From 975d4d97e4b9459c3d21b4dc3ecd807e9c330d9a Mon Sep 17 00:00:00 2001 From: Zack Middleton Date: Tue, 10 Dec 2013 21:14:13 -0600 Subject: Use Opus for VoIP Server/client VoIP protocol is handled by adding new cvars cl_voipProtocol and sv_voipProtocol, sv_voip and cl_voip are used to auto set/clear them. All users need to touch are cl/sv_voip as 0 or 1 just like before. Old Speex VoIP packets in demos are skipped. New VoIP packets are skipped in demos if sv_voipProtocol doesn't match cl_voipProtocol. Notable difference between usage of speex and opus codecs, when using Speex client would be sent 80ms at a time. Using Opus, 60ms is sent at a time. This was changed because the Opus codec supports encoding up to 60ms at a time. (Simpler to send only one codec frame in a packet.) --- src/client/cl_cgame.c | 36 ++++++---------- src/client/cl_input.c | 4 +- src/client/cl_main.c | 108 +++++++++++++++++++++-------------------------- src/client/cl_parse.c | 105 ++++++++++++++++++++++----------------------- src/client/client.h | 21 ++++----- src/client/snd_openal.c | 8 +--- src/qcommon/qcommon.h | 6 ++- src/server/server.h | 3 +- src/server/sv_client.c | 22 +++++++--- src/server/sv_init.c | 3 +- src/server/sv_main.c | 5 ++- src/server/sv_snapshot.c | 2 +- 12 files changed, 154 insertions(+), 169 deletions(-) (limited to 'src') diff --git a/src/client/cl_cgame.c b/src/client/cl_cgame.c index 1c7d91cc..23f3010c 100644 --- a/src/client/cl_cgame.c +++ b/src/client/cl_cgame.c @@ -956,37 +956,27 @@ void CL_FirstSnapshot( void ) { #endif #ifdef USE_VOIP - if (!clc.speexInitialized) { + if (!clc.voipCodecInitialized) { int i; - speex_bits_init(&clc.speexEncoderBits); - speex_bits_reset(&clc.speexEncoderBits); + int error; - clc.speexEncoder = speex_encoder_init(&speex_nb_mode); + clc.opusEncoder = opus_encoder_create(48000, 1, OPUS_APPLICATION_VOIP, &error); - speex_encoder_ctl(clc.speexEncoder, SPEEX_GET_FRAME_SIZE, - &clc.speexFrameSize); - speex_encoder_ctl(clc.speexEncoder, SPEEX_GET_SAMPLING_RATE, - &clc.speexSampleRate); - - clc.speexPreprocessor = speex_preprocess_state_init(clc.speexFrameSize, - clc.speexSampleRate); - - i = 1; - speex_preprocess_ctl(clc.speexPreprocessor, - SPEEX_PREPROCESS_SET_DENOISE, &i); - - i = 1; - speex_preprocess_ctl(clc.speexPreprocessor, - SPEEX_PREPROCESS_SET_AGC, &i); + if ( error ) { + Com_DPrintf("VoIP: Error opus_encoder_create %d\n", error); + return; + } for (i = 0; i < MAX_CLIENTS; i++) { - speex_bits_init(&clc.speexDecoderBits[i]); - speex_bits_reset(&clc.speexDecoderBits[i]); - clc.speexDecoder[i] = speex_decoder_init(&speex_nb_mode); + clc.opusDecoder[i] = opus_decoder_create(48000, 1, &error); + if ( error ) { + Com_DPrintf("VoIP: Error opus_decoder_create(%d) %d\n", i, error); + return; + } clc.voipIgnore[i] = qfalse; clc.voipGain[i] = 1.0f; } - clc.speexInitialized = qtrue; + clc.voipCodecInitialized = qtrue; clc.voipMuteAll = qfalse; Cmd_AddCommand ("voip", CL_Voip_f); Cvar_Set("cl_voipSendTarget", "spatial"); diff --git a/src/client/cl_input.c b/src/client/cl_input.c index fdd0c474..48bf51df 100644 --- a/src/client/cl_input.c +++ b/src/client/cl_input.c @@ -789,7 +789,7 @@ void CL_WritePacket( void ) { { if((clc.voipFlags & VOIP_SPATIAL) || Com_IsVoipTarget(clc.voipTargets, sizeof(clc.voipTargets), -1)) { - MSG_WriteByte (&buf, clc_voip); + MSG_WriteByte (&buf, clc_voipOpus); MSG_WriteByte (&buf, clc.voipOutgoingGeneration); MSG_WriteLong (&buf, clc.voipOutgoingSequence); MSG_WriteByte (&buf, clc.voipOutgoingDataFrames); @@ -810,7 +810,7 @@ void CL_WritePacket( void ) { MSG_Init (&fakemsg, fakedata, sizeof (fakedata)); MSG_Bitstream (&fakemsg); MSG_WriteLong (&fakemsg, clc.reliableAcknowledge); - MSG_WriteByte (&fakemsg, svc_voip); + MSG_WriteByte (&fakemsg, svc_voipOpus); MSG_WriteShort (&fakemsg, clc.clientNum); MSG_WriteByte (&fakemsg, clc.voipOutgoingGeneration); MSG_WriteLong (&fakemsg, clc.voipOutgoingSequence); diff --git a/src/client/cl_main.c b/src/client/cl_main.c index fa47988e..6df57b4d 100644 --- a/src/client/cl_main.c +++ b/src/client/cl_main.c @@ -45,6 +45,7 @@ cvar_t *cl_voipSendTarget; cvar_t *cl_voipGainDuringCapture; cvar_t *cl_voipCaptureMult; cvar_t *cl_voipShowMeter; +cvar_t *cl_voipProtocol; cvar_t *cl_voip; #endif @@ -250,8 +251,8 @@ void CL_Voip_f( void ) if (clc.state != CA_ACTIVE) reason = "Not connected to a server"; - else if (!clc.speexInitialized) - reason = "Speex not initialized"; + else if (!clc.voipCodecInitialized) + reason = "Voip codec not initialized"; else if (!clc.voipEnabled) reason = "Server doesn't support VoIP"; @@ -304,6 +305,8 @@ void CL_VoipNewGeneration(void) clc.voipOutgoingGeneration = 1; clc.voipPower = 0.0f; clc.voipOutgoingSequence = 0; + + opus_encoder_ctl(clc.opusEncoder, OPUS_RESET_STATE); } /* @@ -392,7 +395,7 @@ void CL_VoipParseTargets(void) =============== CL_CaptureVoip -Record more audio from the hardware if required and encode it into Speex +Record more audio from the hardware if required and encode it into Opus data for later transmission. =============== */ @@ -422,11 +425,12 @@ void CL_CaptureVoip(void) Com_Printf("Until then, VoIP is disabled.\n"); Cvar_Set("cl_voip", "0"); } + Cvar_Set("cl_voipProtocol", cl_voip->integer ? "opus" : ""); cl_voip->modified = qfalse; cl_rate->modified = qfalse; } - if (!clc.speexInitialized) + if (!clc.voipCodecInitialized) return; // just in case this gets called at a bad time. if (clc.voipOutgoingDataSize > 0) @@ -479,80 +483,67 @@ void CL_CaptureVoip(void) if ((cl_voipSend->integer) || (finalFrame)) { // user wants to capture audio? int samples = S_AvailableCaptureSamples(); - const int mult = (finalFrame) ? 1 : 4; // 4 == 80ms of audio. + const int packetSamples = (finalFrame) ? VOIP_MAX_FRAME_SAMPLES : VOIP_MAX_PACKET_SAMPLES; // enough data buffered in audio hardware to process yet? - if (samples >= (clc.speexFrameSize * mult)) { - // audio capture is always MONO16 (and that's what speex wants!). - // 2048 will cover 12 uncompressed frames in narrowband mode. - static int16_t sampbuffer[2048]; + if (samples >= packetSamples) { + // audio capture is always MONO16. + static int16_t sampbuffer[VOIP_MAX_PACKET_SAMPLES]; float voipPower = 0.0f; - int speexFrames = 0; - int wpos = 0; - int pos = 0; + int voipFrames; + int i, bytes; - if (samples > (clc.speexFrameSize * 