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fix some voip stuff, opus should be properly supported now. 

fix iqm+events... again.

git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@5043 fc73d0e0-1445-4013-8a0c-d673dee63da5
This commit is contained in:
Spoike 2017-01-17 19:04:09 +00:00
parent 36c0c3302f
commit 15a2038e4e
7 changed files with 69 additions and 52 deletions
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2
engine/client/cl_input.c
View File

@ -1688,7 +1688,7 @@ qboolean CLQW_SendCmd (sizebuf_t *buf, qboolean actuallysend)
void CL_SendCmd (double frametime, qboolean mainloop)
{
sizebuf_t buf;
qbyte data[1024];
qbyte data[MAX_DATAGRAM];
int i, plnum;
usercmd_t *cmd;
float wantfps;

2
engine/client/cl_main.c
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@ -417,7 +417,7 @@ void CL_ConnectToDarkPlaces(char *challenge, netadr_t *adr)
connectinfo.time = realtime; // for retransmit requests
Q_snprintfz(data, sizeof(data), "%c%c%c%cconnect\\protocol\\darkplaces 3\\protocols\\DP7 DP6 DP5 FITZ NEHAHRABJP NEHAHRABJP2 NEHAHRABJP3 QUAKE\\challenge\\%s", 255, 255, 255, 255, challenge);
Q_snprintfz(data, sizeof(data), "%c%c%c%cconnect\\protocol\\darkplaces 3\\protocols\\DP7 DP6 DP5 RMQ FITZ NEHAHRABJP NEHAHRABJP2 NEHAHRABJP3 QUAKE\\challenge\\%s", 255, 255, 255, 255, challenge);
NET_SendPacket (NS_CLIENT, strlen(data), data, adr);

12
engine/client/cl_parse.c
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@ -268,7 +268,7 @@ char *svc_nqstrings[] =
"NEW PROTOCOL(81)", //81
"NEW PROTOCOL(82)", //82
"nqsvcfte_cgamepacket(83)", //83
"NEW PROTOCOL(84)", //84
"nqsvcfte_voicechat", //84
"NEW PROTOCOL(85)", //85
"nqsvcfte_updateentities", //86
"NEW PROTOCOL(87)", //87
@ -3659,6 +3659,10 @@ void CLNQ_ParseServerData(void) //Doesn't change gamedir - use with caution.
if (CPNQ_IS_DP) //DP's protocol requires client+server to have exactly the same data files. this is shit, but in the interests of compatibility...
COM_Effectinfo_Enumerate(CL_Darkplaces_Particle_Precache);
#ifdef VOICECHAT
S_Voip_MapChange();
#endif
#ifdef PEXT_CSQC
CSQC_Shutdown();
CSQC_Init(cls.demoplayback, false, 0);
@ -7823,6 +7827,12 @@ void CLNQ_ParseServerMessage (void)
S_StopSound(i>>3, i&7);
break;
#ifdef PEXT2_VOICECHAT
case svcfte_voicechat:
S_Voip_Parse();
break;
#endif
case svc_temp_entity:
CL_ParseTEnt (true);
break;

8
engine/client/m_options.c
View File

@ -552,19 +552,23 @@ void M_Menu_Audio_f (void)
};
#ifdef VOICECHAT
static const char *voipcodecoptions[] = {
"Auto",
"Speex (ez-compat)",
// "Raw (11025)",
"Opus (external)",
// "Raw16 (11025)",
"Opus",
"Speex (Narrow)",
"Speex (Wide)",
// "Speex (UltraWide)",
NULL
};
static const char *voipcodecvalue[] = {
"",
"0",
// "1",
"2",
"3",
"4",
// "5",
NULL
};

2
engine/client/sbar.c
View File

@ -2684,7 +2684,7 @@ static void Sbar_Voice(int y)
int s, i;
float range = loudness/100.0f;
w = 0;
Font_BeginString(font_default, sbar_rect.x + sbar_rect.width/2, sbar_rect.y + y + sbar_rect.height-SBAR_HEIGHT, &x, &y);
Font_BeginString(font_default, sbar_rect.x + min(320,sbar_rect.width)/2, sbar_rect.y + y + sbar_rect.height-SBAR_HEIGHT, &x, &y);
w += Font_CharWidth(CON_WHITEMASK, 0xe080);
w += Font_CharWidth(CON_WHITEMASK, 0xe081)*16;
w += Font_CharWidth(CON_WHITEMASK, 0xe082);

