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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@ -44,6 +44,7 @@ cvar_t *cl_voipSendTarget;
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cvar_t *cl_voipGainDuringCapture;
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cvar_t *cl_voipCaptureMult;
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cvar_t *cl_voipShowMeter;
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cvar_t *cl_voipProtocol;
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cvar_t *cl_voip;
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#endif
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@ -250,8 +251,8 @@ void CL_Voip_f( void )
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if (clc.state != CA_ACTIVE)
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reason = "Not connected to a server";
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else if (!clc.speexInitialized)
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reason = "Speex not initialized";
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else if (!clc.voipCodecInitialized)
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reason = "Voip codec not initialized";
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else if (!clc.voipEnabled)
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reason = "Server doesn't support VoIP";
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else if (!clc.demoplaying && (Cvar_VariableValue( "g_gametype" ) == GT_SINGLE_PLAYER || Cvar_VariableValue("ui_singlePlayerActive")))
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@ -306,6 +307,8 @@ void CL_VoipNewGeneration(void)
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clc.voipOutgoingGeneration = 1;
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clc.voipPower = 0.0f;
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clc.voipOutgoingSequence = 0;
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opus_encoder_ctl(clc.opusEncoder, OPUS_RESET_STATE);
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}
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/*
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@ -394,7 +397,7 @@ void CL_VoipParseTargets(void)
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===============
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CL_CaptureVoip
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Record more audio from the hardware if required and encode it into Speex
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Record more audio from the hardware if required and encode it into Opus
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data for later transmission.
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===============
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*/
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@ -424,11 +427,12 @@ void CL_CaptureVoip(void)
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Com_Printf("Until then, VoIP is disabled.\n");
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Cvar_Set("cl_voip", "0");
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}
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Cvar_Set("cl_voipProtocol", cl_voip->integer ? "opus" : "");
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cl_voip->modified = qfalse;
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cl_rate->modified = qfalse;
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}
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if (!clc.speexInitialized)
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if (!clc.voipCodecInitialized)
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return; // just in case this gets called at a bad time.
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if (clc.voipOutgoingDataSize > 0)
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@ -481,80 +485,67 @@ void CL_CaptureVoip(void)
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if ((cl_voipSend->integer) || (finalFrame)) { // user wants to capture audio?
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int samples = S_AvailableCaptureSamples();
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const int mult = (finalFrame) ? 1 : 4; // 4 == 80ms of audio.
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const int packetSamples = (finalFrame) ? VOIP_MAX_FRAME_SAMPLES : VOIP_MAX_PACKET_SAMPLES;
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// enough data buffered in audio hardware to process yet?
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if (samples >= (clc.speexFrameSize * mult)) {
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// audio capture is always MONO16 (and that's what speex wants!).
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// 2048 will cover 12 uncompressed frames in narrowband mode.
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static int16_t sampbuffer[2048];
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if (samples >= packetSamples) {
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// audio capture is always MONO16.
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static int16_t sampbuffer[VOIP_MAX_PACKET_SAMPLES];
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float voipPower = 0.0f;
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int speexFrames = 0;
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int wpos = 0;
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int pos = 0;
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int voipFrames;
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int i, bytes;
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if (samples > (clc.speexFrameSize * 4))
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samples = (clc.speexFrameSize * 4);
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if (samples > VOIP_MAX_PACKET_SAMPLES)
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samples = VOIP_MAX_PACKET_SAMPLES;
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// !!! FIXME: maybe separate recording from encoding, so voipPower
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// !!! FIXME: updates faster than 4Hz?
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samples -= samples % clc.speexFrameSize;
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samples -= samples % VOIP_MAX_FRAME_SAMPLES;
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if (samples != 120 && samples != 240 && samples != 480 && samples != 960 && samples != 1920 && samples != 2880 ) {
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Com_Printf("Voip: bad number of samples %d\n", samples);
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return;
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}
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voipFrames = samples / VOIP_MAX_FRAME_SAMPLES;
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S_Capture(samples, (byte *) sampbuffer); // grab from audio card.
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// this will probably generate multiple speex packets each time.
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while (samples > 0) {
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int16_t *sampptr = &sampbuffer[pos];
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int i, bytes;
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// check the "power" of this packet...
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for (i = 0; i < samples; i++) {
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const float flsamp = (float) sampbuffer[i];
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const float s = fabs(flsamp);
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voipPower += s * s;
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sampbuffer[i] = (int16_t) ((flsamp) * audioMult);
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}
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// preprocess samples to remove noise...
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speex_preprocess_run(clc.speexPreprocessor, sampptr);
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// check the "power" of this packet...
