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/*
* Copyright (C) 2006-2007 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 "AudioSystem"
//#define LOG_NDEBUG 0
#include <utils/Log.h>
#include <binder/IServiceManager.h>
#include <media/AudioSystem.h>
#include <math.h>
namespace android {
// client singleton for AudioFlinger binder interface
Mutex AudioSystem::gLock;
sp<IAudioFlinger> AudioSystem::gAudioFlinger;
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
// Cached values
int AudioSystem::gOutSamplingRate[NUM_AUDIO_OUTPUT_TYPES];
int AudioSystem::gOutFrameCount[NUM_AUDIO_OUTPUT_TYPES];
uint32_t AudioSystem::gOutLatency[NUM_AUDIO_OUTPUT_TYPES];
bool AudioSystem::gA2dpEnabled;
// Cached values for recording queries
uint32_t AudioSystem::gPrevInSamplingRate = 16000;
int AudioSystem::gPrevInFormat = AudioSystem::PCM_16_BIT;
int AudioSystem::gPrevInChannelCount = 1;
size_t AudioSystem::gInBuffSize = 0;
// establish binder interface to AudioFlinger service
const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
{
Mutex::Autolock _l(gLock);
if (gAudioFlinger.get() == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
binder = sm->getService(String16("media.audio_flinger"));
if (binder != 0)
break;
LOGW("AudioFlinger not published, waiting...");
usleep(500000); // 0.5 s
} while(true);
if (gAudioFlingerClient == NULL) {
gAudioFlingerClient = new AudioFlingerClient();
} else {
if (gAudioErrorCallback) {
gAudioErrorCallback(NO_ERROR);
}
}
binder->linkToDeath(gAudioFlingerClient);
gAudioFlinger = interface_cast<IAudioFlinger>(binder);
gAudioFlinger->registerClient(gAudioFlingerClient);
// Cache frequently accessed parameters
for (int output = 0; output < NUM_AUDIO_OUTPUT_TYPES; output++) {
gOutFrameCount[output] = (int)gAudioFlinger->frameCount(output);
gOutSamplingRate[output] = (int)gAudioFlinger->sampleRate(output);
gOutLatency[output] = gAudioFlinger->latency(output);
}
gA2dpEnabled = gAudioFlinger->isA2dpEnabled();
}
LOGE_IF(gAudioFlinger==0, "no AudioFlinger!?");
return gAudioFlinger;
}
// routing helper functions
status_t AudioSystem::speakerphone(bool state) {
uint32_t routes = state ? ROUTE_SPEAKER : ROUTE_EARPIECE;
return setRouting(MODE_IN_CALL, routes, ROUTE_ALL);
}
status_t AudioSystem::isSpeakerphoneOn(bool* state) {
uint32_t routes = 0;
status_t s = getRouting(MODE_IN_CALL, &routes);
*state = !!(routes & ROUTE_SPEAKER);
return s;
}
status_t AudioSystem::bluetoothSco(bool state) {
uint32_t mask = ROUTE_BLUETOOTH_SCO;
uint32_t routes = state ? mask : ROUTE_EARPIECE;
return setRouting(MODE_IN_CALL, routes, ROUTE_ALL);
}
status_t AudioSystem::isBluetoothScoOn(bool* state) {
uint32_t routes = 0;
status_t s = getRouting(MODE_IN_CALL, &routes);
*state = !!(routes & ROUTE_BLUETOOTH_SCO);
return s;
}
status_t AudioSystem::muteMicrophone(bool state) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setMicMute(state);
}
status_t AudioSystem::isMicrophoneMuted(bool* state) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*state = af->getMicMute();
return NO_ERROR;
}
status_t AudioSystem::setMasterVolume(float value)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
af->setMasterVolume(value);
return NO_ERROR;
}
status_t AudioSystem::setMasterMute(bool mute)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
af->setMasterMute(mute);
return NO_ERROR;
}
status_t AudioSystem::getMasterVolume(float* volume)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*volume = af->masterVolume();
return NO_ERROR;
}
status_t AudioSystem::getMasterMute(bool* mute)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*mute = af->masterMute();
return NO_ERROR;
}
status_t AudioSystem::setStreamVolume(int stream, float value)
{
if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
af->setStreamVolume(stream, value);
return NO_ERROR;
}
status_t AudioSystem::setStreamMute(int stream, bool mute)
{
if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
af->setStreamMute(stream, mute);
return NO_ERROR;
}
status_t AudioSystem::getStreamVolume(int stream, float* volume)
{
if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*volume = af->streamVolume(stream);
return NO_ERROR;
}
status_t AudioSystem::getStreamMute(int stream, bool* mute)
{
if (uint32_t(stream) >= NUM_STREAM_TYPES) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*mute = af->streamMute(stream);
return NO_ERROR;
}
status_t AudioSystem::setMode(int mode)
{
if (mode >= NUM_MODES) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setMode(mode);
}
status_t AudioSystem::getMode(int* mode)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*mode = af->getMode();
return NO_ERROR;
