/* * Copyright (C) 2008 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ #define LOG_TAG "EffectReverb" //#define LOG_NDEBUG 0 #include #include #include #include #include "EffectReverb.h" #include "EffectsMath.h" // effect_handle_t interface implementation for reverb effect const struct effect_interface_s gReverbInterface = { Reverb_Process, Reverb_Command, Reverb_GetDescriptor, NULL }; // Google auxiliary environmental reverb UUID: 1f0ae2e0-4ef7-11df-bc09-0002a5d5c51b static const effect_descriptor_t gAuxEnvReverbDescriptor = { {0xc2e5d5f0, 0x94bd, 0x4763, 0x9cac, {0x4e, 0x23, 0x4d, 0x06, 0x83, 0x9e}}, {0x1f0ae2e0, 0x4ef7, 0x11df, 0xbc09, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, EFFECT_CONTROL_API_VERSION, // flags other than EFFECT_FLAG_TYPE_AUXILIARY set for test purpose EFFECT_FLAG_TYPE_AUXILIARY | EFFECT_FLAG_DEVICE_IND | EFFECT_FLAG_AUDIO_MODE_IND, 0, // TODO 33, "Aux Environmental Reverb", "The Android Open Source Project" }; // Google insert environmental reverb UUID: aa476040-6342-11df-91a4-0002a5d5c51b static const effect_descriptor_t gInsertEnvReverbDescriptor = { {0xc2e5d5f0, 0x94bd, 0x4763, 0x9cac, {0x4e, 0x23, 0x4d, 0x06, 0x83, 0x9e}}, {0xaa476040, 0x6342, 0x11df, 0x91a4, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, EFFECT_CONTROL_API_VERSION, EFFECT_FLAG_TYPE_INSERT | EFFECT_FLAG_INSERT_FIRST, 0, // TODO 33, "Insert Environmental reverb", "The Android Open Source Project" }; // Google auxiliary preset reverb UUID: 63909320-53a6-11df-bdbd-0002a5d5c51b static const effect_descriptor_t gAuxPresetReverbDescriptor = { {0x47382d60, 0xddd8, 0x11db, 0xbf3a, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, {0x63909320, 0x53a6, 0x11df, 0xbdbd, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, EFFECT_CONTROL_API_VERSION, EFFECT_FLAG_TYPE_AUXILIARY, 0, // TODO 33, "Aux Preset Reverb", "The Android Open Source Project" }; // Google insert preset reverb UUID: d93dc6a0-6342-11df-b128-0002a5d5c51b static const effect_descriptor_t gInsertPresetReverbDescriptor = { {0x47382d60, 0xddd8, 0x11db, 0xbf3a, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, {0xd93dc6a0, 0x6342, 0x11df, 0xb128, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}, EFFECT_CONTROL_API_VERSION, EFFECT_FLAG_TYPE_INSERT | EFFECT_FLAG_INSERT_FIRST, 0, // TODO 33, "Insert Preset Reverb", "The Android Open Source Project" }; // gDescriptors contains pointers to all defined effect descriptor in this library static const effect_descriptor_t * const gDescriptors[] = { &gAuxEnvReverbDescriptor, &gInsertEnvReverbDescriptor, &gAuxPresetReverbDescriptor, &gInsertPresetReverbDescriptor }; /*---------------------------------------------------------------------------- * Effect API implementation *--------------------------------------------------------------------------*/ /*--- Effect Library Interface Implementation ---*/ int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle) { int ret; int i; reverb_module_t *module; const effect_descriptor_t *desc; int aux = 0; int preset = 0; ALOGV("EffectLibCreateEffect start"); if (pHandle == NULL || uuid == NULL) { return -EINVAL; } for (i = 0; gDescriptors[i] != NULL; i++) { desc = gDescriptors[i]; if (memcmp(uuid, &desc->uuid, sizeof(effect_uuid_t)) == 0) { break; } } if (gDescriptors[i] == NULL) { return -ENOENT; } module = malloc(sizeof(reverb_module_t)); module->itfe = &gReverbInterface; module->context.mState = REVERB_STATE_UNINITIALIZED; if (memcmp(&desc->type, SL_IID_PRESETREVERB, sizeof(effect_uuid_t)) == 0) { preset = 1; } if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { aux = 1; } ret = Reverb_Init(module, aux, preset); if (ret < 0) { ALOGW("EffectLibCreateEffect() init failed"); free(module); return ret; } *pHandle = (effect_handle_t) module; module->context.mState = REVERB_STATE_INITIALIZED; ALOGV("EffectLibCreateEffect %p ,size %d", module, sizeof(reverb_module_t)); return 0; } int EffectRelease(effect_handle_t handle) { reverb_module_t *pRvbModule = (reverb_module_t *)handle; ALOGV("EffectLibReleaseEffect %p", handle); if (handle == NULL) { return -EINVAL; } pRvbModule->context.mState = REVERB_STATE_UNINITIALIZED; free(pRvbModule); return 0; } int EffectGetDescriptor(const effect_uuid_t *uuid, effect_descriptor_t *pDescriptor) { int i; int length = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *); if (pDescriptor == NULL || uuid == NULL){ ALOGV("EffectGetDescriptor() called with NULL pointer"); return -EINVAL; } for (i = 0; i < length; i++) { if (memcmp(uuid, &gDescriptors[i]->uuid, sizeof(effect_uuid_t)) == 0) { *pDescriptor = *gDescriptors[i]; ALOGV("EffectGetDescriptor - UUID matched Reverb type %d, UUID = %x", i, gDescriptors[i]->uuid.timeLow); return 0; } } return -EINVAL; } /* end EffectGetDescriptor */ /*--- Effect Control Interface Implementation ---*/ static int Reverb_Process(effect_handle_t self, audio_buffer_t *inBuffer, audio_buffer_t *outBuffer) { reverb_object_t *pReverb; int16_t *pSrc, *pDst; reverb_module_t *pRvbModule = (reverb_module_t *)self; if (pRvbModule == NULL) { return -EINVAL; } if (inBuffer == NULL || inBuffer->raw == NULL || outBuffer == NULL || outBuffer->raw == NULL || inBuffer->frameCount != outBuffer->frameCount) { return -EINVAL; } pReverb = (reverb_object_t*) &pRvbModule->context; if (pReverb->mState == REVERB_STATE_UNINITIALIZED) { return -EINVAL; } if (pReverb->mState == REVERB_STATE_INITIALIZED) { return -ENODATA; } //if bypassed or the preset forces the signal to be completely dry if (pReverb->m_bBypass != 0) { if (inBuffer->raw != outBuffer->raw) { int16_t smp; pSrc = inBuffer->s16; pDst = outBuffer->s16; size_t count = inBuffer->frameCount; if (pRvbModule->config.inputCfg.channels == pRvbModule->config.outputCfg.channels) { count *= 2; while (count--) { *pDst++ = *pSrc++; } } else { while (count--) { smp = *pSrc++; *pDst++ = smp; *pDst++ = smp; } } } return 0; } if (pReverb->m_nNextRoom != pReverb->m_nCurrentRoom) { ReverbUpdateRoom(pReverb, true); } pSrc = inBuffer->s16; pDst = outBuffer->s16; size_t numSamples = outBuffer->frameCount; while (numSamples) { uint32_t processedSamples; if (numSamples > (uint32_t) pReverb->m_nUpdatePeriodInSamples) { processedSamples = (uint32_t) pReverb->m_nUpdatePeriodInSamples; } else { processedSamples = numSamples; } /* increment update counter */ pReverb->m_nUpdateCounter += (int16_t) processedSamples; /* check if update counter needs to be reset */ if (pReverb->m_nUpdateCounter >= pReverb->m_nUpdatePeriodInSamples) { /* update interval has elapsed, so reset counter */ pReverb->m_nUpdateCounter -= pReverb->m_nUpdatePeriodInSamples; ReverbUpdateXfade(pReverb, pReverb->m_nUpdatePeriodInSamples); } /* end if m_nUpdateCounter >= update interval */ Reverb(pReverb, processedSamples, pDst, pSrc); numSamples -= processedSamples; if (pReverb->m_Aux) { pSrc += processedSamples; } else { pSrc += processedSamples * NUM_OUTPUT_CHANNELS; } pDst += processedSamples * NUM_OUTPUT_CHANNELS; } return 0; } static int Reverb_Command(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) { reverb_module_t *pRvbModule = (reverb_module_t *) self; reverb_object_t *pReverb; int retsize; if (pRvbModule == NULL || pRvbModule->context.mState == REVERB_STATE_UNINITIALIZED) { return -EINVAL; } pReverb = (reverb_object_t*) &pRvbModule->context; ALOGV("Reverb_Command command %d cmdSize %d",cmdCode, cmdSize); switch (cmdCode) { case EFFECT_CMD_INIT: if (pReplyData == NULL || *replySize != sizeof(int)) { return -EINVAL; } *(int *) pReplyData = Reverb_Init(pRvbModule, pReverb->m_Aux, pReverb->m_Preset); if (*(int *) pReplyData == 0) { pRvbModule->context.mState = REVERB_STATE_INITIALIZED; } break; case EFFECT_CMD_SET_CONFIG: if (pCmdData == NULL || cmdSize != sizeof(effect_config_t) || pReplyData == NULL || *replySize != sizeof(int)) { return -EINVAL; } *(int *) pReplyData = Reverb_setConfig(pRvbModule, (effect_config_t *)pCmdData, false); break; case EFFECT_CMD_GET_CONFIG: if (pReplyData == NULL || *replySize != sizeof(effect_config_t)) { return -EINVAL; } Reverb_getConfig(pRvbModule, (effect_config_t *) pCmdData); break; case EFFECT_CMD_RESET: Reverb_Reset(pReverb, false); break; case