Mercurial > hg > dmlib
view jssmix.c @ 49:033c660c25f5
Restructure module playing, removing 8bit sample mixing (output can still be
8bit, but samples are internally upconverted to 16bit after module loading.)
Also prepare for floating point mixing support.
author | Matti Hamalainen <ccr@tnsp.org> |
---|---|
date | Mon, 01 Oct 2012 02:51:41 +0300 |
parents | f3407a58e01e |
children | 36e2f910219c |
line wrap: on
line source
/* * miniJSS - Mixing device and channel handling * Programmed and designed by Matti 'ccr' Hamalainen * (C) Copyright 2006-2012 Tecnic Software productions (TNSP) */ #include "jssmix.h" #include <string.h> #ifdef DM_USE_C #define JMIXER_HEADER #include "jmix_c_in.c" #undef JMIXER_HEADER #endif #undef DM_USE_SIMD #ifdef DM_USE_SIMD #define JMIXER_HEADER #include "jmix_mmx_in.c" #undef JMIXER_HEADER #endif typedef struct { int mixerID; int outFormat; int outChannels; int (*jvmMixChannel_FW)(JSSMixer *, JSSChannel *, Sint32 *, const int, const Sint32); int (*jvmMixChannel_BW)(JSSMixer *, JSSChannel *, Sint32 *, const int, const Sint32); void (*jvmPostProcess)(Sint32 *, void *, const int); } JSSMixingRoutine; /* This table should be sorted from fastest to slowest, e.g. MMX/x86 * optimized routines first, pure C versions last. */ static JSSMixingRoutine jvmMixRoutines[] = { #ifdef DM_USE_SIMD { JMIX_MMX, JSS_AUDIO_U8, JSS_AUDIO_MONO, jvmMix_Mono_MMX_FW, jvmMix_Mono_MMX_BW, jvmPostProcess_U8_MMX }, { JMIX_MMX, JSS_AUDIO_S8, JSS_AUDIO_MONO, jvmMix_Mono_MMX_FW, jvmMix_Mono_MMX_BW, jvmPostProcess_S8_MMX }, { JMIX_MMX, JSS_AUDIO_U8, JSS_AUDIO_STEREO, jvmMix_Stereo_MMX_FW, jvmMix_Stereo_MMX_BW, jvmPostProcess_U8_MMX }, { JMIX_MMX, JSS_AUDIO_S8, JSS_AUDIO_STEREO, jvmMix_Stereo_MMX_FW, jvmMix_Stereo_MMX_BW, jvmPostProcess_S8_MMX }, { JMIX_MMX, JSS_AUDIO_U16, JSS_AUDIO_MONO, jvmMix_Mono_MMX_FW, jvmMix_Mono_MMX_BW, jvmPostProcess_U16_MMX }, { JMIX_MMX, JSS_AUDIO_S16, JSS_AUDIO_MONO, jvmMix_Mono_MMX_FW, jvmMix_Mono_MMX_BW, jvmPostProcess_S16_MMX }, { JMIX_MMX, JSS_AUDIO_U16, JSS_AUDIO_STEREO, jvmMix_Stereo_MMX_FW, jvmMix_Stereo_MMX_BW, jvmPostProcess_U16_MMX }, { JMIX_MMX, JSS_AUDIO_S16, JSS_AUDIO_STEREO, jvmMix_Stereo_MMX_FW, jvmMix_Stereo_MMX_BW, jvmPostProcess_S16_MMX }, #endif #ifdef DM_USE_C { JMIX_C, JSS_AUDIO_U8, JSS_AUDIO_MONO, jvmMix_Mono_C_FW, jvmMix_Mono_C_BW, jvmPostProcess_U8_C }, { JMIX_C, JSS_AUDIO_S8, JSS_AUDIO_MONO, jvmMix_Mono_C_FW, jvmMix_Mono_C_BW, jvmPostProcess_S8_C }, { JMIX_C, JSS_AUDIO_U8, JSS_AUDIO_STEREO, jvmMix_Stereo_C_FW, jvmMix_Stereo_C_BW, jvmPostProcess_U8_C }, { JMIX_C, JSS_AUDIO_S8, JSS_AUDIO_STEREO, jvmMix_Stereo_C_FW, jvmMix_Stereo_C_BW, jvmPostProcess_S8_C }, { JMIX_C, JSS_AUDIO_U16, JSS_AUDIO_MONO, jvmMix_Mono_C_FW, jvmMix_Mono_C_BW, jvmPostProcess_U16_C }, { JMIX_C, JSS_AUDIO_S16, JSS_AUDIO_MONO, jvmMix_Mono_C_FW, jvmMix_Mono_C_BW, jvmPostProcess_S16_C }, { JMIX_C, JSS_AUDIO_U16, JSS_AUDIO_STEREO, jvmMix_Stereo_C_FW, jvmMix_Stereo_C_BW, jvmPostProcess_U16_C }, { JMIX_C, JSS_AUDIO_S16, JSS_AUDIO_STEREO, jvmMix_Stereo_C_FW, jvmMix_Stereo_C_BW, jvmPostProcess_S16_C }, #endif }; static const int jvmNMixRoutines = sizeof(jvmMixRoutines) / sizeof(jvmMixRoutines[0]); static int jvmFindMixRoutine(int outFormat, int outChannels, int mixerID) { int i; for (i = 0; i < jvmNMixRoutines; i++) { if (jvmMixRoutines[i].outFormat == outFormat && jvmMixRoutines[i].outChannels == outChannels && (mixerID == JMIX_AUTO || jvmMixRoutines[i].mixerID == mixerID)) return i; } return -1; } JSSMixer *jvmInit(const int outFormat, const int outChannels, const int outFreq, const int mixerID) { JSSMixer *mixer; int mixerIdx; // Check settings if (outChannels < 1) { JSSERROR(DMERR_INVALID_ARGS, NULL, "Invalid number of channels %d\n", outChannels); } if (outFreq < 4000) { JSSERROR(DMERR_INVALID_ARGS, NULL, "Invalid mixing frequency %d\n", outFreq); } /* Select mixing routines: * Here we try to choose the most fitting mixing routines * from the compiled in routines, unless caller is forcing * us to select specific ones. */ if (mixerID == JMIX_AUTO) { mixerIdx = jvmFindMixRoutine(outFormat, outChannels, JMIX_SSE); if (mixerIdx < 0) mixerIdx = jvmFindMixRoutine(outFormat, outChannels, JMIX_MMX); if (mixerIdx < 0) mixerIdx = jvmFindMixRoutine(outFormat, outChannels, JMIX_AUTO); } else { mixerIdx = jvmFindMixRoutine(outFormat, outChannels, mixerID); } if (mixerIdx < 0) { JSSERROR(DMERR_INVALID_ARGS, NULL, "Could not find mixing routine for outFormat=%d, outChannels=%d, outFreq=%d.\n", outFormat, outChannels, outFreq); return NULL; } // Allocate a mixer device structure mixer = dmMalloc0(sizeof(JSSMixer)); if (mixer == NULL) { JSSERROR(DMERR_MALLOC, NULL, "Could not allocate mixing device structure.\n"); } // Initialize variables #ifdef JSS_SUP_THREADS mixer->mutex = dmCreateMutex(); #endif mixer->outFormat = outFormat; mixer->outFreq = outFreq; mixer->outChannels = outChannels; mixer->jvmMixChannel_FW = jvmMixRoutines[mixerIdx].jvmMixChannel_FW; mixer->jvmMixChannel_BW = jvmMixRoutines[mixerIdx].jvmMixChannel_BW; mixer->jvmPostProcess = jvmMixRoutines[mixerIdx].jvmPostProcess; // Allocate addBuffer mixer->addBufSize = outChannels * outFreq; mixer->addBuffer = dmCalloc(mixer->addBufSize, sizeof(Sint32)); if (mixer->addBuffer == NULL) { JSSERROR(DMERR_MALLOC, NULL, "Could not allocate mixing addition buffer.