/* * 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