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android_frameworks_base/media/libmedia/AudioSystem.cpp
Eric Laurent 8ed6ed0b62 Audio policy manager changes for audio effects 
Added methods for audio effects management by audio policy manager.
- control of total CPU load and memory used by effect engines
- selection of output stream for global effects
- added audio session id in parameter list for startOutput() and stopOutput().
this is not used in default audio policy manager implementation.

Modifications of audio effect framework in AudioFlinger to allow moving and reconfiguring
effect engines from one output mixer thread to another when audio tracks in the same session
are moved or when requested by audio policy manager.
Also fixed mutex deadlock problem with effect chains locks.

Change-Id: Ida43484b06e9b890d6b9e53c13958d042720ebdb
2010-07-20 10:31:57 -07:00

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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 <media/IAudioPolicyService.h>
#include <math.h>
// ----------------------------------------------------------------------------
// the sim build doesn't have gettid
#ifndef HAVE_GETTID
# define gettid getpid
#endif
// ----------------------------------------------------------------------------
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
DefaultKeyedVector<int, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
// 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);
}
LOGE_IF(gAudioFlinger==0, "no AudioFlinger!?");
return gAudioFlinger;
}
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, int output)
{
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, output);
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, int output)
{
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, output);
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::isStreamActive(int stream, bool* state) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*state = af->isStreamActive(stream);
return NO_ERROR;
}
status_t AudioSystem::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setParameters(ioHandle, keyValuePairs);
}
String8 AudioSystem::getParameters(audio_io_handle_t ioHandle, const String8& keys) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
String8 result = String8("");
if (af == 0) return result;
result = af->getParameters(ioHandle, keys);
return result;
}
// 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)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
if (streamType == DEFAULT) {
streamType = MUSIC;
}
output = getOutput((stream_type)streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
if (outputDesc == 0) {
LOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*samplingRate = af->sampleRate(output);
} else {
LOGV("getOutputSamplingRate() reading from output desc");
*samplingRate = outputDesc->samplingRate;
gLock.unlock();
}
LOGV("getOutputSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, *samplingRate);
return NO_ERROR;
}
status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
if (streamType == DEFAULT) {
streamType = MUSIC;
}
output = getOutput((stream_type)streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
if (outputDesc == 0) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*frameCount = af->frameCount(output);
} else {
*frameCount = outputDesc->frameCount;
gLock.unlock();
}
LOGV("getOutputFrameCount() streamType %d, output %d, frameCount %d", streamType, output, *frameCount);
return NO_ERROR;
}
status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
if (streamType == DEFAULT) {
streamType = MUSIC;
}
output = getOutput((stream_type)streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
if (outputDesc == 0) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
*latency = af->latency(output);
} else {
*latency = outputDesc->latency;
gLock.unlock();
}
LOGV("getOutputLatency() streamType %d, output %d, latency %d", streamType, output, *latency);
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;
}
status_t AudioSystem::setVoiceVolume(float value)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setVoiceVolume(value);
}
status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
if (stream == DEFAULT) {
stream = MUSIC;
}
return af->getRenderPosition(halFrames, dspFrames, getOutput((stream_type)stream));
}
unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
unsigned int result = 0;
if (af == 0) return result;
if (ioHandle == 0) return result;
result = af->getInputFramesLost(ioHandle);
return result;
}
int AudioSystem::newAudioSessionId() {
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return 0;
return af->newAudioSessionId();
}
// ---------------------------------------------------------------------------
void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who) {
Mutex::Autolock _l(AudioSystem::gLock);
