/* /android/src/frameworks/base/media/libeffects/AudioFormatAdapter.h
**
** Copyright 2009, 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.
*/
#ifndef AUDIOFORMATADAPTER_H_
#define AUDIOFORMATADAPTER_H_
#include <hardware/audio_effect.h>
#define min(x,y) (((x) < (y)) ? (x) : (y))
namespace android {
// An adapter for an audio processor working on audio_sample_t samples with a
// buffer override behavior to arbitrary sample formats and buffer behaviors.
// The adapter may work on any processing class which has a processing function
// with the following signature:
// void process(const audio_sample_t * pIn,
// audio_sample_t * pOut,
// int frameCount);
// It is assumed that the underlying processor works in S7.24 format and an
// overwrite behavior.
//
// Usage is simple: just work with the processor normally, but instead of
// calling its process() function directly, work with the process() function of
// the adapter.
// The adapter supports re-configuration to a different format on the fly.
//
// T The processor class.
// bufSize The maximum number of samples (single channel) to process on a
// single call to the underlying processor. Setting this to a small
// number will save a little memory, but will cost function call
// overhead, resulting from multiple calls to the underlying process()
// per a single call to this class's process().
template<class T, size_t bufSize>
class AudioFormatAdapter {
public:
// Configure the adapter.
// processor The underlying audio processor.
// nChannels Number of input and output channels. The adapter does not do
// channel conversion - this parameter must be in sync with the
// actual processor.
// pcmFormat The desired input/output sample format.
// behavior The desired behavior (overwrite or accumulate).
void configure(T & processor, int nChannels, uint8_t pcmFormat,
uint32_t behavior) {
mpProcessor = &processor;
mNumChannels = nChannels;
mPcmFormat = pcmFormat;
mBehavior = behavior;
mMaxSamplesPerCall = bufSize / nChannels;
}
// Process a block of samples.
// pIn A buffer of samples with the format specified on
// configure().
// pOut A buffer of samples with the format specified on
// configure(). May be the same as pIn.
// numSamples The number of multi-channel samples to process.
void process(const void * pIn, void * pOut, uint32_t numSamples) {
while (numSamples > 0) {
uint32_t numSamplesIter = min(numSamples, mMaxSamplesPerCall);
uint32_t nSamplesChannels = numSamplesIter * mNumChannels;
// This branch of "if" is untested
if (mPcmFormat == AUDIO_FORMAT_PCM_8_24_BIT) {
if (mBehavior == EFFECT_BUFFER_ACCESS_WRITE) {
mpProcessor->process(
reinterpret_cast<const audio_sample_t *> (pIn),
reinterpret_cast<audio_sample_t *> (pOut),
numSamplesIter);
} else if (mBehavior == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
mpProcessor->process(
reinterpret_cast<const audio_sample_t *> (pIn),
mBuffer, numSamplesIter);
MixOutput(pOut, numSamplesIter);
} else {
assert(false);
}
pIn = reinterpret_cast<const audio_sample_t *> (pIn)
+ nSamplesChannels;
pOut = reinterpret_cast<audio_sample_t *> (pOut)
+ nSamplesChannels;
} else {
ConvertInput(pIn, nSamplesChannels);
mpProcessor->process(mBuffer, mBuffer, numSamplesIter);
ConvertOutput(pOut, nSamplesChannels);
}
numSamples -= numSamplesIter;
}
}
private:
// The underlying processor.
T * mpProcessor;
// The number of input/output channels.
int mNumChannels;
// The desired PCM format.
uint8_t mPcmFormat;
// The desired buffer behavior.
uint32_t mBehavior;
// An intermediate buffer for processing.
audio_sample_t mBuffer[bufSize];
// The buffer size, divided by the number of channels - represents the
// maximum number of multi-channel samples that can be stored in the
// intermediate buffer.
size_t mMaxSamplesPerCall;
// Converts a buffer of input samples to audio_sample_t format.
// Output is written to the intermediate buffer.
// pIn The input buffer with the format designated in configure().
// When function exist will point to the next unread input
// sample.
// numSamples The number of single-channel samples to process.
void ConvertInput(const void *& pIn, uint32_t numSamples) {
if (mPcmFormat == AUDIO_FORMAT_PCM_16_BIT) {
const int16_t * pIn16 = reinterpret_cast<const int16_t *>(pIn);
audio_sample_t * pOut = mBuffer;
while (numSamples-- > 0) {
*(pOut++) = s15_to_audio_sample_t(*(pIn16++));
}
pIn = pIn16;
} else {
assert(false);
}
}
// Converts audio_sample_t samples from the intermediate buffer to the
// output buffer, converting to the desired format and buffer behavior.
// pOut The buffer to write the output to.
// When function exist will point to the next output sample.
// numSamples The number of single-channel samples to process.
void ConvertOutput(void *& pOut, uint32_t numSamples) {
if (mPcmFormat == AUDIO_FORMAT_PCM_16_BIT) {
const audio_sample_t * pIn = mBuffer;
int16_t * pOut16 = reinterpret_cast<int16_t *>(pOut);
if (mBehavior == EFFECT_BUFFER_ACCESS_WRITE) {
while (numSamples-- > 0) {
*(pOut16++) = audio_sample_t_to_s15_clip(*(pIn++));
}
} else if (mBehavior == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
while (numSamples-- > 0) {
*(pOut16++) += audio_sample_t_to_s15_clip(*(pIn++));
}
} else {
assert(false);
}
pOut = pOut16;
} else {
assert(false);
}
}
// Accumulate data from the intermediate buffer to the output. Output is
// assumed to be of audio_sample_t type.
// pOut The buffer to mix the output to.
// When function exist will point to the next output sample.
// numSamples The number of single-channel samples to process.
void MixOutput(void *& pOut, uint32_t numSamples) {
const audio_sample_t * pIn = mBuffer;
audio_sample_t * pOut24 = reinterpret_cast<audio_sample_t *>(pOut);
numSamples *= mNumChannels;
while (numSamples-- > 0) {
*(pOut24++) += *(pIn++);
}
pOut = pOut24;
}
};
}
#endif // AUDIOFORMATADAPTER_H_