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.)
This commit is contained in:
Zack Middleton
parent
fe619680f8
commit
615b73288f
13 changed files with 167 additions and 240 deletions
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@ -34,7 +34,8 @@ char *svc_strings[256] = {
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"svc_download",
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"svc_snapshot",
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"svc_EOF",
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"svc_voip",
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"svc_voipSpeex",
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"svc_voipOpus",
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};
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void SHOWNET( msg_t *msg, char *s) {
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@ -359,8 +360,8 @@ void CL_SystemInfoChanged( void ) {
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else
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#endif
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{
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s = Info_ValueForKey( systemInfo, "sv_voip" );
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clc.voipEnabled = atoi(s);
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s = Info_ValueForKey( systemInfo, "sv_voipProtocol" );
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clc.voipEnabled = !Q_stricmp(s, "opus");
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}
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#endif
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@ -680,13 +681,13 @@ static void CL_PlayVoip(int sender, int samplecnt, const byte *data, int flags)
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{
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if(flags & VOIP_DIRECT)
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{
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S_RawSamples(sender + 1, samplecnt, clc.speexSampleRate, 2, 1,
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S_RawSamples(sender + 1, samplecnt, 48000, 2, 1,
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data, clc.voipGain[sender], -1);
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}
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if(flags & VOIP_SPATIAL)
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{
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S_RawSamples(sender + MAX_CLIENTS + 1, samplecnt, clc.speexSampleRate, 2, 1,
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S_RawSamples(sender + MAX_CLIENTS + 1, samplecnt, 48000, 2, 1,
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data, 1.0f, sender);
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}
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}
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@ -699,8 +700,8 @@ A VoIP message has been received from the server
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=====================
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*/
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static
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void CL_ParseVoip ( msg_t *msg ) {
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static short decoded[4096]; // !!! FIXME: don't hardcode.
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void CL_ParseVoip ( msg_t *msg, qboolean ignoreData ) {
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static short decoded[VOIP_MAX_PACKET_SAMPLES*4]; // !!! FIXME: don't hard code
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const int sender = MSG_ReadShort(msg);
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const int generation = MSG_ReadByte(msg);
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@ -708,7 +709,8 @@ void CL_ParseVoip ( msg_t *msg ) {
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const int frames = MSG_ReadByte(msg);
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const int packetsize = MSG_ReadShort(msg);
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const int flags = MSG_ReadBits(msg, VOIP_FLAGCNT);
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char encoded[1024];
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unsigned char encoded[4000];
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int numSamples;
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int seqdiff;
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int written = 0;
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int i;
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@ -738,14 +740,15 @@ void CL_ParseVoip ( msg_t *msg ) {
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return; // overlarge packet, bail.
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}
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if (!clc.speexInitialized) {
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MSG_ReadData(msg, encoded, packetsize); // skip payload.
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return; // can't handle VoIP without libspeex!
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MSG_ReadData(msg, encoded, packetsize);
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if (ignoreData) {
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return; // just ignore legacy speex voip data
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} else if (!clc.voipCodecInitialized) {
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return; // can't handle VoIP without libopus!
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} else if (sender >= MAX_CLIENTS) {
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MSG_ReadData(msg, encoded, packetsize); // skip payload.
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return; // bogus sender.
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} else if (CL_ShouldIgnoreVoipSender(sender)) {
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MSG_ReadData(msg, encoded, packetsize); // skip payload.
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return; // Channel is muted, bail.
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}
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@ -758,70 +761,59 @@ void CL_ParseVoip ( msg_t *msg ) {
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// This is a new "generation" ... a new recording started, reset the bits.
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if (generation != clc.voipIncomingGeneration[sender]) {
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Com_DPrintf("VoIP: new generation %d!\n", generation);
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speex_bits_reset(&clc.speexDecoderBits[sender]);
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opus_decoder_ctl(clc.opusDecoder[sender], OPUS_RESET_STATE);
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clc.voipIncomingGeneration[sender] = generation;
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seqdiff = 0;
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} else if (seqdiff < 0) { // we're ahead of the sequence?!
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// This shouldn't happen unless the packet is corrupted or something.
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Com_DPrintf("VoIP: misordered sequence! %d < %d!\n",
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sequence, clc.voipIncomingSequence[sender]);
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// reset the bits just in case.
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speex_bits_reset(&clc.speexDecoderBits[sender]);
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// reset the decoder just in case.
