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/*
* Copyright (C) 2012 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 "EffectDownmix"
//#define LOG_NDEBUG 0
#include <log/log.h>
#include "EffectDownmix.h"
#include <math.h>
// Do not submit with DOWNMIX_TEST_CHANNEL_INDEX defined, strictly for testing
//#define DOWNMIX_TEST_CHANNEL_INDEX 0
// Do not submit with DOWNMIX_ALWAYS_USE_GENERIC_DOWNMIXER defined, strictly for testing
//#define DOWNMIX_ALWAYS_USE_GENERIC_DOWNMIXER 0
#define MINUS_3_DB_IN_FLOAT M_SQRT1_2 // -3dB = 0.70710678
typedef enum {
DOWNMIX_STATE_UNINITIALIZED,
DOWNMIX_STATE_INITIALIZED,
DOWNMIX_STATE_ACTIVE,
} downmix_state_t;
/* parameters for each downmixer */
typedef struct {
downmix_state_t state;
downmix_type_t type;
bool apply_volume_correction;
uint8_t input_channel_count;
} downmix_object_t;
typedef struct downmix_module_s {
const struct effect_interface_s *itfe;
effect_config_t config;
downmix_object_t context;
} downmix_module_t;
// Audio Effect API
static int32_t DownmixLib_Create(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle);
static int32_t DownmixLib_Release(effect_handle_t handle);
static int32_t DownmixLib_GetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor);
static int32_t Downmix_Process(effect_handle_t self,
audio_buffer_t *inBuffer,
audio_buffer_t *outBuffer);
static int32_t Downmix_Command(effect_handle_t self,
uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t *replySize,
void *pReplyData);
static int32_t Downmix_GetDescriptor(effect_handle_t self,
effect_descriptor_t *pDescriptor);
// Internal methods
static int Downmix_Init(downmix_module_t *pDwmModule);
static int Downmix_Configure(downmix_module_t *pDwmModule, effect_config_t *pConfig, bool init);
static int Downmix_Reset(downmix_object_t *pDownmixer, bool init);
static int Downmix_setParameter(
downmix_object_t *pDownmixer, int32_t param, uint32_t size, void *pValue);
static int Downmix_getParameter(
downmix_object_t *pDownmixer, int32_t param, uint32_t *pSize, void *pValue);
static void Downmix_foldFromQuad(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
static void Downmix_foldFrom5Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
static void Downmix_foldFrom7Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate);
static bool Downmix_foldGeneric(
uint32_t mask, float *pSrc, float *pDst, size_t numFrames, bool accumulate);
// effect_handle_t interface implementation for downmix effect
const struct effect_interface_s gDownmixInterface = {
Downmix_Process,
Downmix_Command,
Downmix_GetDescriptor,
NULL /* no process_reverse function, no reference stream needed */
};
// This is the only symbol that needs to be exported
__attribute__ ((visibility ("default")))
audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = {
.tag = AUDIO_EFFECT_LIBRARY_TAG,
.version = EFFECT_LIBRARY_API_VERSION,
.name = "Downmix Library",
.implementor = "The Android Open Source Project",
.create_effect = DownmixLib_Create,
.release_effect = DownmixLib_Release,
.get_descriptor = DownmixLib_GetDescriptor,
};
// AOSP insert downmix UUID: 93f04452-e4fe-41cc-91f9-e475b6d1d69f
static const effect_descriptor_t gDownmixDescriptor = {
EFFECT_UIID_DOWNMIX__, //type
{0x93f04452, 0xe4fe, 0x41cc, 0x91f9, {0xe4, 0x75, 0xb6, 0xd1, 0xd6, 0x9f}}, // uuid
EFFECT_CONTROL_API_VERSION,
EFFECT_FLAG_TYPE_INSERT | EFFECT_FLAG_INSERT_FIRST,
0, //FIXME what value should be reported? // cpu load
0, //FIXME what value should be reported? // memory usage
"Multichannel Downmix To Stereo", // human readable effect name
"The Android Open Source Project" // human readable effect implementor name
};
// gDescriptors contains pointers to all defined effect descriptor in this library
static const effect_descriptor_t * const gDescriptors[] = {
&gDownmixDescriptor
};
// number of effects in this library
const int kNbEffects = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
static inline float clamp_float(float value) {
return fmin(fmax(value, -1.f), 1.f);
}
/*----------------------------------------------------------------------------
* Test code
*--------------------------------------------------------------------------*/