94
engine/client/snd_dma.c
View File

@ -150,10 +150,10 @@ cvar_t snd_voip_showmeter = CVARAFD("cl_voip_showmeter", "1", NULL, CVAR_ARCHIV
cvar_t snd_voip_play = CVARAFDC("cl_voip_play", "1", NULL, CVAR_ARCHIVE, "Enables voip playback. Value is a volume scaler.", S_Voip_Play_Callback);
cvar_t snd_voip_ducking = CVARAFD("cl_voip_ducking", "0.5", NULL, CVAR_ARCHIVE, "Scales game audio by this much when someone is talking to you. Does not affect your speaker volume when you speak (minimum of cl_voip_capturingvol and cl_voip_ducking is used).");
cvar_t snd_voip_micamp = CVARAFDC("cl_voip_micamp", "2", NULL, CVAR_ARCHIVE, "Amplifies your microphone when using voip.", 0);
cvar_t snd_voip_codec = CVARAFDC("cl_voip_codec", "0", NULL, CVAR_ARCHIVE, "0: speex(@11khz). 1: raw. 2: opus. 3: speex(@8khz). 4: speex(@16). 5:speex(@32).", 0);
cvar_t snd_voip_codec = CVARAFDC("cl_voip_codec", "", NULL, CVAR_ARCHIVE, "0: speex(@11khz). 1: raw. 2: opus. 3: speex(@8khz). 4: speex(@16). 5:speex(@32).", 0);
cvar_t snd_voip_noisefilter = CVARAFDC("cl_voip_noisefilter", "1", NULL, CVAR_ARCHIVE, "Enable the use of the noise cancelation filter.", 0);
cvar_t snd_voip_autogain = CVARAFDC("cl_voip_autogain", "0", NULL, CVAR_ARCHIVE, "Attempts to normalize your voice levels to a standard level. Useful for lazy people, but interferes with voice activation levels.", 0);
cvar_t snd_voip_bitrate = CVARAFDC("cl_voip_bitrate", "0", NULL, CVAR_ARCHIVE, "For codecs with non-specific bitrates, this specifies the target bitrate to use (in kb).", 0);
cvar_t snd_voip_opus_bitrate = CVARAFDC("cl_voip_opus_bitrate", "3000", NULL, CVAR_ARCHIVE, "For codecs with non-specific bitrates, this specifies the target bitrate to use.", 0);
#endif
extern vfsfile_t *rawwritefile;
@ -248,7 +248,7 @@ void S_SoundInfo_f(void)
enum
{
VOIP_SPEEX_OLD = 0, //original supported codec (with needless padding and at the wrong rate to keep quake implementations easy)
VOIP_RAW = 1, //support is not recommended.
VOIP_RAW16 = 1, //support is not recommended.
VOIP_OPUS = 2, //supposed to be better than speex.
VOIP_SPEEX_NARROW = 3, //narrowband speex. packed data.
VOIP_SPEEX_WIDE = 4, //wideband speex. packed data.
@ -256,6 +256,7 @@ enum
VOIP_INVALID = 16 //not currently generating audio.
};
#define VOIP_DEFAULT_CODEC (cls.protocol==CP_QUAKEWORLD?VOIP_SPEEX_OLD:VOIP_OPUS) //opus is preferred, but ezquake is still common and only supports my first attempt at voice compression so favour that for quakeworld.
static struct
{
struct
@ -301,7 +302,7 @@ static struct
unsigned char decgen[MAX_CLIENTS]; /*last generation. if it changes, we flush speex to reset packet loss*/
unsigned int decsamplerate[MAX_CLIENTS];
unsigned int decframesize[MAX_CLIENTS];
float lastspoke[MAX_CLIENTS]; /*time when they're no longer considered talking. if future, they're talking*/
float lastspoke[MAX_CLIENTS]; /*time when they're no longer considered talking. if future, they're talking (timeout avoids flickering, and harder to troll with fake-tourettes when noone is looking)*/
float lastspoke_any;
unsigned char capturebuf[32768]; /*pending data*/
@ -511,8 +512,6 @@ static qboolean S_Opus_Init(void)
}
#endif
Con_Printf("OPUS support is experimental and should not be used\n"); //need to remove the packet length prefix.
s_voip.opus.loaded = true;
return s_voip.opus.loaded;
}
@ -551,7 +550,7 @@ void S_Voip_Decode(unsigned int sender, unsigned int codec, unsigned int gen, un