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opus_decoder_ctl(clc.opusDecoder[sender], OPUS_RESET_STATE);
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seqdiff = 0;
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} else if (seqdiff * clc.speexFrameSize * 2 >= sizeof (decoded)) { // dropped more than we can handle?
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} else if (seqdiff * VOIP_MAX_PACKET_SAMPLES*2 >= sizeof (decoded)) { // dropped more than we can handle?
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// just start over.
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Com_DPrintf("VoIP: Dropped way too many (%d) frames from client #%d\n",
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seqdiff, sender);
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speex_bits_reset(&clc.speexDecoderBits[sender]);
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opus_decoder_ctl(clc.opusDecoder[sender], OPUS_RESET_STATE);
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seqdiff = 0;
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}
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if (seqdiff != 0) {
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Com_DPrintf("VoIP: Dropped %d frames from client #%d\n",
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seqdiff, sender);
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// tell speex that we're missing frames...
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// tell opus that we're missing frames...
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for (i = 0; i < seqdiff; i++) {
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assert((written + clc.speexFrameSize) * 2 < sizeof (decoded));
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speex_decode_int(clc.speexDecoder[sender], NULL, decoded + written);
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written += clc.speexFrameSize;
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assert((written + VOIP_MAX_PACKET_SAMPLES) * 2 < sizeof (decoded));
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numSamples = opus_decode(clc.opusDecoder[sender], NULL, VOIP_MAX_PACKET_SAMPLES * 2, decoded + written, sizeof (decoded) - written, 0);
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if ( numSamples <= 0 ) {
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Com_DPrintf("VoIP: Error decoding frame %d from client #%d\n", i, sender);
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continue;
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}
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written += numSamples;
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}
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}
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for (i = 0; i < frames; i++) {
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const int len = MSG_ReadByte(msg);
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if (len < 0) {
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Com_DPrintf("VoIP: Short packet!\n");
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break;
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}
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MSG_ReadData(msg, encoded, len);
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numSamples = opus_decode(clc.opusDecoder[sender], encoded, packetsize, decoded + written, sizeof (decoded) - written, 0);
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// shouldn't happen, but just in case...
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if ((written + clc.speexFrameSize) * 2 > sizeof (decoded)) {
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Com_DPrintf("VoIP: playback %d bytes, %d samples, %d frames\n",
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written * 2, written, i);
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CL_PlayVoip(sender, written, (const byte *) decoded, flags);
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written = 0;
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}
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speex_bits_read_from(&clc.speexDecoderBits[sender], encoded, len);
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speex_decode_int(clc.speexDecoder[sender],
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&clc.speexDecoderBits[sender], decoded + written);
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#if 0
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static FILE *encio = NULL;
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if (encio == NULL) encio = fopen("voip-incoming-encoded.bin", "wb");
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if (encio != NULL) { fwrite(encoded, len, 1, encio); fflush(encio); }
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static FILE *decio = NULL;
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if (decio == NULL) decio = fopen("voip-incoming-decoded.bin", "wb");
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if (decio != NULL) { fwrite(decoded+written, clc.speexFrameSize*2, 1, decio); fflush(decio); }
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#endif
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written += clc.speexFrameSize;
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if ( numSamples <= 0 ) {
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Com_DPrintf("VoIP: Error decoding voip data from client #%d\n", sender);
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numSamples = 0;
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}
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#if 0
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static FILE *encio = NULL;
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if (encio == NULL) encio = fopen("voip-incoming-encoded.bin", "wb");
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if (encio != NULL) { fwrite(encoded, len, 1, encio); fflush(encio); }
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static FILE *decio = NULL;
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if (decio == NULL) decio = fopen("voip-incoming-decoded.bin", "wb");
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if (decio != NULL) { fwrite(decoded+written, clc.speexFrameSize*2, 1, decio); fflush(decio); }
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#endif
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written += numSamples;
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Com_DPrintf("VoIP: playback %d bytes, %d samples, %d frames\n",
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written * 2, written, i);
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written * 2, written, frames);
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if(written > 0)
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CL_PlayVoip(sender, written, (const byte *) decoded, flags);
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@ -924,9 +916,14 @@ void CL_ParseServerMessage( msg_t *msg ) {
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case svc_download:
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CL_ParseDownload( msg );
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break;
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case svc_voip:
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case svc_voipSpeex:
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#ifdef USE_VOIP
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CL_ParseVoip( msg );
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CL_ParseVoip( msg, qtrue );
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#endif
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break;
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case svc_voipOpus:
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#ifdef USE_VOIP
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CL_ParseVoip( msg, !clc.voipEnabled );
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#endif
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break;
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}
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