summaryrefslogtreecommitdiffstats
path: root/services/audiopolicy
diff options
context:
space:
mode:
Diffstat (limited to 'services/audiopolicy')
-rw-r--r--services/audiopolicy/Android.mk32
-rw-r--r--services/audiopolicy/AudioPolicyClientImpl.cpp261
-rw-r--r--services/audiopolicy/AudioPolicyInterface.h257
-rw-r--r--services/audiopolicy/AudioPolicyInterfaceImpl.cpp489
-rw-r--r--services/audiopolicy/AudioPolicyManager.cpp4104
-rw-r--r--services/audiopolicy/AudioPolicyManager.h582
-rw-r--r--services/audiopolicy/AudioPolicyService.cpp1085
-rw-r--r--services/audiopolicy/AudioPolicyService.h353
8 files changed, 7163 insertions, 0 deletions
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
new file mode 100644
index 0000000..84565bb
--- /dev/null
+++ b/services/audiopolicy/Android.mk
@@ -0,0 +1,32 @@
+LOCAL_PATH:= $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ AudioPolicyService.cpp \
+ AudioPolicyInterfaceImpl.cpp \
+ AudioPolicyClientImpl.cpp
+
+LOCAL_C_INCLUDES := \
+ $(TOPDIR)frameworks/av/services/audioflinger \
+ $(call include-path-for, audio-effects) \
+ $(call include-path-for, audio-utils)
+
+LOCAL_SHARED_LIBRARIES := \
+ libcutils \
+ libutils \
+ liblog \
+ libbinder \
+ libmedia \
+ libhardware \
+ libhardware_legacy
+
+LOCAL_STATIC_LIBRARIES := \
+ libmedia_helper \
+ libserviceutility
+
+LOCAL_MODULE:= libaudiopolicy
+
+LOCAL_CFLAGS += -fvisibility=hidden
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/AudioPolicyClientImpl.cpp
new file mode 100644
index 0000000..53f3e2d
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyClientImpl.cpp
@@ -0,0 +1,261 @@
+/*
+ * Copyright (C) 2009 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 "AudioPolicyService"
+//#define LOG_NDEBUG 0
+
+#include "Configuration.h"
+#undef __STRICT_ANSI__
+#define __STDINT_LIMITS
+#define __STDC_LIMIT_MACROS
+#include <stdint.h>
+
+#include <sys/time.h>
+#include <binder/IServiceManager.h>
+#include <utils/Log.h>
+#include <cutils/properties.h>
+#include <binder/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
+#include <hardware_legacy/power.h>
+#include <media/AudioEffect.h>
+#include <media/EffectsFactoryApi.h>
+//#include <media/IAudioFlinger.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <hardware/audio_policy.h>
+#include <audio_effects/audio_effects_conf.h>
+#include <media/AudioParameter.h>
+
+
+namespace android {
+
+/* implementation of the interface to the policy manager */
+extern "C" {
+
+audio_module_handle_t aps_load_hw_module(void *service __unused,
+ const char *name)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return 0;
+ }
+
+ return af->loadHwModule(name);
+}
+
+// deprecated: replaced by aps_open_output_on_module()
+audio_io_handle_t aps_open_output(void *service __unused,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ uint32_t *pLatencyMs,
+ audio_output_flags_t flags)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return 0;
+ }
+
+ return af->openOutput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask,
+ pLatencyMs, flags);
+}
+
+audio_io_handle_t aps_open_output_on_module(void *service __unused,
+ audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ uint32_t *pLatencyMs,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return 0;
+ }
+ return af->openOutput(module, pDevices, pSamplingRate, pFormat, pChannelMask,
+ pLatencyMs, flags, offloadInfo);
+}
+
+audio_io_handle_t aps_open_dup_output(void *service __unused,
+ audio_io_handle_t output1,
+ audio_io_handle_t output2)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return 0;
+ }
+ return af->openDuplicateOutput(output1, output2);
+}
+
+int aps_close_output(void *service __unused, audio_io_handle_t output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+
+ return af->closeOutput(output);
+}
+
+int aps_suspend_output(void *service __unused, audio_io_handle_t output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return PERMISSION_DENIED;
+ }
+
+ return af->suspendOutput(output);
+}
+
+int aps_restore_output(void *service __unused, audio_io_handle_t output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return PERMISSION_DENIED;
+ }
+
+ return af->restoreOutput(output);
+}
+
+// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored
+audio_io_handle_t aps_open_input(void *service __unused,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ audio_in_acoustics_t acoustics __unused)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return 0;
+ }
+
+ return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask);
+}
+
+audio_io_handle_t aps_open_input_on_module(void *service __unused,
+ audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ ALOGW("%s: could not get AudioFlinger", __func__);
+ return 0;
+ }
+
+ return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
+}
+
+int aps_close_input(void *service __unused, audio_io_handle_t input)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+
+ return af->closeInput(input);
+}
+
+int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+
+ return af->invalidateStream(stream);
+}
+
+int aps_move_effects(void *service __unused, int session,
+ audio_io_handle_t src_output,
+ audio_io_handle_t dst_output)
+{
+ sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+
+ return af->moveEffects(session, src_output, dst_output);
+}
+
+char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
+ const char *keys)
+{
+ String8 result = AudioSystem::getParameters(io_handle, String8(keys));
+ return strdup(result.string());
+}
+
+void aps_set_parameters(void *service, audio_io_handle_t io_handle,
+ const char *kv_pairs, int delay_ms)
+{
+ AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+ audioPolicyService->setParameters(io_handle, kv_pairs, delay_ms);
+}
+
+int aps_set_stream_volume(void *service, audio_stream_type_t stream,
+ float volume, audio_io_handle_t output,
+ int delay_ms)
+{
+ AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+ return audioPolicyService->setStreamVolume(stream, volume, output,
+ delay_ms);
+}
+
+int aps_start_tone(void *service, audio_policy_tone_t tone,
+ audio_stream_type_t stream)
+{
+ AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+ return audioPolicyService->startTone(tone, stream);
+}
+
+int aps_stop_tone(void *service)
+{
+ AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+ return audioPolicyService->stopTone();
+}
+
+int aps_set_voice_volume(void *service, float volume, int delay_ms)
+{
+ AudioPolicyService *audioPolicyService = (AudioPolicyService *)service;
+
+ return audioPolicyService->setVoiceVolume(volume, delay_ms);
+}
+
+}; // extern "C"
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
new file mode 100644
index 0000000..768b13e
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -0,0 +1,257 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYINTERFACE_H
+#define ANDROID_AUDIOPOLICYINTERFACE_H
+
+#include <media/AudioSystem.h>
+#include <utils/String8.h>
+
+#include <hardware/audio_policy.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
+// between the platform specific audio policy manager and Android generic audio policy manager.
+// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class.
+// This implementation makes use of the AudioPolicyClientInterface to control the activity and
+// configuration of audio input and output streams.
+//
+// The platform specific audio policy manager is in charge of the audio routing and volume control
+// policies for a given platform.
+// The main roles of this module are:
+// - keep track of current system state (removable device connections, phone state, user requests...).
+// System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface.
+// - process getOutput() queries received when AudioTrack objects are created: Those queries
+// return a handler on an output that has been selected, configured and opened by the audio policy manager and that
+// must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method.
+// When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide
+// to close or reconfigure the output depending on other streams using this output and current system state.
+// - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs.
+// - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value
+// applicable to each output as a function of platform specific settings and current output route (destination device). It
+// also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries).
+//
+// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so)
+// and is linked with libaudioflinger.so
+
+
+// Audio Policy Manager Interface
+class AudioPolicyInterface
+{
+
+public:
+ virtual ~AudioPolicyInterface() {}
+ //
+ // configuration functions
+ //
+
+ // indicate a change in device connection status
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address) = 0;
+ // retrieve a device connection status
+ virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+ const char *device_address) = 0;
+ // indicate a change in phone state. Valid phones states are defined by audio_mode_t
+ virtual void setPhoneState(audio_mode_t state) = 0;
+ // force using a specific device category for the specified usage
+ virtual void setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0;
+ // retrieve current device category forced for a given usage
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
+ // set a system property (e.g. camera sound always audible)
+ virtual void setSystemProperty(const char* property, const char* value) = 0;
+ // check proper initialization
+ virtual status_t initCheck() = 0;
+
+ //
+ // Audio routing query functions
+ //
+
+ // request an output appropriate for playback of the supplied stream type and parameters
+ virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo) = 0;
+ // indicates to the audio policy manager that the output starts being used by corresponding stream.
+ virtual status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session = 0) = 0;
+ // indicates to the audio policy manager that the output stops being used by corresponding stream.
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session = 0) = 0;
+ // releases the output.
+ virtual void releaseOutput(audio_io_handle_t output) = 0;
+
+ // request an input appropriate for record from the supplied device with supplied parameters.
+ virtual audio_io_handle_t getInput(audio_source_t inputSource,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_in_acoustics_t acoustics) = 0;
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input) = 0;
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input) = 0;
+ // releases the input.
+ virtual void releaseInput(audio_io_handle_t input) = 0;
+
+ //
+ // volume control functions
+ //
+
+ // initialises stream volume conversion parameters by specifying volume index range.
+ virtual void initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax) = 0;
+
+ // sets the new stream volume at a level corresponding to the supplied index for the
+ // supplied device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means
+ // setting volume for all devices
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device) = 0;
+
+ // retrieve current volume index for the specified stream and the
+ // specified device. By convention, specifying AUDIO_DEVICE_OUT_DEFAULT means
+ // querying the volume of the active device.
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device) = 0;
+
+ // return the strategy corresponding to a given stream type
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream) = 0;
+
+ // return the enabled output devices for the given stream type
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream) = 0;
+
+ // Audio effect management
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc) = 0;
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id) = 0;
+ virtual status_t unregisterEffect(int id) = 0;
+ virtual status_t setEffectEnabled(int id, bool enabled) = 0;
+
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const = 0;
+ virtual bool isStreamActiveRemotely(audio_stream_type_t stream,
+ uint32_t inPastMs = 0) const = 0;
+ virtual bool isSourceActive(audio_source_t source) const = 0;
+
+ //dump state
+ virtual status_t dump(int fd) = 0;
+
+ virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo) = 0;
+};
+
+
+// Audio Policy client Interface
+class AudioPolicyClientInterface
+{
+public:
+ virtual ~AudioPolicyClientInterface() {}
+
+ //
+ // Audio HW module functions
+ //
+
+ // loads a HW module.
+ virtual audio_module_handle_t loadHwModule(const char *name) = 0;
+
+ //
+ // Audio output Control functions
+ //
+
+ // opens an audio output with the requested parameters. The parameter values can indicate to use the default values
+ // in case the audio policy manager has no specific requirements for the output being opened.
+ // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream.
+ // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
+ virtual audio_io_handle_t openOutput(audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ uint32_t *pLatencyMs,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo = NULL) = 0;
+ // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
+ // a special mixer thread in the AudioFlinger.
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0;
+ // closes the output stream
+ virtual status_t closeOutput(audio_io_handle_t output) = 0;
+ // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in
+ // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded.
+ virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+ // restores a suspended output.
+ virtual status_t restoreOutput(audio_io_handle_t output) = 0;
+
+ //
+ // Audio input Control functions
+ //
+
+ // opens an audio input
+ virtual audio_io_handle_t openInput(audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask) = 0;
+ // closes an audio input
+ virtual status_t closeInput(audio_io_handle_t input) = 0;
+ //
+ // misc control functions
+ //
+
+ // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
+ // for each output (destination device) it is attached to.
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float volume, audio_io_handle_t output, int delayMs = 0) = 0;
+
+ // invalidate a stream type, causing a reroute to an unspecified new output
+ virtual status_t invalidateStream(audio_stream_type_t stream) = 0;
+
+ // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
+ virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs, int delayMs = 0) = 0;
+ // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
+
+ // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
+ // over a telephony device during a phone call.
+ virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream) = 0;
+ virtual status_t stopTone() = 0;
+
+ // set down link audio volume.
+ virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0;
+
+ // move effect to the specified output
+ virtual status_t moveEffects(int session,
+ audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput) = 0;
+
+};
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface);
+
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYINTERFACE_H
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
new file mode 100644
index 0000000..bb62ab3
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
@@ -0,0 +1,489 @@
+/*
+ * Copyright (C) 2009 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 "AudioPolicyService"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
+
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <hardware/audio_policy.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+
+status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if (!audio_is_output_device(device) && !audio_is_input_device(device)) {
+ return BAD_VALUE;
+ }
+ if (state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE &&
+ state != AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
+ return BAD_VALUE;
+ }
+
+ ALOGV("setDeviceConnectionState()");
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->set_device_connection_state(mpAudioPolicy, device,
+ state, device_address);
+}
+
+audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState(
+ audio_devices_t device,
+ const char *device_address)
+{
+ if (mpAudioPolicy == NULL) {
+ return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ }
+ return mpAudioPolicy->get_device_connection_state(mpAudioPolicy, device,
+ device_address);
+}
+
+status_t AudioPolicyService::setPhoneState(audio_mode_t state)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if (uint32_t(state) >= AUDIO_MODE_CNT) {
+ return BAD_VALUE;
+ }
+
+ ALOGV("setPhoneState()");
+
+ // TODO: check if it is more appropriate to do it in platform specific policy manager
+ AudioSystem::setMode(state);
+
+ Mutex::Autolock _l(mLock);
+ mpAudioPolicy->set_phone_state(mpAudioPolicy, state);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
+ return BAD_VALUE;
+ }
+ if (config < 0 || config >= AUDIO_POLICY_FORCE_CFG_CNT) {
+ return BAD_VALUE;
+ }
+ ALOGV("setForceUse()");
+ Mutex::Autolock _l(mLock);
+ mpAudioPolicy->set_force_use(mpAudioPolicy, usage, config);
+ return NO_ERROR;
+}
+
+audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use_t usage)
+{
+ if (mpAudioPolicy == NULL) {
+ return AUDIO_POLICY_FORCE_NONE;
+ }
+ if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
+ return AUDIO_POLICY_FORCE_NONE;
+ }
+ return mpAudioPolicy->get_force_use(mpAudioPolicy, usage);
+}
+
+audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ if (mpAudioPolicy == NULL) {
+ return 0;
+ }
+ ALOGV("getOutput()");
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->get_output(mpAudioPolicy, stream, samplingRate,
+ format, channelMask, flags, offloadInfo);
+}
+
+status_t AudioPolicyService::startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ ALOGV("startOutput()");
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->start_output(mpAudioPolicy, output, stream, session);
+}
+
+status_t AudioPolicyService::stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ ALOGV("stopOutput()");
+ mOutputCommandThread->stopOutputCommand(output, stream, session);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::doStopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session)
+{
+ ALOGV("doStopOutput from tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->stop_output(mpAudioPolicy, output, stream, session);
+}
+
+void AudioPolicyService::releaseOutput(audio_io_handle_t output)
+{
+ if (mpAudioPolicy == NULL) {
+ return;
+ }
+ ALOGV("releaseOutput()");
+ mOutputCommandThread->releaseOutputCommand(output);
+}
+
+void AudioPolicyService::doReleaseOutput(audio_io_handle_t output)
+{
+ ALOGV("doReleaseOutput from tid %d", gettid());
+ Mutex::Autolock _l(mLock);
+ mpAudioPolicy->release_output(mpAudioPolicy, output);
+}
+
+audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ int audioSession)
+{
+ if (mpAudioPolicy == NULL) {
+ return 0;
+ }
+ // already checked by client, but double-check in case the client wrapper is bypassed
+ if (inputSource >= AUDIO_SOURCE_CNT && inputSource != AUDIO_SOURCE_HOTWORD) {
+ return 0;
+ }
+
+ if ((inputSource == AUDIO_SOURCE_HOTWORD) && !captureHotwordAllowed()) {
+ return 0;
+ }
+
+ Mutex::Autolock _l(mLock);
+ // the audio_in_acoustics_t parameter is ignored by get_input()
+ audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
+ format, channelMask, (audio_in_acoustics_t) 0);
+
+ if (input == 0) {
+ return input;
+ }
+ // create audio pre processors according to input source
+ audio_source_t aliasSource = (inputSource == AUDIO_SOURCE_HOTWORD) ?
+ AUDIO_SOURCE_VOICE_RECOGNITION : inputSource;
+
+ ssize_t index = mInputSources.indexOfKey(aliasSource);
+ if (index < 0) {
+ return input;
+ }
+ ssize_t idx = mInputs.indexOfKey(input);
+ InputDesc *inputDesc;
+ if (idx < 0) {
+ inputDesc = new InputDesc(audioSession);
+ mInputs.add(input, inputDesc);
+ } else {
+ inputDesc = mInputs.valueAt(idx);
+ }
+
+ Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
+ for (size_t i = 0; i < effects.size(); i++) {
+ EffectDesc *effect = effects[i];
+ sp<AudioEffect> fx = new AudioEffect(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
+ status_t status = fx->initCheck();
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGW("Failed to create Fx %s on input %d", effect->mName, input);
+ // fx goes out of scope and strong ref on AudioEffect is released
+ continue;
+ }
+ for (size_t j = 0; j < effect->mParams.size(); j++) {
+ fx->setParameter(effect->mParams[j]);
+ }
+ inputDesc->mEffects.add(fx);
+ }
+ setPreProcessorEnabled(inputDesc, true);
+ return input;
+}
+
+status_t AudioPolicyService::startInput(audio_io_handle_t input)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+
+ return mpAudioPolicy->start_input(mpAudioPolicy, input);
+}
+
+status_t AudioPolicyService::stopInput(audio_io_handle_t input)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+
+ return mpAudioPolicy->stop_input(mpAudioPolicy, input);
+}
+
+void AudioPolicyService::releaseInput(audio_io_handle_t input)
+{
+ if (mpAudioPolicy == NULL) {
+ return;
+ }
+ Mutex::Autolock _l(mLock);
+ mpAudioPolicy->release_input(mpAudioPolicy, input);
+
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ return;
+ }
+ InputDesc *inputDesc = mInputs.valueAt(index);
+ setPreProcessorEnabled(inputDesc, false);
+ delete inputDesc;
+ mInputs.removeItemsAt(index);
+}
+
+status_t AudioPolicyService::initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ return BAD_VALUE;
+ }
+ Mutex::Autolock _l(mLock);
+ mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax);
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ return BAD_VALUE;
+ }
+ Mutex::Autolock _l(mLock);
+ if (mpAudioPolicy->set_stream_volume_index_for_device) {
+ return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy,
+ stream,
+ index,
+ device);
+ } else {
+ return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index);
+ }
+}
+
+status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ return BAD_VALUE;
+ }
+ Mutex::Autolock _l(mLock);
+ if (mpAudioPolicy->get_stream_volume_index_for_device) {
+ return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy,
+ stream,
+ index,
+ device);
+ } else {
+ return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index);
+ }
+}
+
+uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream)
+{
+ if (mpAudioPolicy == NULL) {
+ return 0;
+ }
+ return mpAudioPolicy->get_strategy_for_stream(mpAudioPolicy, stream);
+}
+
+//audio policy: use audio_device_t appropriately
+
+audio_devices_t AudioPolicyService::getDevicesForStream(audio_stream_type_t stream)
+{
+ if (mpAudioPolicy == NULL) {
+ return (audio_devices_t)0;
+ }
+ return mpAudioPolicy->get_devices_for_stream(mpAudioPolicy, stream);
+}
+
+audio_io_handle_t AudioPolicyService::getOutputForEffect(const effect_descriptor_t *desc)
+{
+ // FIXME change return type to status_t, and return NO_INIT here
+ if (mpAudioPolicy == NULL) {
+ return 0;
+ }
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->get_output_for_effect(mpAudioPolicy, desc);
+}
+
+status_t AudioPolicyService::registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ return mpAudioPolicy->register_effect(mpAudioPolicy, desc, io, strategy, session, id);
+}
+
+status_t AudioPolicyService::unregisterEffect(int id)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ return mpAudioPolicy->unregister_effect(mpAudioPolicy, id);
+}
+
+status_t AudioPolicyService::setEffectEnabled(int id, bool enabled)
+{
+ if (mpAudioPolicy == NULL) {
+ return NO_INIT;
+ }
+ return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled);
+}
+
+bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+ if (mpAudioPolicy == NULL) {
+ return 0;
+ }
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->is_stream_active(mpAudioPolicy, stream, inPastMs);
+}
+
+bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+ if (mpAudioPolicy == NULL) {
+ return 0;
+ }
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs);
+}
+
+bool AudioPolicyService::isSourceActive(audio_source_t source) const
+{
+ if (mpAudioPolicy == NULL) {
+ return false;
+ }
+ if (mpAudioPolicy->is_source_active == 0) {
+ return false;
+ }
+ Mutex::Autolock _l(mLock);
+ return mpAudioPolicy->is_source_active(mpAudioPolicy, source);
+}
+
+status_t AudioPolicyService::queryDefaultPreProcessing(int audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count)
+{
+
+ if (mpAudioPolicy == NULL) {
+ *count = 0;
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+ status_t status = NO_ERROR;
+
+ size_t index;
+ for (index = 0; index < mInputs.size(); index++) {
+ if (mInputs.valueAt(index)->mSessionId == audioSession) {
+ break;
+ }
+ }
+ if (index == mInputs.size()) {
+ *count = 0;
+ return BAD_VALUE;
+ }
+ Vector< sp<AudioEffect> > effects = mInputs.valueAt(index)->mEffects;
+
+ for (size_t i = 0; i < effects.size(); i++) {
+ effect_descriptor_t desc = effects[i]->descriptor();
+ if (i < *count) {
+ descriptors[i] = desc;
+ }
+ }
+ if (effects.size() > *count) {
+ status = NO_MEMORY;
+ }
+ *count = effects.size();
+ return status;
+}
+
+bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
+{
+ if (mpAudioPolicy == NULL) {
+ ALOGV("mpAudioPolicy == NULL");
+ return false;
+ }
+
+ if (mpAudioPolicy->is_offload_supported == NULL) {
+ ALOGV("HAL does not implement is_offload_supported");
+ return false;
+ }
+
+ return mpAudioPolicy->is_offload_supported(mpAudioPolicy, &info);
+}
+
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
new file mode 100644
index 0000000..5ac9d9e
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -0,0 +1,4104 @@
+/*
+ * Copyright (C) 2009 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 "AudioPolicyManager"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+// A device mask for all audio input devices that are considered "virtual" when evaluating
+// active inputs in getActiveInput()
+#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+
+#include <utils/Log.h>
+#include "AudioPolicyManager.h"
+#include <hardware/audio_effect.h>
+#include <hardware/audio.h>
+#include <math.h>
+#include <hardware_legacy/audio_policy_conf.h>
+#include <cutils/properties.h>
+#include <media/AudioParameter.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// AudioPolicyInterface implementation
+// ----------------------------------------------------------------------------
+
+
+status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address)
+{
+ SortedVector <audio_io_handle_t> outputs;
+
+ ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
+
+ // connect/disconnect only 1 device at a time
+ if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+
+ if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
+ ALOGE("setDeviceConnectionState() invalid address: %s", device_address);
+ return BAD_VALUE;
+ }
+
+ // handle output devices
+ if (audio_is_output_device(device)) {
+
+ if (!mHasA2dp && audio_is_a2dp_device(device)) {
+ ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
+ return BAD_VALUE;
+ }
+ if (!mHasUsb && audio_is_usb_device(device)) {
+ ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
+ return BAD_VALUE;
+ }
+ if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
+ ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
+ return BAD_VALUE;
+ }
+
+ // save a copy of the opened output descriptors before any output is opened or closed
+ // by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
+ mPreviousOutputs = mOutputs;
+ String8 paramStr;
+ switch (state)
+ {
+ // handle output device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE:
+ if (mAvailableOutputDevices & device) {
+ ALOGW("setDeviceConnectionState() device already connected: %x", device);
+ return INVALID_OPERATION;
+ }
+ ALOGV("setDeviceConnectionState() connecting device %x", device);
+
+ if (mHasA2dp && audio_is_a2dp_device(device)) {
+ // handle A2DP device connection
+ AudioParameter param;
+ param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
+ paramStr = param.toString();
+ } else if (mHasUsb && audio_is_usb_device(device)) {
+ // handle USB device connection
+ paramStr = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+ }
+
+ if (checkOutputsForDevice(device, state, outputs, paramStr) != NO_ERROR) {
+ return INVALID_OPERATION;
+ }
+ ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs",
+ outputs.size());
+ // register new device as available
+ mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
+
+ if (mHasA2dp && audio_is_a2dp_device(device)) {
+ // handle A2DP device connection
+ mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+ mA2dpSuspended = false;
+ } else if (audio_is_bluetooth_sco_device(device)) {
+ // handle SCO device connection
+ mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+ } else if (mHasUsb && audio_is_usb_device(device)) {
+ // handle USB device connection
+ mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
+ }
+
+ break;
+ // handle output device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (!(mAvailableOutputDevices & device)) {
+ ALOGW("setDeviceConnectionState() device not connected: %x", device);
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("setDeviceConnectionState() disconnecting device %x", device);
+ // remove device from available output devices
+ mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
+
+ checkOutputsForDevice(device, state, outputs, paramStr);
+ if (mHasA2dp && audio_is_a2dp_device(device)) {
+ // handle A2DP device disconnection
+ mA2dpDeviceAddress = "";
+ mA2dpSuspended = false;
+ } else if (audio_is_bluetooth_sco_device(device)) {
+ // handle SCO device disconnection
+ mScoDeviceAddress = "";
+ } else if (mHasUsb && audio_is_usb_device(device)) {
+ // handle USB device disconnection
+ mUsbCardAndDevice = "";
+ }
+ // not currently handling multiple simultaneous submixes: ignoring remote submix
+ // case and address
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ // outputs must be closed after checkOutputForAllStrategies() is executed
+ if (!outputs.isEmpty()) {
+ for (size_t i = 0; i < outputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
+ // close unused outputs after device disconnection or direct outputs that have been
+ // opened by checkOutputsForDevice() to query dynamic parameters
+ if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+ (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+ (desc->mDirectOpenCount == 0))) {
+ closeOutput(outputs[i]);
+ }
+ }
+ }
+
+ updateDevicesAndOutputs();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ // do not force device change on duplicated output because if device is 0, it will
+ // also force a device 0 for the two outputs it is duplicated to which may override
+ // a valid device selection on those outputs.