EFFECT_CMD_GET_PARAM: ALOGV("Reverb_Command EFFECT_CMD_GET_PARAM pCmdData %p, *replySize %d, pReplyData: %p",pCmdData, *replySize, pReplyData); if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) || pReplyData == NULL || *replySize < (int) sizeof(effect_param_t)) { return -EINVAL; } effect_param_t *rep = (effect_param_t *) pReplyData; memcpy(pReplyData, pCmdData, sizeof(effect_param_t) + sizeof(int32_t)); ALOGV("Reverb_Command EFFECT_CMD_GET_PARAM param %d, replySize %d",*(int32_t *)rep->data, rep->vsize); rep->status = Reverb_getParameter(pReverb, *(int32_t *)rep->data, &rep->vsize, rep->data + sizeof(int32_t)); *replySize = sizeof(effect_param_t) + sizeof(int32_t) + rep->vsize; break; case EFFECT_CMD_SET_PARAM: ALOGV("Reverb_Command EFFECT_CMD_SET_PARAM cmdSize %d pCmdData %p, *replySize %d, pReplyData %p", cmdSize, pCmdData, *replySize, pReplyData); if (pCmdData == NULL || (cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t))) || pReplyData == NULL || *replySize != (int)sizeof(int32_t)) { return -EINVAL; } effect_param_t *cmd = (effect_param_t *) pCmdData; *(int *)pReplyData = Reverb_setParameter(pReverb, *(int32_t *)cmd->data, cmd->vsize, cmd->data + sizeof(int32_t)); break; case EFFECT_CMD_ENABLE: if (pReplyData == NULL || *replySize != sizeof(int)) { return -EINVAL; } if (pReverb->mState != REVERB_STATE_INITIALIZED) { return -ENOSYS; } pReverb->mState = REVERB_STATE_ACTIVE; ALOGV("EFFECT_CMD_ENABLE() OK"); *(int *)pReplyData = 0; break; case EFFECT_CMD_DISABLE: if (pReplyData == NULL || *replySize != sizeof(int)) { return -EINVAL; } if (pReverb->mState != REVERB_STATE_ACTIVE) { return -ENOSYS; } pReverb->mState = REVERB_STATE_INITIALIZED; ALOGV("EFFECT_CMD_DISABLE() OK"); *(int *)pReplyData = 0; break; case EFFECT_CMD_SET_DEVICE: if (pCmdData == NULL || cmdSize != (int)sizeof(uint32_t)) { return -EINVAL; } ALOGV("Reverb_Command EFFECT_CMD_SET_DEVICE: 0x%08x", *(uint32_t *)pCmdData); break; case EFFECT_CMD_SET_VOLUME: { // audio output is always stereo => 2 channel volumes if (pCmdData == NULL || cmdSize != (int)sizeof(uint32_t) * 2) { return -EINVAL; } float left = (float)(*(uint32_t *)pCmdData) / (1 << 24); float right = (float)(*((uint32_t *)pCmdData + 1)) / (1 << 24); ALOGV("Reverb_Command EFFECT_CMD_SET_VOLUME: left %f, right %f ", left, right); break; } case EFFECT_CMD_SET_AUDIO_MODE: if (pCmdData == NULL || cmdSize != (int)sizeof(uint32_t)) { return -EINVAL; } ALOGV("Reverb_Command EFFECT_CMD_SET_AUDIO_MODE: %d", *(uint32_t *)pCmdData); break; default: ALOGW("Reverb_Command invalid command %d",cmdCode); return -EINVAL; } return 0; } int Reverb_GetDescriptor(effect_handle_t self, effect_descriptor_t *pDescriptor) { reverb_module_t *pRvbModule = (reverb_module_t *) self; reverb_object_t *pReverb; const effect_descriptor_t *desc; if (pRvbModule == NULL || pRvbModule->context.mState == REVERB_STATE_UNINITIALIZED) { return -EINVAL; } pReverb = (reverb_object_t*) &pRvbModule->context; if (pReverb->m_Aux) { if (pReverb->m_Preset) { desc = &gAuxPresetReverbDescriptor; } else { desc = &gAuxEnvReverbDescriptor; } } else { if (pReverb->m_Preset) { desc = &gInsertPresetReverbDescriptor; } else { desc = &gInsertEnvReverbDescriptor; } } *pDescriptor = *desc; return 0; } /* end Reverb_getDescriptor */ /*---------------------------------------------------------------------------- * Reverb internal functions *--------------------------------------------------------------------------*/ /*---------------------------------------------------------------------------- * Reverb_Init() *---------------------------------------------------------------------------- * Purpose: * Initialize reverb context and apply default parameters * * Inputs: * pRvbModule - pointer to reverb effect module * aux - indicates if the reverb is used as auxiliary (1) or insert (0) * preset - indicates if the reverb is used in preset (1) or environmental (0) mode * * Outputs: * * Side Effects: * *---------------------------------------------------------------------------- */ int Reverb_Init(reverb_module_t *pRvbModule, int aux, int preset) { int ret; ALOGV("Reverb_Init module %p, aux: %d, preset: %d", pRvbModule,aux, preset); memset(&pRvbModule->context, 0, sizeof(reverb_object_t)); pRvbModule->context.m_Aux = (uint16_t)aux; pRvbModule->context.m_Preset = (uint16_t)preset; pRvbModule->config.inputCfg.samplingRate = 44100; if (aux) { pRvbModule->config.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; } else { pRvbModule->config.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO; } pRvbModule->config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; pRvbModule->config.inputCfg.bufferProvider.getBuffer = NULL; pRvbModule->config.inputCfg.bufferProvider.releaseBuffer = NULL; pRvbModule->config.inputCfg.bufferProvider.cookie = NULL; pRvbModule->config.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; pRvbModule->config.inputCfg.mask = EFFECT_CONFIG_ALL; pRvbModule->config.outputCfg.samplingRate = 44100; pRvbModule->config.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO; pRvbModule->config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; pRvbModule->config.outputCfg.bufferProvider.getBuffer = NULL; pRvbModule->config.outputCfg.bufferProvider.releaseBuffer = NULL; pRvbModule->config.outputCfg.bufferProvider.cookie = NULL; pRvbModule->config.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; pRvbModule->config.outputCfg.mask = EFFECT_CONFIG_ALL; ret = Reverb_setConfig(pRvbModule, &pRvbModule->config, true); if (ret < 0) { ALOGV("Reverb_Init error %d on module %p", ret, pRvbModule); } return ret; } /*---------------------------------------------------------------------------- * Reverb_setConfig() *---------------------------------------------------------------------------- * Purpose: * Set input and output audio configuration. * * Inputs: * pRvbModule - pointer to reverb effect module * pConfig - pointer to effect_config_t structure containing input * and output audio parameters configuration * init - true if called from init function * Outputs: * * Side Effects: * *---------------------------------------------------------------------------- */ int Reverb_setConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig, bool init) { reverb_object_t *pReverb = &pRvbModule->context; int bufferSizeInSamples; int updatePeriodInSamples; int xfadePeriodInSamples; // Check configuration compatibility with build options if (pConfig->inputCfg.samplingRate != pConfig->outputCfg.samplingRate || pConfig->outputCfg.channels != OUTPUT_CHANNELS || pConfig->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT || pConfig->outputCfg.format != AUDIO_FORMAT_PCM_16_BIT) { ALOGV("Reverb_setConfig invalid config"); return -EINVAL; } if ((pReverb->m_Aux && (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_MONO)) || (!pReverb->m_Aux && (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO))) { ALOGV("Reverb_setConfig invalid config"); return -EINVAL; } pRvbModule->config = *pConfig; pReverb->m_nSamplingRate = pRvbModule->config.outputCfg.samplingRate; switch (pReverb->m_nSamplingRate) { case 8000: pReverb->m_nUpdatePeriodInBits = 5; bufferSizeInSamples = 4096; pReverb->m_nCosWT_5KHz = -23170; break; case 16000: pReverb->m_nUpdatePeriodInBits = 6; bufferSizeInSamples = 8192; pReverb->m_nCosWT_5KHz = -12540; break; case 22050: pReverb->m_nUpdatePeriodInBits = 7; bufferSizeInSamples = 8192; pReverb->m_nCosWT_5KHz = 4768; break; case 32000: pReverb->m_nUpdatePeriodInBits = 7; bufferSizeInSamples = 16384; pReverb->m_nCosWT_5KHz = 18205; break; case 44100: pReverb->m_nUpdatePeriodInBits = 8; bufferSizeInSamples = 16384; pReverb->m_nCosWT_5KHz = 24799; break; case 48000: pReverb->m_nUpdatePeriodInBits = 8; bufferSizeInSamples = 16384; pReverb->m_nCosWT_5KHz = 25997; break; default: ALOGV("Reverb_setConfig invalid sampling rate %d", pReverb->m_nSamplingRate); return -EINVAL; } // Define a mask for circular addressing, so that array index // can wraparound and stay in array boundary of 0, 1, ..., (buffer size -1) // The buffer size MUST be a power of two pReverb->m_nBufferMask = (int32_t) (bufferSizeInSamples - 1); /* reverb parameters are updated every 2^(pReverb->m_nUpdatePeriodInBits) samples */ updatePeriodInSamples = (int32_t) (0x1L << pReverb->m_nUpdatePeriodInBits); /* calculate the update counter by bitwise ANDING with this value to generate a 2^n modulo value */ pReverb->m_nUpdatePeriodInSamples = (int32_t) updatePeriodInSamples; xfadePeriodInSamples = (int32_t) (REVERB_XFADE_PERIOD_IN_SECONDS * (double) pReverb->m_nSamplingRate); // set xfade parameters pReverb->m_nPhaseIncrement = (int16_t) (65536 / ((int16_t) xfadePeriodInSamples / (int16_t) updatePeriodInSamples)); if (init) { ReverbReadInPresets(pReverb); // for debugging purposes, allow noise generator pReverb->m_bUseNoise = true; // for debugging purposes, allow bypass pReverb->m_bBypass = 0; pReverb->m_nNextRoom = 1; pReverb->m_nNoise = (int16_t) 0xABCD; } Reverb_Reset(pReverb, init); return 0; } /*---------------------------------------------------------------------------- * Reverb_getConfig() *---------------------------------------------------------------------------- * Purpose: * Get input and output audio configuration. * * Inputs: * pRvbModule - pointer to reverb effect module * pConfig - pointer to effect_config_t structure containing input * and output audio parameters configuration * Outputs: * * Side Effects: * *---------------------------------------------------------------------------- */ void Reverb_getConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig) { *pConfig = pRvbModule->config; } /*---------------------------------------------------------------------------- * Reverb_Reset() *---------------------------------------------------------------------------- * Purpose: * Reset internal states and clear delay lines. * * Inputs: * pReverb - pointer to reverb context * init - true if called from init function * * Outputs: * * Side Effects: * *---------------------------------------------------------------------------- */ void Reverb_Reset(reverb_object_t *pReverb, bool init) { int bufferSizeInSamples = (int32_t) (pReverb->m_nBufferMask + 1); int maxApSamples; int maxDelaySamples; int maxEarlySamples; int ap1In; int delay0In; int delay1In; int32_t i; uint16_t nOffset; maxApSamples = ((int32_t) (MAX_AP_TIME * pReverb->m_nSamplingRate) >> 16); maxDelaySamples = ((int32_t) (MAX_DELAY_TIME * pReverb->m_nSamplingRate) >> 16); maxEarlySamples = ((int32_t) (MAX_EARLY_TIME * pReverb->m_nSamplingRate) >> 16); ap1In = (AP0_IN + maxApSamples + GUARD); delay0In = (ap1In + maxApSamples + GUARD); delay1In = (delay0In + maxDelaySamples + GUARD); // Define the max offsets for the end points of each section // i.e., we don't expect a given section's taps to go beyond // the following limits pReverb->m_nEarly0in = (delay1In + maxDelaySamples + GUARD); pReverb->m_nEarly1in = (pReverb->m_nEarly0in + maxEarlySamples + GUARD); pReverb->m_sAp0.m_zApIn = AP0_IN; pReverb->m_zD0In = delay0In; pReverb->m_sAp1.m_zApIn = ap1In; pReverb->m_zD1In = delay1In; pReverb->m_zOutLpfL = 0; pReverb->m_zOutLpfR = 0; pReverb->m_nRevFbkR = 0; pReverb->m_nRevFbkL = 0; // set base index into circular buffer pReverb->m_nBaseIndex = 0; // clear the reverb delay line for (i = 0; i < bufferSizeInSamples; i++) { pReverb->m_nDelayLine[i] = 0; } ReverbUpdateRoom(pReverb, init); pReverb->m_nUpdateCounter = 0; pReverb->m_nPhase = -32768; pReverb->m_nSin = 0; pReverb->m_nCos = 0; pReverb->m_nSinIncrement = 0; pReverb->m_nCosIncrement = 0; // set delay tap lengths nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD1Cross = pReverb->m_nDelay1Out - pReverb->m_nMaxExcursion + nOffset; nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD0Cross = pReverb->m_nDelay0Out - pReverb->m_nMaxExcursion - nOffset; nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD0Self = pReverb->m_nDelay0Out - pReverb->m_nMaxExcursion - nOffset; nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD1Self = pReverb->m_nDelay1Out - pReverb->m_nMaxExcursion + nOffset; } /*---------------------------------------------------------------------------- * Reverb_getParameter() *---------------------------------------------------------------------------- * Purpose: * Get a Reverb parameter * * Inputs: * pReverb - handle to instance data * param - parameter * pValue - pointer to variable to hold retrieved value * pSize - pointer to value size: maximum size as input * * Outputs: * *pValue updated with parameter value * *pSize updated with actual value size * * * Side Effects: * *---------------------------------------------------------------------------- */ int Reverb_getParameter(reverb_object_t *pReverb, int32_t param, uint32_t *pSize, void *pValue) { int32_t *pValue32; int16_t *pValue16; t_reverb_settings *pProperties; int32_t i; int32_t temp; int32_t temp2; uint32_t size; if (pReverb->m_Preset) { if (param != REVERB_PARAM_PRESET || *pSize < sizeof(int16_t)) { return -EINVAL; } size = sizeof(int16_t); pValue16 = (int16_t *)pValue; // REVERB_PRESET_NONE is mapped to bypass if (pReverb->m_bBypass != 0) { *pValue16 = (int16_t)REVERB_PRESET_NONE; } else { *pValue16 = (int16_t)(pReverb->m_nNextRoom + 1); } ALOGV("get REVERB_PARAM_PRESET, preset %d", *pValue16); } else { switch (param) { case REVERB_PARAM_ROOM_LEVEL: case REVERB_PARAM_ROOM_HF_LEVEL: case REVERB_PARAM_DECAY_HF_RATIO: case REVERB_PARAM_REFLECTIONS_LEVEL: case REVERB_PARAM_REVERB_LEVEL: case REVERB_PARAM_DIFFUSION: case REVERB_PARAM_DENSITY: size = sizeof(int16_t); break; case REVERB_PARAM_BYPASS: case REVERB_PARAM_DECAY_TIME: case REVERB_PARAM_REFLECTIONS_DELAY: case REVERB_PARAM_REVERB_DELAY: size = sizeof(int32_t); break; case REVERB_PARAM_PROPERTIES: size = sizeof(t_reverb_settings); break; default: return -EINVAL; } if (*pSize < size) { return -EINVAL; } pValue32 = (int32_t *) pValue; pValue16 = (int16_t *) pValue; pProperties = (t_reverb_settings *) pValue; switch (param) { case REVERB_PARAM_BYPASS: *pValue32 = (int32_t) pReverb->m_bBypass; break; case REVERB_PARAM_PROPERTIES: pValue16 = &pProperties->roomLevel; /* FALL THROUGH */ case REVERB_PARAM_ROOM_LEVEL: // Convert m_nRoomLpfFwd to millibels temp = (pReverb->m_nRoomLpfFwd << 15) / (32767 - pReverb->m_nRoomLpfFbk); *pValue16 = Effects_Linear16ToMillibels(temp); ALOGV("get REVERB_PARAM_ROOM_LEVEL %d, gain %d, m_nRoomLpfFwd %d, m_nRoomLpfFbk %d", *pValue16, temp, pReverb->m_nRoomLpfFwd, pReverb->m_nRoomLpfFbk); if (param == REVERB_PARAM_ROOM_LEVEL) { break; } pValue16 = &pProperties->roomHFLevel; /* FALL THROUGH */ case REVERB_PARAM_ROOM_HF_LEVEL: // The ratio between linear gain at 0Hz and at 5000Hz for the room low pass is: // (1 + a1) / sqrt(a1^2 + 2*C*a1 + 1) where: // - a1 is minus the LP feedback gain: -pReverb->m_nRoomLpfFbk // - C is cos(2piWT) @ 5000Hz: pReverb->m_nCosWT_5KHz temp = MULT_EG1_EG1(pReverb->m_nRoomLpfFbk, pReverb->m_nRoomLpfFbk); ALOGV("get REVERB_PARAM_ROOM_HF_LEVEL, a1^2 %d", temp); temp2 = MULT_EG1_EG1(pReverb->m_nRoomLpfFbk, pReverb->m_nCosWT_5KHz) << 1; ALOGV("get REVERB_PARAM_ROOM_HF_LEVEL, 2 Cos a1 %d", temp2); temp = 32767 + temp - temp2; ALOGV("get REVERB_PARAM_ROOM_HF_LEVEL, a1^2 + 2 Cos a1 + 1 %d", temp); temp = Effects_Sqrt(temp) * 181; ALOGV("get REVERB_PARAM_ROOM_HF_LEVEL, SQRT(a1^2 + 2 Cos a1 + 1) %d", temp); temp = ((32767 - pReverb->m_nRoomLpfFbk) << 15) / temp; ALOGV("get REVERB_PARAM_ROOM_HF_LEVEL, gain %d, m_nRoomLpfFwd %d, m_nRoomLpfFbk %d", temp, pReverb->m_nRoomLpfFwd, pReverb->m_nRoomLpfFbk); *pValue16 = Effects_Linear16ToMillibels(temp); if (param == REVERB_PARAM_ROOM_HF_LEVEL) { break; } pValue32 = (int32_t *)&pProperties->decayTime; /* FALL THROUGH */ case REVERB_PARAM_DECAY_TIME: // Calculate reverb feedback path gain temp = (pReverb->m_nRvbLpfFwd << 15) / (32767 - pReverb->m_nRvbLpfFbk); temp = Effects_Linear16ToMillibels(temp); // Calculate decay time: g = -6000 d/DT , g gain in millibels, d reverb delay, DT decay time temp = (-6000 * pReverb->m_nLateDelay) / temp; // Convert samples to ms *pValue32 = (temp * 1000) / pReverb->m_nSamplingRate; ALOGV("get REVERB_PARAM_DECAY_TIME, samples %d, ms %d", temp, *pValue32); if (param == REVERB_PARAM_DECAY_TIME) { break; } pValue16 = &pProperties->decayHFRatio; /* FALL