AudioSystem::gAudioFlinger.clear();
// clear output handles and stream to output map caches
AudioSystem::gStreamOutputMap.clear();
AudioSystem::gOutputs.clear();
if (gAudioErrorCallback) {
gAudioErrorCallback(DEAD_OBJECT);
}
LOGW("AudioFlinger server died!");
}
void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, void *param2) {
LOGV("ioConfigChanged() event %d", event);
OutputDescriptor *desc;
uint32_t stream;
if (ioHandle == 0) return;
Mutex::Autolock _l(AudioSystem::gLock);
switch (event) {
case STREAM_CONFIG_CHANGED:
if (param2 == 0) break;
stream = *(uint32_t *)param2;
LOGV("ioConfigChanged() STREAM_CONFIG_CHANGED stream %d, output %d", stream, ioHandle);
if (gStreamOutputMap.indexOfKey(stream) >= 0) {
gStreamOutputMap.replaceValueFor(stream, ioHandle);
}
break;
case OUTPUT_OPENED: {
if (gOutputs.indexOfKey(ioHandle) >= 0) {
LOGV("ioConfigChanged() opening already existing output! %d", ioHandle);
break;
}
if (param2 == 0) break;
desc = (OutputDescriptor *)param2;
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
LOGV("ioConfigChanged() new output samplingRate %d, format %d channels %d frameCount %d latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channels, outputDesc->frameCount, outputDesc->latency);
} break;
case OUTPUT_CLOSED: {
if (gOutputs.indexOfKey(ioHandle) < 0) {
LOGW("ioConfigChanged() closing unknow output! %d", ioHandle);
break;
}
LOGV("ioConfigChanged() output %d closed", ioHandle);
gOutputs.removeItem(ioHandle);
for (int i = gStreamOutputMap.size() - 1; i >= 0 ; i--) {
if (gStreamOutputMap.valueAt(i) == ioHandle) {
gStreamOutputMap.removeItemsAt(i);
}
}
} break;
case OUTPUT_CONFIG_CHANGED: {
int index = gOutputs.indexOfKey(ioHandle);
if (index < 0) {
LOGW("ioConfigChanged() modifying unknow output! %d", ioHandle);
break;
}
if (param2 == 0) break;
desc = (OutputDescriptor *)param2;
LOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channels, desc->frameCount, desc->latency);
OutputDescriptor *outputDesc = gOutputs.valueAt(index);
delete outputDesc;
outputDesc = new OutputDescriptor(*desc);
gOutputs.replaceValueFor(ioHandle, outputDesc);
} break;
case INPUT_OPENED:
case INPUT_CLOSED:
case INPUT_CONFIG_CHANGED:
break;
}
}
void AudioSystem::setErrorCallback(audio_error_callback cb) {
Mutex::Autolock _l(gLock);
gAudioErrorCallback = cb;
}
bool AudioSystem::routedToA2dpOutput(int streamType) {
switch(streamType) {
case MUSIC:
case VOICE_CALL:
case BLUETOOTH_SCO:
case SYSTEM:
return true;
default:
return false;
}
}
// client singleton for AudioPolicyService binder interface
sp<IAudioPolicyService> AudioSystem::gAudioPolicyService;
sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient;
// establish binder interface to AudioFlinger service
const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
{
gLock.lock();
if (gAudioPolicyService.get() == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
binder = sm->getService(String16("media.audio_policy"));
if (binder != 0)
break;
LOGW("AudioPolicyService not published, waiting...");
usleep(500000); // 0.5 s
} while(true);
if (gAudioPolicyServiceClient == NULL) {
gAudioPolicyServiceClient = new AudioPolicyServiceClient();
}
binder->linkToDeath(gAudioPolicyServiceClient);
gAudioPolicyService = interface_cast<IAudioPolicyService>(binder);
gLock.unlock();
} else {
gLock.unlock();
}
return gAudioPolicyService;
}
status_t AudioSystem::setDeviceConnectionState(audio_devices device,
device_connection_state state,
const char *device_address)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setDeviceConnectionState(device, state, device_address);
}
AudioSystem::device_connection_state AudioSystem::getDeviceConnectionState(audio_devices device,
const char *device_address)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return DEVICE_STATE_UNAVAILABLE;
return aps->getDeviceConnectionState(device, device_address);
}
status_t AudioSystem::setPhoneState(int state)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setPhoneState(state);
}
status_t AudioSystem::setRingerMode(uint32_t mode, uint32_t mask)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setRingerMode(mode, mask);
}
status_t AudioSystem::setForceUse(force_use usage, forced_config config)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setForceUse(usage, config);
}
AudioSystem::forced_config AudioSystem::getForceUse(force_use usage)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return FORCE_NONE;
return aps->getForceUse(usage);
}
audio_io_handle_t AudioSystem::getOutput(stream_type stream,
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
output_flags flags)
{
audio_io_handle_t output = 0;
// Do not use stream to output map cache if the direct output
// flag is set or if we are likely to use a direct output
// (e.g voice call stream @ 8kHz could use BT SCO device and be routed to
// a direct output on some platforms).
