| /* |
| * 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 <cutils/log.h> |
| #include <stdlib.h> |
| #include <string.h> |
| #include <stdbool.h> |
| #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, size_t *pSize, |
| void *pValue) { |
| int32_t *pValue32; |
| int16_t *pValue16; |
| t_reverb_settings *pProperties; |
| int32_t i; |
| int32_t temp; |
| int32_t temp2; |
| size_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, size_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; |
| size_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, |
| }; |