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for (i = 0; i < clc.speexFrameSize; i++) {
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const float flsamp = (float) sampptr[i];
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const float s = fabs(flsamp);
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voipPower += s * s;
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sampptr[i] = (int16_t) ((flsamp) * audioMult);
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}
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// encode raw audio samples into Speex data...
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speex_bits_reset(&clc.speexEncoderBits);
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speex_encode_int(clc.speexEncoder, sampptr,
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&clc.speexEncoderBits);
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bytes = speex_bits_write(&clc.speexEncoderBits,
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(char *) &clc.voipOutgoingData[wpos+1],
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sizeof (clc.voipOutgoingData) - (wpos+1));
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assert((bytes > 0) && (bytes < 256));
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clc.voipOutgoingData[wpos] = (byte) bytes;
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wpos += bytes + 1;
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// look at the data for the next packet...
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pos += clc.speexFrameSize;
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samples -= clc.speexFrameSize;
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speexFrames++;
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// encode raw audio samples into Opus data...
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bytes = opus_encode(clc.opusEncoder, sampbuffer, samples,
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(unsigned char *) clc.voipOutgoingData,
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sizeof (clc.voipOutgoingData));
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if ( bytes <= 0 ) {
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Com_DPrintf("VoIP: Error encoding %d samples\n", samples);
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bytes = 0;
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}
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clc.voipPower = (voipPower / (32768.0f * 32768.0f *
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((float) (clc.speexFrameSize * speexFrames)))) *
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100.0f;
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((float) samples))) * 100.0f;
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if ((useVad) && (clc.voipPower < cl_voipVADThreshold->value)) {
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CL_VoipNewGeneration(); // no "talk" for at least 1/4 second.
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} else {
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clc.voipOutgoingDataSize = wpos;
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clc.voipOutgoingDataFrames = speexFrames;
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clc.voipOutgoingDataSize = bytes;
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clc.voipOutgoingDataFrames = voipFrames;
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Com_DPrintf("VoIP: Send %d frames, %d bytes, %f power\n",
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speexFrames, wpos, clc.voipPower);
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voipFrames, bytes, clc.voipPower);
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#if 0
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static FILE *encio = NULL;
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if (encio == NULL) encio = fopen("voip-outgoing-encoded.bin", "wb");
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if (encio != NULL) { fwrite(clc.voipOutgoingData, wpos, 1, encio); fflush(encio); }
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if (encio != NULL) { fwrite(clc.voipOutgoingData, bytes, 1, encio); fflush(encio); }
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static FILE *decio = NULL;
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if (decio == NULL) decio = fopen("voip-outgoing-decoded.bin", "wb");
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if (decio != NULL) { fwrite(sampbuffer, speexFrames * clc.speexFrameSize * 2, 1, decio); fflush(decio); }
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if (decio != NULL) { fwrite(sampbuffer, voipFrames * VOIP_MAX_FRAME_SAMPLES * 2, 1, decio); fflush(decio); }
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#endif
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}
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}
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@ -1420,14 +1411,11 @@ void CL_Disconnect( qboolean showMainMenu ) {
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cl_voipUseVAD->integer = tmp;
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}
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if (clc.speexInitialized) {
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if (clc.voipCodecInitialized) {
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int i;
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speex_bits_destroy(&clc.speexEncoderBits);
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speex_encoder_destroy(clc.speexEncoder);
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speex_preprocess_state_destroy(clc.speexPreprocessor);
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opus_encoder_destroy(clc.opusEncoder);
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for (i = 0; i < MAX_CLIENTS; i++) {
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speex_bits_destroy(&clc.speexDecoderBits[i]);
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speex_decoder_destroy(clc.speexDecoder[i]);
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opus_decoder_destroy(clc.opusDecoder[i]);
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}
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}
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Cmd_RemoveCommand ("voip");
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@ -3653,9 +3641,9 @@ void CL_Init( void ) {
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cl_voipVADThreshold = Cvar_Get ("cl_voipVADThreshold", "0.25", CVAR_ARCHIVE);
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cl_voipShowMeter = Cvar_Get ("cl_voipShowMeter", "1", CVAR_ARCHIVE);
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// This is a protocol version number.
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cl_voip = Cvar_Get ("cl_voip", "1", CVAR_USERINFO | CVAR_ARCHIVE);
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cl_voip = Cvar_Get ("cl_voip", "1", CVAR_ARCHIVE);
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Cvar_CheckRange( cl_voip, 0, 1, qtrue );
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cl_voipProtocol = Cvar_Get ("cl_voipProtocol", cl_voip->integer ? "opus" : "", CVAR_USERINFO | CVAR_ROM);
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#endif
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