+ setOutputDevice(mOutputs.keyAt(i),
+ getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
+ !mOutputs.valueAt(i)->isDuplicated(),
+ 0);
+ }
+
+ if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO ||
+ device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
+ device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else {
+ return NO_ERROR;
+ }
+ }
+ // handle input devices
+ if (audio_is_input_device(device)) {
+
+ switch (state)
+ {
+ // handle input device connection
+ case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
+ if (mAvailableInputDevices & device) {
+ ALOGW("setDeviceConnectionState() device already connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN);
+ }
+ break;
+
+ // handle input device disconnection
+ case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
+ if (!(mAvailableInputDevices & device)) {
+ ALOGW("setDeviceConnectionState() device not connected: %d", device);
+ return INVALID_OPERATION;
+ }
+ mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device);
+ } break;
+
+ default:
+ ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ return BAD_VALUE;
+ }
+
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+ audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+ if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+ ALOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
+ inputDesc->mDevice, newDevice, activeInput);
+ inputDesc->mDevice = newDevice;
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+ mpClientInterface->setParameters(activeInput, param.toString());
+ }
+ }
+
+ return NO_ERROR;
+ }
+
+ ALOGW("setDeviceConnectionState() invalid device: %x", device);
+ return BAD_VALUE;
+}
+
+audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
+ const char *device_address)
+{
+ audio_policy_dev_state_t state = AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
+ String8 address = String8(device_address);
+ if (audio_is_output_device(device)) {
+ if (device & mAvailableOutputDevices) {
+ if (audio_is_a2dp_device(device) &&
+ (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
+ return state;
+ }
+ if (audio_is_bluetooth_sco_device(device) &&
+ address != "" && mScoDeviceAddress != address) {
+ return state;
+ }
+ if (audio_is_usb_device(device) &&
+ (!mHasUsb || (address != "" && mUsbCardAndDevice != address))) {
+ ALOGE("getDeviceConnectionState() invalid device: %x", device);
+ return state;
+ }
+ if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) {
+ return state;
+ }
+ state = AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
+ }
+ } else if (audio_is_input_device(device)) {
+ if (device & mAvailableInputDevices) {
+ state = AUDIO_POLICY_DEVICE_STATE_AVAILABLE;
+ }
+ }
+
+ return state;
+}
+
+void AudioPolicyManager::setPhoneState(audio_mode_t state)
+{
+ ALOGV("setPhoneState() state %d", state);
+ audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+ if (state < 0 || state >= AUDIO_MODE_CNT) {
+ ALOGW("setPhoneState() invalid state %d", state);
+ return;
+ }
+
+ if (state == mPhoneState ) {
+ ALOGW("setPhoneState() setting same state %d", state);
+ return;
+ }
+
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isInCall()) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ handleIncallSonification((audio_stream_type_t)stream, false, true);
+ }
+ }
+
+ // store previous phone state for management of sonification strategy below
+ int oldState = mPhoneState;
+ mPhoneState = state;
+ bool force = false;
+
+ // are we entering or starting a call
+ if (!isStateInCall(oldState) && isStateInCall(state)) {
+ ALOGV(" Entering call in setPhoneState()");
+ // force routing command to audio hardware when starting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+ }
+ } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+ ALOGV(" Exiting call in setPhoneState()");
+ // force routing command to audio hardware when exiting a call
+ // even if no device change is needed
+ force = true;
+ for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
+ sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+ }
+ } else if (isStateInCall(state) && (state != oldState)) {
+ ALOGV(" Switching between telephony and VoIP in setPhoneState()");
+ // force routing command to audio hardware when switching between telephony and VoIP
+ // even if no device change is needed
+ force = true;
+ }
+
+ // check for device and output changes triggered by new phone state
+ newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+
+ AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
+ newDevice = hwOutputDesc->device();
+ }
+
+ int delayMs = 0;
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((desc->isStrategyActive(STRATEGY_MEDIA,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ desc->isStrategyActive(STRATEGY_SONIFICATION,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->mLatency*2)) {
+ delayMs = desc->mLatency*2;
+ }
+ setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+ setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+ }
+ }
+
+ // change routing is necessary
+ setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
+
+ // if entering in call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isStateInCall(state)) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ handleIncallSonification((audio_stream_type_t)stream, true, true);
+ }
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ if (state == AUDIO_MODE_RINGTONE &&
+ isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+}
+
+void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config)
+{
+ ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
+
+ bool forceVolumeReeval = false;
+ switch(usage) {
+ case AUDIO_POLICY_FORCE_FOR_COMMUNICATION:
+ if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
+ return;
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_MEDIA:
+ if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_NO_BT_A2DP) {
+ ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_RECORD:
+ if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_NONE) {
+ ALOGW("setForceUse() invalid config %d for FOR_RECORD", config);
+ return;
+ }
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_DOCK:
+ if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK &&
+ config != AUDIO_POLICY_FORCE_BT_DESK_DOCK &&
+ config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY &&
+ config != AUDIO_POLICY_FORCE_ANALOG_DOCK &&
+ config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) {
+ ALOGW("setForceUse() invalid config %d for FOR_DOCK", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ case AUDIO_POLICY_FORCE_FOR_SYSTEM:
+ if (config != AUDIO_POLICY_FORCE_NONE &&
+ config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config);
+ }
+ forceVolumeReeval = true;
+ mForceUse[usage] = config;
+ break;
+ default:
+ ALOGW("setForceUse() invalid usage %d", usage);
+ break;
+ }
+
+ // check for device and output changes triggered by new force usage
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t output = mOutputs.keyAt(i);
+ audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
+ setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
+ if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
+ applyStreamVolumes(output, newDevice, 0, true);
+ }
+ }
+
+ audio_io_handle_t activeInput = getActiveInput();
+ if (activeInput != 0) {
+ AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
+ audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+ if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+ ALOGV("setForceUse() changing device from %x to %x for input %d",
+ inputDesc->mDevice, newDevice, activeInput);
+ inputDesc->mDevice = newDevice;
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
+ mpClientInterface->setParameters(activeInput, param.toString());
+ }
+ }
+
+}
+
+audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage)
+{
+ return mForceUse[usage];
+}
+
+void AudioPolicyManager::setSystemProperty(const char* property, const char* value)
+{
+ ALOGV("setSystemProperty() property %s, value %s", property, value);
+}
+
+// Find a direct output profile compatible with the parameters passed, even if the input flags do
+// not explicitly request a direct output
+AudioPolicyManager::IOProfile *AudioPolicyManager::getProfileForDirectOutput(
+ audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags)
+{
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
+ IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+ if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (profile->isCompatibleProfile(device, samplingRate, format,
+ channelMask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
+ if (mAvailableOutputDevices & profile->mSupportedDevices) {
+ return mHwModules[i]->mOutputProfiles[j];
+ }
+ }
+ } else {
+ if (profile->isCompatibleProfile(device, samplingRate, format,
+ channelMask,
+ AUDIO_OUTPUT_FLAG_DIRECT)) {
+ if (mAvailableOutputDevices & profile->mSupportedDevices) {
+ return mHwModules[i]->mOutputProfiles[j];
+ }
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
+{
+ audio_io_handle_t output = 0;
+ uint32_t latency = 0;
+ routing_strategy strategy = getStrategy(stream);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
+ device, stream, samplingRate, format, channelMask, flags);
+
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mSamplingRate = mTestSamplingRate;
+ outputDesc->mFormat = mTestFormat;
+ outputDesc->mChannelMask = mTestChannels;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags =
+ (audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags,
+ offloadInfo);
+ if (mTestOutputs[mCurOutput]) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
+ }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+
+ // open a direct output if required by specified parameters
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ IOProfile *profile = NULL;
+ if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
+ !isNonOffloadableEffectEnabled()) {
+ profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ }
+
+ if (profile != NULL) {
+ AudioOutputDescriptor *outputDesc = NULL;
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mId);
+ }
+ outputDesc = new AudioOutputDescriptor(profile);
+ outputDesc->mDevice = device;
+ outputDesc->mSamplingRate = samplingRate;
+ outputDesc->mFormat = format;
+ outputDesc->mChannelMask = channelMask;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+ output = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags,
+ offloadInfo);
+
+ // only accept an output with the requested parameters
+ if (output == 0 ||
+ (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+ (format != AUDIO_FORMAT_DEFAULT && format != outputDesc->mFormat) ||
+ (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != 0) {
+ mpClientInterface->closeOutput(output);
+ }
+ delete outputDesc;
+ return 0;
+ }
+ audio_io_handle_t srcOutput = getOutputForEffect();
+ addOutput(output, outputDesc);
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput == output) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
+ }
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ return output;
+ }
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // for non direct outputs, only PCM is supported
+ if (audio_is_linear_pcm(format)) {
+ // get which output is suitable for the specified stream. The actual
+ // routing change will happen when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ output = selectOutput(outputs, flags);
+ }
+ ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGV("getOutput() returns output %d", output);
+
+ return output;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags)
+{
+ // select one output among several that provide a path to a particular device or set of
+ // devices (the list was previously build by getOutputsForDevice()).
+ // The priority is as follows:
+ // 1: the output with the highest number of requested policy flags
+ // 2: the primary output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+ if (outputs.size() == 1) {
+ return outputs[0];
+ }
+
+ int maxCommonFlags = 0;
+ audio_io_handle_t outputFlags = 0;
+ audio_io_handle_t outputPrimary = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
+ if (!outputDesc->isDuplicated()) {
+ int commonFlags = popcount(outputDesc->mProfile->mFlags & flags);
+ if (commonFlags > maxCommonFlags) {
+ outputFlags = outputs[i];
+ maxCommonFlags = commonFlags;
+ ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
+ }
+ if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ outputPrimary = outputs[i];
+ }
+ }
+ }
+
+ if (outputFlags != 0) {
+ return outputFlags;
+ }
+ if (outputPrimary != 0) {
+ return outputPrimary;
+ }
+
+ return outputs[0];
+}
+
+status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session)
+{
+ ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("startOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ if (outputDesc->mRefCount[stream] == 1) {
+ audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+ routing_strategy strategy = getStrategy(stream);
+ bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+ (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
+ uint32_t waitMs = 0;
+ bool force = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // force a device change if any other output is managed by the same hw
+ // module and has a current device selection that differs from selected device.
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other active output.
+ if (outputDesc->sharesHwModuleWith(desc) &&
+ desc->device() != newDevice) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate.
+ uint32_t latency = desc->latency();
+ if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+ waitMs = latency;
+ }
+ }
+ }
+ uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream,
+ mStreams[stream].getVolumeIndex(newDevice),
+ output,
+ newDevice);
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+ if (waitMs > muteWaitMs) {
+ usleep((waitMs - muteWaitMs) * 2 * 1000);
+ }
+ }
+ return NO_ERROR;
+}
+
+
+status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session)
+{
+ ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("stopOutput() unknown output %d", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, false, false);
+ }
+
+ if (outputDesc->mRefCount[stream] > 0) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->mRefCount[stream] == 0) {
+ outputDesc->mStopTime[stream] = systemTime();
+ audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (curOutput != output &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ (newDevice != desc->device())) {
+ setOutputDevice(curOutput,
+ getNewDevice(curOutput, false /*fromCache*/),
+ true,
+ outputDesc->mLatency*2);
+ }
+ }
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
+ }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0 for output %d", output);
+ return INVALID_OPERATION;
+ }
+}
+
+void AudioPolicyManager::releaseOutput(audio_io_handle_t output)
+{
+ ALOGV("releaseOutput() %d", output);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("releaseOutput() releasing unknown output %d", output);
+ return;
+ }
+
+#ifdef AUDIO_POLICY_TEST
+ int testIndex = testOutputIndex(output);
+ if (testIndex != 0) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->isActive()) {
+ mpClientInterface->closeOutput(output);
+ delete mOutputs.valueAt(index);
+ mOutputs.removeItem(output);
+ mTestOutputs[testIndex] = 0;
+ }
+ return;
+ }
+#endif //AUDIO_POLICY_TEST
+
+ AudioOutputDescriptor *desc = mOutputs.valueAt(index);
+ if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ if (desc->mDirectOpenCount <= 0) {
+ ALOGW("releaseOutput() invalid open count %d for output %d",
+ desc->mDirectOpenCount, output);
+ return;
+ }
+ if (--desc->mDirectOpenCount == 0) {
+ closeOutput(output);
+ // If effects where present on the output, audioflinger moved them to the primary
+ // output by default: move them back to the appropriate output.