#ifdef DOWNMIX_TEST_CHANNEL_INDEX
// strictly for testing, logs the indices of the channels for a given mask,
// uses the same code as Downmix_foldGeneric()
void Downmix_testIndexComputation(uint32_t mask) {
ALOGI("Testing index computation for %#x:", mask);
// check against unsupported channels
if (mask & kUnsupported) {
ALOGE("Unsupported channels (top or front left/right of center)");
return;
}
// verify has FL/FR
if ((mask & AUDIO_CHANNEL_OUT_STEREO) != AUDIO_CHANNEL_OUT_STEREO) {
ALOGE("Front channels must be present");
return;
}
// verify uses SIDE as a pair (ok if not using SIDE at all)
bool hasSides = false;
if ((mask & kSides) != 0) {
if ((mask & kSides) != kSides) {
ALOGE("Side channels must be used as a pair");
return;
}
hasSides = true;
}
// verify uses BACK as a pair (ok if not using BACK at all)
bool hasBacks = false;
if ((mask & kBacks) != 0) {
if ((mask & kBacks) != kBacks) {
ALOGE("Back channels must be used as a pair");
return;
}
hasBacks = true;
}
const int numChan = audio_channel_count_from_out_mask(mask);
const bool hasFC = ((mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) == AUDIO_CHANNEL_OUT_FRONT_CENTER);
const bool hasLFE =
((mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY);
const bool hasBC = ((mask & AUDIO_CHANNEL_OUT_BACK_CENTER) == AUDIO_CHANNEL_OUT_BACK_CENTER);
// compute at what index each channel is: samples will be in the following order:
// FL FR FC LFE BL BR BC SL SR
// when a channel is not present, its index is set to the same as the index of the preceding
// channel
const int indexFC = hasFC ? 2 : 1; // front center
const int indexLFE = hasLFE ? indexFC + 1 : indexFC; // low frequency
const int indexBL = hasBacks ? indexLFE + 1 : indexLFE; // back left
const int indexBR = hasBacks ? indexBL + 1 : indexBL; // back right
const int indexBC = hasBC ? indexBR + 1 : indexBR; // back center
const int indexSL = hasSides ? indexBC + 1 : indexBC; // side left
const int indexSR = hasSides ? indexSL + 1 : indexSL; // side right
ALOGI(" FL FR FC LFE BL BR BC SL SR");
ALOGI(" %d %d %d %d %d %d %d %d %d",
0, 1, indexFC, indexLFE, indexBL, indexBR, indexBC, indexSL, indexSR);
}
#endif
static bool Downmix_validChannelMask(uint32_t mask)
{
if (!mask) {
return false;
}
// check against unsupported channels
if (mask & ~AUDIO_CHANNEL_OUT_22POINT2) {
ALOGE("Unsupported channels in %u", mask & ~AUDIO_CHANNEL_OUT_22POINT2);
return false;
}
return true;
}
/*----------------------------------------------------------------------------
* Effect API implementation
*--------------------------------------------------------------------------*/
/*--- Effect Library Interface Implementation ---*/
static int32_t DownmixLib_Create(const effect_uuid_t *uuid,
int32_t /* sessionId */,
int32_t /* ioId */,
effect_handle_t *pHandle) {
int ret;
int i;
downmix_module_t *module;
const effect_descriptor_t *desc;
ALOGV("DownmixLib_Create()");
#ifdef DOWNMIX_TEST_CHANNEL_INDEX
// should work (won't log an error)
ALOGI("DOWNMIX_TEST_CHANNEL_INDEX: should work:");
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT |
AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_BACK_CENTER);
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_QUAD_SIDE | AUDIO_CHANNEL_OUT_QUAD_BACK);
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_5POINT1_SIDE | AUDIO_CHANNEL_OUT_BACK_CENTER);
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_5POINT1_BACK | AUDIO_CHANNEL_OUT_BACK_CENTER);
// shouldn't work (will log an error, won't display channel indices)
ALOGI("DOWNMIX_TEST_CHANNEL_INDEX: should NOT work:");
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT |
AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_BACK_LEFT);
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_FRONT_LEFT | AUDIO_CHANNEL_OUT_FRONT_RIGHT |
AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_SIDE_LEFT);
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_FRONT_LEFT |
AUDIO_CHANNEL_OUT_BACK_LEFT | AUDIO_CHANNEL_OUT_BACK_RIGHT);
Downmix_testIndexComputation(AUDIO_CHANNEL_OUT_FRONT_LEFT |
AUDIO_CHANNEL_OUT_SIDE_LEFT | AUDIO_CHANNEL_OUT_SIDE_RIGHT);
#endif
if (pHandle == NULL || uuid == NULL) {
return -EINVAL;
}