// TODO: the output cache and stream to output mapping implementation needs to
// be reworked for proper operation with direct outputs. This code is too specific
// to the first use case we want to cover (Voice Recognition and Voice Dialer over
// Bluetooth SCO
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0 &&
((stream != AudioSystem::VOICE_CALL && stream != AudioSystem::BLUETOOTH_SCO) ||
channels != AudioSystem::CHANNEL_OUT_MONO ||
(samplingRate != 8000 && samplingRate != 16000))) {
Mutex::Autolock _l(gLock);
output = AudioSystem::gStreamOutputMap.valueFor(stream);
LOGV_IF((output != 0), "getOutput() read %d from cache for stream %d", output, stream);
}
if (output == 0) {
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
output = aps->getOutput(stream, samplingRate, format, channels, flags);
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) == 0) {
Mutex::Autolock _l(gLock);
AudioSystem::gStreamOutputMap.add(stream, output);
}
}
return output;
}
status_t AudioSystem::startOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->startOutput(output, stream, session);
}
status_t AudioSystem::stopOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
int session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->stopOutput(output, stream, session);
}
void AudioSystem::releaseOutput(audio_io_handle_t output)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return;
aps->releaseOutput(output);
}
audio_io_handle_t AudioSystem::getInput(int inputSource,
uint32_t samplingRate,
uint32_t format,
uint32_t channels,
audio_in_acoustics acoustics)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
return aps->getInput(inputSource, samplingRate, format, channels, acoustics);
}
status_t AudioSystem::startInput(audio_io_handle_t input)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->startInput(input);
}
status_t AudioSystem::stopInput(audio_io_handle_t input)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->stopInput(input);
}
void AudioSystem::releaseInput(audio_io_handle_t input)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return;
aps->releaseInput(input);
}
status_t AudioSystem::initStreamVolume(stream_type stream,
int indexMin,
int indexMax)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->initStreamVolume(stream, indexMin, indexMax);
}
status_t AudioSystem::setStreamVolumeIndex(stream_type stream, int index)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setStreamVolumeIndex(stream, index);
}
status_t AudioSystem::getStreamVolumeIndex(stream_type stream, int *index)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->getStreamVolumeIndex(stream, index);
}
uint32_t AudioSystem::getStrategyForStream(AudioSystem::stream_type stream)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return 0;
return aps->getStrategyForStream(stream);
}
audio_io_handle_t AudioSystem::getOutputForEffect(effect_descriptor_t *desc)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->getOutputForEffect(desc);
}
status_t AudioSystem::registerEffect(effect_descriptor_t *desc,
audio_io_handle_t output,
uint32_t strategy,
int session,
int id)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->registerEffect(desc, output, strategy, session, id);
}
status_t AudioSystem::unregisterEffect(int id)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->unregisterEffect(id);
}
// ---------------------------------------------------------------------------
void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who) {
Mutex::Autolock _l(AudioSystem::gLock);
AudioSystem::gAudioPolicyService.clear();
LOGW("AudioPolicyService server died!");
}
// ---------------------------------------------------------------------------
// use emulated popcount optimization
// http://www.df.lth.se/~john_e/gems/gem002d.html
uint32_t AudioSystem::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;
}
bool AudioSystem::isOutputDevice(audio_devices device)
{
if ((popCount(device) == 1 ) &&
((device & ~AudioSystem::DEVICE_OUT_ALL) == 0)) {
return true;
} else {
return false;
}
}
bool AudioSystem::isInputDevice(audio_devices device)
{
if ((popCount(device) == 1 ) &&
((device & ~AudioSystem::DEVICE_IN_ALL) == 0)) {
return true;
} else {
return false;
}
}
bool AudioSystem::isA2dpDevice(audio_devices device)
{
if ((popCount(device) == 1 ) &&
(device & (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER))) {