+ audio_io_handle_t dstOutput = getOutputForEffect();
+ if (dstOutput != mPrimaryOutput) {
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput);
+ }
+ }
+ }
+}
+
+
+audio_io_handle_t AudioPolicyManager::getInput(audio_source_t inputSource,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_in_acoustics_t acoustics)
+{
+ audio_io_handle_t input = 0;
+ audio_devices_t device = getDeviceForInputSource(inputSource);
+
+ ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
+ inputSource, samplingRate, format, channelMask, acoustics);
+
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGW("getInput() could not find device for inputSource %d", inputSource);
+ return 0;
+ }
+
+ // adapt channel selection to input source
+ switch(inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ default:
+ break;
+ }
+
+ IOProfile *profile = getInputProfile(device,
+ samplingRate,
+ format,
+ channelMask);
+ if (profile == NULL) {
+ ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d, "
+ "channelMask %04x",
+ device, samplingRate, format, channelMask);
+ return 0;
+ }
+
+ if (profile->mModule->mHandle == 0) {
+ ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
+ return 0;
+ }
+
+ AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
+
+ inputDesc->mInputSource = inputSource;
+ inputDesc->mDevice = device;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = format;
+ inputDesc->mChannelMask = channelMask;
+ inputDesc->mRefCount = 0;
+ input = mpClientInterface->openInput(profile->mModule->mHandle,
+ &inputDesc->mDevice,
+ &inputDesc->mSamplingRate,
+ &inputDesc->mFormat,
+ &inputDesc->mChannelMask);
+
+ // only accept input with the exact requested set of parameters
+ if (input == 0 ||
+ (samplingRate != inputDesc->mSamplingRate) ||
+ (format != inputDesc->mFormat) ||
+ (channelMask != inputDesc->mChannelMask)) {
+ ALOGI("getInput() failed opening input: samplingRate %d, format %d, channelMask %x",
+ samplingRate, format, channelMask);
+ if (input != 0) {
+ mpClientInterface->closeInput(input);
+ }
+ delete inputDesc;
+ return 0;
+ }
+ mInputs.add(input, inputDesc);
+ return input;
+}
+
+status_t AudioPolicyManager::startInput(audio_io_handle_t input)
+{
+ ALOGV("startInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("startInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+#ifdef AUDIO_POLICY_TEST
+ if (mTestInput == 0)
+#endif //AUDIO_POLICY_TEST
+ {
+ // refuse 2 active AudioRecord clients at the same time except if the active input
+ // uses AUDIO_SOURCE_HOTWORD in which case it is closed.
+ audio_io_handle_t activeInput = getActiveInput();
+ if (!isVirtualInputDevice(inputDesc->mDevice) && activeInput != 0) {
+ AudioInputDescriptor *activeDesc = mInputs.valueFor(activeInput);
+ if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
+ ALOGW("startInput() preempting already started low-priority input %d", activeInput);
+ stopInput(activeInput);
+ releaseInput(activeInput);
+ } else {
+ ALOGW("startInput() input %d failed: other input already started", input);
+ return INVALID_OPERATION;
+ }
+ }
+ }
+
+ audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+ if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+ inputDesc->mDevice = newDevice;
+ }
+
+ // automatically enable the remote submix output when input is started
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+ }
+
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
+
+ int aliasSource = (inputDesc->mInputSource == AUDIO_SOURCE_HOTWORD) ?
+ AUDIO_SOURCE_VOICE_RECOGNITION : inputDesc->mInputSource;
+
+ param.addInt(String8(AudioParameter::keyInputSource), aliasSource);
+ ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
+
+ mpClientInterface->setParameters(input, param.toString());
+
+ inputDesc->mRefCount = 1;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::stopInput(audio_io_handle_t input)
+{
+ ALOGV("stopInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("stopInput() unknown input %d", input);
+ return BAD_VALUE;
+ }
+ AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+ if (inputDesc->mRefCount == 0) {
+ ALOGW("stopInput() input %d already stopped", input);
+ return INVALID_OPERATION;
+ } else {
+ // automatically disable the remote submix output when input is stopped
+ if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+ }
+
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), 0);
+ mpClientInterface->setParameters(input, param.toString());
+ inputDesc->mRefCount = 0;
+ return NO_ERROR;
+ }
+}
+
+void AudioPolicyManager::releaseInput(audio_io_handle_t input)
+{
+ ALOGV("releaseInput() %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("releaseInput() releasing unknown input %d", input);
+ return;
+ }
+ mpClientInterface->closeInput(input);
+ delete mInputs.valueAt(index);
+ mInputs.removeItem(input);
+ ALOGV("releaseInput() exit");
+}
+
+void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax)
+{
+ ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ if (indexMin < 0 || indexMin >= indexMax) {
+ ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
+ return;
+ }
+ mStreams[stream].mIndexMin = indexMin;
+ mStreams[stream].mIndexMax = indexMax;
+}
+
+status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
+{
+
+ if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+
+ // Force max volume if stream cannot be muted
+ if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+ ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
+ stream, device, index);
+
+ // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
+ // clear all device specific values
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ mStreams[stream].mIndexCur.clear();
+ }
+ mStreams[stream].mIndexCur.add(device, index);
+
+ // compute and apply stream volume on all outputs according to connected device
+ status_t status = NO_ERROR;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_devices_t curDevice =
+ getDeviceForVolume(mOutputs.valueAt(i)->device());
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) {
+ status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+ if (volStatus != NO_ERROR) {
+ status = volStatus;
+ }
+ }
+ }
+ return status;
+}
+
+status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
+{
+ if (index == NULL) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+ // if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
+ // the strategy the stream belongs to.
+ if (device == AUDIO_DEVICE_OUT_DEFAULT) {
+ device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
+ }
+ device = getDeviceForVolume(device);
+
+ *index = mStreams[stream].getVolumeIndex(device);
+ ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
+ return NO_ERROR;
+}
+
+audio_io_handle_t AudioPolicyManager::selectOutputForEffects(
+ const SortedVector<audio_io_handle_t>& outputs)
+{
+ // select one output among several suitable for global effects.
+ // The priority is as follows:
+ // 1: An offloaded output. If the effect ends up not being offloadable,
+ // AudioFlinger will invalidate the track and the offloaded output
+ // will be closed causing the effect to be moved to a PCM output.
+ // 2: A deep buffer output
+ // 3: the first output in the list
+
+ if (outputs.size() == 0) {
+ return 0;
+ }
+
+ audio_io_handle_t outputOffloaded = 0;
+ audio_io_handle_t outputDeepBuffer = 0;
+
+ for (size_t i = 0; i < outputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
+ ALOGV("selectOutputForEffects outputs[%d] flags %x", i, desc->mFlags);
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ outputOffloaded = outputs[i];
+ }
+ if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) != 0) {
+ outputDeepBuffer = outputs[i];
+ }
+ }
+
+ ALOGV("selectOutputForEffects outputOffloaded %d outputDeepBuffer %d",
+ outputOffloaded, outputDeepBuffer);
+ if (outputOffloaded != 0) {
+ return outputOffloaded;
+ }
+ if (outputDeepBuffer != 0) {
+ return outputDeepBuffer;
+ }
+
+ return outputs[0];
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForEffect(const effect_descriptor_t *desc)
+{
+ // apply simple rule where global effects are attached to the same output as MUSIC streams
+
+ routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
+
+ audio_io_handle_t output = selectOutputForEffects(dstOutputs);
+ ALOGV("getOutputForEffect() got output %d for fx %s flags %x",
+ output, (desc == NULL) ? "unspecified" : desc->name, (desc == NULL) ? 0 : desc->flags);
+
+ return output;
+}
+
+status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id)
+{
+ ssize_t index = mOutputs.indexOfKey(io);
+ if (index < 0) {
+ index = mInputs.indexOfKey(io);
+ if (index < 0) {
+ ALOGW("registerEffect() unknown io %d", io);
+ return INVALID_OPERATION;
+ }
+ }
+
+ if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
+ ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
+ desc->name, desc->memoryUsage);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsMemory += desc->memoryUsage;
+ ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
+ desc->name, io, strategy, session, id);
+ ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
+
+ EffectDescriptor *pDesc = new EffectDescriptor();
+ memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
+ pDesc->mIo = io;
+ pDesc->mStrategy = (routing_strategy)strategy;
+ pDesc->mSession = session;
+ pDesc->mEnabled = false;
+
+ mEffects.add(id, pDesc);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterEffect(int id)
+{
+ ssize_t index = mEffects.indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ EffectDescriptor *pDesc = mEffects.valueAt(index);
+
+ setEffectEnabled(pDesc, false);
+
+ if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
+ ALOGW("unregisterEffect() memory %d too big for total %d",
+ pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+ pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
+ }
+ mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
+ ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
+ pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
+
+ mEffects.removeItem(id);
+ delete pDesc;
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled)
+{
+ ssize_t index = mEffects.indexOfKey(id);
+ if (index < 0) {
+ ALOGW("unregisterEffect() unknown effect ID %d", id);
+ return INVALID_OPERATION;
+ }
+
+ return setEffectEnabled(mEffects.valueAt(index), enabled);
+}
+
+status_t AudioPolicyManager::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
+{
+ if (enabled == pDesc->mEnabled) {
+ ALOGV("setEffectEnabled(%s) effect already %s",
+ enabled?"true":"false", enabled?"enabled":"disabled");
+ return INVALID_OPERATION;
+ }
+
+ if (enabled) {
+ if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
+ ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
+ pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
+ return INVALID_OPERATION;
+ }
+ mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
+ } else {
+ if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
+ ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
+ pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
+ pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
+ }
+ mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
+ ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
+ }
+ pDesc->mEnabled = enabled;
+ return NO_ERROR;
+}
+
+bool AudioPolicyManager::isNonOffloadableEffectEnabled()
+{
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ const EffectDescriptor * const pDesc = mEffects.valueAt(i);
+ if (pDesc->mEnabled && (pDesc->mStrategy == STRATEGY_MEDIA) &&
+ ((pDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) {
+ ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d",
+ pDesc->mDesc.name, pDesc->mSession);
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream,
+ uint32_t inPastMs) const
+{
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+ outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::isSourceActive(audio_source_t source) const
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
+ if ((inputDescriptor->mInputSource == (int)source ||
+ (source == AUDIO_SOURCE_VOICE_RECOGNITION &&
+ inputDescriptor->mInputSource == AUDIO_SOURCE_HOTWORD))
+ && (inputDescriptor->mRefCount > 0)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+
+status_t AudioPolicyManager::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbCardAndDevice.string());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for communications %d\n",
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+
+ snprintf(buffer, SIZE, "\nHW Modules dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ snprintf(buffer, SIZE, "- HW Module %d:\n", i + 1);
+ write(fd, buffer, strlen(buffer));
+ mHwModules[i]->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nOutputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mOutputs.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nInputs dump:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mInputs.valueAt(i)->dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nStreams dump:\n");
+ write(fd, buffer, strlen(buffer));
+ snprintf(buffer, SIZE,
+ " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
+ write(fd, buffer, strlen(buffer));
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02d ", i);
+ write(fd, buffer, strlen(buffer));
+ mStreams[i].dump(fd);
+ }
+
+ snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
+ (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
+ write(fd, buffer, strlen(buffer));
+
+ snprintf(buffer, SIZE, "Registered effects:\n");
+ write(fd, buffer, strlen(buffer));
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
+ write(fd, buffer, strlen(buffer));
+ mEffects.valueAt(i)->dump(fd);
+ }
+
+
+ return NO_ERROR;
+}
+
+// This function checks for the parameters which can be offloaded.
+// This can be enhanced depending on the capability of the DSP and policy
+// of the system.
+bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadInfo)
+{
+ ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
+ " BitRate=%u, duration=%lld us, has_video=%d",
+ offloadInfo.sample_rate, offloadInfo.channel_mask,
+ offloadInfo.format,
+ offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
+ offloadInfo.has_video);
+
+ // Check if offload has been disabled
+ char propValue[PROPERTY_VALUE_MAX];
+ if (property_get("audio.offload.disable", propValue, "0")) {
+ if (atoi(propValue) != 0) {
+ ALOGV("offload disabled by audio.offload.disable=%s", propValue );
+ return false;
+ }
+ }
+
+ // Check if stream type is music, then only allow offload as of now.
+ if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
+ {
+ ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
+ return false;
+ }
+
+ //TODO: enable audio offloading with video when ready
+ if (offloadInfo.has_video)
+ {
+ ALOGV("isOffloadSupported: has_video == true, returning false");
+ return false;
+ }
+
+ //If duration is less than minimum value defined in property, return false
+ if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
+ if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
+ ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
+ return false;
+ }
+ } else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
+ ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
+ return false;
+ }
+
+ // Do not allow offloading if one non offloadable effect is enabled. This prevents from
+ // creating an offloaded track and tearing it down immediately after start when audioflinger
+ // detects there is an active non offloadable effect.
