From: notaz Date: Mon, 28 Feb 2011 22:48:20 +0000 (+0200) Subject: spu: drop unused mono code X-Git-Tag: r7~8 X-Git-Url: https://notaz.gp2x.de/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=97ea407715c0eafc8d73ffc5477f6d7e6ad8f20b;p=pcsx_rearmed.git spu: drop unused mono code also merge the mix buffers - the idea was to make use of neon there but I've decided againt it, it wouldn't improve things much according to profiling. --- diff --git a/plugins/dfsound/alsa.c b/plugins/dfsound/alsa.c index 2eba878a..7e594338 100644 --- a/plugins/dfsound/alsa.c +++ b/plugins/dfsound/alsa.c @@ -40,8 +40,7 @@ void SetupSound(void) unsigned int period_time = buffer_time / 4; int err; - if (iDisStereo) pchannels = 1; - else pchannels=2; + pchannels=2; pspeed = 44100; format = SND_PCM_FORMAT_S16; @@ -153,6 +152,5 @@ void SoundFeedStreamData(unsigned char* pSound,long lBytes) if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN) snd_pcm_prepare(handle); - snd_pcm_writei(handle,pSound, - iDisStereo ? lBytes / 2 : lBytes / 4); + snd_pcm_writei(handle,pSound, lBytes / 4); } diff --git a/plugins/dfsound/cfg.c b/plugins/dfsound/cfg.c index 7d087bb8..5801453e 100644 --- a/plugins/dfsound/cfg.c +++ b/plugins/dfsound/cfg.c @@ -141,11 +141,6 @@ static void ReadConfigFile(void) if(iUseInterpolation<0) iUseInterpolation=0; if(iUseInterpolation>3) iUseInterpolation=3; - strcpy(t,"\nDisStereo");p=strstr(pB,t);if(p) {p=strstr(p,"=");len=1;} - if(p) iDisStereo=atoi(p+len); - if(iDisStereo<0) iDisStereo=0; - if(iDisStereo>1) iDisStereo=1; - free(pB); } @@ -161,7 +156,6 @@ void ReadConfigSPU(void) iUseTimer=2; iUseReverb=2; iUseInterpolation=2; - iDisStereo=0; ReadConfigFile(); } diff --git a/plugins/dfsound/externals.h b/plugins/dfsound/externals.h index 92683a95..90e26cec 100644 --- a/plugins/dfsound/externals.h +++ b/plugins/dfsound/externals.h @@ -205,7 +205,6 @@ extern int iDebugMode; extern int iRecordMode; extern int iUseReverb; extern int iUseInterpolation; -extern int iDisStereo; // MISC extern int iSpuAsyncWait; diff --git a/plugins/dfsound/oss.c b/plugins/dfsound/oss.c index 929d9213..b2685fbf 100644 --- a/plugins/dfsound/oss.c +++ b/plugins/dfsound/oss.c @@ -46,9 +46,7 @@ void SetupSound(void) int myfrag; int oss_speed, oss_stereo; - if(iDisStereo) pstereo=OSS_MODE_MONO; - else pstereo=OSS_MODE_STEREO; - + pstereo = OSS_MODE_STEREO; oss_speed = pspeed; oss_stereo = pstereo; @@ -90,17 +88,12 @@ void SetupSound(void) return; } - if(ioctl(oss_audio_fd,SNDCTL_DSP_STEREO,&oss_stereo)==-1) + if(ioctl(oss_audio_fd,SNDCTL_DSP_STEREO,&oss_stereo)==-1 || !oss_stereo) { printf("Stereo mode not supported!\n"); return; } - if(oss_stereo!=1) - { - iDisStereo=1; - } - if(ioctl(oss_audio_fd,SNDCTL_DSP_SPEED,&oss_speed)==-1) { printf("Sound frequency not supported\n"); diff --git a/plugins/dfsound/pulseaudio.c b/plugins/dfsound/pulseaudio.c index 60051557..e5ffb59e 100644 --- a/plugins/dfsound/pulseaudio.c +++ b/plugins/dfsound/pulseaudio.c @@ -195,10 +195,7 @@ void SetupSound (void) // Set sample spec //////////////////////////////////////////////////////// device.spec.format = PA_SAMPLE_S16NE; - if (iDisStereo) - device.spec.channels = 1; - else - device.spec.channels = 2; + device.spec.channels = 2; device.spec.rate = settings.frequency; pa_buffer_attr buffer_attributes; diff --git a/plugins/dfsound/sdl.c b/plugins/dfsound/sdl.c index 45ccba2d..f6777440 100644 --- a/plugins/dfsound/sdl.c +++ b/plugins/dfsound/sdl.c @@ -70,7 +70,7 @@ void SetupSound(void) { spec.freq = 44100; spec.format = AUDIO_S16SYS; - spec.channels = iDisStereo ? 1 : 2; + spec.channels = 2; spec.samples = 512; spec.callback = SOUND_FillAudio; @@ -80,7 +80,6 @@ void SetupSound(void) { } iBufSize = BUFFER_SIZE; - if (iDisStereo) iBufSize /= 2; pSndBuffer = (short *)malloc(iBufSize * sizeof(short)); if (pSndBuffer == NULL) { diff --git a/plugins/dfsound/spu.c b/plugins/dfsound/spu.c index ca0b97b9..dc0f5845 100644 --- a/plugins/dfsound/spu.c +++ b/plugins/dfsound/spu.c @@ -73,7 +73,6 @@ int iDebugMode=0; int iRecordMode=0; int iUseReverb=2; int iUseInterpolation=2; -int iDisStereo=0; // MAIN infos struct for each channel @@ -107,8 +106,7 @@ static const int f[8][2] = { { 0, 0 }, { 115, -52 }, { 98, -55 }, { 122, -60 } }; -int SSumR[NSSIZE]; -int SSumL[NSSIZE]; +int SSumLR[NSSIZE*2]; int iFMod[NSSIZE]; int iCycle = 0; short * pS; @@ -623,8 +621,8 @@ static void *MAINThread(void *arg) ////////////////////////////////////////////// // ok, left/right sound volume (psx volume goes from 0 ... 