4)) - samples = (clc.speexFrameSize * 4); + if (samples > VOIP_MAX_PACKET_SAMPLES) + samples = VOIP_MAX_PACKET_SAMPLES; // !!! FIXME: maybe separate recording from encoding, so voipPower // !!! FIXME: updates faster than 4Hz? - samples -= samples % clc.speexFrameSize; - S_Capture(samples, (byte *) sampbuffer); // grab from audio card. - - // this will probably generate multiple speex packets each time. - while (samples > 0) { - int16_t *sampptr = &sampbuffer[pos]; - int i, bytes; + samples -= samples % VOIP_MAX_FRAME_SAMPLES; + if (samples != 120 && samples != 240 && samples != 480 && samples != 960 && samples != 1920 && samples != 2880 ) { + Com_Printf("Voip: bad number of samples %d\n", samples); + return; + } + voipFrames = samples / VOIP_MAX_FRAME_SAMPLES; - // preprocess samples to remove noise... - speex_preprocess_run(clc.speexPreprocessor, sampptr); + S_Capture(samples, (byte *) sampbuffer); // grab from audio card. - // check the "power" of this packet... - for (i = 0; i < clc.speexFrameSize; i++) { - const float flsamp = (float) sampptr[i]; - const float s = fabs(flsamp); - voipPower += s * s; - sampptr[i] = (int16_t) ((flsamp) * audioMult); - } + // check the "power" of this packet... + for (i = 0; i < samples; i++) { + const float flsamp = (float) sampbuffer[i]; + const float s = fabs(flsamp); + voipPower += s * s; + sampbuffer[i] = (int16_t) ((flsamp) * audioMult); + } - // encode raw audio samples into Speex data... - speex_bits_reset(&clc.speexEncoderBits); - speex_encode_int(clc.speexEncoder, sampptr, - &clc.speexEncoderBits); - bytes = speex_bits_write(&clc.speexEncoderBits, - (char *) &clc.voipOutgoingData[wpos+1], - sizeof (clc.voipOutgoingData) - (wpos+1)); - assert((bytes > 0) && (bytes < 256)); - clc.voipOutgoingData[wpos] = (byte) bytes; - wpos += bytes + 1; - - // look at the data for the next packet... - pos += clc.speexFrameSize; - samples -= clc.speexFrameSize; - speexFrames++; + // encode raw audio samples into Opus data... + bytes = opus_encode(clc.opusEncoder, sampbuffer, samples, + (unsigned char *) clc.voipOutgoingData, + sizeof (clc.voipOutgoingData)); + if ( bytes <= 0 ) { + Com_DPrintf("VoIP: Error encoding %d samples\n", samples); + bytes = 0; } clc.voipPower = (voipPower / (32768.0f * 32768.0f * - ((float) (clc.speexFrameSize * speexFrames)))) * - 100.0f; + ((float) samples))) * 100.0f; if ((useVad) && (clc.voipPower < cl_voipVADThreshold->value)) { CL_VoipNewGeneration(); // no "talk" for at least 1/4 second. } else { - clc.voipOutgoingDataSize = wpos; - clc.voipOutgoingDataFrames = speexFrames; + clc.voipOutgoingDataSize = bytes; + clc.voipOutgoingDataFrames = voipFrames; Com_DPrintf("VoIP: Send %d frames, %d bytes, %f power\n", - speexFrames, wpos, clc.voipPower); + voipFrames, bytes, clc.voipPower); #if 0 static FILE *encio = NULL; if (encio == NULL) encio = fopen("voip-outgoing-encoded.bin", "wb"); - if (encio != NULL) { fwrite(clc.voipOutgoingData, wpos, 1, encio); fflush(encio); } + if (encio != NULL) { fwrite(clc.voipOutgoingData, bytes, 1, encio); fflush(encio); } static FILE *decio = NULL; if (decio == NULL) decio = fopen("voip-outgoing-decoded.bin", "wb"); - if (decio != NULL) { fwrite(sampbuffer, speexFrames * clc.speexFrameSize * 2, 1, decio); fflush(decio); } + if (decio != NULL) { fwrite(sampbuffer, voipFrames * VOIP_MAX_FRAME_SAMPLES * 2, 1, decio); fflush(decio); } #endif } } @@ -1419,14 +1410,11 @@ void CL_Disconnect( qboolean showMainMenu ) { cl_voipUseVAD->integer = tmp; } - if (clc.speexInitialized) { + if (clc.voipCodecInitialized) { int i; - speex_bits_destroy(&clc.speexEncoderBits); - speex_encoder_destroy(clc.speexEncoder); - speex_preprocess_state_destroy(clc.speexPreprocessor); + opus_encoder_destroy(clc.opusEncoder); for (i = 0; i < MAX_CLIENTS; i++) { - speex_bits_destroy(&clc.speexDecoderBits[i]); - speex_decoder_destroy(clc.speexDecoder[i]); + opus_decoder_destroy(clc.opusDecoder[i]); } } Cmd_RemoveCommand ("voip"); @@ -3706,9 +3694,9 @@ void CL_Init( void ) { cl_voipVADThreshold = Cvar_Get ("cl_voipVADThreshold", "0.25", CVAR_ARCHIVE); cl_voipShowMeter = Cvar_Get ("cl_voipShowMeter", "1", CVAR_ARCHIVE); - // This is a protocol version number. - cl_voip = Cvar_Get ("cl_voip", "1", CVAR_USERINFO | CVAR_ARCHIVE); + cl_voip = Cvar_Get ("cl_voip", "1", CVAR_ARCHIVE); Cvar_CheckRange( cl_voip, 0, 1, qtrue ); + cl_voipProtocol = Cvar_Get ("cl_voipProtocol", cl_voip->integer ? "opus" : "", CVAR_USERINFO | CVAR_ROM); #endif diff --git a/src/client/cl_parse.c b/src/client/cl_parse.c index a8b68ca5..4230c04f 100644 --- a/src/client/cl_parse.c +++ b/src/client/cl_parse.c @@ -35,7 +35,8 @@ char *svc_strings[256] = { "svc_download", "svc_snapshot", "svc_EOF", - "svc_voip", + "svc_voipSpeex", + "svc_voipOpus", }; void SHOWNET( msg_t *msg, char *s) { @@ -354,8 +355,8 @@ void CL_SystemInfoChanged( void ) { cl.serverId = atoi( Info_ValueForKey( systemInfo, "sv_serverid" ) ); #ifdef USE_VOIP - s = Info_ValueForKey( systemInfo, "sv_voip" ); - clc.voipEnabled = atoi(s); + s = Info_ValueForKey( systemInfo, "sv_voipProtocol" ); + clc.voipEnabled = !Q_stricmp(s, "opus"); #endif // don't set any vars when playing a demo @@ -674,13 +675,13 @@ static void CL_PlayVoip(int sender, int samplecnt, const byte *data, int flags) { if(flags & VOIP_DIRECT) { - S_RawSamples(sender + 1, samplecnt, clc.speexSampleRate, 2, 1, + S_RawSamples(sender + 1, samplecnt, 48000, 2, 1, data, clc.voipGain[sender], -1); } if(flags & VOIP_SPATIAL) { - S_RawSamples(sender + MAX_CLIENTS + 1, samplecnt, clc.speexSampleRate, 2, 1, + S_RawSamples(sender + MAX_CLIENTS + 1, samplecnt, 48000, 2, 1, data, 1.0f, sender); } } @@ -693,8 +694,8 @@ A VoIP message has been received from the server ===================== */ static -void CL_ParseVoip ( msg_t *msg ) { - static short decoded[4096]; // !!! FIXME: don't hardcode. +void CL_ParseVoip ( msg_t *msg, qboolean ignoreData ) { + static short decoded[VOIP_MAX_PACKET_SAMPLES*4]; // !!! FIXME: don't hard code const int sender = MSG_ReadShort(msg); const int generation = MSG_ReadByte(msg); @@ -702,7 +703,8 @@ void CL_ParseVoip ( msg_t *msg ) { const int frames = MSG_ReadByte(msg); const int packetsize = MSG_ReadShort(msg); const int flags = MSG_ReadBits(msg, VOIP_FLAGCNT); - char encoded[1024]; + unsigned char encoded[4000]; + int numSamples; int seqdiff; int