+ // FIXME: We should check the audio session here but we do not have it in this context.
+ // This may prevent offloading in rare situations where effects are left active by apps
+ // in the background.
+ if (isNonOffloadableEffectEnabled()) {
+ return false;
+ }
+
+ // See if there is a profile to support this.
+ // AUDIO_DEVICE_NONE
+ IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ offloadInfo.sample_rate,
+ offloadInfo.format,
+ offloadInfo.channel_mask,
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
+ return (profile != NULL);
+}
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager
+// ----------------------------------------------------------------------------
+
+AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+ :
+#ifdef AUDIO_POLICY_TEST
+ Thread(false),
+#endif //AUDIO_POLICY_TEST
+ mPrimaryOutput((audio_io_handle_t)0),
+ mAvailableOutputDevices(AUDIO_DEVICE_NONE),
+ mPhoneState(AUDIO_MODE_NORMAL),
+ mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
+ mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
+ mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false),
+ mSpeakerDrcEnabled(false)
+{
+ mpClientInterface = clientInterface;
+
+ for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) {
+ mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
+ }
+
+ mA2dpDeviceAddress = String8("");
+ mScoDeviceAddress = String8("");
+ mUsbCardAndDevice = String8("");
+
+ if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
+ if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
+ ALOGE("could not load audio policy configuration file, setting defaults");
+ defaultAudioPolicyConfig();
+ }
+ }
+
+ // must be done after reading the policy
+ initializeVolumeCurves();
+
+ // open all output streams needed to access attached devices
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
+ if (mHwModules[i]->mHandle == 0) {
+ ALOGW("could not open HW module %s", mHwModules[i]->mName);
+ continue;
+ }
+ // open all output streams needed to access attached devices
+ // except for direct output streams that are only opened when they are actually
+ // required by an app.
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j];
+
+ if ((outProfile->mSupportedDevices & mAttachedOutputDevices) &&
+ ((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
+ outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice &
+ outProfile->mSupportedDevices);
+ audio_io_handle_t output = mpClientInterface->openOutput(
+ outProfile->mModule->mHandle,
+ &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (output == 0) {
+ delete outputDesc;
+ } else {
+ mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices |
+ (outProfile->mSupportedDevices & mAttachedOutputDevices));
+ if (mPrimaryOutput == 0 &&
+ outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ mPrimaryOutput = output;
+ }
+ addOutput(output, outputDesc);
+ setOutputDevice(output,
+ (audio_devices_t)(mDefaultOutputDevice &
+ outProfile->mSupportedDevices),
+ true);
+ }
+ }
+ }
+ }
+
+ ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices),
+ "Not output found for attached devices %08x",
+ (mAttachedOutputDevices & ~mAvailableOutputDevices));
+
+ ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
+
+ updateDevicesAndOutputs();
+
+#ifdef AUDIO_POLICY_TEST
+ if (mPrimaryOutput != 0) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+
+ mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ mTestSamplingRate = 44100;
+ mTestFormat = AUDIO_FORMAT_PCM_16_BIT;
+ mTestChannels = AUDIO_CHANNEL_OUT_STEREO;
+ mTestLatencyMs = 0;
+ mCurOutput = 0;
+ mDirectOutput = false;
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ mTestOutputs[i] = 0;
+ }
+
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ snprintf(buffer, SIZE, "AudioPolicyManagerTest");
+ run(buffer, ANDROID_PRIORITY_AUDIO);
+ }
+#endif //AUDIO_POLICY_TEST
+}
+
+AudioPolicyManager::~AudioPolicyManager()
+{
+#ifdef AUDIO_POLICY_TEST
+ exit();
+#endif //AUDIO_POLICY_TEST
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ mpClientInterface->closeOutput(mOutputs.keyAt(i));
+ delete mOutputs.valueAt(i);
+ }
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ mpClientInterface->closeInput(mInputs.keyAt(i));
+ delete mInputs.valueAt(i);
+ }
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ delete mHwModules[i];
+ }
+}
+
+status_t AudioPolicyManager::initCheck()
+{
+ return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
+}
+
+#ifdef AUDIO_POLICY_TEST
+bool AudioPolicyManager::threadLoop()
+{
+ ALOGV("entering threadLoop()");
+ while (!exitPending())
+ {
+ String8 command;
+ int valueInt;
+ String8 value;
+
+ Mutex::Autolock _l(mLock);
+ mWaitWorkCV.waitRelative(mLock, milliseconds(50));
+
+ command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
+ AudioParameter param = AudioParameter(command);
+
+ if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
+ valueInt != 0) {
+ ALOGV("Test command %s received", command.string());
+ String8 target;
+ if (param.get(String8("target"), target) != NO_ERROR) {
+ target = "Manager";
+ }
+ if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_output"));
+ mCurOutput = valueInt;
+ }
+ if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_direct"));
+ if (value == "false") {
+ mDirectOutput = false;
+ } else if (value == "true") {
+ mDirectOutput = true;
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_input"));
+ mTestInput = valueInt;
+ }
+
+ if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_format"));
+ int format = AUDIO_FORMAT_INVALID;
+ if (value == "PCM 16 bits") {
+ format = AUDIO_FORMAT_PCM_16_BIT;
+ } else if (value == "PCM 8 bits") {
+ format = AUDIO_FORMAT_PCM_8_BIT;
+ } else if (value == "Compressed MP3") {
+ format = AUDIO_FORMAT_MP3;
+ }
+ if (format != AUDIO_FORMAT_INVALID) {
+ if (target == "Manager") {
+ mTestFormat = format;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("format"), format);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_channels"));
+ int channels = 0;
+
+ if (value == "Channels Stereo") {
+ channels = AUDIO_CHANNEL_OUT_STEREO;
+ } else if (value == "Channels Mono") {
+ channels = AUDIO_CHANNEL_OUT_MONO;
+ }
+ if (channels != 0) {
+ if (target == "Manager") {
+ mTestChannels = channels;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("channels"), channels);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+ if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_sampleRate"));
+ if (valueInt >= 0 && valueInt <= 96000) {
+ int samplingRate = valueInt;
+ if (target == "Manager") {
+ mTestSamplingRate = samplingRate;
+ } else if (mTestOutputs[mCurOutput] != 0) {
+ AudioParameter outputParam = AudioParameter();
+ outputParam.addInt(String8("sampling_rate"), samplingRate);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
+ }
+ }
+ }
+
+ if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
+ param.remove(String8("test_cmd_policy_reopen"));
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ mpClientInterface->closeOutput(mPrimaryOutput);
+
+ audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
+
+ delete mOutputs.valueFor(mPrimaryOutput);
+ mOutputs.removeItem(mPrimaryOutput);
+
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
+ &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (mPrimaryOutput == 0) {
+ ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
+ outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
+ } else {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"), 0);
+ mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
+ addOutput(mPrimaryOutput, outputDesc);
+ }
+ }
+
+
+ mpClientInterface->setParameters(0, String8("test_cmd_policy="));
+ }
+ }
+ return false;
+}
+
+void AudioPolicyManager::exit()
+{
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+int AudioPolicyManager::testOutputIndex(audio_io_handle_t output)
+{
+ for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
+ if (output == mTestOutputs[i]) return i;
+ }
+ return 0;
+}
+#endif //AUDIO_POLICY_TEST
+
+// ---
+
+void AudioPolicyManager::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
+{
+ outputDesc->mId = id;
+ mOutputs.add(id, outputDesc);
+}
+
+
+status_t AudioPolicyManager::checkOutputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 paramStr)
+{
+ AudioOutputDescriptor *desc;
+
+ if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
+ // first list already open outputs that can be routed to this device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) {
+ ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ }
+ }
+ // then look for output profiles that can be routed to this device
+ SortedVector<IOProfile *> profiles;
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) {
+ ALOGV("checkOutputsForDevice(): adding profile %d from module %d", j, i);
+ profiles.add(mHwModules[i]->mOutputProfiles[j]);
+ }
+ }
+ }
+
+ if (profiles.isEmpty() && outputs.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+
+ // open outputs for matching profiles if needed. Direct outputs are also opened to
+ // query for dynamic parameters and will be closed later by setDeviceConnectionState()
+ for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
+ IOProfile *profile = profiles[profile_index];
+
+ // nothing to do if one output is already opened for this profile
+ size_t j;
+ for (j = 0; j < mOutputs.size(); j++) {
+ desc = mOutputs.valueAt(j);
+ if (!desc->isDuplicated() && desc->mProfile == profile) {
+ break;
+ }
+ }
+ if (j != mOutputs.size()) {
+ continue;
+ }
+
+ ALOGV("opening output for device %08x with params %s", device, paramStr.string());
+ desc = new AudioOutputDescriptor(profile);
+ desc->mDevice = device;
+ audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
+ offloadInfo.sample_rate = desc->mSamplingRate;
+ offloadInfo.format = desc->mFormat;
+ offloadInfo.channel_mask = desc->mChannelMask;
+
+ audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &desc->mDevice,
+ &desc->mSamplingRate,
+ &desc->mFormat,
+ &desc->mChannelMask,
+ &desc->mLatency,
+ desc->mFlags,
+ &offloadInfo);
+ if (output != 0) {
+ if (!paramStr.isEmpty()) {
+ mpClientInterface->setParameters(output, paramStr);
+ }
+
+ if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+ String8 reply;
+ char *value;
+ if (profile->mSamplingRates[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
+ ALOGV("checkOutputsForDevice() direct output sup sampling rates %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ loadSamplingRates(value + 1, profile);
+ }
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
+ ALOGV("checkOutputsForDevice() direct output sup formats %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ loadFormats(value + 1, profile);
+ }
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ reply = mpClientInterface->getParameters(output,
+ String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
+ ALOGV("checkOutputsForDevice() direct output sup channel masks %s",
+ reply.string());
+ value = strpbrk((char *)reply.string(), "=");
+ if (value != NULL) {
+ loadOutChannels(value + 1, profile);
+ }
+ }
+ if (((profile->mSamplingRates[0] == 0) &&
+ (profile->mSamplingRates.size() < 2)) ||
+ ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+ (profile->mFormats.size() < 2)) ||
+ ((profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) &&
+ (profile->mChannelMasks.size() < 2))) {
+ ALOGW("checkOutputsForDevice() direct output missing param");
+ mpClientInterface->closeOutput(output);
+ output = 0;
+ } else {
+ addOutput(output, desc);
+ }
+ } else {
+ audio_io_handle_t duplicatedOutput = 0;
+ // add output descriptor
+ addOutput(output, desc);
+ // set initial stream volume for device
+ applyStreamVolumes(output, device, 0, true);
+
+ //TODO: configure audio effect output stage here
+
+ // open a duplicating output thread for the new output and the primary output
+ duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
+ mPrimaryOutput);
+ if (duplicatedOutput != 0) {
+ // add duplicated output descriptor
+ AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL);
+ dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
+ dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
+ dupOutputDesc->mSamplingRate = desc->mSamplingRate;
+ dupOutputDesc->mFormat = desc->mFormat;
+ dupOutputDesc->mChannelMask = desc->mChannelMask;
+ dupOutputDesc->mLatency = desc->mLatency;
+ addOutput(duplicatedOutput, dupOutputDesc);
+ applyStreamVolumes(duplicatedOutput, device, 0, true);
+ } else {
+ ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
+ mPrimaryOutput, output);
+ mpClientInterface->closeOutput(output);
+ mOutputs.removeItem(output);
+ output = 0;
+ }
+ }
+ }
+ if (output == 0) {
+ ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+ delete desc;
+ profiles.removeAt(profile_index);
+ profile_index--;
+ } else {
+ outputs.add(output);
+ ALOGV("checkOutputsForDevice(): adding output %d", output);
+ }
+ }
+
+ if (profiles.isEmpty()) {
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ return BAD_VALUE;
+ }
+ } else {
+ // check if one opened output is not needed any more after disconnecting one device
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() &&
+ !(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) {
+ ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i));
+ outputs.add(mOutputs.keyAt(i));
+ }
+ }
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
+ {
+ IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+ if ((profile->mSupportedDevices & device) &&
+ (profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
+ ALOGV("checkOutputsForDevice(): clearing direct output profile %d on module %d",
+ j, i);
+ if (profile->mSamplingRates[0] == 0) {
+ profile->mSamplingRates.clear();
+ profile->mSamplingRates.add(0);
+ }
+ if (profile->mFormats[0] == AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.clear();
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ }
+ if (profile->mChannelMasks[0] == 0) {
+ profile->mChannelMasks.clear();
+ profile->mChannelMasks.add(0);
+ }
+ }
+ }
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::closeOutput(audio_io_handle_t output)
+{
+ ALOGV("closeOutput(%d)", output);
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ if (outputDesc == NULL) {
+ ALOGW("closeOutput() unknown output %d", output);
+ return;
+ }
+
+ // look for duplicated outputs connected to the output being removed.