for (i = 0 ; i < kNbEffects ; i++) {
desc = gDescriptors[i];
if (memcmp(uuid, &desc->uuid, sizeof(effect_uuid_t)) == 0) {
break;
}
}
if (i == kNbEffects) {
return -ENOENT;
}
module = new downmix_module_t{};
module->itfe = &gDownmixInterface;
module->context.state = DOWNMIX_STATE_UNINITIALIZED;
ret = Downmix_Init(module);
if (ret < 0) {
ALOGW("DownmixLib_Create() init failed");
free(module);
return ret;
}
*pHandle = (effect_handle_t) module;
ALOGV("DownmixLib_Create() %p , size %zu", module, sizeof(downmix_module_t));
return 0;
}
static int32_t DownmixLib_Release(effect_handle_t handle) {
downmix_module_t *pDwmModule = (downmix_module_t *)handle;
ALOGV("DownmixLib_Release() %p", handle);
if (handle == NULL) {
return -EINVAL;
}
pDwmModule->context.state = DOWNMIX_STATE_UNINITIALIZED;
delete pDwmModule;
return 0;
}
static int32_t DownmixLib_GetDescriptor(
const effect_uuid_t *uuid, effect_descriptor_t *pDescriptor) {
ALOGV("DownmixLib_GetDescriptor()");
int i;
if (pDescriptor == NULL || uuid == NULL){
ALOGE("DownmixLib_Create() called with NULL pointer");
return -EINVAL;
}
ALOGV("DownmixLib_GetDescriptor() nb effects=%d", kNbEffects);
for (i = 0; i < kNbEffects; i++) {
ALOGV("DownmixLib_GetDescriptor() i=%d", i);
if (memcmp(uuid, &gDescriptors[i]->uuid, sizeof(effect_uuid_t)) == 0) {
memcpy(pDescriptor, gDescriptors[i], sizeof(effect_descriptor_t));
ALOGV("EffectGetDescriptor - UUID matched downmix type %d, UUID = %#x",
i, gDescriptors[i]->uuid.timeLow);
return 0;
}
}
return -EINVAL;
}
/*--- Effect Control Interface Implementation ---*/
static int32_t Downmix_Process(effect_handle_t self,
audio_buffer_t *inBuffer, audio_buffer_t *outBuffer) {
downmix_object_t *pDownmixer;
float *pSrc, *pDst;
downmix_module_t *pDwmModule = (downmix_module_t *)self;
if (pDwmModule == NULL) {
return -EINVAL;
}
if (inBuffer == NULL || inBuffer->raw == NULL ||
outBuffer == NULL || outBuffer->raw == NULL ||
inBuffer->frameCount != outBuffer->frameCount) {
return -EINVAL;
}
pDownmixer = (downmix_object_t*) &pDwmModule->context;
if (pDownmixer->state == DOWNMIX_STATE_UNINITIALIZED) {
ALOGE("Downmix_Process error: trying to use an uninitialized downmixer");
return -EINVAL;
} else if (pDownmixer->state == DOWNMIX_STATE_INITIALIZED) {
ALOGE("Downmix_Process error: trying to use a non-configured downmixer");
return -ENODATA;
}
pSrc = inBuffer->f32;
pDst = outBuffer->f32;
size_t numFrames = outBuffer->frameCount;
const bool accumulate =
(pDwmModule->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE);
const uint32_t downmixInputChannelMask = pDwmModule->config.inputCfg.channels;
switch(pDownmixer->type) {
case DOWNMIX_TYPE_STRIP:
if (accumulate) {
while (numFrames) {
pDst[0] = clamp_float(pDst[0] + pSrc[0]);
pDst[1] = clamp_float(pDst[1] + pSrc[1]);
pSrc += pDownmixer->input_channel_count;
pDst += 2;
numFrames--;
}
} else {
while (numFrames) {
pDst[0] = pSrc[0];
pDst[1] = pSrc[1];
pSrc += pDownmixer->input_channel_count;
pDst += 2;
numFrames--;
}
}
break;
case DOWNMIX_TYPE_FOLD:
#ifdef DOWNMIX_ALWAYS_USE_GENERIC_DOWNMIXER
// bypass the optimized downmix routines for the common formats
if (!Downmix_foldGeneric(
downmixInputChannelMask, pSrc, pDst, numFrames, accumulate)) {
ALOGE("Multichannel configuration %#x is not supported",
downmixInputChannelMask);
return -EINVAL;
}
break;
#endif
// optimize for the common formats
switch (downmixInputChannelMask) {
case AUDIO_CHANNEL_OUT_QUAD_BACK:
case AUDIO_CHANNEL_OUT_QUAD_SIDE:
Downmix_foldFromQuad(pSrc, pDst, numFrames, accumulate);
break;
case AUDIO_CHANNEL_OUT_5POINT1_BACK:
case AUDIO_CHANNEL_OUT_5POINT1_SIDE:
Downmix_foldFrom5Point1(pSrc, pDst, numFrames, accumulate);
break;
case AUDIO_CHANNEL_OUT_7POINT1:
Downmix_foldFrom7Point1(pSrc, pDst, numFrames, accumulate);
break;
default:
if (!Downmix_foldGeneric(
downmixInputChannelMask, pSrc, pDst, numFrames, accumulate)) {
ALOGE("Multichannel configuration %#x is not supported",
downmixInputChannelMask);
return -EINVAL;
}
break;
}
break;
default:
return -EINVAL;
}
return 0;
}
static int32_t Downmix_Command(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize,
void *pCmdData, uint32_t *replySize, void *pReplyData) {
downmix_module_t *pDwmModule = (downmix_module_t *) self;