written = 0; int i; @@ -732,14 +734,15 @@ void CL_ParseVoip ( msg_t *msg ) { return; // overlarge packet, bail. } - if (!clc.speexInitialized) { - MSG_ReadData(msg, encoded, packetsize); // skip payload. - return; // can't handle VoIP without libspeex! + MSG_ReadData(msg, encoded, packetsize); + + if (ignoreData) { + return; // just ignore legacy speex voip data + } else if (!clc.voipCodecInitialized) { + return; // can't handle VoIP without libopus! } else if (sender >= MAX_CLIENTS) { - MSG_ReadData(msg, encoded, packetsize); // skip payload. return; // bogus sender. } else if (CL_ShouldIgnoreVoipSender(sender)) { - MSG_ReadData(msg, encoded, packetsize); // skip payload. return; // Channel is muted, bail. } @@ -752,70 +755,59 @@ void CL_ParseVoip ( msg_t *msg ) { // This is a new "generation" ... a new recording started, reset the bits. if (generation != clc.voipIncomingGeneration[sender]) { Com_DPrintf("VoIP: new generation %d!\n", generation); - speex_bits_reset(&clc.speexDecoderBits[sender]); + opus_decoder_ctl(clc.opusDecoder[sender], OPUS_RESET_STATE); clc.voipIncomingGeneration[sender] = generation; seqdiff = 0; } else if (seqdiff < 0) { // we're ahead of the sequence?! // This shouldn't happen unless the packet is corrupted or something. Com_DPrintf("VoIP: misordered sequence! %d < %d!\n", sequence, clc.voipIncomingSequence[sender]); - // reset the bits just in case. - speex_bits_reset(&clc.speexDecoderBits[sender]); + // reset the decoder just in case. + opus_decoder_ctl(clc.opusDecoder[sender], OPUS_RESET_STATE); seqdiff = 0; - } else if (seqdiff * clc.speexFrameSize * 2 >= sizeof (decoded)) { // dropped more than we can handle? + } else if (seqdiff * VOIP_MAX_PACKET_SAMPLES*2 >= sizeof (decoded)) { // dropped more than we can handle? // just start over. Com_DPrintf("VoIP: Dropped way too many (%d) frames from client #%d\n", seqdiff, sender); - speex_bits_reset(&clc.speexDecoderBits[sender]); + opus_decoder_ctl(clc.opusDecoder[sender], OPUS_RESET_STATE); seqdiff = 0; } if (seqdiff != 0) { Com_DPrintf("VoIP: Dropped %d frames from client #%d\n", seqdiff, sender); - // tell speex that we're missing frames... + // tell opus that we're missing frames... for (i = 0; i < seqdiff; i++) { - assert((written + clc.speexFrameSize) * 2 < sizeof (decoded)); - speex_decode_int(clc.speexDecoder[sender], NULL, decoded + written); - written += clc.speexFrameSize; + assert((written + VOIP_MAX_PACKET_SAMPLES) * 2 < sizeof (decoded)); + numSamples = opus_decode(clc.opusDecoder[sender], NULL, VOIP_MAX_PACKET_SAMPLES * 2, decoded + written, sizeof (decoded) - written, 0); + if ( numSamples <= 0 ) { + Com_DPrintf("VoIP: Error decoding frame %d from client #%d\n", i, sender); + continue; + } + written += numSamples; } } - for (i = 0; i < frames; i++) { - const int len = MSG_ReadByte(msg); - if (len < 0) { - Com_DPrintf("VoIP: Short packet!