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
+ if (dupOutputDesc->isDuplicated() &&
+ (dupOutputDesc->mOutput1 == outputDesc ||
+ dupOutputDesc->mOutput2 == outputDesc)) {
+ AudioOutputDescriptor *outputDesc2;
+ if (dupOutputDesc->mOutput1 == outputDesc) {
+ outputDesc2 = dupOutputDesc->mOutput2;
+ } else {
+ outputDesc2 = dupOutputDesc->mOutput1;
+ }
+ // As all active tracks on duplicated output will be deleted,
+ // and as they were also referenced on the other output, the reference
+ // count for their stream type must be adjusted accordingly on
+ // the other output.
+ for (int j = 0; j < AUDIO_STREAM_CNT; j++) {
+ int refCount = dupOutputDesc->mRefCount[j];
+ outputDesc2->changeRefCount((audio_stream_type_t)j,-refCount);
+ }
+ audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
+ ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
+
+ mpClientInterface->closeOutput(duplicatedOutput);
+ delete mOutputs.valueFor(duplicatedOutput);
+ mOutputs.removeItem(duplicatedOutput);
+ }
+ }
+
+ AudioParameter param;
+ param.add(String8("closing"), String8("true"));
+ mpClientInterface->setParameters(output, param.toString());
+
+ mpClientInterface->closeOutput(output);
+ delete outputDesc;
+ mOutputs.removeItem(output);
+ mPreviousOutputs = mOutputs;
+}
+
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device,
+ DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs)
+{
+ SortedVector<audio_io_handle_t> outputs;
+
+ ALOGVV("getOutputsForDevice() device %04x", device);
+ for (size_t i = 0; i < openOutputs.size(); i++) {
+ ALOGVV("output %d isDuplicated=%d device=%04x",
+ i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
+ if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
+ ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+ outputs.add(openOutputs.keyAt(i));
+ }
+ }
+ return outputs;
+}
+
+bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2)
+{
+ if (outputs1.size() != outputs2.size()) {
+ return false;
+ }
+ for (size_t i = 0; i < outputs1.size(); i++) {
+ if (outputs1[i] != outputs2[i]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
+{
+ audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
+ audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+
+ if (!vectorsEqual(srcOutputs,dstOutputs)) {
+ ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
+ strategy, srcOutputs[0], dstOutputs[0]);
+ // mute strategy while moving tracks from one output to another
+ for (size_t i = 0; i < srcOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]);
+ if (desc->isStrategyActive(strategy)) {
+ setStrategyMute(strategy, true, srcOutputs[i]);
+ setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
+ }
+ }
+
+ // Move effects associated to this strategy from previous output to new output
+ if (strategy == STRATEGY_MEDIA) {
+ audio_io_handle_t fxOutput = selectOutputForEffects(dstOutputs);
+ SortedVector<audio_io_handle_t> moved;
+ for (size_t i = 0; i < mEffects.size(); i++) {
+ EffectDescriptor *desc = mEffects.valueAt(i);
+ if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
+ desc->mIo != fxOutput) {
+ if (moved.indexOf(desc->mIo) < 0) {
+ ALOGV("checkOutputForStrategy() moving effect %d to output %d",
+ mEffects.keyAt(i), fxOutput);
+ mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
+ fxOutput);
+ moved.add(desc->mIo);
+ }
+ desc->mIo = fxOutput;
+ }
+ }
+ }
+ // Move tracks associated to this strategy from previous output to new output
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ if (getStrategy((audio_stream_type_t)i) == strategy) {
+ mpClientInterface->invalidateStream((audio_stream_type_t)i);
+ }
+ }
+ }
+}
+
+void AudioPolicyManager::checkOutputForAllStrategies()
+{
+ checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
+ checkOutputForStrategy(STRATEGY_PHONE);
+ checkOutputForStrategy(STRATEGY_SONIFICATION);
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ checkOutputForStrategy(STRATEGY_MEDIA);
+ checkOutputForStrategy(STRATEGY_DTMF);
+}
+
+audio_io_handle_t AudioPolicyManager::getA2dpOutput()
+{
+ if (!mHasA2dp) {
+ return 0;
+ }
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
+ if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ return mOutputs.keyAt(i);
+ }
+ }
+
+ return 0;
+}
+
+void AudioPolicyManager::checkA2dpSuspend()
+{
+ if (!mHasA2dp) {
+ return;
+ }
+ audio_io_handle_t a2dpOutput = getA2dpOutput();
+ if (a2dpOutput == 0) {
+ return;
+ }
+
+ // suspend A2DP output if:
+ // (NOT already suspended) &&
+ // ((SCO device is connected &&
+ // (forced usage for communication || for record is SCO))) ||
+ // (phone state is ringing || in call)
+ //
+ // restore A2DP output if:
+ // (Already suspended) &&
+ // ((SCO device is NOT connected ||
+ // (forced usage NOT for communication && NOT for record is SCO))) &&
+ // (phone state is NOT ringing && NOT in call)
+ //
+ if (mA2dpSuspended) {
+ if (((mScoDeviceAddress == "") ||
+ ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) &&
+ ((mPhoneState != AUDIO_MODE_IN_CALL) &&
+ (mPhoneState != AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->restoreOutput(a2dpOutput);
+ mA2dpSuspended = false;
+ }
+ } else {
+ if (((mScoDeviceAddress != "") &&
+ ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) ||
+ ((mPhoneState == AUDIO_MODE_IN_CALL) ||
+ (mPhoneState == AUDIO_MODE_RINGTONE))) {
+
+ mpClientInterface->suspendOutput(a2dpOutput);
+ mA2dpSuspended = true;
+ }
+ }
+}
+
+audio_devices_t AudioPolicyManager::getNewDevice(audio_io_handle_t output, bool fromCache)
+{
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: the strategy enforced audible is active on the output:
+ // use device for strategy enforced audible
+ // 2: we are in call or the strategy phone is active on the output:
+ // use device for strategy phone
+ // 3: the strategy sonification is active on the output:
+ // use device for strategy sonification
+ // 4: the strategy "respectful" sonification is active on the output:
+ // use device for strategy "respectful" sonification
+ // 5: the strategy media is active on the output:
+ // use device for strategy media
+ // 6: the strategy DTMF is active on the output:
+ // use device for strategy DTMF
+ if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isInCall() ||
+ outputDesc->isStrategyActive(STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ }
+
+ ALOGV("getNewDevice() selected device %x", device);
+ return device;
+}
+
+uint32_t AudioPolicyManager::getStrategyForStream(audio_stream_type_t stream) {
+ return (uint32_t)getStrategy(stream);
+}
+
+audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
+ audio_devices_t devices;
+ // By checking the range of stream before calling getStrategy, we avoid
+ // getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
+ // and then return STRATEGY_MEDIA, but we want to return the empty set.
+ if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_CNT) {
+ devices = AUDIO_DEVICE_NONE;
+ } else {
+ AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
+ devices = getDeviceForStrategy(strategy, true /*fromCache*/);
+ }
+ return devices;
+}
+
+AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
+ audio_stream_type_t stream) {
+ // stream to strategy mapping
+ switch (stream) {
+ case AUDIO_STREAM_VOICE_CALL:
+ case AUDIO_STREAM_BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AUDIO_STREAM_RING:
+ case AUDIO_STREAM_ALARM:
+ return STRATEGY_SONIFICATION;
+ case AUDIO_STREAM_NOTIFICATION:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+ case AUDIO_STREAM_DTMF:
+ return STRATEGY_DTMF;
+ default:
+ ALOGE("unknown stream type");
+ case AUDIO_STREAM_SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AUDIO_STREAM_TTS:
+ case AUDIO_STREAM_MUSIC:
+ return STRATEGY_MEDIA;
+ case AUDIO_STREAM_ENFORCED_AUDIBLE:
+ return STRATEGY_ENFORCED_AUDIBLE;
+ }
+}
+
+void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
+ switch(stream) {
+ case AUDIO_STREAM_MUSIC:
+ checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
+ updateDevicesAndOutputs();
+ break;
+ default:
+ break;
+ }
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ if (fromCache) {
+ ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+ strategy, mDeviceForStrategy[strategy]);
+ return mDeviceForStrategy[strategy];
+ }
+
+ switch (strategy) {
+
+ case STRATEGY_SONIFICATION_RESPECTFUL:
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC,
+ SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing on a remote device, use the the sonification behavior.
+ // Note that we test this usecase before testing if media is playing because
+ // the isStreamActive() method only informs about the activity of a stream, not
+ // if it's for local playback. Note also that we use the same delay between both tests
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing (or has recently played), use the same device
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ } else {
+ // when media is not playing anymore, fall back on the sonification behavior
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ }
+
+ break;
+
+ case STRATEGY_DTMF:
+ if (!isInCall()) {
+ // when off call, DTMF strategy follows the same rules as MEDIA strategy
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ break;
+ }
+ // when in call, DTMF and PHONE strategies follow the same rules
+ // FALL THROUGH
+
+ case STRATEGY_PHONE:
+ // for phone strategy, we first consider the forced use and then the available devices by order
+ // of priority
+ switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
+ case AUDIO_POLICY_FORCE_BT_SCO:
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ if (device) break;
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ // FALL THROUGH
+
+ default: // FORCE_NONE
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+ if (mHasA2dp && !isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0) && !mA2dpSuspended) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ if (device) break;
+ if (mPhoneState != AUDIO_MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+ }
+ break;
+
+ case AUDIO_POLICY_FORCE_SPEAKER:
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+ // A2DP speaker when forcing to speaker output
+ if (mHasA2dp && !isInCall() &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0) && !mA2dpSuspended) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ if (device) break;
+ }
+ if (mPhoneState != AUDIO_MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+ }
+ break;
+ }
+ break;
+
+ case STRATEGY_SONIFICATION:
+
+ // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+ // handleIncallSonification().
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+ break;
+ }
+ // FALL THROUGH
+
+ case STRATEGY_ENFORCED_AUDIBLE:
+ // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+ // except:
+ // - when in call where it doesn't default to STRATEGY_PHONE behavior
+ // - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+ if ((strategy == STRATEGY_SONIFICATION) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+ }
+ }
+ // The second device used for sonification is the same as the device used by media strategy
+ // FALL THROUGH
+
+ case STRATEGY_MEDIA: {
+ uint32_t device2 = AUDIO_DEVICE_NONE;
+ if (strategy != STRATEGY_SONIFICATION) {
+ // no sonification on remote submix (e.g. WFD)
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ mHasA2dp &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ (getA2dpOutput() != 0) && !mA2dpSuspended) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ }
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+ // no sonification on aux digital (e.g. HDMI)
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+ // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+ device |= device2;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+ }
+ } break;
+
+ default:
+ ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ break;
+ }
+
+ ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+ return device;
+}
+
+void AudioPolicyManager::updateDevicesAndOutputs()
+{
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ }
+ mPreviousOutputs = mOutputs;
+}
+
+uint32_t AudioPolicyManager::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs)
+{
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ if (outputDesc->isDuplicated()) {
+ return 0;
+ }
+
+ uint32_t muteWaitMs = 0;
+ audio_devices_t device = outputDesc->device();
+ bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
+ // temporary mute output if device selection changes to avoid volume bursts due to
+ // different per device volumes
+ bool tempMute = outputDesc->isActive() && (device != prevDevice);
+
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+ bool doMute = false;
+
+ if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = true;
+ } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = false;
+ }
+ if (doMute || tempMute) {
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(j);
+ // skip output if it does not share any device with current output
+ if ((desc->supportedDevices() & outputDesc->supportedDevices())
+ == AUDIO_DEVICE_NONE) {
+ continue;
+ }
+ audio_io_handle_t curOutput = mOutputs.keyAt(j);
+ ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
+ mute ? "muting" : "unmuting", i, curDevice, curOutput);
+ setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+ if (desc->isStrategyActive((routing_strategy)i)) {
+ // do tempMute only for current output
+ if (tempMute && (desc == outputDesc)) {
+ setStrategyMute((routing_strategy)i, true, curOutput);
+ setStrategyMute((routing_strategy)i, false, curOutput,
+ desc->latency() * 2, device);
+ }
+ if ((tempMute && (desc == outputDesc)) || mute) {
+ if (muteWaitMs < desc->latency()) {
+ muteWaitMs = desc->latency();
+ }
+ }
+ }
+ }
+ }
+ }
+
+ // FIXME: should not need to double latency if volume could be applied immediately by the
+ // audioflinger mixer. We must account for the delay between now and the next time
+ // the audioflinger thread for this output will process a buffer (which corresponds to
+ // one buffer size, usually 1/2 or 1/4 of the latency).