return true;
} else {
return false;
}
}
bool AudioSystem::isBluetoothScoDevice(audio_devices device)
{
if ((popCount(device) == 1 ) &&
(device & (AudioSystem::DEVICE_OUT_BLUETOOTH_SCO |
AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT))) {
return true;
} else {
return false;
}
}
bool AudioSystem::isLowVisibility(stream_type stream)
{
if (stream == AudioSystem::SYSTEM ||
stream == AudioSystem::NOTIFICATION ||
stream == AudioSystem::RING) {
return true;
} else {
return false;
}
}
bool AudioSystem::isInputChannel(uint32_t channel)
{
if ((channel & ~AudioSystem::CHANNEL_IN_ALL) == 0) {
return true;
} else {
return false;
}
}
bool AudioSystem::isOutputChannel(uint32_t channel)
{
if ((channel & ~AudioSystem::CHANNEL_OUT_ALL) == 0) {
return true;
} else {
return false;
}
}
bool AudioSystem::isValidFormat(uint32_t format)
{
switch (format & MAIN_FORMAT_MASK) {
case PCM:
case MP3:
case AMR_NB:
case AMR_WB:
case AAC:
case HE_AAC_V1:
case HE_AAC_V2:
case VORBIS:
return true;
default:
return false;
}
}
bool AudioSystem::isLinearPCM(uint32_t format)
{
switch (format) {
case PCM_16_BIT:
case PCM_8_BIT:
return true;
default:
return false;
}
}
//------------------------- AudioParameter class implementation ---------------
const char *AudioParameter::keyRouting = "routing";
const char *AudioParameter::keySamplingRate = "sampling_rate";
const char *AudioParameter::keyFormat = "format";
const char *AudioParameter::keyChannels = "channels";
const char *AudioParameter::keyFrameCount = "frame_count";
AudioParameter::AudioParameter(const String8& keyValuePairs)
{
char *str = new char[keyValuePairs.length()+1];
mKeyValuePairs = keyValuePairs;
strcpy(str, keyValuePairs.string());
char *pair = strtok(str, ";");
while (pair != NULL) {
if (strlen(pair) != 0) {
size_t eqIdx = strcspn(pair, "=");
String8 key = String8(pair, eqIdx);
String8 value;
if (eqIdx == strlen(pair)) {
value = String8("");
} else {
value = String8(pair + eqIdx + 1);
}
if (mParameters.indexOfKey(key) < 0) {
mParameters.add(key, value);
} else {
mParameters.replaceValueFor(key, value);
}
} else {
LOGV("AudioParameter() cstor empty key value pair");
}
pair = strtok(NULL, ";");
}
delete[] str;
}
AudioParameter::~AudioParameter()
{
mParameters.clear();
}
String8 AudioParameter::toString()
{
String8 str = String8("");
size_t size = mParameters.size();
for (size_t i = 0; i < size; i++) {
str += mParameters.keyAt(i);
str += "=";
str += mParameters.valueAt(i);
if (i < (size - 1)) str += ";";
}
return str;
}
status_t AudioParameter::add(const String8& key, const String8& value)
{
if (mParameters.indexOfKey(key) < 0) {
mParameters.add(key, value);
return NO_ERROR;
} else {
mParameters.replaceValueFor(key, value);
return ALREADY_EXISTS;
}
}
status_t AudioParameter::addInt(const String8& key, const int value)
{
char str[12];
if (snprintf(str, 12, "%d", value) > 0) {
String8 str8 = String8(str);
return add(key, str8);
} else {
return BAD_VALUE;
}
}
status_t AudioParameter::addFloat(const String8& key, const float value)
{
char str[23];
if (snprintf(str, 23, "%.10f", value) > 0) {
String8 str8 = String8(str);
return add(key, str8);
} else {
return BAD_VALUE;
}
}
status_t AudioParameter::remove(const String8& key)
{
if (mParameters.indexOfKey(key) >= 0) {
mParameters.removeItem(key);
return NO_ERROR;
} else {
return BAD_VALUE;
}
}
status_t AudioParameter::get(const String8& key, String8& value)
{
if (mParameters.indexOfKey(key) >= 0) {
value = mParameters.valueFor(key);
return NO_ERROR;
} else {
return BAD_VALUE;
}
}
status_t AudioParameter::getInt(const String8& key, int& value)
{
String8 str8;
status_t result = get(key, str8);
value = 0;
if (result == NO_ERROR) {
int val;
if (sscanf(str8.string(), "%d", &val) == 1) {
value = val;
} else {
result = INVALID_OPERATION;
}
}
return result;
}
status_t AudioParameter::getFloat(const String8& key, float& value)
{
String8 str8;
status_t result = get(key, str8);
value = 0;
if (result == NO_ERROR) {
float val;
if (sscanf(str8.string(), "%f", &val) == 1) {
value = val;
} else {
result = INVALID_OPERATION;
}
}
return result;
}
status_t AudioParameter::getAt(size_t index, String8& key, String8& value)
{
if (mParameters.size() > index) {
key = mParameters.keyAt(index);
value = mParameters.valueAt(index);
return NO_ERROR;
} else {
return BAD_VALUE;
}
}
}; // namespace android
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