downmix_object_t *pDownmixer;
if (pDwmModule == NULL || pDwmModule->context.state == DOWNMIX_STATE_UNINITIALIZED) {
return -EINVAL;
}
pDownmixer = (downmix_object_t*) &pDwmModule->context;
ALOGV("Downmix_Command command %u cmdSize %u", cmdCode, cmdSize);
switch (cmdCode) {
case EFFECT_CMD_INIT:
if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
*(int *) pReplyData = Downmix_Init(pDwmModule);
break;
case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
|| pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
*(int *) pReplyData = Downmix_Configure(pDwmModule,
(effect_config_t *)pCmdData, false);
break;
case EFFECT_CMD_RESET:
Downmix_Reset(pDownmixer, false);
break;
case EFFECT_CMD_GET_PARAM: {
ALOGV("Downmix_Command EFFECT_CMD_GET_PARAM pCmdData %p, *replySize %u, pReplyData: %p",
pCmdData, *replySize, pReplyData);
if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) ||
pReplyData == NULL || replySize == NULL ||
*replySize < (int) sizeof(effect_param_t) + 2 * sizeof(int32_t)) {
return -EINVAL;
}
effect_param_t *rep = (effect_param_t *) pReplyData;
memcpy(pReplyData, pCmdData, sizeof(effect_param_t) + sizeof(int32_t));
ALOGV("Downmix_Command EFFECT_CMD_GET_PARAM param %d, replySize %u",
*(int32_t *)rep->data, rep->vsize);
rep->status = Downmix_getParameter(pDownmixer, *(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("Downmix_Command EFFECT_CMD_SET_PARAM cmdSize %d pCmdData %p, *replySize %u"
", pReplyData %p", cmdSize, pCmdData, *replySize, pReplyData);
if (pCmdData == NULL || (cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)))
|| pReplyData == NULL || replySize == NULL || *replySize != (int)sizeof(int32_t)) {
return -EINVAL;
}
effect_param_t *cmd = (effect_param_t *) pCmdData;
if (cmd->psize != sizeof(int32_t)) {
android_errorWriteLog(0x534e4554, "63662938");
return -EINVAL;
}
*(int *)pReplyData = Downmix_setParameter(pDownmixer, *(int32_t *)cmd->data,
cmd->vsize, cmd->data + sizeof(int32_t));
break;
}
case EFFECT_CMD_SET_PARAM_DEFERRED:
//FIXME implement
ALOGW("Downmix_Command command EFFECT_CMD_SET_PARAM_DEFERRED not supported, FIXME");
break;
case EFFECT_CMD_SET_PARAM_COMMIT:
//FIXME implement
ALOGW("Downmix_Command command EFFECT_CMD_SET_PARAM_COMMIT not supported, FIXME");
break;
case EFFECT_CMD_ENABLE:
if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
if (pDownmixer->state != DOWNMIX_STATE_INITIALIZED) {
return -ENOSYS;
}
pDownmixer->state = DOWNMIX_STATE_ACTIVE;
ALOGV("EFFECT_CMD_ENABLE() OK");
*(int *)pReplyData = 0;
break;
case EFFECT_CMD_DISABLE:
if (pReplyData == NULL || replySize == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
if (pDownmixer->state != DOWNMIX_STATE_ACTIVE) {
return -ENOSYS;
}
pDownmixer->state = DOWNMIX_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;
}
// FIXME change type if playing on headset vs speaker
ALOGV("Downmix_Command EFFECT_CMD_SET_DEVICE: %#x", *(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;
}
// FIXME change volume
ALOGW("Downmix_Command command EFFECT_CMD_SET_VOLUME not supported, FIXME");
float left = (float)(*(uint32_t *)pCmdData) / (1 << 24);
float right = (float)(*((uint32_t *)pCmdData + 1)) / (1 << 24);
ALOGV("Downmix_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("Downmix_Command EFFECT_CMD_SET_AUDIO_MODE: %u", *(uint32_t *)pCmdData);
break;
case EFFECT_CMD_SET_CONFIG_REVERSE:
case EFFECT_CMD_SET_INPUT_DEVICE:
// these commands are ignored by a downmix effect
break;
default:
ALOGW("Downmix_Command invalid command %u", cmdCode);
return -EINVAL;
}
return 0;
}
static int32_t Downmix_GetDescriptor(effect_handle_t self, effect_descriptor_t *pDescriptor)
{
downmix_module_t *pDwnmxModule = (downmix_module_t *) self;
if (pDwnmxModule == NULL ||
pDwnmxModule->context.state == DOWNMIX_STATE_UNINITIALIZED) {
return -EINVAL;
}
memcpy(pDescriptor, &gDownmixDescriptor, sizeof(effect_descriptor_t));
return 0;
}
/*----------------------------------------------------------------------------
* Downmix internal functions
*--------------------------------------------------------------------------*/
/*----------------------------------------------------------------------------
* Downmix_Init()
