Reuse buffers between function calls

src/mumble/AudioInput.cpp

@@ -357,7 +357,7 @@ static void inMixerFloatMask(float *RESTRICT buffer, const void *RESTRICT ipt, u
 	const float *RESTRICT input = reinterpret_cast< const float * >(ipt);
 
 	unsigned int chancount = 0;
-	std::vector< unsigned int > chanindex;
+	static std::vector< unsigned int > chanindex;
 	chanindex.resize(N);
 	for (unsigned int j = 0; j < N; ++j) {
 		if ((mask & (1ULL << j)) == 0) {
@@ -382,7 +382,7 @@ static void inMixerShortMask(float *RESTRICT buffer, const void *RESTRICT ipt, u
 	const short *RESTRICT input = reinterpret_cast< const short * >(ipt);
 
 	unsigned int chancount = 0;
-	std::vector< unsigned int > chanindex;
+	static std::vector< unsigned int > chanindex;
 	chanindex.resize(N);
 	for (unsigned int j = 0; j < N; ++j) {
 		if ((mask & (1ULL << j)) == 0) {

src/mumble/AudioOutput.cpp

@@ -469,16 +469,16 @@ bool AudioOutput::mix(void *outbuff, unsigned int frameCount) {
 
 	// If the audio backend uses a float-array we can sample and mix the audio sources directly into the output.
 	// Otherwise we'll have to use an intermediate buffer which we will convert to an array of shorts later
-	std::vector< float > fOutput;
+	static std::vector< float > fOutput;
 	fOutput.resize(iChannels * frameCount);
 	float *output = (eSampleFormat == SampleFloat) ? reinterpret_cast< float * >(outbuff) : fOutput.data();
 	memset(output, 0, sizeof(float) * frameCount * iChannels);
 
 	if (!qlMix.isEmpty()) {
 		// There are audio sources available -> mix those sources together and feed them into the audio backend
-		std::vector< float > speaker;
+		static std::vector< float > speaker;
 		speaker.resize(iChannels * 3);
-		std::vector< float > svol;
+		static std::vector< float > svol;
 		svol.resize(iChannels);
 
 		bool validListener = false;

src/mumble/AudioOutputSample.cpp

@@ -243,7 +243,7 @@ bool AudioOutputSample::prepareSampleBuffer(unsigned int frameCount) {
 			  / static_cast< float >(iOutSampleRate)));
 	unsigned int iInputSamples = iInputFrames * channels;
 
-	std::vector< float > fOut;
+	static std::vector< float > fOut;
 	fOut.resize(iInputSamples);
 
 	bool eof = false;

src/mumble/AudioStats.cpp

@@ -153,9 +153,9 @@ void AudioEchoWidget::paintEvent(QPaintEvent *) {
 	spx_int32_t sz;
 	speex_echo_ctl(ai->sesEcho, SPEEX_ECHO_GET_IMPULSE_RESPONSE_SIZE, &sz);
 
-	std::vector< spx_int32_t > w;
+	static std::vector< spx_int32_t > w;
 	w.resize(static_cast< std::size_t >(sz));
-	std::vector< float > W;
+	static std::vector< float > W;
 	W.resize(static_cast< std::size_t >(sz));
 
 	speex_echo_ctl(ai->sesEcho, SPEEX_ECHO_GET_IMPULSE_RESPONSE, w.data());
@@ -226,9 +226,9 @@ void AudioNoiseWidget::paintEvent(QPaintEvent *) {
 	spx_int32_t ps_size = 0;
 	speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD_SIZE, &ps_size);
 
-	std::vector< spx_int32_t > noise;
+	static std::vector< spx_int32_t > noise;
 	noise.resize(static_cast< std::size_t >(ps_size));
-	std::vector< spx_int32_t > ps;
+	static std::vector< spx_int32_t > ps;
 	ps.resize(static_cast< std::size_t >(ps_size));
 
 	speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD, ps.data());
@@ -339,9 +339,9 @@ void AudioStats::on_Tick_timeout() {
 	spx_int32_t ps_size = 0;
 	speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD_SIZE, &ps_size);
 
-	std::vector< spx_int32_t > noise;
+	static std::vector< spx_int32_t > noise;
 	noise.resize(static_cast< std::size_t >(ps_size));
-	std::vector< spx_int32_t > ps;
+	static std::vector< spx_int32_t > ps;
 	ps.resize(static_cast< std::size_t >(ps_size));
 
 	speex_preprocess_ctl(ai->sppPreprocess, SPEEX_PREPROCESS_GET_PSD, ps.data());

src/mumble/OSS.cpp

@@ -323,7 +323,7 @@ void OSSOutput::run() {
 	qWarning("OSSOutput: Starting audio playback to %s", device.constData());
 
 	std::size_t blocklen = iOutputBlock * iChannels * sizeof(short);
-	std::vector< short > mbuffer;
+	static std::vector< short > mbuffer;
 	mbuffer.resize(iOutputBlock * iChannels);
 
 	while (bRunning) {

src/mumble/PulseAudio.cpp

@@ -601,7 +601,7 @@ void PulseAudioSystem::write_callback(pa_stream *s, size_t bytes, void *userdata
 			pao->eSampleFormat = PulseAudioOutput::SampleShort;
 		pao->iMixerFreq = pss->rate;
 		pao->iChannels  = pss->channels;
-		std::vector< unsigned int > chanmasks;
+		static std::vector< unsigned int > chanmasks;
 		chanmasks.resize(pss->channels);
 		for (int i = 0; i < pss->channels; ++i) {
 			unsigned int cm = 0;
@@ -652,7 +652,7 @@ void PulseAudioSystem::write_callback(pa_stream *s, size_t bytes, void *userdata
 	const unsigned int samples     = static_cast< unsigned int >(bytes) / iSampleSize;
 	bool oldAttenuation            = pas->bAttenuating;
 
-	std::vector< unsigned char > buffer;
+	static std::vector< unsigned char > buffer;
 	buffer.resize(bytes);
 	// do we have some mixed output?
 	if (pao->mix(buffer.data(), samples)) {

src/mumble/ServerHandler.cpp

@@ -290,7 +290,7 @@ void ServerHandler::handleVoicePacket(const Mumble::Protocol::AudioData &audioDa
 }
 
 void ServerHandler::sendMessage(const unsigned char *data, int len, bool force) {
-	std::vector< unsigned char > crypto;
+	static std::vector< unsigned char > crypto;
 	crypto.resize(static_cast< std::size_t >(len + 4));
 
 	QMutexLocker qml(&qmUdp);

src/murmur/Server.cpp

@@ -1035,7 +1035,7 @@ void Server::sendMessage(ServerUser &u, const unsigned char *data, int len, QByt
 	if ((u.aiUdpFlag.load() == 1 || force) && (u.sUdpSocket != INVALID_SOCKET)) {
 #endif
 #if defined(__LP64__)
-		std::vector< char > ebuffer;
+		static std::vector< char > ebuffer;
 		ebuffer.resize(static_cast< std::size_t >(len + 4 + 16));
 		char *buffer = reinterpret_cast< char * >(
 			((reinterpret_cast< quint64 >(ebuffer.data()) + 8) & static_cast< quint64 >(~7)) + 4);