blob: c77f551f119ad125af06798c39b59ce5cf8339ac [file] [log] [blame]
/*
* 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)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->startOutput(output, stream);
}
status_t AudioSystem::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->stopOutput(output, stream);
}
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);
}
// ---------------------------------------------------------------------------
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