android_device_samsung_c1-c.../libaudio/ALSAStreamOps.cpp
2011-06-12 00:07:25 +02:00

325 lines
8.9 KiB
C++

/* ALSAStreamOps.cpp
**
** Copyright 2008-2009 Wind River Systems
**
** 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.
*/
#include <errno.h>
#include <stdarg.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdlib.h>
#include <unistd.h>
#include <dlfcn.h>
#define LOG_TAG "AudioHardwareALSA"
#include <utils/Log.h>
#include <utils/String8.h>
#include <cutils/properties.h>
#include <media/AudioRecord.h>
#include <hardware_legacy/power.h>
#include "AudioHardwareALSA.h"
namespace android
{
// ----------------------------------------------------------------------------
ALSAStreamOps::ALSAStreamOps(AudioHardwareALSA *parent, alsa_handle_t *handle) :
mParent(parent),
mHandle(handle),
mPowerLock(false)
{
}
ALSAStreamOps::~ALSAStreamOps()
{
AutoMutex lock(mLock);
close();
}
// use emulated popcount optimization
// http://www.df.lth.se/~john_e/gems/gem002d.html
static inline uint32_t popCount(uint32_t u)
{
u = ((u&0x55555555) + ((u>>1)&0x55555555));
u = ((u&0x33333333) + ((u>>2)&0x33333333));
u = ((u&0x0f0f0f0f) + ((u>>4)&0x0f0f0f0f));
u = ((u&0x00ff00ff) + ((u>>8)&0x00ff00ff));
u = ( u&0x0000ffff) + (u>>16);
return u;
}
acoustic_device_t *ALSAStreamOps::acoustics()
{
return mParent->mAcousticDevice;
}
ALSAMixer *ALSAStreamOps::mixer()
{
return mParent->mMixer;
}
status_t ALSAStreamOps::set(int *format,
uint32_t *channels,
uint32_t *rate)
{
status_t status = NO_ERROR;
if (channels && *channels != 0) {
if (mHandle->channels != popCount(*channels)) {
//updating default value
mHandle->channels = popCount(*channels);
status = BAD_VALUE;
}
} else if (channels) {
*channels = 0;
if (mHandle->devices & AudioSystem::DEVICE_OUT_ALL)
switch(mHandle->channels) {
case 4:
*channels |= AudioSystem::CHANNEL_OUT_BACK_LEFT;
*channels |= AudioSystem::CHANNEL_OUT_BACK_RIGHT;
// Fall through...
default:
case 2:
*channels |= AudioSystem::CHANNEL_OUT_FRONT_RIGHT;
// Fall through...
case 1:
*channels |= AudioSystem::CHANNEL_OUT_FRONT_LEFT;
break;
}
else
switch(mHandle->channels) {
default:
case 2:
*channels |= AudioSystem::CHANNEL_IN_RIGHT;
// Fall through...
case 1:
*channels |= AudioSystem::CHANNEL_IN_LEFT;
break;
}
}
if (rate && *rate > 0) {
if (mHandle->sampleRate != *rate){
//updating default value
mHandle->sampleRate = *rate;
status = BAD_VALUE;
}
} else if (rate)
*rate = mHandle->sampleRate;
snd_pcm_format_t iformat = mHandle->format;
if (format) {
switch(*format) {
case AudioSystem::FORMAT_DEFAULT:
break;
case AudioSystem::PCM_16_BIT:
iformat = SND_PCM_FORMAT_S16_LE;
break;
case AudioSystem::PCM_8_BIT:
iformat = SND_PCM_FORMAT_S8;
break;
default:
LOGE("Unknown PCM format %i. Forcing default", *format);
break;
}
if (mHandle->format != iformat){
//updating default value
mHandle->format = iformat;
status = BAD_VALUE;
} else {
switch(iformat) {
default:
case SND_PCM_FORMAT_S16_LE:
*format = AudioSystem::PCM_16_BIT;
break;
case SND_PCM_FORMAT_S8:
*format = AudioSystem::PCM_8_BIT;
break;
}
}
}
if (status == BAD_VALUE) {
/* resetting the default values */
if (mParent->mALSADevice->resetDefaults) {
status = mParent->mALSADevice->resetDefaults(mHandle);
if (status != NO_ERROR) {
LOGE("reset defaults return failure");
return BAD_VALUE;
}
}
}
return status;
}
status_t ALSAStreamOps::setParameters(const String8& keyValuePairs)
{
AudioParameter param = AudioParameter(keyValuePairs);