\n"); } return mixer; } int jvmClose(JSSMixer * mixer) { if (mixer == NULL) return DMERR_NULLPTR; // Deallocate resources #ifdef JSS_SUP_THREADS dmDestroyMutex(mixer->mutex); #endif dmFree(mixer->addBuffer); memset(mixer, 0, sizeof(JSSMixer)); dmFree(mixer); return DMERR_OK; } int jvmGetSampleSize(JSSMixer *mixer) { int sampSize = 1; assert(mixer); switch (mixer->outChannels) { case JSS_AUDIO_STEREO: case JSS_AUDIO_MONO: sampSize = mixer->outChannels; break; default: JSSERROR(DMERR_INVALID_ARGS, -1, "outChannels=%d not stereo or mono!\n", mixer->outChannels); break; } switch (mixer->outFormat) { case JSS_AUDIO_U16: sampSize *= sizeof(Uint16); break; case JSS_AUDIO_S16: sampSize *= sizeof(Sint16); break; case JSS_AUDIO_U8: sampSize *= sizeof(Uint8); break; case JSS_AUDIO_S8: sampSize *= sizeof(Sint8); break; default: JSSERROR(DMERR_INVALID_ARGS, -1, "outFormat=%d is not supported!\n", mixer->outFormat); } return sampSize; } int jvmGetSampleRes(JSSMixer *mixer) { int sampRes = 1; assert(mixer); switch (mixer->outFormat) { case JSS_AUDIO_U16: case JSS_AUDIO_S16: sampRes = 16; break; case JSS_AUDIO_U8: case JSS_AUDIO_S8: sampRes = 8; break; default: JSSERROR(DMERR_INVALID_ARGS, -1, "outFormat=%d is not supported!\n", mixer->outFormat); } return sampRes; } static void jvmMixChannel(JSSMixer *mixer, JSSChannel *chn, Sint32 *addBuffer, const int mixLength) { int mixDone = mixLength, mixResult; Sint32 *ab = addBuffer; if (!chn->chPlaying || chn->chMute) return; DBG("%s(%p, %d)\n", __FUNCTION__, chn, mixLength); while (mixDone > 0) { if (chn->chDirection) { // Channel is playing FORWARDS if (chn->chFlags & jsfLooped) { // Sample is looped if (chn->chFlags & jsfBiDi) { // Bi-directional loop if (FP_GETH(chn->chPos) >= chn->chLoopE) { FP_SETH(chn->chPos, chn->chLoopE - 1); FP_SETL(chn->chPos, 0); chn->chDirection = FALSE; } } else { // Normal forward loop if (FP_GETH(chn->chPos) >= chn->chLoopE) { FP_SETH(chn->chPos, chn->chLoopS /* + (FP_GETH(chn->chPos) - chn->chLoopE) */); FP_SETL(chn->chPos, 0); } } } else { // Normal (non-looped) sample if (FP_GETH(chn->chPos) >= chn->chSize) { chn->chPlaying = FALSE; return; } } } else { // Channel is playing BACKWARDS if (chn->chFlags & jsfLooped) { // Sample is looped if (chn->chFlags & jsfBiDi) { // Bi-directional loop if (FP_GETH(chn->chPos) <= chn->chLoopS) { FP_SETH(chn->chPos, chn->chLoopS); FP_SETL(chn->chPos, 0); chn->chDirection = TRUE; } } else { // Normal forward loop if (FP_GETH(chn->chPos) <= chn->chLoopS) { FP_SETH(chn->chPos, chn->chLoopE - 1); FP_SETL(chn->chPos, 0); } } } else { // Normal (non-looped) sample if (FP_GETH(chn->chPos) <= 0) { chn->chPlaying = FALSE; return; } } } // Call the mixing innerloop functions if (chn->chDirection) { DBG("MIX_FW[%p : %d : ", ab, mixDone); if (chn->chFlags & jsfLooped) { DBG("%d (%x)] {loop}\n", chn->chLoopE, chn->chLoopE); mixResult = mixer->jvmMixChannel_FW((void *) mixer, chn, ab, mixDone, chn->chLoopE); } else { DBG("%d (%x)]\n", chn->chSize, chn->chSize); mixResult = mixer->jvmMixChannel_FW((void *) mixer, chn, ab, mixDone, chn->chSize); } } else { DBG("MIX_BW[%p : %d : ", ab, mixDone); if (chn->chFlags & jsfLooped) { DBG("%d (%x)] {loop}\n", chn->chLoopS, chn->chLoopS); mixResult = mixer->jvmMixChannel_BW(mixer, chn, ab, mixDone, chn->chLoopS); } else { DBG("%d (%x)]\n", 0, 0); mixResult = mixer->jvmMixChannel_BW(mixer, chn, ab, mixDone, 0); } } mixDone -= mixResult; ab += mixResult * mixer->outChannels; } #ifdef JSS_DEBUG if (mixDone < 0) JSSWARNING(DMERR_BOUNDS,, "mixDone < 0 in mixing logic loop.\n"); #endif } void jvmRenderAudio(JSSMixer *mixer, void *mixBuffer, const int mixLength) { int i, blockLength, mixLeft; Sint32 *ab; JSS_LOCK(mixer); assert(mixer != NULL); assert(mixBuffer != NULL); assert(mixLength > 0); assert(mixLength * mixer->outChannels < mixer->addBufSize); // Clear mixer->addBuffer memset(mixer->addBuffer, 0, mixLength * mixer->outChannels * sizeof(Sint32)); ab = mixer->addBuffer; mixLeft = mixLength; while (mixLeft > 0) { // Check for callbacks blockLength = mixLeft; if (mixer->cbFunction) { if (mixer->cbCounter <= 0) { mixer->cbFunction(mixer, mixer->cbData); mixer->cbCounter = mixer->cbFreq; } if (mixer->cbCounter < blockLength) blockLength = mixer->cbCounter; } // Do mixing for (i = 0; i < jsetNChannels; i++) { JSSChannel *chn = &(mixer->channels[i]); if (chn->chPlaying && !chn->chMute) jvmMixChannel(mixer, chn, ab, blockLength); } /* if (chn->chPlaying) { if (!chn->chMute) jvmMixChannel(mixer, chn, ab, blockLength); else jvmAdvanceChannel(mixer, chn, blockLength); } */ ab += blockLength * mixer->outChannels; mixLeft -= blockLength; mixer->cbCounter -= blockLength; } // Post-process mixer->jvmPostProcess(mixer->addBuffer, mixBuffer, mixLength * mixer->outChannels); JSS_UNLOCK(mixer); } int jvmSetCallback(JSSMixer * mixer, void (*cbFunction) (void *, void *), void *cbData) { assert(mixer); if (cbFunction == NULL) JSSERROR(DMERR_NULLPTR, DMERR_NULLPTR, "NULL pointer given as cbFunction"); JSS_LOCK(mixer); mixer->cbFunction = cbFunction; mixer->cbData = cbData; JSS_UNLOCK(mixer); return DMERR_OK; } void jvmRemoveCallback(JSSMixer * mixer) { assert(mixer); JSS_LOCK(mixer); mixer->cbFunction = NULL; mixer->cbData = NULL; mixer->cbFreq = mixer->cbCounter = 0; JSS_UNLOCK(mixer); } int jvmSetCallbackFreq(JSSMixer * mixer, const int cbFreq) { assert(mixer); if ((cbFreq < 1) || (cbFreq >= mixer->outFreq)) JSSERROR(DMERR_INVALID_ARGS, DMERR_INVALID_ARGS, "Invalid callback frequency given (%i / %i)\n", cbFreq, mixer->outFreq); JSS_LOCK(mixer); mixer->cbFreq = (mixer->outFreq / cbFreq); mixer->cbCounter = 0; //fprintf(stderr, "set(outFreq = %d, cbFreq = %d) = %d\n", mixer->outFreq, cbFreq, mixer->cbFreq); JSS_UNLOCK(mixer); return