*----------------------------------------------------------------------------
* Purpose:
* Initialize downmix context and apply default parameters
*
* Inputs:
* pDwmModule pointer to downmix effect module
*
* Outputs:
*
* Returns:
* 0 indicates success
*
* Side Effects:
* updates:
* pDwmModule->context.type
* pDwmModule->context.apply_volume_correction
* pDwmModule->config.inputCfg
* pDwmModule->config.outputCfg
* pDwmModule->config.inputCfg.samplingRate
* pDwmModule->config.outputCfg.samplingRate
* pDwmModule->context.state
* doesn't set:
* pDwmModule->itfe
*
*----------------------------------------------------------------------------
*/
static int Downmix_Init(downmix_module_t *pDwmModule) {
ALOGV("Downmix_Init module %p", pDwmModule);
int ret = 0;
memset(&pDwmModule->context, 0, sizeof(downmix_object_t));
pDwmModule->config.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
pDwmModule->config.inputCfg.format = AUDIO_FORMAT_PCM_FLOAT;
pDwmModule->config.inputCfg.channels = AUDIO_CHANNEL_OUT_7POINT1;
pDwmModule->config.inputCfg.bufferProvider.getBuffer = NULL;
pDwmModule->config.inputCfg.bufferProvider.releaseBuffer = NULL;
pDwmModule->config.inputCfg.bufferProvider.cookie = NULL;
pDwmModule->config.inputCfg.mask = EFFECT_CONFIG_ALL;
pDwmModule->config.inputCfg.samplingRate = 44100;
pDwmModule->config.outputCfg.samplingRate = pDwmModule->config.inputCfg.samplingRate;
// set a default value for the access mode, but should be overwritten by caller
pDwmModule->config.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
pDwmModule->config.outputCfg.format = AUDIO_FORMAT_PCM_FLOAT;
pDwmModule->config.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
pDwmModule->config.outputCfg.bufferProvider.getBuffer = NULL;
pDwmModule->config.outputCfg.bufferProvider.releaseBuffer = NULL;
pDwmModule->config.outputCfg.bufferProvider.cookie = NULL;
pDwmModule->config.outputCfg.mask = EFFECT_CONFIG_ALL;
ret = Downmix_Configure(pDwmModule, &pDwmModule->config, true);
if (ret != 0) {
ALOGV("Downmix_Init error %d on module %p", ret, pDwmModule);
} else {
pDwmModule->context.state = DOWNMIX_STATE_INITIALIZED;
}
return ret;
}
/*----------------------------------------------------------------------------
* Downmix_Configure()
*----------------------------------------------------------------------------
* Purpose:
* Set input and output audio configuration.
*
* Inputs:
* pDwmModule pointer to downmix effect module
* pConfig pointer to effect_config_t structure containing input
* and output audio parameters configuration
* init true if called from init function
*
* Outputs:
*
* Returns:
* 0 indicates success
*
* Side Effects:
*
*----------------------------------------------------------------------------
*/
static int Downmix_Configure(downmix_module_t *pDwmModule, effect_config_t *pConfig, bool init) {
downmix_object_t *pDownmixer = &pDwmModule->context;
// Check configuration compatibility with build options, and effect capabilities
if (pConfig->inputCfg.samplingRate != pConfig->outputCfg.samplingRate
|| pConfig->outputCfg.channels != AUDIO_CHANNEL_OUT_STEREO
|| pConfig->inputCfg.format != AUDIO_FORMAT_PCM_FLOAT
|| pConfig->outputCfg.format != AUDIO_FORMAT_PCM_FLOAT) {
ALOGE("Downmix_Configure error: invalid config");
return -EINVAL;
}
if (&pDwmModule->config != pConfig) {
memcpy(&pDwmModule->config, pConfig, sizeof(effect_config_t));
}
if (init) {
pDownmixer->type = DOWNMIX_TYPE_FOLD;
pDownmixer->apply_volume_correction = false;
pDownmixer->input_channel_count = 8; // matches default input of AUDIO_CHANNEL_OUT_7POINT1
} else {
// when configuring the effect, do not allow a blank or unsupported channel mask
if (!Downmix_validChannelMask(pConfig->inputCfg.channels)) {
ALOGE("Downmix_Configure error: input channel mask(0x%x) not supported",
pConfig->inputCfg.channels);
return -EINVAL;
}
pDownmixer->input_channel_count =
audio_channel_count_from_out_mask(pConfig->inputCfg.channels);
}
Downmix_Reset(pDownmixer, init);
return 0;
}
/*----------------------------------------------------------------------------
* Downmix_Reset()
*----------------------------------------------------------------------------
* Purpose:
* Reset internal states.