0x3fff) - SSumL[ns]+=(sval*s_chan[ch].iLeftVolume)/0x4000L; - SSumR[ns]+=(sval*s_chan[ch].iRightVolume)/0x4000L; + SSumLR[ns*2] +=(sval*s_chan[ch].iLeftVolume)/0x4000L; + SSumLR[ns*2+1]+=(sval*s_chan[ch].iRightVolume)/0x4000L; ////////////////////////////////////////////// // now let us store sound data for reverb @@ -670,37 +668,21 @@ ENDX: ; /////////////////////////////////////////////////////// // mix all channels (including reverb) into one buffer - if(iDisStereo) // no stereo? + for (ns = 0; ns < NSSIZE*2; ) { - int dl, dr; - for (ns = 0; ns < NSSIZE; ns++) - { - SSumL[ns] += MixREVERBLeft(ns); - - dl = SSumL[ns] / voldiv; SSumL[ns] = 0; - if (dl < -32767) dl = -32767; if (dl > 32767) dl = 32767; - - SSumR[ns] += MixREVERBRight(); - - dr = SSumR[ns] / voldiv; SSumR[ns] = 0; - if (dr < -32767) dr = -32767; if (dr > 32767) dr = 32767; - *pS++ = (dl + dr) / 2; - } - } - else // stereo: - for (ns = 0; ns < NSSIZE; ns++) - { - SSumL[ns] += MixREVERBLeft(ns); + SSumLR[ns] += MixREVERBLeft(ns/2); - d = SSumL[ns] / voldiv; SSumL[ns] = 0; + d = SSumLR[ns] / voldiv; SSumLR[ns] = 0; if (d < -32767) d = -32767; if (d > 32767) d = 32767; *pS++ = d; + ns++; - SSumR[ns] += MixREVERBRight(); + SSumLR[ns] += MixREVERBRight(); - d = SSumR[ns] / voldiv; SSumR[ns] = 0; + d = SSumLR[ns] / voldiv; SSumLR[ns] = 0; if(d < -32767) d = -32767; if(d > 32767) d = 32767; *pS++ = d; + ns++; } ////////////////////////////////////////////////////// @@ -818,8 +800,7 @@ void CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes) // SETUPTIMER: init of certain buffers and threads/timers void SetupTimer(void) { - memset(SSumR,0,NSSIZE*sizeof(int)); // init some mixing buffers - memset(SSumL,0,NSSIZE*sizeof(int)); + memset(SSumLR,0,sizeof(SSumLR)); // init some mixing buffers memset(iFMod,0,NSSIZE*sizeof(int)); pS=(short *)pSpuBuffer; // setup soundbuffer pointer diff --git a/plugins/dfsound/xa.c b/plugins/dfsound/xa.c index df60346a..bdea89a3 100644 --- a/plugins/dfsound/xa.c +++ b/plugins/dfsound/xa.c @@ -22,6 +22,8 @@ // will be included from spu.c #ifdef _IN_SPU +#define XA_HACK + //////////////////////////////////////////////////////////////////////// // XA GLOBALS //////////////////////////////////////////////////////////////////////// @@ -61,40 +63,40 @@ INLINE void MixXA(void) int ns; uint32_t l; - for(ns=0;ns>16)&0xffff)) * iRightXAVol)/32768; + SSumLR[ns++]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32768; + SSumLR[ns++]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32768; #else - SSumL[ns]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32767; - SSumR[ns]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32767; + SSumLR[ns++]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32767; + SSumLR[ns++]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32767; #endif } if(XAPlay==XAFeed && XARepeat) { XARepeat--; - for(;ns>16)&0xffff)) * iRightXAVol)/32768; + SSumLR[ns++]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32768; + SSumLR[ns++]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32768; #else - SSumL[ns]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32767; - SSumR[ns]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32767; + SSumLR[ns++]+=(((short)(XALastVal&0xffff)) * iLeftXAVol)/32767; + SSumLR[ns++]+=(((short)((XALastVal>>16)&0xffff)) * iRightXAVol)/32767; #endif } } - for(ns=0;ns>16)&0xffff)) * iRightXAVol)/32767; + SSumLR[ns++]+=(((short)(l&0xffff)) * iLeftXAVol)/32767; + SSumLR[ns++]+=(((short)((l>>16)&0xffff)) * iRightXAVol)/32767; } } @@ -122,7 +124,7 @@ INLINE void FeedXA(xa_decode_t *xap) xapGlobal = xap; // store info for save states XARepeat = 100; // set up repeat -#ifdef XA_HACK +#if 0//def XA_HACK iSize=((45500*xap->nsamples)/xap->freq); // get size #else iSize=((44100*xap->nsamples)/xap->freq); // get size