String8 key = String8(AudioParameter::keyRouting);
status_t status = NO_ERROR;
int device;
LOGV("setParameters() %s", keyValuePairs.string());
String8 keyfm = String8("fm_off");
String8 valuefm;
if (param.get(keyfm, valuefm) == NO_ERROR) {
mParent->mALSADevice->standby(mHandle);
param.remove(keyfm);
return status;
}
if (param.getInt(key, device) == NO_ERROR) {
mParent->mALSADevice->route(mHandle, (uint32_t)device, mParent->mode());
param.remove(key);
}
if (param.size()) {
// default action: fwd the kvp's to the module incase it wants to take action
if (mParent->mALSADevice->set) {
status = mParent->mALSADevice->set(keyValuePairs);
LOGI("setParameters() %s, %d", keyValuePairs.string(), (int)status);
return status;
} else {
LOGI("setParameters() :: BAD_VALUE");
return BAD_VALUE;
}
}
return status;
}
String8 ALSAStreamOps::getParameters(const String8& keys)
{
AudioParameter param = AudioParameter(keys);
String8 value;
String8 key = String8(AudioParameter::keyRouting);
if (param.get(key, value) == NO_ERROR) {
param.addInt(key, (int)mHandle->curDev);
}
LOGV("getParameters() %s", param.toString().string());
return param.toString();
}
uint32_t ALSAStreamOps::sampleRate() const
{
return mHandle->sampleRate;
}
//
// Return the number of bytes (not frames)
//
size_t ALSAStreamOps::bufferSize() const
{
snd_pcm_uframes_t bufferSize = mHandle->bufferSize;
size_t bytes = static_cast<size_t>(snd_pcm_frames_to_bytes(mHandle->handle, bufferSize));
// Not sure when this happened, but unfortunately it now
// appears that the bufferSize must be reported as a
// power of 2. This might be for OSS compatibility.
for (size_t i = 1; (bytes & ~i) != 0; i<<=1)
bytes &= ~i;
LOGI("buffer size (bytes) to AF = %d", bytes);
return bytes;
}
int ALSAStreamOps::format() const
{
int pcmFormatBitWidth;
int audioSystemFormat;
snd_pcm_format_t ALSAFormat = mHandle->format;
pcmFormatBitWidth = snd_pcm_format_physical_width(ALSAFormat);
switch(pcmFormatBitWidth) {
case 8:
audioSystemFormat = AudioSystem::PCM_8_BIT;
break;
default:
LOG_FATAL("Unknown AudioSystem bit width %i!", pcmFormatBitWidth);
case 16:
audioSystemFormat = AudioSystem::PCM_16_BIT;
break;
}
return audioSystemFormat;
}
uint32_t ALSAStreamOps::channels() const
{
unsigned int count = mHandle->channels;
uint32_t channels = 0;
if (mHandle->curDev & AudioSystem::DEVICE_OUT_ALL)
switch(count) {
case 4:
channels |= AudioSystem::CHANNEL_OUT_BACK_LEFT;
channels |= AudioSystem::CHANNEL_OUT_BACK_RIGHT;
// Fall through...
default:
case 2:
channels |= AudioSystem::CHANNEL_OUT_FRONT_RIGHT;
// Fall through...
case 1:
channels |= AudioSystem::CHANNEL_OUT_FRONT_LEFT;
break;
}
else
switch(count) {
default:
case 2:
channels |= AudioSystem::CHANNEL_IN_RIGHT;
// Fall through...
case 1:
channels |= AudioSystem::CHANNEL_IN_LEFT;
break;
}
return channels;
}
void ALSAStreamOps::close()
{
mParent->mALSADevice->close(mHandle);
}
//
// Set playback or capture PCM device. It's possible to support audio output
// or input from multiple devices by using the ALSA plugins, but this is
// not supported for simplicity.
//
// The AudioHardwareALSA API does not allow one to set the input routing.
//
// If the "routes" value does not map to a valid device, the default playback
// device is used.
//
status_t ALSAStreamOps::open(int mode)
{
return mParent->mALSADevice->open(mHandle, mHandle->curDev, mode);
}
} // namespace android