THROUGH */ case REVERB_PARAM_DECAY_HF_RATIO: // If r is the decay HF ratio (r = REVERB_PARAM_DECAY_HF_RATIO/1000) we have: // DT_5000Hz = DT_0Hz * r // and G_5000Hz = -6000 * d / DT_5000Hz and G_0Hz = -6000 * d / DT_0Hz in millibels so : // r = G_0Hz/G_5000Hz in millibels // The linear gain at 5000Hz is b0 / sqrt(a1^2 + 2*C*a1 + 1) where: // - a1 is minus the LP feedback gain: -pReverb->m_nRvbLpfFbk // - b0 is the LP forward gain: pReverb->m_nRvbLpfFwd // - C is cos(2piWT) @ 5000Hz: pReverb->m_nCosWT_5KHz if (pReverb->m_nRvbLpfFbk == 0) { *pValue16 = 1000; ALOGV("get REVERB_PARAM_DECAY_HF_RATIO, pReverb->m_nRvbLpfFbk == 0, ratio %d", *pValue16); } else { temp = MULT_EG1_EG1(pReverb->m_nRvbLpfFbk, pReverb->m_nRvbLpfFbk); temp2 = MULT_EG1_EG1(pReverb->m_nRvbLpfFbk, pReverb->m_nCosWT_5KHz) << 1; temp = 32767 + temp - temp2; temp = Effects_Sqrt(temp) * 181; temp = (pReverb->m_nRvbLpfFwd << 15) / temp; // The linear gain at 0Hz is b0 / (a1 + 1) temp2 = (pReverb->m_nRvbLpfFwd << 15) / (32767 - pReverb->m_nRvbLpfFbk); temp = Effects_Linear16ToMillibels(temp); temp2 = Effects_Linear16ToMillibels(temp2); ALOGV("get REVERB_PARAM_DECAY_HF_RATIO, gain 5KHz %d mB, gain DC %d mB", temp, temp2); if (temp == 0) temp = 1; temp = (int16_t) ((1000 * temp2) / temp); if (temp > 1000) temp = 1000; *pValue16 = temp; ALOGV("get REVERB_PARAM_DECAY_HF_RATIO, ratio %d", *pValue16); } if (param == REVERB_PARAM_DECAY_HF_RATIO) { break; } pValue16 = &pProperties->reflectionsLevel; /* FALL THROUGH */ case REVERB_PARAM_REFLECTIONS_LEVEL: *pValue16 = Effects_Linear16ToMillibels(pReverb->m_nEarlyGain); ALOGV("get REVERB_PARAM_REFLECTIONS_LEVEL, %d", *pValue16); if (param == REVERB_PARAM_REFLECTIONS_LEVEL) { break; } pValue32 = (int32_t *)&pProperties->reflectionsDelay; /* FALL THROUGH */ case REVERB_PARAM_REFLECTIONS_DELAY: // convert samples to ms *pValue32 = (pReverb->m_nEarlyDelay * 1000) / pReverb->m_nSamplingRate; ALOGV("get REVERB_PARAM_REFLECTIONS_DELAY, samples %d, ms %d", pReverb->m_nEarlyDelay, *pValue32); if (param == REVERB_PARAM_REFLECTIONS_DELAY) { break; } pValue16 = &pProperties->reverbLevel; /* FALL THROUGH */ case REVERB_PARAM_REVERB_LEVEL: // Convert linear gain to millibels *pValue16 = Effects_Linear16ToMillibels(pReverb->m_nLateGain << 2); ALOGV("get REVERB_PARAM_REVERB_LEVEL %d", *pValue16); if (param == REVERB_PARAM_REVERB_LEVEL) { break; } pValue32 = (int32_t *)&pProperties->reverbDelay; /* FALL THROUGH */ case REVERB_PARAM_REVERB_DELAY: // convert samples to ms *pValue32 = (pReverb->m_nLateDelay * 1000) / pReverb->m_nSamplingRate; ALOGV("get REVERB_PARAM_REVERB_DELAY, samples %d, ms %d", pReverb->m_nLateDelay, *pValue32); if (param == REVERB_PARAM_REVERB_DELAY) { break; } pValue16 = &pProperties->diffusion; /* FALL THROUGH */ case REVERB_PARAM_DIFFUSION: temp = (int16_t) ((1000 * (pReverb->m_sAp0.m_nApGain - AP0_GAIN_BASE)) / AP0_GAIN_RANGE); if (temp < 0) temp = 0; if (temp > 1000) temp = 1000; *pValue16 = temp; ALOGV("get REVERB_PARAM_DIFFUSION, %d, AP0 gain %d", *pValue16, pReverb->m_sAp0.m_nApGain); if (param == REVERB_PARAM_DIFFUSION) { break; } pValue16 = &pProperties->density; /* FALL THROUGH */ case REVERB_PARAM_DENSITY: // Calculate AP delay in time units temp = ((pReverb->m_sAp0.m_zApOut - pReverb->m_sAp0.m_zApIn) << 16) / pReverb->m_nSamplingRate; temp = (int16_t) ((1000 * (temp - AP0_TIME_BASE)) / AP0_TIME_RANGE); if (temp < 0) temp = 0; if (temp > 1000) temp = 1000; *pValue16 = temp; ALOGV("get REVERB_PARAM_DENSITY, %d, AP0 delay smps %d", *pValue16, pReverb->m_sAp0.m_zApOut - pReverb->m_sAp0.m_zApIn); break; default: break; } } *pSize = size; ALOGV("Reverb_getParameter, context %p, param %d, value %d", pReverb, param, *(int *)pValue); return 0; } /* end Reverb_getParameter */ /*---------------------------------------------------------------------------- * Reverb_setParameter() *---------------------------------------------------------------------------- * Purpose: * Set a Reverb parameter * * Inputs: * pReverb - handle to instance data * param - parameter * pValue - pointer to parameter value * size - value size * * Outputs: * * * Side Effects: * *---------------------------------------------------------------------------- */ int Reverb_setParameter(reverb_object_t *pReverb, int32_t param, uint32_t size, void *pValue) { int32_t value32; int16_t value16; t_reverb_settings *pProperties; int32_t i; int32_t temp; int32_t temp2; reverb_preset_t *pPreset; int maxSamples; int32_t averageDelay; uint32_t paramSize; ALOGV("Reverb_setParameter, context %p, param %d, value16 %d, value32 %d", pReverb, param, *(int16_t *)pValue, *(int32_t *)pValue); if (pReverb->m_Preset) { if (param != REVERB_PARAM_PRESET || size != sizeof(int16_t)) { return -EINVAL; } value16 = *(int16_t *)pValue; ALOGV("set REVERB_PARAM_PRESET, preset %d", value16); if (value16 < REVERB_PRESET_NONE || value16 > REVERB_PRESET_PLATE) { return -EINVAL; } // REVERB_PRESET_NONE is mapped to bypass if (value16 == REVERB_PRESET_NONE) { pReverb->m_bBypass = 1; } else { pReverb->m_bBypass = 0; pReverb->m_nNextRoom = value16 - 1; } } else { switch (param) { case REVERB_PARAM_ROOM_LEVEL: case REVERB_PARAM_ROOM_HF_LEVEL: case REVERB_PARAM_DECAY_HF_RATIO: case REVERB_PARAM_REFLECTIONS_LEVEL: case REVERB_PARAM_REVERB_LEVEL: case REVERB_PARAM_DIFFUSION: case REVERB_PARAM_DENSITY: paramSize = sizeof(int16_t); break; case REVERB_PARAM_BYPASS: case REVERB_PARAM_DECAY_TIME: case REVERB_PARAM_REFLECTIONS_DELAY: case REVERB_PARAM_REVERB_DELAY: paramSize = sizeof(int32_t); break; case REVERB_PARAM_PROPERTIES: paramSize = sizeof(t_reverb_settings); break; default: return -EINVAL; } if (size != paramSize) { return -EINVAL; } if (paramSize == sizeof(int16_t)) { value16 = *(int16_t *) pValue; } else if (paramSize == sizeof(int32_t)) { value32 = *(int32_t *) pValue; } else { pProperties = (t_reverb_settings *) pValue; } pPreset = &pReverb->m_sPreset.m_sPreset[pReverb->m_nNextRoom]; switch (param) { case REVERB_PARAM_BYPASS: pReverb->m_bBypass = (uint16_t)value32; break; case REVERB_PARAM_PROPERTIES: value16 = pProperties->roomLevel; /* FALL THROUGH */ case REVERB_PARAM_ROOM_LEVEL: // Convert millibels to linear 16 bit signed => m_nRoomLpfFwd if (value16 > 0) return -EINVAL; temp = Effects_MillibelsToLinear16(value16); pReverb->m_nRoomLpfFwd = MULT_EG1_EG1(temp, (32767 - pReverb->m_nRoomLpfFbk)); ALOGV("REVERB_PARAM_ROOM_LEVEL, gain %d, new m_nRoomLpfFwd %d, m_nRoomLpfFbk %d", temp, pReverb->m_nRoomLpfFwd, pReverb->m_nRoomLpfFbk); if (param == REVERB_PARAM_ROOM_LEVEL) break; value16 = pProperties->roomHFLevel; /* FALL THROUGH */ case REVERB_PARAM_ROOM_HF_LEVEL: // Limit to 0 , -40dB range because of low pass implementation if (value16 > 0 || value16 < -4000) return -EINVAL; // Convert attenuation @ 5000H expressed in millibels to => m_nRoomLpfFbk // m_nRoomLpfFbk is -a1 where a1 is the solution of: // a1^2 + 2*(C-dG^2)/(1-dG^2)*a1 + 1 = 0 where: // - C is cos(2*pi*5000/Fs) (pReverb->m_nCosWT_5KHz) // - dG is G0/Gf (G0 is the linear gain at DC and Gf is the wanted gain at 5000Hz) // Save current DC gain m_nRoomLpfFwd / (32767 - m_nRoomLpfFbk) to keep it unchanged // while changing HF level temp2 = (pReverb->m_nRoomLpfFwd << 15) / (32767 - pReverb->m_nRoomLpfFbk); if (value16 == 0) { pReverb->m_nRoomLpfFbk = 0; } else { int32_t dG2, b, delta; // dG^2 temp = Effects_MillibelsToLinear16(value16); ALOGV("REVERB_PARAM_ROOM_HF_LEVEL, HF gain %d", temp); temp = (1 << 30) / temp; ALOGV("REVERB_PARAM_ROOM_HF_LEVEL, 1/ HF gain %d", temp); dG2 = (int32_t) (((int64_t) temp * (int64_t) temp) >> 15); ALOGV("REVERB_PARAM_ROOM_HF_LEVEL, 1/ HF gain ^ 2 %d", dG2); // b = 2*(C-dG^2)/(1-dG^2) b = (int32_t) ((((int64_t) 1 << (15 + 1)) * ((int64_t) pReverb->m_nCosWT_5KHz - (int64_t) dG2)) / ((int64_t) 32767 - (int64_t) dG2)); // delta = b^2 - 4 delta = (int32_t) ((((int64_t) b * (int64_t) b) >> 15) - (1 << (15 + 2))); ALOGV_IF(delta > (1<<30), " delta overflow %d", delta); ALOGV("REVERB_PARAM_ROOM_HF_LEVEL, dG2 %d, b %d, delta %d, m_nCosWT_5KHz %d", dG2, b, delta, pReverb->m_nCosWT_5KHz); // m_nRoomLpfFbk = -a1 = - (- b + sqrt(delta)) / 2 pReverb->m_nRoomLpfFbk = (b - Effects_Sqrt(delta) * 181) >> 1; } ALOGV("REVERB_PARAM_ROOM_HF_LEVEL, olg DC gain %d new m_nRoomLpfFbk %d, old m_nRoomLpfFwd %d", temp2, pReverb->m_nRoomLpfFbk, pReverb->m_nRoomLpfFwd); pReverb->m_nRoomLpfFwd = MULT_EG1_EG1(temp2, (32767 - pReverb->m_nRoomLpfFbk)); ALOGV("REVERB_PARAM_ROOM_HF_LEVEL, new m_nRoomLpfFwd %d", pReverb->m_nRoomLpfFwd); if (param == REVERB_PARAM_ROOM_HF_LEVEL) break; value32 = pProperties->decayTime; /* FALL THROUGH */ case REVERB_PARAM_DECAY_TIME: // Convert milliseconds to => m_nRvbLpfFwd (function of m_nRvbLpfFbk) // convert ms to samples value32 = (value32 * pReverb->m_nSamplingRate) / 1000; // calculate valid decay time range as a function of current reverb delay and // max feed back gain. Min value <=> -40dB in one pass, Max value <=> feedback gain = -1 dB // Calculate attenuation for each round in late reverb given a total attenuation of -6000 millibels. // g = -6000 d/DT , g gain in millibels, d reverb delay, DT decay time averageDelay = pReverb->m_nLateDelay - pReverb->m_nMaxExcursion; averageDelay += ((pReverb->m_sAp0.m_zApOut - pReverb->m_sAp0.m_zApIn) + (pReverb->m_sAp1.m_zApOut - pReverb->m_sAp1.m_zApIn)) >> 1; temp = (-6000 * averageDelay) / value32; ALOGV("REVERB_PARAM_DECAY_TIME, delay smps %d, DT smps %d, gain mB %d",averageDelay, value32, temp); if (temp < -4000 || temp > -100) return -EINVAL; // calculate low pass gain by adding reverb input attenuation (pReverb->m_nLateGain) and substrating output // xfade and sum gain (max +9dB) temp -= Effects_Linear16ToMillibels(pReverb->m_nLateGain) + 900; temp = Effects_MillibelsToLinear16(temp); // DC gain (temp) = b0 / (1 + a1) = pReverb->m_nRvbLpfFwd / (32767 - pReverb->m_nRvbLpfFbk) pReverb->m_nRvbLpfFwd = MULT_EG1_EG1(temp, (32767 - pReverb->m_nRvbLpfFbk)); ALOGV("REVERB_PARAM_DECAY_TIME, gain %d, new m_nRvbLpfFwd %d, old m_nRvbLpfFbk %d, reverb gain %d", temp, pReverb->m_nRvbLpfFwd, pReverb->m_nRvbLpfFbk, Effects_Linear16ToMillibels(pReverb->m_nLateGain)); if (param == REVERB_PARAM_DECAY_TIME) break; value16 = pProperties->decayHFRatio; /* FALL THROUGH */ case REVERB_PARAM_DECAY_HF_RATIO: // We limit max value to 1000 because reverb filter is lowpass only if (value16 < 100 || value16 > 1000) return -EINVAL; // Convert per mille to => m_nLpfFwd, m_nLpfFbk // Save current DC gain m_nRoomLpfFwd / (32767 - m_nRoomLpfFbk) to keep it unchanged // while changing HF level temp2 = (pReverb->m_nRvbLpfFwd << 15) / (32767 - pReverb->m_nRvbLpfFbk); if (value16 == 1000) { pReverb->m_nRvbLpfFbk = 0; } else { int32_t dG2, b, delta; temp = Effects_Linear16ToMillibels(temp2); // G_5000Hz = G_DC * (1000/REVERB_PARAM_DECAY_HF_RATIO) in millibels value32 = ((int32_t) 1000 << 15) / (int32_t) value16; ALOGV("REVERB_PARAM_DECAY_HF_RATIO, DC gain %d, DC gain mB %d, 1000/R %d", temp2, temp, value32); temp = (int32_t) (((int64_t) temp * (int64_t) value32) >> 15); if (temp < -4000) { ALOGV("REVERB_PARAM_DECAY_HF_RATIO HF gain overflow %d mB", temp); temp = -4000; } temp = Effects_MillibelsToLinear16(temp); ALOGV("REVERB_PARAM_DECAY_HF_RATIO, HF gain %d", temp); // dG^2 temp = (temp2 << 15) / temp; dG2 = (int32_t) (((int64_t) temp * (int64_t) temp) >> 15); // b = 2*(C-dG^2)/(1-dG^2) b = (int32_t) ((((int64_t) 1 << (15 + 1)) * ((int64_t) pReverb->m_nCosWT_5KHz - (int64_t) dG2)) / ((int64_t) 32767 - (int64_t) dG2)); // delta = b^2 - 4 delta = (int32_t) ((((int64_t) b * (int64_t) b) >> 15) - (1 << (15 + 2))); // m_nRoomLpfFbk = -a1 = - (- b + sqrt(delta)) / 2 pReverb->m_nRvbLpfFbk = (b - Effects_Sqrt(delta) * 181) >> 1; ALOGV("REVERB_PARAM_DECAY_HF_RATIO, dG2 %d, b %d, delta %d", dG2, b, delta); } ALOGV("REVERB_PARAM_DECAY_HF_RATIO, gain %d, m_nRvbLpfFbk %d, m_nRvbLpfFwd %d", temp2, pReverb->m_nRvbLpfFbk, pReverb->m_nRvbLpfFwd); pReverb->m_nRvbLpfFwd = MULT_EG1_EG1(temp2, (32767 - pReverb->m_nRvbLpfFbk)); if (param == REVERB_PARAM_DECAY_HF_RATIO) break; value16 = pProperties->reflectionsLevel; /* FALL THROUGH */ case REVERB_PARAM_REFLECTIONS_LEVEL: // We limit max value to 0 because gain is limited to 0dB if (value16 > 0 || value16 < -6000) return -EINVAL; // Convert millibels to linear 16 bit signed and recompute m_sEarlyL.m_nGain[i] and m_sEarlyR.m_nGain[i]. value16 = Effects_MillibelsToLinear16(value16); for (i = 0; i < REVERB_MAX_NUM_REFLECTIONS; i++) { pReverb->m_sEarlyL.m_nGain[i] = MULT_EG1_EG1(pPreset->m_sEarlyL.m_nGain[i],value16); pReverb->m_sEarlyR.m_nGain[i] = MULT_EG1_EG1(pPreset->m_sEarlyR.m_nGain[i],value16); } pReverb->m_nEarlyGain = value16; ALOGV("REVERB_PARAM_REFLECTIONS_LEVEL, m_nEarlyGain %d", pReverb->m_nEarlyGain); if (param == REVERB_PARAM_REFLECTIONS_LEVEL) break; value32 = pProperties->reflectionsDelay; /* FALL THROUGH */ case REVERB_PARAM_REFLECTIONS_DELAY: // We limit max value MAX_EARLY_TIME // convert ms to time units temp = (value32 * 65536) / 1000; if (temp < 0 || temp > MAX_EARLY_TIME) return -EINVAL; maxSamples = (int32_t) (MAX_EARLY_TIME * pReverb->m_nSamplingRate) >> 16; temp = (temp * pReverb->m_nSamplingRate) >> 16; for (i = 0; i < REVERB_MAX_NUM_REFLECTIONS; i++) { temp2 = temp + (((int32_t) pPreset->m_sEarlyL.m_zDelay[i] * pReverb->m_nSamplingRate) >> 16); if (temp2 > maxSamples) temp2 = maxSamples; pReverb->m_sEarlyL.m_zDelay[i] = pReverb->m_nEarly0in + temp2; temp2 = temp + (((int32_t) pPreset->m_sEarlyR.m_zDelay[i] * pReverb->m_nSamplingRate) >> 16); if (temp2 > maxSamples) temp2 = maxSamples; pReverb->m_sEarlyR.m_zDelay[i] = pReverb->m_nEarly1in + temp2; } pReverb->m_nEarlyDelay = temp; ALOGV("REVERB_PARAM_REFLECTIONS_DELAY, m_nEarlyDelay smps %d max smp delay %d", pReverb->m_nEarlyDelay, maxSamples); // Convert milliseconds to sample count => m_nEarlyDelay if (param == REVERB_PARAM_REFLECTIONS_DELAY) break; value16 = pProperties->reverbLevel; /* FALL THROUGH */ case REVERB_PARAM_REVERB_LEVEL: // We limit max value to 0 because gain is limited to 0dB if (value16 > 0 || value16 < -6000) return -EINVAL; // Convert millibels to linear 16 bits (gange 0 - 8191) => m_nLateGain. pReverb->m_nLateGain = Effects_MillibelsToLinear16(value16) >> 2; ALOGV("REVERB_PARAM_REVERB_LEVEL, m_nLateGain %d", pReverb->m_nLateGain); if (param == REVERB_PARAM_REVERB_LEVEL) break; value32 = pProperties->reverbDelay; /* FALL THROUGH */ case REVERB_PARAM_REVERB_DELAY: // We limit max value to MAX_DELAY_TIME // convert ms to time units temp = (value32 * 65536) / 1000; if (temp < 0 || temp > MAX_DELAY_TIME) return -EINVAL; maxSamples = (int32_t) (MAX_DELAY_TIME * pReverb->m_nSamplingRate) >> 16; temp = (temp * pReverb->m_nSamplingRate) >> 16; if ((temp + pReverb->m_nMaxExcursion) > maxSamples) { temp = maxSamples - pReverb->m_nMaxExcursion; } if (temp < pReverb->m_nMaxExcursion) { temp = pReverb->m_nMaxExcursion; } temp -= pReverb->m_nLateDelay; pReverb->m_nDelay0Out += temp; pReverb->m_nDelay1Out += temp; pReverb->m_nLateDelay += temp; ALOGV("REVERB_PARAM_REVERB_DELAY, m_nLateDelay smps %d max smp delay %d", pReverb->m_nLateDelay, maxSamples); // Convert milliseconds to sample count => m_nDelay1Out + m_nMaxExcursion if (param == REVERB_PARAM_REVERB_DELAY) break; value16 = pProperties->diffusion; /* FALL THROUGH */ case REVERB_PARAM_DIFFUSION: if (value16 < 0 || value16 > 1000) return -EINVAL; // Convert per mille to m_sAp0.m_nApGain, m_sAp1.m_nApGain pReverb->m_sAp0.m_nApGain = AP0_GAIN_BASE + ((int32_t) value16 * AP0_GAIN_RANGE) / 1000; pReverb->m_sAp1.m_nApGain = AP1_GAIN_BASE + ((int32_t) value16 * AP1_GAIN_RANGE) / 1000; ALOGV("REVERB_PARAM_DIFFUSION, m_sAp0.m_nApGain %d m_sAp1.m_nApGain %d", pReverb->m_sAp0.m_nApGain, pReverb->m_sAp1.m_nApGain); if (param == REVERB_PARAM_DIFFUSION) break; value16 = pProperties->density; /* FALL THROUGH */ case REVERB_PARAM_DENSITY: if (value16 < 0 || value16 > 1000) return -EINVAL; // Convert per mille to m_sAp0.m_zApOut, m_sAp1.m_zApOut maxSamples = (int32_t) (MAX_AP_TIME * pReverb->m_nSamplingRate) >> 16; temp = AP0_TIME_BASE + ((int32_t) value16 * AP0_TIME_RANGE) / 1000; /*lint -e{702} shift for performance */ temp = (temp * pReverb->m_nSamplingRate) >> 16; if (temp > maxSamples) temp = maxSamples; pReverb->m_sAp0.m_zApOut = (uint16_t) (pReverb->m_sAp0.m_zApIn + temp); ALOGV("REVERB_PARAM_DENSITY, Ap0 delay smps %d", temp); temp = AP1_TIME_BASE + ((int32_t) value16 * AP1_TIME_RANGE) / 1000; /*lint -e{702} shift for performance */ temp = (temp * pReverb->m_nSamplingRate) >> 16; if (temp > maxSamples) temp = maxSamples; pReverb->m_sAp1.m_zApOut = (uint16_t) (pReverb->m_sAp1.m_zApIn + temp); ALOGV("Ap1 delay smps %d", temp); break; default: break; } } return 0; } /* end Reverb_setParameter */ /*---------------------------------------------------------------------------- * ReverbUpdateXfade *---------------------------------------------------------------------------- * Purpose: * Update the xfade parameters as required * * Inputs: * nNumSamplesToAdd - number of samples to write to buffer * * Outputs: * * * Side Effects: * - xfade parameters will be changed * *---------------------------------------------------------------------------- */ static int ReverbUpdateXfade(reverb_object_t *pReverb, int nNumSamplesToAdd) { uint16_t nOffset; int16_t tempCos; int16_t tempSin; if (pReverb->m_nXfadeCounter >= pReverb->m_nXfadeInterval) { /* update interval has elapsed, so reset counter */ pReverb->m_nXfadeCounter = 0; // Pin the sin,cos values to min / max values to ensure that the // modulated taps' coefs are zero (thus no clicks) if (pReverb->m_nPhaseIncrement > 0) { // if phase increment > 0, then sin -> 1, cos -> 0 pReverb->m_nSin = 32767; pReverb->m_nCos = 0; // reset the phase to match the sin, cos values pReverb->m_nPhase = 32767; // modulate the cross taps because their tap coefs are zero nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD1Cross = pReverb->m_nDelay1Out - pReverb->m_nMaxExcursion + nOffset; nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD0Cross = pReverb->m_nDelay0Out - pReverb->m_nMaxExcursion - nOffset; } else { // if phase increment < 0, then sin -> 0, cos -> 1 pReverb->m_nSin = 0; pReverb->m_nCos = 32767; // reset the phase to match the sin, cos values pReverb->m_nPhase = -32768; // modulate the self taps because their tap coefs are zero nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD0Self = pReverb->m_nDelay0Out - pReverb->m_nMaxExcursion - nOffset; nOffset = ReverbCalculateNoise(pReverb); pReverb->m_zD1Self = pReverb->m_nDelay1Out - pReverb->m_nMaxExcursion + nOffset; } // end if-else (pReverb->m_nPhaseIncrement > 0) // Reverse the direction of the sin,cos so that the // tap whose coef was previously increasing now decreases // and vice versa pReverb->m_nPhaseIncrement = -pReverb->m_nPhaseIncrement; } // end if counter >= update interval //compute what phase will be next time pReverb->m_nPhase += pReverb->m_nPhaseIncrement; //calculate what the new sin and cos need to reach by the next update ReverbCalculateSinCos(pReverb->m_nPhase, &tempSin, &tempCos); //calculate the per-sample increment required to get there by the next update /*lint -e{702} shift for performance */ pReverb->m_nSinIncrement = (tempSin - pReverb->m_nSin) >> pReverb->m_nUpdatePeriodInBits; /*lint -e{702} shift for performance */ pReverb->m_nCosIncrement = (tempCos - pReverb->m_nCos) >> pReverb->m_nUpdatePeriodInBits; /* increment update counter */ pReverb->m_nXfadeCounter += (uint16_t) nNumSamplesToAdd; return 0; } /* end ReverbUpdateXfade */ /*---------------------------------------------------------------------------- * ReverbCalculateNoise *---------------------------------------------------------------------------- * Purpose: * Calculate a noise sample and limit its value * * Inputs: * nMaxExcursion - noise value is limited to this value * pnNoise - return new noise sample in this (not limited) * * Outputs: * new limited noise value * * Side Effects: * - *pnNoise noise value is updated * *---------------------------------------------------------------------------- */ static uint16_t ReverbCalculateNoise(reverb_object_t *pReverb) { int16_t nNoise = pReverb->m_nNoise; // calculate new noise value if (pReverb->m_bUseNoise) { nNoise = (int16_t) (nNoise * 5 + 1); } else { nNoise = 0; } pReverb->m_nNoise = nNoise; // return the limited noise value return (pReverb->m_nMaxExcursion & nNoise); } /* end ReverbCalculateNoise */ /*---------------------------------------------------------------------------- * ReverbCalculateSinCos *---------------------------------------------------------------------------- * Purpose: * Calculate a new sin and cosine value based on the given phase * * Inputs: * nPhase - phase angle * pnSin - input old value, output new value * pnCos - input old value, output new value * * Outputs: * * Side Effects: * - *pnSin, *pnCos are updated * *---------------------------------------------------------------------------- */ static int ReverbCalculateSinCos(int16_t nPhase, int16_t *pnSin, int16_t *pnCos) { int32_t nTemp; int32_t nNetAngle; // -1 <= nPhase < 1 // However, for the calculation, we need a value // that ranges from -1/2 to +1/2, so divide the phase by 2 /*lint -e{702} shift for performance */ nNetAngle = nPhase >> 1; /* Implement the following sin(x) = (2-4*c)*x^2 + c + x cos(x) = (2-4*c)*x^2 + c - x where c = 1/sqrt(2) using the a0 + x*(a1 + x*a2) approach */ /* limit the input "angle" to be between -0.5 and +0.5 */ if (nNetAngle > EG1_HALF) { nNetAngle = EG1_HALF; } else if (nNetAngle < EG1_MINUS_HALF) { nNetAngle = EG1_MINUS_HALF; } /* calculate sin */ nTemp = EG1_ONE + MULT_EG1_EG1(REVERB_PAN_G2, nNetAngle); nTemp = REVERB_PAN_G0 + MULT_EG1_EG1(nTemp, nNetAngle); *pnSin = (int16_t) SATURATE_EG1(nTemp); /* calculate cos */ nTemp = -EG1_ONE + MULT_EG1_EG1(REVERB_PAN_G2, nNetAngle); nTemp = REVERB_PAN_G0 + MULT_EG1_EG1(nTemp, nNetAngle); *pnCos = (int16_t) SATURATE_EG1(nTemp); return 0; } /* end ReverbCalculateSinCos */ /*---------------------------------------------------------------------------- * Reverb *---------------------------------------------------------------------------- * Purpose: * apply reverb to the given signal * * Inputs: * nNu * pnSin - input old value, output new value * pnCos - input old value, output new value * * Outputs: * number of samples actually reverberated * * Side Effects: * *---------------------------------------------------------------------------- */ static int Reverb(reverb_object_t *pReverb, int nNumSamplesToAdd, short *pOutputBuffer, short *pInputBuffer) { int32_t i; int32_t nDelayOut0; int32_t nDelayOut1; uint16_t nBase; uint32_t nAddr; int32_t nTemp1; int32_t nTemp2; int32_t nApIn; int32_t nApOut; int32_t j; int32_t nEarlyOut; int32_t tempValue; // get the base address nBase = pReverb->m_nBaseIndex; for (i = 0; i < nNumSamplesToAdd; i++) { // ********** Left Allpass - start nApIn = *pInputBuffer; if (!pReverb->m_Aux) { pInputBuffer++; } // store to early delay line nAddr = CIRCULAR(nBase, pReverb->m_nEarly0in, pReverb->m_nBufferMask); pReverb->m_nDelayLine[nAddr] = (short) nApIn; // left input = (left dry * m_nLateGain) + right feedback from previous period nApIn = SATURATE(nApIn + pReverb->m_nRevFbkR); nApIn = MULT_EG1_EG1(nApIn, pReverb->m_nLateGain); // fetch allpass delay line out //nAddr = CIRCULAR(nBase, psAp0->m_zApOut, pReverb->m_nBufferMask); nAddr = CIRCULAR(nBase, pReverb->m_sAp0.m_zApOut, pReverb->m_nBufferMask); nDelayOut0 = pReverb->m_nDelayLine[nAddr]; // calculate allpass feedforward; subtract the feedforward result nTemp1 = MULT_EG1_EG1(nApIn, pReverb->m_sAp0.m_nApGain); nApOut = SATURATE(nDelayOut0 - nTemp1); // allpass output // calculate allpass feedback; add the feedback result nTemp1 = MULT_EG1_EG1(nApOut, pReverb->m_sAp0.m_nApGain); nTemp1 = SATURATE(nApIn + nTemp1); // inject into allpass delay nAddr = CIRCULAR(nBase, pReverb->m_sAp0.m_zApIn, pReverb->m_nBufferMask); pReverb->m_nDelayLine[nAddr] = (short) nTemp1; // inject allpass output into delay line nAddr = CIRCULAR(nBase, pReverb->m_zD0In, pReverb->m_nBufferMask); pReverb->m_nDelayLine[nAddr] = (short) nApOut; // ********** Left Allpass - end // ********** Right Allpass - start nApIn = (*pInputBuffer++); // store to early delay line nAddr = CIRCULAR(nBase, pReverb->m_nEarly1in, pReverb->m_nBufferMask); pReverb->m_nDelayLine[nAddr] = (short) nApIn; // right input = (right dry * m_nLateGain) + left feedback from