\n"); - break; - } - MSG_ReadData(msg, encoded, len); + numSamples = opus_decode(clc.opusDecoder[sender], encoded, packetsize, decoded + written, sizeof (decoded) - written, 0); - // shouldn't happen, but just in case... - if ((written + clc.speexFrameSize) * 2 > sizeof (decoded)) { - Com_DPrintf("VoIP: playback %d bytes, %d samples, %d frames\n", - written * 2, written, i); - - CL_PlayVoip(sender, written, (const byte *) decoded, flags); - written = 0; - } - - speex_bits_read_from(&clc.speexDecoderBits[sender], encoded, len); - speex_decode_int(clc.speexDecoder[sender], - &clc.speexDecoderBits[sender], decoded + written); + if ( numSamples <= 0 ) { + Com_DPrintf("VoIP: Error decoding voip data from client #%d\n", sender); + numSamples = 0; + } - #if 0 - static FILE *encio = NULL; - if (encio == NULL) encio = fopen("voip-incoming-encoded.bin", "wb"); - if (encio != NULL) { fwrite(encoded, len, 1, encio); fflush(encio); } - static FILE *decio = NULL; - if (decio == NULL) decio = fopen("voip-incoming-decoded.bin", "wb"); - if (decio != NULL) { fwrite(decoded+written, clc.speexFrameSize*2, 1, decio); fflush(decio); } - #endif + #if 0 + static FILE *encio = NULL; + if (encio == NULL) encio = fopen("voip-incoming-encoded.bin", "wb"); + if (encio != NULL) { fwrite(encoded, len, 1, encio); fflush(encio); } + static FILE *decio = NULL; + if (decio == NULL) decio = fopen("voip-incoming-decoded.bin", "wb"); + if (decio != NULL) { fwrite(decoded+written, clc.speexFrameSize*2, 1, decio); fflush(decio); } + #endif - written += clc.speexFrameSize; - } + written += numSamples; Com_DPrintf("VoIP: playback %d bytes, %d samples, %d frames\n", - written * 2, written, i); + written * 2, written, frames); if(written > 0) CL_PlayVoip(sender, written, (const byte *) decoded, flags); @@ -918,9 +910,14 @@ void CL_ParseServerMessage( msg_t *msg ) { case svc_download: CL_ParseDownload( msg ); break; - case svc_voip: + case svc_voipSpeex: +#ifdef USE_VOIP + CL_ParseVoip( msg, qtrue ); +#endif + break; + case svc_voipOpus: #ifdef USE_VOIP - CL_ParseVoip( msg ); + CL_ParseVoip( msg, !clc.voipEnabled ); #endif break; } diff --git a/src/client/client.h b/src/client/client.h index 541dca40..b4016467 100644 --- a/src/client/client.h +++ b/src/client/client.h @@ -36,8 +36,7 @@ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA #endif /* USE_CURL */ #ifdef USE_VOIP -#include "speex/speex.h" -#include "speex/speex_preprocess.h" +#include #endif // file full of random crap that gets used to create cl_guid @@ -241,14 +240,11 @@ typedef struct { #ifdef USE_VOIP qboolean voipEnabled; - qboolean speexInitialized; - int speexFrameSize; - int speexSampleRate; + qboolean voipCodecInitialized; // incoming data... // !!! FIXME: convert from parallel arrays to array of a struct. - SpeexBits speexDecoderBits[MAX_CLIENTS]; - void *speexDecoder[MAX_CLIENTS]; + OpusDecoder *opusDecoder[MAX_CLIENTS]; byte voipIncomingGeneration[MAX_CLIENTS]; int voipIncomingSequence[MAX_CLIENTS]; float voipGain[MAX_CLIENTS]; @@ -260,9 +256,7 @@ typedef struct { // then we are sending to clientnum i. uint8_t voipTargets[(MAX_CLIENTS + 7) / 8]; uint8_t voipFlags; - SpeexPreprocessState *speexPreprocessor; - SpeexBits speexEncoderBits; - void *speexEncoder; + OpusEncoder *opusEncoder; int voipOutgoingDataSize; int voipOutgoingDataFrames; int voipOutgoingSequence; @@ -449,6 +443,13 @@ extern cvar_t *cl_voipGainDuringCapture; extern cvar_t *cl_voipCaptureMult; extern cvar_t *cl_voipShowMeter; extern cvar_t *cl_voip; + +// 20ms at 48k +#define VOIP_MAX_FRAME_SAMPLES ( 20 * 48 ) + +// 3 frame is 60ms of audio, the max opus will encode at once +#define VOIP_MAX_PACKET_FRAMES 3 +#define VOIP_MAX_PACKET_SAMPLES ( VOIP_MAX_FRAME_SAMPLES * VOIP_MAX_PACKET_FRAMES ) #endif //================================================= diff --git a/src/client/snd_openal.c b/src/client/snd_openal.c index d90e6d23..b5c89918 100644 --- a/src/client/snd_openal.c +++ b/src/client/snd_openal.c @@ -2700,16 +2700,12 @@ qboolean S_AL_Init( soundInterface_t *si ) s_alAvailableInputDevices = Cvar_Get("s_alAvailableInputDevices", inputdevicenames, CVAR_ROM | CVAR_NORESTART); - // !!! FIXME: 8000Hz is what Speex narrowband mode needs, but we - // !!! FIXME: should probably open the capture device after - // !!! FIXME: initializing Speex so we can change to wideband - // !!! FIXME: if we like. Com_Printf("OpenAL default capture device is '%s'\n", defaultinputdevice ? defaultinputdevice : "none"); - alCaptureDevice = qalcCaptureOpenDevice(inputdevice, 8000, AL_FORMAT_MONO16, 4096); + alCaptureDevice = qalcCaptureOpenDevice(inputdevice, 48000, AL_FORMAT_MONO16, VOIP_MAX_PACKET_SAMPLES*4); if( !alCaptureDevice && inputdevice ) { Com_Printf( "Failed to open OpenAL Input device '%s', trying default.\n", inputdevice ); - alCaptureDevice = qalcCaptureOpenDevice(NULL, 8000, AL_FORMAT_MONO16, 4096); + alCaptureDevice = qalcCaptureOpenDevice(NULL, 48000, AL_FORMAT_MONO16, VOIP_MAX_PACKET_SAMPLES*4); } Com_Printf( "OpenAL capture device %s.\n", (alCaptureDevice == NULL) ? "failed to open" : "opened"); diff --git a/src/qcommon/qcommon.h b/src/qcommon/qcommon.h index d71531b6..1b4a5a16 100644 --- a/src/qcommon/qcommon.h +++ b/src/qcommon/qcommon.h @@ -279,7 +279,8 @@ enum svc_ops_e { svc_EOF, // new commands, supported only by ioquake3 protocol but not legacy - svc_voip, // not wrapped in USE_VOIP, so this value is reserved. + svc_voipSpeex, // not wrapped in USE_VOIP, so this value is reserved. + svc_voipOpus, // }; @@ -295,7 +296,8 @@ enum clc_ops_e { clc_EOF, // new commands, supported only by ioquake3 protocol but not legacy - clc_voip, // not wrapped in USE_VOIP, so this value is reserved. + clc_voipSpeex, // not wrapped in USE_VOIP, so this value is reserved. + clc_voipOpus, // }; /* diff --git a/src/server/server.h b/src/server/server.h index 59b3ff8b..57d34b5b 100644 --- a/src/server/server.h +++ b/src/server/server.h @@ -45,7 +45,7 @@ typedef struct voipServerPacket_s int len; int sender; int flags; - byte data[1024]; + byte data[4000]; } voipServerPacket_t; #endif @@ -281,6 +281,7 @@ extern cvar_t *sv_banFile; #ifdef USE_VOIP extern cvar_t *sv_voip; +extern cvar_t *sv_voipProtocol; #endif diff --git a/src/server/sv_client.c b/src/server/sv_client.c index ba4392f7..dcf6311e 100644 --- a/src/server/sv_client.c +++ b/src/server/sv_client.c @@ -1200,8 +1200,8 @@ void SV_UserinfoChanged( client_t *cl ) { } #ifdef USE_VOIP - val = Info_ValueForKey(cl->userinfo, "cl_voip"); - cl->hasVoip = atoi(val); + val = Info_ValueForKey(cl->userinfo, "cl_voipProtocol"); + cl->hasVoip = !Q_stricmp( val, "opus" ); #endif // TTimo @@ -1536,7 +1536,7 @@ static qboolean SV_ShouldIgnoreVoipSender(const client_t *cl) } static -void SV_UserVoip(client_t *cl, msg_t *msg) +void SV_UserVoip(client_t *cl, msg_t *msg, qboolean ignoreData) { int sender, generation, sequence, frames, packetsize; uint8_t recips[(MAX_CLIENTS + 7) / 8]; @@ -1571,12 +1571,12 @@ void SV_UserVoip(client_t *cl, msg_t *msg) MSG_ReadData(msg, encoded, packetsize); - if (SV_ShouldIgnoreVoipSender(cl)) + if (ignoreData || SV_ShouldIgnoreVoipSender(cl)) return; // Blacklisted, disabled, etc. // !!! FIXME: see if we read past end of msg... - // !!! FIXME: reject if not speex narrowband codec. + // !!! FIXME: reject if not opus data. // !!! FIXME: decide if this is bogus data? // decide who needs this VoIP packet sent to them... @@ -1725,10 +1725,18 @@ void SV_ExecuteClientMessage( client_t *cl, msg_t *msg ) { } } while ( 1 ); + // skip legacy speex voip data + if ( c == clc_voipSpeex ) { +#ifdef USE_VOIP + SV_UserVoip( cl, msg, qtrue ); + c = MSG_ReadByte( msg ); +#endif + } + // read optional voip data - if ( c == clc_voip ) { + if ( c == clc_voipOpus ) { #ifdef USE_VOIP - SV_UserVoip( cl, msg ); + SV_UserVoip( cl, msg, qfalse ); c = MSG_ReadByte( msg ); #endif } diff --git a/src/server/sv_init.c b/src/server/sv_init.c index 30fd3936..997d58f6 100644 --- a/src/server/sv_init.c +++ b/src/server/sv_init.c @@ -646,8 +646,9 @@ void SV_Init (void) sv_serverid = Cvar_Get ("sv_serverid", "0", CVAR_SYSTEMINFO | CVAR_ROM ); sv_pure = Cvar_Get ("sv_pure", "1", CVAR_SYSTEMINFO ); #ifdef USE_VOIP - sv_voip = Cvar_Get("sv_voip", "1", CVAR_SYSTEMINFO | CVAR_LATCH); + sv_voip = Cvar_Get("sv_voip", "1", CVAR_LATCH); Cvar_CheckRange(sv_voip, 0, 1, qtrue); + sv_voipProtocol = Cvar_Get("sv_voipProtocol", sv_voip->integer ? "opus" : "", CVAR_SYSTEMINFO | CVAR_ROM ); #endif Cvar_Get ("sv_paks", "", CVAR_SYSTEMINFO | CVAR_ROM ); Cvar_Get ("sv_pakNames", "", CVAR_SYSTEMINFO | CVAR_ROM ); diff --git a/src/server/sv_main.c b/src/server/sv_main.c index 77d1a35d..0c4c7f46 100644 --- a/src/server/sv_main.c +++ b/src/server/sv_main.c @@ -25,6 +25,7 @@ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA #ifdef USE_VOIP cvar_t *sv_voip; +cvar_t *sv_voipProtocol; #endif serverStatic_t svs; // persistant server info @@ -656,8 +657,8 @@ void SVC_Info( netadr_t from ) { Info_SetValueForKey( infostring, "pure", va("%i", sv_pure->integer ) ); #ifdef USE_VOIP - if (sv_voip->integer) { - Info_SetValueForKey( infostring, "voip", va("%i", sv_voip->integer ) ); + if (sv_voipProtocol->string && *sv_voipProtocol->string) { + Info_SetValueForKey( infostring, "voip", sv_voipProtocol->string ); } #endif diff --git a/src/server/sv_snapshot.c b/src/server/sv_snapshot.c index 71c4b86c..58136388 100644 --- a/src/server/sv_snapshot.c +++ b/src/server/sv_snapshot.c @@ -548,7 +548,7 @@ static void SV_WriteVoipToClient(client_t *cl, msg_t *msg) if (totalbytes > (msg->maxsize - msg->cursize) / 2) break; - MSG_WriteByte(msg, svc_voip); + MSG_WriteByte(msg, svc_voipOpus); MSG_WriteShort(msg, packet->sender); MSG_WriteByte(msg, (byte) packet->generation); MSG_WriteLong(msg, packet->sequence); -- cgit