case VOIP_SPEEX_ULTRAWIDE:
qspeex_decoder_destroy(s_voip.decoder[sender]);
break;
case VOIP_RAW:
case VOIP_RAW16:
break;
case VOIP_OPUS:
qopus_decoder_destroy(s_voip.decoder[sender]);
@ -565,7 +564,7 @@ void S_Voip_Decode(unsigned int sender, unsigned int codec, unsigned int gen, un
{
default: //codec not supported.
return;
case VOIP_RAW:
case VOIP_RAW16:
s_voip.decsamplerate[sender] = 11025;
break;
case VOIP_SPEEX_OLD:
@ -634,7 +633,7 @@ void S_Voip_Decode(unsigned int sender, unsigned int codec, unsigned int gen, un
if (!s_voip.decoder[sender])
return;
s_voip.decframesize[sender] = (sizeof(decodebuf) / sizeof(decodebuf[0])) / 2; //this is the maximum size in a single frame.
s_voip.decframesize[sender] = s_voip.decsamplerate[sender]/400; //this is the maximum size in a single frame.
}
else
qopus_decoder_ctl(s_voip.decoder[sender], OPUS_RESET_STATE);
@ -715,7 +714,7 @@ void S_Voip_Decode(unsigned int sender, unsigned int codec, unsigned int gen, un
}
}
break;
case VOIP_RAW:
case VOIP_RAW16:
len = min(bytes, sizeof(decodebuf)-(sizeof(decodebuf[0])*decodesamps));
memcpy(decodebuf+decodesamps, start, len);
decodesamps += len / sizeof(decodebuf[0]);
@ -734,9 +733,10 @@ void S_Voip_Decode(unsigned int sender, unsigned int codec, unsigned int gen, un
// Con_Printf("Decoded %i frames from %i bytes\n", r, len);
if (r > 0)
{
int frames = r / s_voip.decframesize[sender];
decodesamps += r;
s_voip.decseq[sender] = (s_voip.decseq[sender] + 1) & 0xff;//r / s_voip.decframesize[sender];
seq = (seq+1)&0xff;//r / s_voip.decframesize[sender];
s_voip.decseq[sender] = (s_voip.decseq[sender] + frames) & 0xff;
seq = (seq+frames)&0xff;
}
else if (r < 0)
Con_Printf("Opus decoding error %i\n", r);
@ -909,7 +909,7 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
int len;
float micamp = snd_voip_micamp.value;
qboolean voipsendenable = true;
int voipcodec = snd_voip_codec.ival;
int voipcodec = *snd_voip_codec.string?snd_voip_codec.ival:VOIP_DEFAULT_CODEC;
qboolean rtpstream = NET_RTP_Active();
if (buf)
@ -1023,7 +1023,7 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
qspeex_encoder_ctl(s_voip.encoder, SPEEX_SET_SAMPLING_RATE, &s_voip.encsamplerate);
}
break;
case VOIP_RAW:
case VOIP_RAW16:
s_voip.encsamplerate = 11025;
s_voip.encframesize = 256;
break;
@ -1037,7 +1037,6 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
//use whatever is convienient.
s_voip.encsamplerate = 48000;
s_voip.encframesize = s_voip.encsamplerate / 400; //2.5ms frames, at a minimum.
s_voip.encframesize *= 4; //go for 10ms
s_voip.encoder = qopus_encoder_create(s_voip.encsamplerate, 1, OPUS_APPLICATION_VOIP, NULL);
if (!s_voip.encoder)
return;
@ -1100,7 +1099,7 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
case VOIP_SPEEX_ULTRAWIDE:
qspeex_bits_reset(&s_voip.speex.encbits);
break;
case VOIP_RAW:
case VOIP_RAW16:
break;
case VOIP_OPUS:
qopus_encoder_ctl(s_voip.encoder, OPUS_RESET_STATE);
@ -1196,7 +1195,7 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
len = qspeex_bits_write(&s_voip.speex.encbits, outbuf+outpos, sizeof(outbuf) - outpos);
outpos += len;
break;
case VOIP_RAW:
case VOIP_RAW16:
len = s_voip.capturepos-encpos; //amount of data to be eaten in this frame
len = min(len, sizeof(outbuf)-outpos);
len &= ~((s_voip.encframesize*2)-1);
@ -1216,38 +1215,36 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
//densely pack the frames.
start = (short*)(s_voip.capturebuf + encpos);
frames = (s_voip.capturepos-encpos)/2;