previous period /*lint -e{702} use shift for performance */ nApIn = SATURATE(nApIn + pReverb->m_nRevFbkL); nApIn = MULT_EG1_EG1(nApIn, pReverb->m_nLateGain); // fetch allpass delay line out nAddr = CIRCULAR(nBase, pReverb->m_sAp1.m_zApOut, pReverb->m_nBufferMask); nDelayOut1 = pReverb->m_nDelayLine[nAddr]; // calculate allpass feedforward; subtract the feedforward result nTemp1 = MULT_EG1_EG1(nApIn, pReverb->m_sAp1.m_nApGain); nApOut = SATURATE(nDelayOut1 - nTemp1); // allpass output // calculate allpass feedback; add the feedback result nTemp1 = MULT_EG1_EG1(nApOut, pReverb->m_sAp1.m_nApGain); nTemp1 = SATURATE(nApIn + nTemp1); // inject into allpass delay nAddr = CIRCULAR(nBase, pReverb->m_sAp1.m_zApIn, pReverb->m_nBufferMask); pReverb->m_nDelayLine[nAddr] = (short) nTemp1; // inject allpass output into delay line nAddr = CIRCULAR(nBase, pReverb->m_zD1In, pReverb->m_nBufferMask); pReverb->m_nDelayLine[nAddr] = (short) nApOut; // ********** Right Allpass - end // ********** D0 output - start // fetch delay line self out nAddr = CIRCULAR(nBase, pReverb->m_zD0Self, pReverb->m_nBufferMask); nDelayOut0 = pReverb->m_nDelayLine[nAddr]; // calculate delay line self out nTemp1 = MULT_EG1_EG1(nDelayOut0, pReverb->m_nSin); // fetch delay line cross out nAddr = CIRCULAR(nBase, pReverb->m_zD1Cross, pReverb->m_nBufferMask); nDelayOut0 = pReverb->m_nDelayLine[nAddr]; // calculate delay line self out nTemp2 = MULT_EG1_EG1(nDelayOut0, pReverb->m_nCos); // calculate unfiltered delay out nDelayOut0 = SATURATE(nTemp1 + nTemp2); // ********** D0 output - end // ********** D1 output - start // fetch delay line self out nAddr = CIRCULAR(nBase, pReverb->m_zD1Self, pReverb->m_nBufferMask); nDelayOut1 = pReverb->m_nDelayLine[nAddr]; // calculate delay line self out nTemp1 = MULT_EG1_EG1(nDelayOut1, pReverb->m_nSin); // fetch delay line cross out nAddr = CIRCULAR(nBase, pReverb->m_zD0Cross, pReverb->m_nBufferMask); nDelayOut1 = pReverb->m_nDelayLine[nAddr]; // calculate delay line self out nTemp2 = MULT_EG1_EG1(nDelayOut1, pReverb->m_nCos); // calculate unfiltered delay out nDelayOut1 = SATURATE(nTemp1 + nTemp2); // ********** D1 output - end // ********** mixer and feedback - start // sum is fedback to right input (R + L) nDelayOut0 = (short) SATURATE(nDelayOut0 + nDelayOut1); // difference is feedback to left input (R - L) /*lint -e{685} lint complains that it can't saturate negative */ nDelayOut1 = (short) SATURATE(nDelayOut1 - nDelayOut0); // ********** mixer and feedback - end // calculate lowpass filter (mixer scale factor included in LPF feedforward) nTemp1 = MULT_EG1_EG1(nDelayOut0, pReverb->m_nRvbLpfFwd); nTemp2 = MULT_EG1_EG1(pReverb->m_nRevFbkL, pReverb->m_nRvbLpfFbk); // calculate filtered delay out and simultaneously update LPF state variable // filtered delay output is stored in m_nRevFbkL pReverb->m_nRevFbkL = (short) SATURATE(nTemp1 + nTemp2); // calculate lowpass filter (mixer scale factor included in LPF feedforward) nTemp1 = MULT_EG1_EG1(nDelayOut1, pReverb->m_nRvbLpfFwd); nTemp2 = MULT_EG1_EG1(pReverb->m_nRevFbkR, pReverb->m_nRvbLpfFbk); // calculate filtered delay out and simultaneously update LPF state variable // filtered delay output is stored in m_nRevFbkR pReverb->m_nRevFbkR = (short) SATURATE(nTemp1 + nTemp2); // ********** start early reflection generator, left //psEarly = &(pReverb->m_sEarlyL); for (j = 0; j < REVERB_MAX_NUM_REFLECTIONS; j++) { // fetch delay line out //nAddr = CIRCULAR(nBase, psEarly->m_zDelay[j], pReverb->m_nBufferMask); nAddr = CIRCULAR(nBase, pReverb->m_sEarlyL.m_zDelay[j], pReverb->m_nBufferMask); nTemp1 = pReverb->m_nDelayLine[nAddr]; // calculate reflection //nTemp1 = MULT_EG1_EG1(nDelayOut0, psEarly->m_nGain[j]); nTemp1 = MULT_EG1_EG1(nTemp1, pReverb->m_sEarlyL.m_nGain[j]); nDelayOut0 = SATURATE(nDelayOut0 + nTemp1); } // end for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++) // apply lowpass to early reflections and reverb output //nTemp1 = MULT_EG1_EG1(nEarlyOut, psEarly->m_nRvbLpfFwd); nTemp1 = MULT_EG1_EG1(nDelayOut0, pReverb->m_nRoomLpfFwd); //nTemp2 = MULT_EG1_EG1(psEarly->m_zLpf, psEarly->m_nLpfFbk); nTemp2 = MULT_EG1_EG1(pReverb->m_zOutLpfL, pReverb->m_nRoomLpfFbk); // calculate filtered out and simultaneously update LPF state variable // filtered output is stored in m_zOutLpfL pReverb->m_zOutLpfL = (short) SATURATE(nTemp1 + nTemp2); //sum with output buffer tempValue = *pOutputBuffer; *pOutputBuffer++ = (short) SATURATE(tempValue+pReverb->m_zOutLpfL); // ********** end early reflection generator, left // ********** start early reflection generator, right //psEarly = &(pReverb->m_sEarlyR); for (j = 0; j < REVERB_MAX_NUM_REFLECTIONS; j++) { // fetch delay line out nAddr = CIRCULAR(nBase, pReverb->m_sEarlyR.m_zDelay[j], pReverb->m_nBufferMask); nTemp1 = pReverb->m_nDelayLine[nAddr]; // calculate reflection nTemp1 = MULT_EG1_EG1(nTemp1, pReverb->m_sEarlyR.m_nGain[j]); nDelayOut1 = SATURATE(nDelayOut1 + nTemp1); } // end for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++) // apply lowpass to early reflections nTemp1 = MULT_EG1_EG1(nDelayOut1, pReverb->m_nRoomLpfFwd); nTemp2 = MULT_EG1_EG1(pReverb->m_zOutLpfR, pReverb->m_nRoomLpfFbk); // calculate filtered out and simultaneously update LPF state variable // filtered output is stored in m_zOutLpfR pReverb->m_zOutLpfR = (short) SATURATE(nTemp1 + nTemp2); //sum with output buffer tempValue = *pOutputBuffer; *pOutputBuffer++ = (short) SATURATE(tempValue + pReverb->m_zOutLpfR); // ********** end early reflection generator, right // decrement base addr for next sample period nBase--; pReverb->m_nSin += pReverb->m_nSinIncrement; pReverb->m_nCos += pReverb->m_nCosIncrement; } // end for (i=0; i < nNumSamplesToAdd; i++) // store the most up to date version pReverb->m_nBaseIndex = nBase; return 0; } /* end Reverb */ /*---------------------------------------------------------------------------- * ReverbUpdateRoom *---------------------------------------------------------------------------- * Purpose: * Update the room's preset parameters as required * * Inputs: * * Outputs: * * * Side Effects: * - reverb paramters (fbk, fwd, etc) will be changed * - m_nCurrentRoom := m_nNextRoom *---------------------------------------------------------------------------- */ static int ReverbUpdateRoom(reverb_object_t *pReverb, bool fullUpdate) { int temp; int i; int maxSamples; int earlyDelay; int earlyGain; reverb_preset_t *pPreset = &pReverb->m_sPreset.m_sPreset[pReverb->m_nNextRoom]; if (fullUpdate) { pReverb->m_nRvbLpfFwd = pPreset->m_nRvbLpfFwd; pReverb->m_nRvbLpfFbk = pPreset->m_nRvbLpfFbk; pReverb->m_nEarlyGain = pPreset->m_nEarlyGain; //stored as time based, convert to sample based pReverb->m_nLateGain = pPreset->m_nLateGain; pReverb->m_nRoomLpfFbk = pPreset->m_nRoomLpfFbk; pReverb->m_nRoomLpfFwd = pPreset->m_nRoomLpfFwd; // set the early reflections gains earlyGain = pPreset->m_nEarlyGain; for (i = 0; i < REVERB_MAX_NUM_REFLECTIONS; i++) { pReverb->m_sEarlyL.m_nGain[i] = MULT_EG1_EG1(pPreset->m_sEarlyL.m_nGain[i],earlyGain); pReverb->m_sEarlyR.m_nGain[i] = MULT_EG1_EG1(pPreset->m_sEarlyR.m_nGain[i],earlyGain); } pReverb->m_nMaxExcursion = pPreset->m_nMaxExcursion; pReverb->m_sAp0.m_nApGain = pPreset->m_nAp0_ApGain; pReverb->m_sAp1.m_nApGain = pPreset->m_nAp1_ApGain; // set the early reflections delay earlyDelay = ((int) pPreset->m_nEarlyDelay * pReverb->m_nSamplingRate) >> 16; pReverb->m_nEarlyDelay = earlyDelay; maxSamples = (int32_t) (MAX_EARLY_TIME * pReverb->m_nSamplingRate) >> 16; for (i = 0; i < REVERB_MAX_NUM_REFLECTIONS; i++) { //stored as time based, convert to sample based temp = earlyDelay + (((int) pPreset->m_sEarlyL.m_zDelay[i] * pReverb->m_nSamplingRate) >> 16); if (temp > maxSamples) temp = maxSamples; pReverb->m_sEarlyL.m_zDelay[i] = pReverb->m_nEarly0in + temp; //stored as time based, convert to sample based temp = earlyDelay + (((int) pPreset->m_sEarlyR.m_zDelay[i] * pReverb->m_nSamplingRate) >> 16); if (temp > maxSamples) temp = maxSamples; pReverb->m_sEarlyR.m_zDelay[i] = pReverb->m_nEarly1in + temp; } maxSamples = (int32_t) (MAX_DELAY_TIME * pReverb->m_nSamplingRate) >> 16; //stored as