*
* Inputs:
* pDownmixer pointer to downmix context
* init true if called from init function
*
* Outputs:
*
* Returns:
* 0 indicates success
*
* Side Effects:
*
*----------------------------------------------------------------------------
*/
static int Downmix_Reset(downmix_object_t* /* pDownmixer */, bool /* init */) {
// nothing to do here
return 0;
}
/*----------------------------------------------------------------------------
* Downmix_setParameter()
*----------------------------------------------------------------------------
* Purpose:
* Set a Downmix parameter
*
* Inputs:
* pDownmixer handle to instance data
* param parameter
* pValue pointer to parameter value
* size value size
*
* Outputs:
*
* Returns:
* 0 indicates success
*
* Side Effects:
*
*----------------------------------------------------------------------------
*/
static int Downmix_setParameter(
downmix_object_t *pDownmixer, int32_t param, uint32_t size, void *pValue) {
int16_t value16;
ALOGV("Downmix_setParameter, context %p, param %d, value16 %d, value32 %d",
pDownmixer, param, *(int16_t *)pValue, *(int32_t *)pValue);
switch (param) {
case DOWNMIX_PARAM_TYPE:
if (size != sizeof(downmix_type_t)) {
ALOGE("Downmix_setParameter(DOWNMIX_PARAM_TYPE) invalid size %u, should be %zu",
size, sizeof(downmix_type_t));
return -EINVAL;
}
value16 = *(int16_t *)pValue;
ALOGV("set DOWNMIX_PARAM_TYPE, type %d", value16);
if (!((value16 > DOWNMIX_TYPE_INVALID) && (value16 <= DOWNMIX_TYPE_LAST))) {
ALOGE("Downmix_setParameter invalid DOWNMIX_PARAM_TYPE value %d", value16);
return -EINVAL;
} else {
pDownmixer->type = (downmix_type_t) value16;
break;
default:
ALOGE("Downmix_setParameter unknown parameter %d", param);
return -EINVAL;
}
}
return 0;
} /* end Downmix_setParameter */
/*----------------------------------------------------------------------------
* Downmix_getParameter()
*----------------------------------------------------------------------------
* Purpose:
* Get a Downmix parameter
*
* Inputs:
* pDownmixer 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
*
* Returns:
* 0 indicates success
*
* Side Effects:
*
*----------------------------------------------------------------------------
*/
static int Downmix_getParameter(
downmix_object_t *pDownmixer, int32_t param, uint32_t *pSize, void *pValue) {
int16_t *pValue16;
switch (param) {
case DOWNMIX_PARAM_TYPE:
if (*pSize < sizeof(int16_t)) {
ALOGE("Downmix_getParameter invalid parameter size %u for DOWNMIX_PARAM_TYPE", *pSize);
return -EINVAL;
}
pValue16 = (int16_t *)pValue;
*pValue16 = (int16_t) pDownmixer->type;
*pSize = sizeof(int16_t);
ALOGV("Downmix_getParameter DOWNMIX_PARAM_TYPE is %d", *pValue16);
break;
default:
ALOGE("Downmix_getParameter unknown parameter %d", param);
return -EINVAL;
}
return 0;
} /* end Downmix_getParameter */
/*----------------------------------------------------------------------------
* Downmix_foldFromQuad()
*----------------------------------------------------------------------------
* Purpose:
* downmix a quad signal to stereo
*
* Inputs:
* pSrc quad audio samples to downmix
* numFrames the number of quad frames to downmix
* accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
* or overwrite pDst (when false)
*
* Outputs:
* pDst downmixed stereo audio samples
*
*----------------------------------------------------------------------------
*/
void Downmix_foldFromQuad(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
// sample at index 0 is FL
// sample at index 1 is FR
// sample at index 2 is RL
// sample at index 3 is RR
if (accumulate) {
while (numFrames) {
// FL + RL
pDst[0] = clamp_float(pDst[0] + ((pSrc[0] + pSrc[2]) / 2.0f));
// FR + RR
pDst[1] = clamp_float(pDst[1] + ((pSrc[1] + pSrc[3]) / 2.0f));
pSrc += 4;
pDst += 2;
numFrames--;
}
} else { // same code as above but without adding and clamping pDst[i] to itself
while (numFrames) {
// FL + RL
pDst[0] = clamp_float((pSrc[0] + pSrc[2]) / 2.0f);
// FR + RR
pDst[1] = clamp_float((pSrc[1] + pSrc[3]) / 2.0f);
pSrc += 4;
pDst += 2;
numFrames--;
}
}
}
/*----------------------------------------------------------------------------
* Downmix_foldFrom5Point1()
*----------------------------------------------------------------------------
* Purpose:
* downmix a 5.1 signal to stereo
*
* Inputs:
* pSrc 5.1 audio samples to downmix
* numFrames the number of 5.1 frames to downmix
* accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
* or overwrite pDst (when false)
*
* Outputs:
* pDst downmixed stereo audio samples
*
*----------------------------------------------------------------------------
*/
void Downmix_foldFrom5Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
float lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
// sample at index 0 is FL
// sample at index 1 is FR
// sample at index 2 is FC
// sample at index 3 is LFE
// sample at index 4 is RL
// sample at index 5 is RR
// code is mostly duplicated between the two values of accumulate to avoid repeating the test
// for every sample
if (accumulate) {
while (numFrames) {
// centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
+ (pSrc[3] * MINUS_3_DB_IN_FLOAT);
// FL + centerPlusLfeContrib + RL
lt = pSrc[0] + centerPlusLfeContrib + pSrc[4];
// FR + centerPlusLfeContrib + RR
rt = pSrc[1] + centerPlusLfeContrib + pSrc[5];
// accumulate in destination
pDst[0] = clamp_float(pDst[0] + (lt / 2.0f));
pDst[1] = clamp_float(pDst[1] + (rt / 2.0f));
pSrc += 6;
pDst += 2;
numFrames--;
}
} else { // same code as above but without adding and clamping pDst[i] to itself
while (numFrames) {
// centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
+ (pSrc[3] * MINUS_3_DB_IN_FLOAT);
// FL + centerPlusLfeContrib + RL
lt = pSrc[0] + centerPlusLfeContrib + pSrc[4];
// FR + centerPlusLfeContrib + RR
rt = pSrc[1] + centerPlusLfeContrib + pSrc[5];
// store in destination
pDst[0] = clamp_float(lt / 2.0f); // differs from when accumulate is true above
pDst[1] = clamp_float(rt / 2.0f); // differs from when accumulate is true above
pSrc += 6;
pDst += 2;
numFrames--;
}
}
}
/*----------------------------------------------------------------------------
* Downmix_foldFrom7Point1()
*----------------------------------------------------------------------------
* Purpose:
* downmix a 7.1 signal to stereo
*
* Inputs:
* pSrc 7.1 audio samples to downmix
* numFrames the number of 7.1 frames to downmix
* accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
* or overwrite pDst (when false)
*
* Outputs:
* pDst downmixed stereo audio samples
*
*----------------------------------------------------------------------------
*/
void Downmix_foldFrom7Point1(float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
float lt, rt, centerPlusLfeContrib; // samples in Q19.12 format
// sample at index 0 is FL
// sample at index 1 is FR
// sample at index 2 is FC
// sample at index 3 is LFE
// sample at index 4 is RL
// sample at index 5 is RR
// sample at index 6 is SL
// sample at index 7 is SR
// code is mostly duplicated between the two values of accumulate to avoid repeating the test
// for every sample
if (accumulate) {
while (numFrames) {
// centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
+ (pSrc[3] * MINUS_3_DB_IN_FLOAT);
// FL + centerPlusLfeContrib + SL + RL
lt = pSrc[0] + centerPlusLfeContrib + pSrc[6] + pSrc[4];
// FR + centerPlusLfeContrib + SR + RR
rt = pSrc[1] + centerPlusLfeContrib + pSrc[7] + pSrc[5];
//accumulate in destination
pDst[0] = clamp_float(pDst[0] + (lt / 2.0f));
pDst[1] = clamp_float(pDst[1] + (rt / 2.0f));
pSrc += 8;
pDst += 2;
numFrames--;
}
} else { // same code as above but without adding and clamping pDst[i] to itself
while (numFrames) {
// centerPlusLfeContrib = FC(-3dB) + LFE(-3dB)
centerPlusLfeContrib = (pSrc[2] * MINUS_3_DB_IN_FLOAT)
+ (pSrc[3] * MINUS_3_DB_IN_FLOAT);
// FL + centerPlusLfeContrib + SL + RL
lt = pSrc[0] + centerPlusLfeContrib + pSrc[6] + pSrc[4];
// FR + centerPlusLfeContrib + SR + RR
rt = pSrc[1] + centerPlusLfeContrib + pSrc[7] + pSrc[5];
// store in destination
pDst[0] = clamp_float(lt / 2.0f); // differs from when accumulate is true above
pDst[1] = clamp_float(rt / 2.0f); // differs from when accumulate is true above
pSrc += 8;
pDst += 2;
numFrames--;
}
}
}
/*----------------------------------------------------------------------------
* Downmix_foldGeneric()
*----------------------------------------------------------------------------
* Purpose:
* downmix to stereo a multichannel signal of arbitrary channel position mask.