frames = s_voip.encframesize;
if (frames >= 2880)
frames = 2880;
else if (frames >= 1920)
frames = 1920;
else if (frames >= 960)
frames = 960;
else if (frames >= 480)
frames = 480;
else if (frames >= 240)
frames = 240;
else if (frames >= 120)
frames = 120;
else
{
Con_Printf("invalid Opus frame size\n");
frames = 0;
}
nrate = snd_voip_bitrate.value * 1000;
nrate = snd_voip_opus_bitrate.value;
if (nrate != s_voip.curbitrate)
{
s_voip.curbitrate = nrate;
if (nrate == 0)
nrate = -1000;
qopus_encoder_ctl(s_voip.encoder, OPUS_SET_BITRATE_REQUEST, (int)nrate);
nrate = 10000;
}
// Con_Printf("Encoding %i frames", frames);
if (frames >= 2880)
frames = 2880;
else if (frames >= 1920 && nrate > 100)
frames = 1920;
else if (frames >= 960 && nrate > 500)
frames = 960;
else if (frames >= 480 && nrate > 1000)
frames = 480;
else if (snd_voip_send.ival & 4)
break; //don't send small rtp packets, its abusive.
else if (frames >= 240 && nrate > 2000)
frames = 240;
else if (frames >= 120 && nrate > 4000)
frames = 120;
else
break; //invalid size, wait for more.
level += S_Voip_Preprocess(start, frames, micamp);
len = qopus_encode(s_voip.encoder, start, frames, outbuf+outpos, sizeof(outbuf) - outpos);
// Con_Printf(" (%i bytes)\n", len);
if (len >= 0)
{
s_voip.encsequence += frames / s_voip.encframesize;
@ -1301,15 +1298,20 @@ void S_Voip_Transmit(unsigned char clc, sizebuf_t *buf)
}
}
if (outpos && (!buf || buf->maxsize - buf->cursize >= outpos+4))
if (outpos)
{
if (buf && (snd_voip_send.ival != 4))
if (buf && !(snd_voip_send.ival & 4))
{
MSG_WriteByte(buf, clc);
MSG_WriteByte(buf, (s_voip.enccodec<<4) | (s_voip.generation & 0x0f)); /*gonna leave that nibble clear here... in this version, the client will ignore packets with those bits set. can use them for codec or something*/
MSG_WriteByte(buf, initseq&0xff);
MSG_WriteShort(buf, outpos);
SZ_Write(buf, outbuf, outpos);
if (buf->maxsize - buf->cursize >= 5+outpos)
{
MSG_WriteByte(buf, clc);
MSG_WriteByte(buf, (s_voip.enccodec<<4) | (s_voip.generation & 0x0f)); /*gonna leave that nibble clear here... in this version, the client will ignore packets with those bits set. can use them for codec or something*/
MSG_WriteByte(buf, initseq&0xff);
MSG_WriteShort(buf, outpos);
SZ_Write(buf, outbuf, outpos);
}
else
Con_Printf("Audio frame too small %i vs %i\n", outpos+4, buf->maxsize - buf->cursize);
}
#ifdef SUPPORT_ICE
@ -1379,6 +1381,7 @@ static void QDECL S_Voip_Play_Callback(cvar_t *var, char *oldval)
}
void S_Voip_MapChange(void)
{
voipbutton = false;
Cvar_ForceCallback(&snd_voip_play);
}
int S_Voip_Loudness(qboolean ignorevad)
@ -1417,6 +1420,7 @@ void S_Voip_Init(void)
Cvar_Register(&snd_voip_codec, "Voice Chat");
Cvar_Register(&snd_voip_noisefilter, "Voice Chat");
Cvar_Register(&snd_voip_autogain, "Voice Chat");
Cvar_Register(&snd_voip_opus_bitrate, "Voice Chat");
Cmd_AddCommand("+voip", S_Voip_Enable_f);
Cmd_AddCommand("-voip", S_Voip_Disable_f);
Cmd_AddCommand("voip", S_Voip_f);

1
engine/common/com_mesh.c
View File

@ -7217,7 +7217,6 @@ galiasinfo_t *Mod_ParseIQMMeshModel(model_t *mod, const char *buffer, size_t fsi
oevent->code = fteevents->evcode;
oevent->data = ZG_Malloc(&mod->memgroup, strlen(strings+fteevents->evdata_str)+1);
strcpy(oevent->data, strings+fteevents->evdata_str);
oevent->timestamp /= fgroup[fteevents->anim].rate;
link = &fgroup[fteevents->anim].events;
while (*link && (*link)->timestamp <= oevent->timestamp)
link = &(*link)->next;