time based, convert to sample based /*lint -e{702} shift for performance */ temp = (pPreset->m_nLateDelay * pReverb->m_nSamplingRate) >> 16; if ((temp + pReverb->m_nMaxExcursion) > maxSamples) { temp = maxSamples - pReverb->m_nMaxExcursion; } temp -= pReverb->m_nLateDelay; pReverb->m_nDelay0Out += temp; pReverb->m_nDelay1Out += temp; pReverb->m_nLateDelay += temp; maxSamples = (int32_t) (MAX_AP_TIME * pReverb->m_nSamplingRate) >> 16; //stored as time based, convert to absolute sample value temp = pPreset->m_nAp0_ApOut; /*lint -e{702} shift for performance */ temp = (temp * pReverb->m_nSamplingRate) >> 16; if (temp > maxSamples) temp = maxSamples; pReverb->m_sAp0.m_zApOut = (uint16_t) (pReverb->m_sAp0.m_zApIn + temp); //stored as time based, convert to absolute sample value temp = pPreset->m_nAp1_ApOut; /*lint -e{702} shift for performance */ temp = (temp * pReverb->m_nSamplingRate) >> 16; if (temp > maxSamples) temp = maxSamples; pReverb->m_sAp1.m_zApOut = (uint16_t) (pReverb->m_sAp1.m_zApIn + temp); //gpsReverbObject->m_sAp1.m_zApOut = pPreset->m_nAp1_ApOut; } //stored as time based, convert to sample based temp = pPreset->m_nXfadeInterval; /*lint -e{702} shift for performance */ temp = (temp * pReverb->m_nSamplingRate) >> 16; pReverb->m_nXfadeInterval = (uint16_t) temp; //gsReverbObject.m_nXfadeInterval = pPreset->m_nXfadeInterval; pReverb->m_nXfadeCounter = pReverb->m_nXfadeInterval + 1; // force update on first iteration pReverb->m_nCurrentRoom = pReverb->m_nNextRoom; return 0; } /* end ReverbUpdateRoom */ /*---------------------------------------------------------------------------- * ReverbReadInPresets() *---------------------------------------------------------------------------- * Purpose: sets global reverb preset bank to defaults * * Inputs: * * Outputs: * *---------------------------------------------------------------------------- */ static int ReverbReadInPresets(reverb_object_t *pReverb) { int preset; // this is for test only. OpenSL ES presets are mapped to 4 presets. // REVERB_PRESET_NONE is mapped to bypass for (preset = 0; preset < REVERB_NUM_PRESETS; preset++) { reverb_preset_t *pPreset = &pReverb->m_sPreset.m_sPreset[preset]; switch (preset + 1) { case REVERB_PRESET_PLATE: case REVERB_PRESET_SMALLROOM: pPreset->m_nRvbLpfFbk = 5077; pPreset->m_nRvbLpfFwd = 11076; pPreset->m_nEarlyGain = 27690; pPreset->m_nEarlyDelay = 1311; pPreset->m_nLateGain = 8191; pPreset->m_nLateDelay = 3932; pPreset->m_nRoomLpfFbk = 3692; pPreset->m_nRoomLpfFwd = 20474; pPreset->m_sEarlyL.m_zDelay[0] = 1376; pPreset->m_sEarlyL.m_nGain[0] = 22152; pPreset->m_sEarlyL.m_zDelay[1] = 1462; pPreset->m_sEarlyL.m_nGain[1] = 17537; pPreset->m_sEarlyL.m_zDelay[2] = 0; pPreset->m_sEarlyL.m_nGain[2] = 14768; pPreset->m_sEarlyL.m_zDelay[3] = 1835; pPreset->m_sEarlyL.m_nGain[3] = 14307; pPreset->m_sEarlyL.m_zDelay[4] = 0; pPreset->m_sEarlyL.m_nGain[4] = 13384; pPreset->m_sEarlyR.m_zDelay[0] = 721; pPreset->m_sEarlyR.m_nGain[0] = 20306; pPreset->m_sEarlyR.m_zDelay[1] = 2621; pPreset->m_sEarlyR.m_nGain[1] = 17537; pPreset->m_sEarlyR.m_zDelay[2] = 0; pPreset->m_sEarlyR.m_nGain[2] = 14768; pPreset->m_sEarlyR.m_zDelay[3] = 0; pPreset->m_sEarlyR.m_nGain[3] = 16153; pPreset->m_sEarlyR.m_zDelay[4] = 0; pPreset->m_sEarlyR.m_nGain[4] = 13384; pPreset->m_nMaxExcursion = 127; pPreset->m_nXfadeInterval = 6470; //6483; pPreset->m_nAp0_ApGain = 14768; pPreset->m_nAp0_ApOut = 792; pPreset->m_nAp1_ApGain = 14777; pPreset->m_nAp1_ApOut = 1191; pPreset->m_rfu4 = 0; pPreset->m_rfu5 = 0; pPreset->m_rfu6 = 0; pPreset->m_rfu7 = 0; pPreset->m_rfu8 = 0; pPreset->m_rfu9 = 0; pPreset->m_rfu10 = 0; break; case REVERB_PRESET_MEDIUMROOM: case REVERB_PRESET_LARGEROOM: pPreset->m_nRvbLpfFbk = 5077; pPreset->m_nRvbLpfFwd = 12922; pPreset->m_nEarlyGain = 27690; pPreset->m_nEarlyDelay = 1311; pPreset->m_nLateGain = 8191; pPreset->m_nLateDelay = 3932; pPreset->m_nRoomLpfFbk = 3692; pPreset->m_nRoomLpfFwd = 21703; pPreset->m_sEarlyL.m_zDelay[0] = 1376; pPreset->m_sEarlyL.m_nGain[0] = 22152; pPreset->m_sEarlyL.m_zDelay[1] = 1462; pPreset->m_sEarlyL.m_nGain[1] = 17537; pPreset->m_sEarlyL.m_zDelay[2] = 0; pPreset->m_sEarlyL.m_nGain[2] = 14768; pPreset->m_sEarlyL.m_zDelay[3] = 1835; pPreset->m_sEarlyL.m_nGain[3] = 14307; pPreset->m_sEarlyL.m_zDelay[4] = 0; pPreset->m_sEarlyL.m_nGain[4] = 13384; pPreset->m_sEarlyR.m_zDelay[0] = 721; pPreset->m_sEarlyR.m_nGain[0] = 20306; pPreset->m_sEarlyR.m_zDelay[1] = 2621; pPreset->m_sEarlyR.m_nGain[1] = 17537; pPreset->m_sEarlyR.m_zDelay[2] = 0; pPreset->m_sEarlyR.m_nGain[2] = 14768; pPreset->m_sEarlyR.m_zDelay[3] = 0; pPreset->m_sEarlyR.m_nGain[3] = 16153; pPreset->m_sEarlyR.m_zDelay[4] = 0; pPreset->m_sEarlyR.m_nGain[4] = 13384; pPreset->m_nMaxExcursion = 127; pPreset->m_nXfadeInterval = 6449; pPreset->m_nAp0_ApGain = 15691; pPreset->m_nAp0_ApOut = 774; pPreset->m_nAp1_ApGain = 16317; pPreset->m_nAp1_ApOut = 1155; pPreset->m_rfu4 = 0; pPreset->m_rfu5 = 0; pPreset->m_rfu6 = 0; pPreset->m_rfu7 = 0; pPreset->m_rfu8 = 0; pPreset->m_rfu9 = 0; pPreset->m_rfu10 = 0; break; case REVERB_PRESET_MEDIUMHALL: pPreset->m_nRvbLpfFbk = 6461; pPreset->m_nRvbLpfFwd = 14307; pPreset->m_nEarlyGain = 27690; pPreset->m_nEarlyDelay = 1311; pPreset->m_nLateGain = 8191; pPreset->m_nLateDelay = 3932; pPreset->m_nRoomLpfFbk = 3692; pPreset->m_nRoomLpfFwd = 24569; pPreset->m_sEarlyL.m_zDelay[0] = 1376; pPreset->m_sEarlyL.m_nGain[0] = 22152; pPreset->m_sEarlyL.m_zDelay[1] = 1462; pPreset->m_sEarlyL.m_nGain[1] = 17537; pPreset->m_sEarlyL.m_zDelay[2] = 0; pPreset->m_sEarlyL.m_nGain[2] = 14768; pPreset->m_sEarlyL.m_zDelay[3] = 1835; pPreset->m_sEarlyL.m_nGain[3] = 14307; pPreset->m_sEarlyL.m_zDelay[4] = 0; pPreset->m_sEarlyL.m_nGain[4] = 13384; pPreset->m_sEarlyR.m_zDelay[0] = 721; pPreset->m_sEarlyR.m_nGain[0] = 20306; pPreset->m_sEarlyR.m_zDelay[1] = 2621; pPreset->m_sEarlyR.m_nGain[1] = 17537; pPreset->m_sEarlyR.m_zDelay[2] = 0; pPreset->m_sEarlyR.m_nGain[2] = 14768; pPreset->m_sEarlyR.m_zDelay[3] = 0; pPreset->m_sEarlyR.m_nGain[3] = 16153; pPreset->m_sEarlyR.m_zDelay[4] = 0; pPreset->m_sEarlyR.m_nGain[4] = 13384; pPreset->m_nMaxExcursion = 127; pPreset->m_nXfadeInterval = 6391; pPreset->m_nAp0_ApGain = 15230; pPreset->m_nAp0_ApOut = 708; pPreset->m_nAp1_ApGain = 15547; pPreset->m_nAp1_ApOut = 1023; pPreset->m_rfu4 = 0; pPreset->m_rfu5 = 0; pPreset->m_rfu6 = 0; pPreset->m_rfu7 = 0; pPreset->m_rfu8 = 0; pPreset->m_rfu9 = 0; pPreset->m_rfu10 = 0; break; case REVERB_PRESET_LARGEHALL: pPreset->m_nRvbLpfFbk = 8307; pPreset->m_nRvbLpfFwd = 14768; pPreset->m_nEarlyGain = 27690; pPreset->m_nEarlyDelay = 1311; pPreset->m_nLateGain = 8191; pPreset->m_nLateDelay = 3932; pPreset->m_nRoomLpfFbk = 3692; pPreset->m_nRoomLpfFwd = 24569; pPreset->m_sEarlyL.m_zDelay[0] = 1376; pPreset->m_sEarlyL.m_nGain[0] = 22152; pPreset->m_sEarlyL.m_zDelay[1] = 2163; pPreset->m_sEarlyL.m_nGain[1] = 17537; pPreset->m_sEarlyL.m_zDelay[2] = 0; pPreset->m_sEarlyL.m_nGain[2] = 14768; pPreset->m_sEarlyL.m_zDelay[3] = 1835; pPreset->m_sEarlyL.m_nGain[3] = 14307; pPreset->m_sEarlyL.m_zDelay[4] = 0; pPreset->m_sEarlyL.m_nGain[4] = 13384; pPreset->m_sEarlyR.m_zDelay[0] = 721; pPreset->m_sEarlyR.m_nGain[0] = 20306; pPreset->m_sEarlyR.m_zDelay[1] = 2621; pPreset->m_sEarlyR.m_nGain[1] = 17537; pPreset->m_sEarlyR.m_zDelay[2] = 0; pPreset->m_sEarlyR.m_nGain[2] = 14768; pPreset->m_sEarlyR.m_zDelay[3] = 0; pPreset->m_sEarlyR.m_nGain[3] = 16153; pPreset->m_sEarlyR.m_zDelay[4] = 0; pPreset->m_sEarlyR.m_nGain[4] = 13384; pPreset->m_nMaxExcursion = 127; pPreset->m_nXfadeInterval = 6388; pPreset->m_nAp0_ApGain = 15691; pPreset->m_nAp0_ApOut = 711; pPreset->m_nAp1_ApGain = 16317; pPreset->m_nAp1_ApOut = 1029; pPreset->m_rfu4 = 0; pPreset->m_rfu5 = 0; pPreset->m_rfu6 = 0; pPreset->m_rfu7 = 0; pPreset->m_rfu8 = 0; pPreset->m_rfu9 = 0; pPreset->m_rfu10 = 0; break; } } return 0; } audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { .tag = AUDIO_EFFECT_LIBRARY_TAG, .version = EFFECT_LIBRARY_API_VERSION, .name = "Test Equalizer Library", .implementor = "The Android Open Source Project", .create_effect = EffectCreate, .release_effect = EffectRelease, .get_descriptor = EffectGetDescriptor, };