}
status_t AudioSystem::setRouting(int mode, uint32_t routes, uint32_t mask)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setRouting(mode, routes, mask);
}
status_t AudioSystem::getRouting(int mode, uint32_t* routes)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
uint32_t r = af->getRouting(mode);
*routes = r;
return NO_ERROR;
}
status_t AudioSystem::isMusicActive(bool* state) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*state = af->isMusicActive();
return NO_ERROR;
}
// Temporary interface, do not use
// TODO: Replace with a more generic key:value get/set mechanism
status_t AudioSystem::setParameter(const char* key, const char* value) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setParameter(key, value);
}
// convert volume steps to natural log scale
// change this value to change volume scaling
static const float dBPerStep = 0.5f;
// shouldn't need to touch these
static const float dBConvert = -dBPerStep * 2.302585093f / 20.0f;
static const float dBConvertInverse = 1.0f / dBConvert;
float AudioSystem::linearToLog(int volume)
{
// float v = volume ? exp(float(100 - volume) * dBConvert) : 0;
// LOGD("linearToLog(%d)=%f", volume, v);
// return v;
return volume ? exp(float(100 - volume) * dBConvert) : 0;
}
int AudioSystem::logToLinear(float volume)
{
// int v = volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
// LOGD("logTolinear(%d)=%f", v, volume);
// return v;
return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
}
status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
{
int output = getOutput(streamType);
if (output == NUM_AUDIO_OUTPUT_TYPES) return PERMISSION_DENIED;
// gOutSamplingRate[] is updated by getOutput() which calls get_audio_flinger()
LOGV("getOutputSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, gOutSamplingRate[output]);
*samplingRate = gOutSamplingRate[output];
return NO_ERROR;
}
status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
{
int output = getOutput(streamType);
if (output == NUM_AUDIO_OUTPUT_TYPES) return PERMISSION_DENIED;
// gOutFrameCount[] is updated by getOutput() which calls get_audio_flinger()
LOGV("getOutputFrameCount() streamType %d, output %d, frame count %d", streamType, output, gOutFrameCount[output]);
*frameCount = gOutFrameCount[output];
return NO_ERROR;
}
status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
{
int output = getOutput(streamType);
if (output == NUM_AUDIO_OUTPUT_TYPES) return PERMISSION_DENIED;
// gOutLatency[] is updated by getOutput() which calls get_audio_flinger()
LOGV("getOutputLatency() streamType %d, output %d, latency %d", streamType, output, gOutLatency[output]);
*latency = gOutLatency[output];
return NO_ERROR;
}
status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
size_t* buffSize)
{
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
|| (channelCount != gPrevInChannelCount)) {
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
gPrevInChannelCount = channelCount;
gInBuffSize = 0;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) {
return PERMISSION_DENIED;
}
gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
}
*buffSize = gInBuffSize;
return NO_ERROR;
}
// ---------------------------------------------------------------------------
void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
Mutex::Autolock _l(AudioSystem::gLock);
AudioSystem::gAudioFlinger.clear();
for (int output = 0; output < NUM_AUDIO_OUTPUT_TYPES; output++) {
gOutFrameCount[output] = 0;
gOutSamplingRate[output] = 0;
gOutLatency[output] = 0;
}
AudioSystem::gInBuffSize = 0;
if (gAudioErrorCallback) {
gAudioErrorCallback(DEAD_OBJECT);
}
LOGW("AudioFlinger server died!");
}
void AudioSystem::AudioFlingerClient::a2dpEnabledChanged(bool enabled) {
gA2dpEnabled = enabled;
LOGV("AudioFlinger A2DP enabled status changed! %d", enabled);
}
void AudioSystem::setErrorCallback(audio_error_callback cb) {
Mutex::Autolock _l(AudioSystem::gLock);
gAudioErrorCallback = cb;
}
int AudioSystem::getOutput(int streamType)
{
// make sure that gA2dpEnabled is valid by calling get_audio_flinger() which in turn
// will call gAudioFlinger->isA2dpEnabled()
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return NUM_AUDIO_OUTPUT_TYPES;
if (streamType == DEFAULT) {
streamType = MUSIC;
}
if (gA2dpEnabled && routedToA2dpOutput(streamType)) {
return AUDIO_OUTPUT_A2DP;
} else {
return AUDIO_OUTPUT_HARDWARE;
}
}
bool AudioSystem::routedToA2dpOutput(int streamType) {
switch(streamType) {
case MUSIC:
case VOICE_CALL:
case BLUETOOTH_SCO:
case SYSTEM:
return true;
default:
return false;
}
}
}; // namespace android
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