFFER );
if( s_listener.paused ) return;
@@ -935,6 +939,7 @@ static void MIX_MixRawSamplesBuffer( int end )
// copy from the streaming sound source
rawchan_t *ch = raw_channels[i];
qboolean stream;
qboolean is_voice;
if( !ch )
continue;
@@ -943,8 +948,18 @@ static void MIX_MixRawSamplesBuffer( int end )
if( !ch->leftvol && !ch->rightvol )
continue;
is_voice = (ch->entnum > 0 && ch->entnum <= MAX_CLIENTS) ||
(ch->entnum == VOICE_LOOPBACK_INDEX) ||
(ch->entnum == VOICE_LOCALCLIENT_INDEX);
stream = ch->entnum == S_RAW_SOUND_BACKGROUNDTRACK || CL_IsPlayerIndex( ch->entnum );
pbuf = stream ? streambuf : roombuf;
if( is_voice )
pbuf = voicebuf;
else if( stream )
pbuf = streambuf;
else
pbuf = roombuf;
stop = (end < ch->s_rawend) ? end : ch->s_rawend;
@@ -1045,7 +1060,10 @@ void MIX_PaintChannels( int endtime )
MIX_MixPaintbuffers( IPAINTBUFFER, IROOMBUFFER, IPAINTBUFFER, count, S_GetMasterVolume() );
// add music or soundtrack from movie (no dsp)
MIX_MixPaintbuffers( IPAINTBUFFER, ISTREAMBUFFER, IPAINTBUFFER, count, S_GetMusicVolume() );
MIX_MixPaintbuffers( IPAINTBUFFER, ISTREAMBUFFER, IPAINTBUFFER, count, 1.0f );
// voice chat
MIX_MixPaintbuffers( IPAINTBUFFER, IVOICEBUFFER, IPAINTBUFFER, count, 1.0f );
// clip all values > 16 bit down to 16 bit
MIX_CompressPaintbuffer( IPAINTBUFFER, count );
engine/client/s_stream.c
+2 -1
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@@ -242,7 +242,8 @@ void S_StreamBackgroundTrack( void )
if( r > 0 )
{
// add to raw buffer
S_RawEntSamples( S_RAW_SOUND_BACKGROUNDTRACK, fileSamples, info->rate, info->width, info->channels, raw, 255 );
int music_vol = (int)(255.0f * S_GetMusicVolume());
S_RawEntSamples( S_RAW_SOUND_BACKGROUNDTRACK, fileSamples, info->rate, info->width, info->channels, raw, music_vol );
}
else
{
engine/client/voice.c
+729 -200
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File diff suppressed because it is too large Load Diff
engine/client/voice.h
+51 -22
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@@ -24,46 +24,81 @@ GNU General Public License for more details.
typedef struct OpusCustomEncoder OpusCustomEncoder;
typedef struct OpusCustomDecoder OpusCustomDecoder;
typedef struct OpusCustomMode OpusCustomMode;
typedef struct OpusEncoder OpusEncoder;
typedef struct OpusDecoder OpusDecoder;
#define VOICE_LOOPBACK_INDEX (-2)
#define VOICE_LOCALCLIENT_INDEX (-1)
#define VOICE_LOOPBACK_INDEX ( -2 )
#define VOICE_LOCALCLIENT_INDEX ( -1 )
#define VOICE_PCM_CHANNELS 1 // always mono
#define VOICE_PCM_CHANNELS 1 // always mono
#define VOICE_MAX_DATA_SIZE 8192
#define VOICE_MAX_GS_DATA_SIZE 4096
// never change these parameters when using opuscustom
#define VOICE_OPUS_CUSTOM_SAMPLERATE 44100
// must follow opus custom requirements
// also be divisible with MAX_RAW_SAMPLES
#define VOICE_OPUS_CUSTOM_FRAME_SIZE 1024
#define VOICE_OPUS_CUSTOM_CODEC "opus_custom_44k_512"
#define VOICE_OPUS_CUSTOM_CODEC "opus_custom_44k_512"
// a1ba: do not change, we don't have any re-encoding support now
#define VOICE_DEFAULT_CODEC VOICE_OPUS_CUSTOM_CODEC
#define VOICE_DEFAULT_CODEC VOICE_OPUS_CUSTOM_CODEC
#define VOICE_DEFAULT_SILENCE_FRAME_SIZE 160
// GoldSrc voice configuration
#define GS_MAX_DECOMPRESSED_SAMPLES 32768
#define GS_DEFAULT_SAMPLE_RATE 24000
#define GS_DEFAULT_FRAME_SIZE 480
// VPC (Voice Packet Control) types
enum gs_vpc_type
{
GS_VPC_VDATA_SILENCE = 0,
GS_VPC_VDATA_MILES = 1,
GS_VPC_VDATA_SPEEX = 2,
GS_VPC_VDATA_RAW = 3,
GS_VPC_VDATA_SILK = 4,
GS_VPC_VDATA_OPUS_PLC = 6,
GS_VPC_SETSAMPLERATE = 11,
GS_VPC_UNKNOWN = 10
};
typedef struct voice_status_s
{
qboolean talking_ack;
double talking_timeout;
double talking_timeout;
} voice_status_t;
typedef struct voice_autogain_s
{
int block_size;
float current_gain;
float next_gain;
float gain_multiplier;
} voice_autogain_t;
typedef struct voice_state_s
{
string codec;
int quality;
string codec;
int quality;
qboolean goldsrc;
qboolean initialized;
qboolean is_recording;
qboolean device_opened;
double start_time;
double start_time;
voice_status_t local;
voice_status_t players_status[MAX_CLIENTS];
voice_status_t local;
voice_status_t players_status[MAX_CLIENTS];
// opus stuff
OpusCustomMode *custom_mode;
OpusCustomEncoder *encoder;
OpusCustomDecoder *decoders[MAX_CLIENTS];
OpusEncoder *gs_encoder;
OpusDecoder *gs_decoders[MAX_CLIENTS];
// audio info
uint width;
uint samplerate;
@@ -73,25 +108,18 @@ typedef struct voice_state_s
byte input_buffer[MAX_RAW_SAMPLES];
byte compress_buffer[MAX_RAW_SAMPLES];
byte decompress_buffer[MAX_RAW_SAMPLES];
fs_offset_t input_buffer_pos; // in bytes
fs_offset_t input_buffer_pos; // in bytes
// input from file
wavdata_t *input_file;
fs_offset_t input_file_pos; // in bytes
wavdata_t *input_file;
fs_offset_t input_file_pos; // in bytes
// automatic gain control
struct {
int block_size;
float current_gain;
float next_gain;
float gain_multiplier;
} autogain;
voice_autogain_t autogain;
} voice_state_t;
extern voice_state_t voice;
void CL_AddVoiceToDatagram( void );
void Voice_RegisterCvars( void );
qboolean Voice_Init( const char *pszCodecName, int quality, qboolean preinit );
void Voice_Idle( double frametime );
@@ -101,5 +129,6 @@ void Voice_RecordStart( void );
void Voice_Disconnect( void );
void Voice_AddIncomingData( int ent, const byte *data, uint size, uint frames );
void Voice_StatusAck( voice_status_t *status, int playerIndex );
void Voice_StartChannel( uint samples, byte *data, int entnum );
#endif // VOICE_H
engine/platform/sdl2/s_sdl2.c
+1
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@@ -342,6 +342,7 @@ void VoiceCapture_Shutdown( void )
return;
SDL_CloseAudioDevice( in_dev );
in_dev = 0;
}
#endif // XASH_SOUND == SOUND_SDL