DMERR_OK; } /* Channel manipulation routines */ void jvmPlay(JSSMixer * mixer, const int channel) { JSS_LOCK(mixer); mixer->channels[channel].chPlaying = TRUE; JSS_UNLOCK(mixer); } void jvmStop(JSSMixer * mixer, const int channel) { JSS_LOCK(mixer); mixer->channels[channel].chPlaying = FALSE; JSS_UNLOCK(mixer); } void jvmSetSample(JSSMixer * mixer, const int channel, void *data, const Sint32 size, const Sint32 loopS, const Sint32 loopE, const int flags) { JSSChannel *c; JSS_LOCK(mixer); c = &mixer->channels[channel]; c->chData = data; c->chSize = size; c->chLoopS = loopS; c->chLoopE = loopE; c->chFlags = flags; c->chDirection = TRUE; c->chPrevL = c->chPrevR = 0; c->chPos.dw = c->chDeltaO.dw = 0; JSS_UNLOCK(mixer); } void jvmSetFreq(JSSMixer * mixer, const int channel, const int freq) { JSS_LOCK(mixer); mixer->channels[channel].chFreq = freq; if (mixer->outFreq > 0) { DMFixedPoint a, b; FP_SETHL(a, freq, 0); FP_CONV(b, mixer->outFreq); FP_DIV_R(mixer->channels[channel].chDeltaO, a, b); } else { FP_SET(mixer->channels[channel].chDeltaO, 0); } JSS_UNLOCK(mixer); } int jvmGetFreq(JSSMixer * mixer, const int channel) { int tmp; JSS_LOCK(mixer); tmp = mixer->channels[channel].chFreq; JSS_UNLOCK(mixer); return tmp; } void jvmSetVolume(JSSMixer * mixer, const int channel, const int volume) { JSS_LOCK(mixer); mixer->channels[channel].chVolume = volume; JSS_UNLOCK(mixer); } int jvmGetVolume(JSSMixer * mixer, const int channel) { int tmp; JSS_LOCK(mixer); tmp = mixer->channels[channel].chVolume; JSS_UNLOCK(mixer); return tmp; } void jvmSetPos(JSSMixer * mixer, const int channel, const Sint32 pos) { JSS_LOCK(mixer); FP_SETH(mixer->channels[channel].chPos, pos); FP_SETL(mixer->channels[channel].chPos, 0); JSS_UNLOCK(mixer); } Sint32 jvmGetPos(JSSMixer * mixer, const int channel) { Sint32 tmp; JSS_LOCK(mixer); tmp = FP_GETH(mixer->channels[channel].chPos); JSS_UNLOCK(mixer); return tmp; } void jvmSetPan(JSSMixer * mixer, const int channel, const int panning) { JSS_LOCK(mixer); mixer->channels[channel].chPanning = panning; JSS_UNLOCK(mixer); } int jvmGetPan(JSSMixer * mixer, const int channel) { int tmp; JSS_LOCK(mixer); tmp = mixer->channels[channel].chPanning; JSS_UNLOCK(mixer); return tmp; } void jvmMute(JSSMixer * mixer, const int channel, const BOOL mute) { JSS_LOCK(mixer); mixer->channels[channel].chMute = mute; JSS_UNLOCK(mixer); } void jvmClear(JSSMixer * mixer, const int channel) { JSS_LOCK(mixer); memset(&mixer->channels[channel], 0, sizeof(JSSChannel)); JSS_UNLOCK(mixer); } void jvmSetGlobalVol(JSSMixer * mixer, const int volume) { JSS_LOCK(mixer); mixer->globalVol = volume; JSS_UNLOCK(mixer); } int jvmGetGlobalVol(JSSMixer * mixer) { int tmp; JSS_LOCK(mixer); tmp = mixer->globalVol; JSS_UNLOCK(mixer); return tmp; }