*
* Inputs:
* mask the channel mask of pSrc
* pSrc multichannel audio buffer to downmix
* numFrames the number of multichannel frames to downmix
* accumulate whether to mix (when true) the result of the downmix with the contents of pDst,
* or overwrite pDst (when false)
*
* Outputs:
* pDst downmixed stereo audio samples
*
* Returns: false if multichannel format is not supported
*
*----------------------------------------------------------------------------
*/
bool Downmix_foldGeneric(
uint32_t mask, float *pSrc, float *pDst, size_t numFrames, bool accumulate) {
if (!Downmix_validChannelMask(mask)) {
return false;
}
const int numChan = audio_channel_count_from_out_mask(mask);
// compute at what index each channel is: samples will be in the following order:
// FL FR FC LFE BL BR BC SL SR
//
// (transfer matrix)
// FL FR FC LFE BL BR BC SL SR
// 0.5 0.353 0.353 0.5 0.353 0.5
// 0.5 0.353 0.353 0.5 0.353 0.5
// derive the indices for the transfer matrix columns that have non-zero values.
int indexFL = -1;
int indexFR = -1;
int indexFC = -1;
int indexLFE = -1;
int indexBL = -1;
int indexBR = -1;
int indexBC = -1;
int indexSL = -1;
int indexSR = -1;
int index = 0;
for (unsigned tmp = mask;
(tmp & (AUDIO_CHANNEL_OUT_7POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER)) != 0;
++index) {
const unsigned lowestBit = tmp & -(signed)tmp;
switch (lowestBit) {
case AUDIO_CHANNEL_OUT_FRONT_LEFT:
indexFL = index;
break;
case AUDIO_CHANNEL_OUT_FRONT_RIGHT:
indexFR = index;
break;
case AUDIO_CHANNEL_OUT_FRONT_CENTER:
indexFC = index;
break;
case AUDIO_CHANNEL_OUT_LOW_FREQUENCY:
indexLFE = index;
break;
case AUDIO_CHANNEL_OUT_BACK_LEFT:
indexBL = index;
break;
case AUDIO_CHANNEL_OUT_BACK_RIGHT:
indexBR = index;
break;
case AUDIO_CHANNEL_OUT_BACK_CENTER:
indexBC = index;
break;
case AUDIO_CHANNEL_OUT_SIDE_LEFT:
indexSL = index;
break;
case AUDIO_CHANNEL_OUT_SIDE_RIGHT:
indexSR = index;
break;
}
tmp ^= lowestBit;
}
// With good branch prediction, this should run reasonably fast.
// Also consider using a transfer matrix form.
while (numFrames) {
// compute contribution of FC, BC and LFE
float centersLfeContrib = 0;
if (indexFC >= 0) centersLfeContrib = pSrc[indexFC];
if (indexLFE >= 0) centersLfeContrib += pSrc[indexLFE];
if (indexBC >= 0) centersLfeContrib += pSrc[indexBC];
centersLfeContrib *= MINUS_3_DB_IN_FLOAT;
float ch[2];
ch[0] = centersLfeContrib;
ch[1] = centersLfeContrib;
// mix in left / right channels
if (indexFL >= 0) ch[0] += pSrc[indexFL];
if (indexFR >= 0) ch[1] += pSrc[indexFR];
if (indexSL >= 0) ch[0] += pSrc[indexSL];
if (indexSR >= 0) ch[1] += pSrc[indexSR]; // note pair checks enforce this if indexSL != 0
if (indexBL >= 0) ch[0] += pSrc[indexBL];
if (indexBR >= 0) ch[1] += pSrc[indexBR]; // note pair checks enforce this if indexBL != 0
// scale to prevent overflow.
ch[0] *= 0.5f;
ch[1] *= 0.5f;
if (accumulate) {
ch[0] += pDst[0];
ch[1] += pDst[1];
}
pDst[0] = clamp_float(ch[0]);
pDst[1] = clamp_float(ch[1]);
pSrc += numChan;
pDst += 2;
numFrames--;
}
return true;
}