+ muteWaitMs *= 2;
+ // wait for the PCM output buffers to empty before proceeding with the rest of the command
+ if (muteWaitMs > delayMs) {
+ muteWaitMs -= delayMs;
+ usleep(muteWaitMs * 1000);
+ return muteWaitMs;
+ }
+ return 0;
+}
+
+uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force,
+ int delayMs)
+{
+ ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ AudioParameter param;
+ uint32_t muteWaitMs;
+
+ if (outputDesc->isDuplicated()) {
+ muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
+ muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
+ return muteWaitMs;
+ }
+ // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+ // output profile
+ if ((device != AUDIO_DEVICE_NONE) &&
+ ((device & outputDesc->mProfile->mSupportedDevices) == 0)) {
+ return 0;
+ }
+
+ // filter devices according to output selected
+ device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices);
+
+ audio_devices_t prevDevice = outputDesc->mDevice;
+
+ ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
+
+ if (device != AUDIO_DEVICE_NONE) {
+ outputDesc->mDevice = device;
+ }
+ muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+
+ // Do not change the routing if:
+ // - the requested device is AUDIO_DEVICE_NONE
+ // - the requested device is the same as current device and force is not specified.
+ // Doing this check here allows the caller to call setOutputDevice() without conditions
+ if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
+ ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
+ return muteWaitMs;
+ }
+
+ ALOGV("setOutputDevice() changing device");
+ // do the routing
+ param.addInt(String8(AudioParameter::keyRouting), (int)device);
+ mpClientInterface->setParameters(output, param.toString(), delayMs);
+
+ // update stream volumes according to new device
+ applyStreamVolumes(output, device, delayMs);
+
+ return muteWaitMs;
+}
+
+AudioPolicyManager::IOProfile *AudioPolicyManager::getInputProfile(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask)
+{
+ // Choose an input profile based on the requested capture parameters: select the first available
+ // profile supporting all requested parameters.
+
+ for (size_t i = 0; i < mHwModules.size(); i++)
+ {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
+ {
+ IOProfile *profile = mHwModules[i]->mInputProfiles[j];
+ if (profile->isCompatibleProfile(device, samplingRate, format,
+ channelMask, AUDIO_OUTPUT_FLAG_NONE)) {
+ return profile;
+ }
+ }
+ }
+ return NULL;
+}
+
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ break;
+ }
+ // FALL THROUGH
+
+ case AUDIO_SOURCE_DEFAULT:
+ case AUDIO_SOURCE_MIC:
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ case AUDIO_SOURCE_HOTWORD:
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO &&
+ mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_CAMCORDER:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ }
+ break;
+ case AUDIO_SOURCE_REMOTE_SUBMIX:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ default:
+ ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+ break;
+ }
+ ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ return device;
+}
+
+bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device)
+{
+ if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+ device &= ~AUDIO_DEVICE_BIT_IN;
+ if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
+ return true;
+ }
+ return false;
+}
+
+audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
+{
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
+ if ((input_descriptor->mRefCount > 0)
+ && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
+ return mInputs.keyAt(i);
+ }
+ }
+ return 0;
+}
+
+
+audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
+{
+ if (device == AUDIO_DEVICE_NONE) {
+ // this happens when forcing a route update and no track is active on an output.
+ // In this case the returned category is not important.
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (popcount(device) > 1) {
+ // Multiple device selection is either:
+ // - speaker + one other device: give priority to speaker in this case.
+ // - one A2DP device + another device: happens with duplicated output. In this case
+ // retain the device on the A2DP output as the other must not correspond to an active
+ // selection if not the speaker.
+ if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else {
+ device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+ }
+ }
+
+ ALOGW_IF(popcount(device) != 1,
+ "getDeviceForVolume() invalid device combination: %08x",
+ device);
+
+ return device;
+}
+
+AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
+{
+ switch(getDeviceForVolume(device)) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ return DEVICE_CATEGORY_EARPIECE;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+ return DEVICE_CATEGORY_HEADSET;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+ default:
+ return DEVICE_CATEGORY_SPEAKER;
+ }
+}
+
+float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi)
+{
+ device_category deviceCategory = getDeviceCategory(device);
+ const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+ // the volume index in the UI is relative to the min and max volume indices for this stream type
+ int nbSteps = 1 + curve[VOLMAX].mIndex -
+ curve[VOLMIN].mIndex;
+ int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+ (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+ // find what part of the curve this index volume belongs to, or if it's out of bounds
+ int segment = 0;
+ if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
+ return 0.0f;
+ } else if (volIdx < curve[VOLKNEE1].mIndex) {
+ segment = 0;
+ } else if (volIdx < curve[VOLKNEE2].mIndex) {
+ segment = 1;
+ } else if (volIdx <= curve[VOLMAX].mIndex) {
+ segment = 2;
+ } else { // out of bounds
+ return 1.0f;
+ }
+
+ // linear interpolation in the attenuation table in dB
+ float decibels = curve[segment].mDBAttenuation +
+ ((float)(volIdx - curve[segment].mIndex)) *
+ ( (curve[segment+1].mDBAttenuation -
+ curve[segment].mDBAttenuation) /
+ ((float)(curve[segment+1].mIndex -
+ curve[segment].mIndex)) );
+
+ float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+ ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+ curve[segment].mIndex, volIdx,
+ curve[segment+1].mIndex,
+ curve[segment].mDBAttenuation,
+ decibels,
+ curve[segment+1].mDBAttenuation,
+ amplification);
+
+ return amplification;
+}
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
+ {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
+ {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
+ {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const AudioPolicyManager::VolumeCurvePoint
+ *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
+ [AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
+ { // AUDIO_STREAM_VOICE_CALL
+ sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_SYSTEM
+ sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_RING
+ sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_MUSIC
+ sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_ALARM
+ sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_NOTIFICATION
+ sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_BLUETOOTH_SCO
+ sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_ENFORCED_AUDIBLE
+ sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_DTMF
+ sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+ { // AUDIO_STREAM_TTS
+ sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
+ },
+};
+
+void AudioPolicyManager::initializeVolumeCurves()
+{
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ mStreams[i].mVolumeCurve[j] =
+ sVolumeProfiles[i][j];
+ }
+ }
+
+ // Check availability of DRC on speaker path: if available, override some of the speaker curves
+ if (mSpeakerDrcEnabled) {
+ mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+ sDefaultSystemVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+ sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+ sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
+ sSpeakerSonificationVolumeCurveDrc;
+ }
+}
+
+float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device)
+{
+ float volume = 1.0;
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ StreamDescriptor &streamDesc = mStreams[stream];
+
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ // if volume is not 0 (not muted), force media volume to max on digital output
+ if (stream == AUDIO_STREAM_MUSIC &&
+ index != mStreams[stream].mIndexMin &&
+ (device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+ device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET ||
+ device == AUDIO_DEVICE_OUT_USB_ACCESSORY ||
+ device == AUDIO_DEVICE_OUT_USB_DEVICE)) {
+ return 1.0;
+ }
+
+ volume = volIndexToAmpl(device, streamDesc, index);
+
+ // if a headset is connected, apply the following rules to ring tones and notifications
+ // to avoid sound level bursts in user's ears:
+ // - always attenuate ring tones and notifications volume by 6dB
+ // - if music is playing, always limit the volume to current music volume,
+ // with a minimum threshold at -36dB so that notification is always perceived.
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+ AUDIO_DEVICE_OUT_WIRED_HEADSET |
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
+ ((stream_strategy == STRATEGY_SONIFICATION)
+ || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
+ || (stream == AUDIO_STREAM_SYSTEM)
+ || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) &&
+ streamDesc.mCanBeMuted) {
+ volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
+ // when the phone is ringing we must consider that music could have been paused just before
+ // by the music application and behave as if music was active if the last music track was
+ // just stopped
+ if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
+ mLimitRingtoneVolume) {
+ audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
+ float musicVol = computeVolume(AUDIO_STREAM_MUSIC,
+ mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice),
+ output,
+ musicDevice);
+ float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
+ musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
+ if (volume > minVol) {
+ volume = minVol;
+ ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
+ }
+ }
+ }
+
+ return volume;
+}
+
+status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+
+ // do not change actual stream volume if the stream is muted
+ if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
+ ALOGVV("checkAndSetVolume() stream %d muted count %d",
+ stream, mOutputs.valueFor(output)->mMuteCount[stream]);
+ return NO_ERROR;
+ }
+
+ // do not change in call volume if bluetooth is connected and vice versa
+ if ((stream == AUDIO_STREAM_VOICE_CALL &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) ||
+ (stream == AUDIO_STREAM_BLUETOOTH_SCO &&
+ mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) {
+ ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
+ stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]);
+ return INVALID_OPERATION;
+ }
+
+ float volume = computeVolume(stream, index, output, device);
+ // We actually change the volume if:
+ // - the float value returned by computeVolume() changed
+ // - the force flag is set
+ if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
+ force) {
+ mOutputs.valueFor(output)->mCurVolume[stream] = volume;
+ ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
+ // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+ // enabled
+ if (stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs);
+ }
+ mpClientInterface->setStreamVolume(stream, volume, output, delayMs);
+ }
+
+ if (stream == AUDIO_STREAM_VOICE_CALL ||
+ stream == AUDIO_STREAM_BLUETOOTH_SCO) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AUDIO_STREAM_VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
+{
+ ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
+
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ checkAndSetVolume((audio_stream_type_t)stream,
+ mStreams[stream].getVolumeIndex(device),
+ output,
+ device,
+ delayMs,
+ force);
+ }
+}
+
+void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+ for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
+ if (getStrategy((audio_stream_type_t)stream) == strategy) {
+ setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device);
+ }
+ }
+}
+
+void AudioPolicyManager::setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ StreamDescriptor &streamDesc = mStreams[stream];
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
+ stream, on, output, outputDesc->mMuteCount[stream], device);
+
+ if (on) {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ if (streamDesc.mCanBeMuted &&
+ ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) ||
+ (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) {
+ checkAndSetVolume(stream, 0, output, device, delayMs);
+ }
+ }
+ // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+ outputDesc->mMuteCount[stream]++;
+ } else {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ ALOGV("setStreamMute() unmuting non muted stream!");
+ return;
+ }
+ if (--outputDesc->mMuteCount[stream] == 0) {
+ checkAndSetVolume(stream,
+ streamDesc.getVolumeIndex(device),
+ output,
+ device,
+ delayMs);
+ }
+ }
+}
+
+void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
+ bool starting, bool stateChange)
+{
+ // if the stream pertains to sonification strategy and we are in call we must
+ // mute the stream if it is low visibility. If it is high visibility, we must play a tone
+ // in the device used for phone strategy and play the tone if the selected device does not
+ // interfere with the device used for phone strategy
+ // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
+ // many times as there are active tracks on the output
+ const routing_strategy stream_strategy = getStrategy(stream);
+ if ((stream_strategy == STRATEGY_SONIFICATION) ||
+ ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
+ ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
+ stream, starting, outputDesc->mDevice, stateChange);
+ if (outputDesc->mRefCount[stream]) {
+ int muteCount = 1;
+ if (stateChange) {
+ muteCount = outputDesc->mRefCount[stream];
+ }
+ if (audio_is_low_visibility(stream)) {
+ ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ } else {
+ ALOGV("handleIncallSonification() high visibility");
+ if (outputDesc->device() &
+ getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
+ ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
+ for (int i = 0; i < muteCount; i++) {
+ setStreamMute(stream, starting, mPrimaryOutput);
+ }
+ }
+ if (starting) {
+ mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
+ AUDIO_STREAM_VOICE_CALL);
+ } else {
+ mpClientInterface->stopTone();
+ }
+ }
+ }
+ }
+}
+
+bool AudioPolicyManager::isInCall()
+{
+ return isStateInCall(mPhoneState);
+}
+
+bool AudioPolicyManager::isStateInCall(int state) {
+ return ((state == AUDIO_MODE_IN_CALL) ||
+ (state == AUDIO_MODE_IN_COMMUNICATION));
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsCpuLoad()
+{
+ return MAX_EFFECTS_CPU_LOAD;
+}
+
+uint32_t AudioPolicyManager::getMaxEffectsMemory()
+{
+ return MAX_EFFECTS_MEMORY;
+}
+
+// --- AudioOutputDescriptor class implementation
+
+AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
+ const IOProfile *profile)
+ : mId(0), mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT),
+ mChannelMask(0), mLatency(0),
+ mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE),
+ mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+ // clear usage count for all stream types
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ mRefCount[i] = 0;
+ mCurVolume[i] = -1.0;
+ mMuteCount[i] = 0;
+ mStopTime[i] = 0;
+ }
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mStrategyMutedByDevice[i] = false;
+ }
+ if (profile != NULL) {
+ mSamplingRate = profile->mSamplingRates[0];
+ mFormat = profile->mFormats[0];
+ mChannelMask = profile->mChannelMasks[0];
+ mFlags = profile->mFlags;
+ }
+}
+
+audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+ } else {
+ return mDevice;
+ }
+}
+
+uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
+{
+ if (isDuplicated()) {
+ return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+ } else {
+ return mLatency;
+ }
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
+ const AudioOutputDescriptor *outputDesc)
+{
+ if (isDuplicated()) {
+ return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+ } else if (outputDesc->isDuplicated()){
+ return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+ } else {
+ return (mProfile->mModule == outputDesc->mProfile->mModule);
+ }
+}
+
+void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+ int delta)
+{
+ // forward usage count change to attached outputs
+ if (isDuplicated()) {
+ mOutput1->changeRefCount(stream, delta);
+ mOutput2->changeRefCount(stream, delta);
+ }
+ if ((delta + (int)mRefCount[stream]) < 0) {
+ ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+ delta, stream, mRefCount[stream]);
+ mRefCount[stream] = 0;
+ return;
+ }
+ mRefCount[stream] += delta;
+ ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+ } else {
+ return mProfile->mSupportedDevices ;
+ }
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+ return isStrategyActive(NUM_STRATEGIES, inPastMs);
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if ((sysTime == 0) && (inPastMs != 0)) {
+ sysTime = systemTime();
+ }
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ if (((getStrategy((audio_stream_type_t)i) == strategy) ||
+ (NUM_STRATEGIES == strategy)) &&
+ isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if (mRefCount[stream] != 0) {
+ return true;
+ }
+ if (inPastMs == 0) {
+ return false;
+ }
+ if (sysTime == 0) {
+ sysTime = systemTime();
+ }
+ if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+ return true;
+ }
+ return false;
+}
+
+
+status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", device());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+ result.append(buffer);
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
+ i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+// --- AudioInputDescriptor class implementation
+
+AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const IOProfile *profile)
+ : mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(0),
+ mDevice(AUDIO_DEVICE_NONE), mRefCount(0),
+ mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile)
+{
+}
+
+status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+// --- StreamDescriptor class implementation
+
+AudioPolicyManager::StreamDescriptor::StreamDescriptor()
+ : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+ mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+ device = AudioPolicyManager::getDeviceForVolume(device);
+ // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+ if (mIndexCur.indexOfKey(device) < 0) {
+ device = AUDIO_DEVICE_OUT_DEFAULT;
+ }
+ return mIndexCur.valueFor(device);
+}
+
+void AudioPolicyManager::StreamDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%s %02d %02d ",
+ mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+ result.append(buffer);
+ for (size_t i = 0; i < mIndexCur.size(); i++) {
+ snprintf(buffer, SIZE, "%04x : %02d, ",
+ mIndexCur.keyAt(i),
+ mIndexCur.valueAt(i));
+ result.append(buffer);
+ }
+ result.append("\n");
+
+ write(fd, result.string(), result.size());
+}
+
+// --- EffectDescriptor class implementation
+
+status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " I/O: %d\n", mIo);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Session: %d\n", mSession);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+// --- IOProfile class implementation
+
+AudioPolicyManager::HwModule::HwModule(const char *name)
+ : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0)
+{
+}
+
+AudioPolicyManager::HwModule::~HwModule()
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ delete mOutputProfiles[i];
+ }
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ delete mInputProfiles[i];
+ }
+ free((void *)mName);
+}
+
+void AudioPolicyManager::HwModule::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " - name: %s\n", mName);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ if (mOutputProfiles.size()) {
+ write(fd, " - outputs:\n", strlen(" - outputs:\n"));
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " output %d:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mOutputProfiles[i]->dump(fd);
+ }
+ }
+ if (mInputProfiles.size()) {
+ write(fd, " - inputs:\n", strlen(" - inputs:\n"));
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " input %d:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mInputProfiles[i]->dump(fd);
+ }
+ }
+}
+
+AudioPolicyManager::IOProfile::IOProfile(HwModule *module)
+ : mFlags((audio_output_flags_t)0), mModule(module)
+{
+}
+
+AudioPolicyManager::IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags) const
+{
+ if (samplingRate == 0 || !audio_is_valid_format(format) || channelMask == 0) {
+ return false;
+ }
+
+ if ((mSupportedDevices & device) != device) {
+ return false;
+ }
+ if ((mFlags & flags) != flags) {
+ return false;
+ }
+ size_t i;
+ for (i = 0; i < mSamplingRates.size(); i++)
+ {
+ if (mSamplingRates[i] == samplingRate) {
+ break;
+ }
+ }
+ if (i == mSamplingRates.size()) {
+ return false;
+ }
+ for (i = 0; i < mFormats.size(); i++)
+ {
+ if (mFormats[i] == format) {
+ break;
+ }
+ }
+ if (i == mFormats.size()) {
+ return false;
+ }
+ for (i = 0; i < mChannelMasks.size(); i++)
+ {
+ if (mChannelMasks[i] == channelMask) {
+ break;
+ }
+ }
+ if (i == mChannelMasks.size()) {
+ return false;
+ }
+ return true;
+}
+
+void AudioPolicyManager::IOProfile::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " - sampling rates: ");
+ result.append(buffer);
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ result.append(buffer);
+ result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", ");
+ }
+
+ snprintf(buffer, SIZE, " - channel masks: ");
+ result.append(buffer);
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ result.append(buffer);
+ result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", ");
+ }
+
+ snprintf(buffer, SIZE, " - formats: ");
+ result.append(buffer);
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ snprintf(buffer, SIZE, "0x%08x", mFormats[i]);
+ result.append(buffer);
+ result.append(i == (mFormats.size() - 1) ? "\n" : ", ");
+ }
+
+ snprintf(buffer, SIZE, " - devices: 0x%04x\n", mSupportedDevices);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+}
+
+// --- audio_policy.conf file parsing
+
+struct StringToEnum {
+ const char *name;
+ uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+const struct StringToEnum sDeviceNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+};
+
+const struct StringToEnum sFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+};
+
+const struct StringToEnum sFormatNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+ STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+ STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+};
+
+const struct StringToEnum sOutChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const struct StringToEnum sInChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+
+uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (strcmp(table[i].name, name) == 0) {
+ ALOGV("stringToEnum() found %s", table[i].name);
+ return table[i].value;
+ }
+ }
+ return 0;
+}
+
+bool AudioPolicyManager::stringToBool(const char *value)
+{
+ return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+audio_output_flags_t AudioPolicyManager::parseFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= stringToEnum(sFlagNameToEnumTable,
+ ARRAY_SIZE(sFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+
+ return (audio_output_flags_t)flag;
+}
+
+audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
+{
+ uint32_t device = 0;
+
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ device |= stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ }
+ devName = strtok(NULL, "|");
+ }
+ return device;
+}
+
+void AudioPolicyManager::loadSamplingRates(char *name, IOProfile *profile)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+ // rates should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ profile->mSamplingRates.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ uint32_t rate = atoi(str);
+ if (rate != 0) {
+ ALOGV("loadSamplingRates() adding rate %d", rate);
+ profile->mSamplingRates.add(rate);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+void AudioPolicyManager::loadFormats(char *name, IOProfile *profile)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mFormats indicates the supported formats
+ // should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ profile->mFormats.add(AUDIO_FORMAT_DEFAULT);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ str);
+ if (format != AUDIO_FORMAT_DEFAULT) {
+ profile->mFormats.add(format);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+void AudioPolicyManager::loadInChannels(char *name, IOProfile *profile)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadInChannels() %s", name);
+
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ profile->mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+ profile->mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+void AudioPolicyManager::loadOutChannels(char *name, IOProfile *profile)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadOutChannels() %s", name);
+
+ // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+ // masks should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ profile->mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ profile->mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+status_t AudioPolicyManager::loadInput(cnode *root, HwModule *module)
+{
+ cnode *node = root->first_child;
+
+ IOProfile *profile = new IOProfile(module);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ loadSamplingRates((char *)node->value, profile);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ loadFormats((char *)node->value, profile);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ loadInChannels((char *)node->value, profile);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices = parseDeviceNames((char *)node->value);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
+ "loadInput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadInput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadInput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadInput() invalid supported formats");
+ if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadInput() adding input mSupportedDevices %04x", profile->mSupportedDevices);
+
+ module->mInputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ delete profile;
+ return BAD_VALUE;
+ }
+}
+
+status_t AudioPolicyManager::loadOutput(cnode *root, HwModule *module)
+{
+ cnode *node = root->first_child;
+
+ IOProfile *profile = new IOProfile(module);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ loadSamplingRates((char *)node->value, profile);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ loadFormats((char *)node->value, profile);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ loadOutChannels((char *)node->value, profile);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices = parseDeviceNames((char *)node->value);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = parseFlagNames((char *)node->value);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
+ "loadOutput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadOutput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadOutput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadOutput() invalid supported formats");
+ if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x",
+ profile->mSupportedDevices, profile->mFlags);
+
+ module->mOutputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ delete profile;
+ return BAD_VALUE;
+ }
+}
+
+void AudioPolicyManager::loadHwModule(cnode *root)
+{
+ cnode *node = config_find(root, OUTPUTS_TAG);
+ status_t status = NAME_NOT_FOUND;
+
+ HwModule *module = new HwModule(root->name);
+
+ if (node != NULL) {
+ if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
+ mHasA2dp = true;
+ } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
+ mHasUsb = true;
+ } else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
+ mHasRemoteSubmix = true;
+ }
+
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading output %s", node->name);
+ status_t tmpStatus = loadOutput(node, module);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ node = config_find(root, INPUTS_TAG);
+ if (node != NULL) {
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModule() loading input %s", node->name);
+ status_t tmpStatus = loadInput(node, module);
+ if (status == NAME_NOT_FOUND || status == NO_ERROR) {
+ status = tmpStatus;
+ }
+ node = node->next;
+ }
+ }
+ if (status == NO_ERROR) {
+ mHwModules.add(module);
+ } else {
+ delete module;
+ }
+}
+
+void AudioPolicyManager::loadHwModules(cnode *root)
+{
+ cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
+ if (node == NULL) {
+ return;
+ }
+
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadHwModules() loading module %s", node->name);
+ loadHwModule(node);
+ node = node->next;
+ }
+}
+
+void AudioPolicyManager::loadGlobalConfig(cnode *root)
+{
+ cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
+ if (node == NULL) {
+ return;
+ }
+ node = node->first_child;
+ while (node) {
+ if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
+ mAttachedOutputDevices = parseDeviceNames((char *)node->value);
+ ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE,
+ "loadGlobalConfig() no attached output devices");
+ ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices);
+ } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
+ mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ (char *)node->value);
+ ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE,
+ "loadGlobalConfig() default device not specified");
+ ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice);
+ } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
+ mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN;
+ ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices);
+ } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
+ mSpeakerDrcEnabled = stringToBool((char *)node->value);
+ ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
+ }
+ node = node->next;
+ }
+}
+
+status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
+{
+ cnode *root;
+ char *data;
+
+ data = (char *)load_file(path, NULL);
+ if (data == NULL) {
+ return -ENODEV;
+ }
+ root = config_node("", "");
+ config_load(root, data);
+
+ loadGlobalConfig(root);
+ loadHwModules(root);
+
+ config_free(root);
+ free(root);
+ free(data);
+
+ ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::defaultAudioPolicyConfig(void)
+{
+ HwModule *module;
+ IOProfile *profile;
+
+ mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER;
+ mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER;
+ mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN;
+
+ module = new HwModule("primary");
+
+ profile = new IOProfile(module);
+ profile->mSamplingRates.add(44100);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
+ profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER;
+ profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
+ module->mOutputProfiles.add(profile);
+
+ profile = new IOProfile(module);
+ profile->mSamplingRates.add(8000);
+ profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
+ profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
+ profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ module->mInputProfiles.add(profile);
+
+ mHwModules.add(module);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
new file mode 100644
index 0000000..e00d8ab
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -0,0 +1,582 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include "AudioPolicyInterface.h"
+
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager implements audio policy manager behavior common to all platforms.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManager: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+ , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+ AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+ virtual ~AudioPolicyManager();
+
+ // AudioPolicyInterface
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address);
+ virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+ const char *device_address);
+ virtual void setPhoneState(audio_mode_t state);
+ virtual void setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config);
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+ virtual void setSystemProperty(const char* property, const char* value);
+ virtual status_t initCheck();
+ virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ virtual status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session = 0);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session = 0);
+ virtual void releaseOutput(audio_io_handle_t output);
+ virtual audio_io_handle_t getInput(audio_source_t inputSource,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_in_acoustics_t acoustics);
+
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input);
+
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input);
+ virtual void releaseInput(audio_io_handle_t input);
+ virtual void initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax);
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
+
+ // return the strategy corresponding to a given stream type
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+
+ // return the enabled output devices for the given stream type
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id);
+ virtual status_t unregisterEffect(int id);
+ virtual status_t setEffectEnabled(int id, bool enabled);
+
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ // return whether a stream is playing remotely, override to change the definition of
+ // local/remote playback, used for instance by notification manager to not make
+ // media players lose audio focus when not playing locally
+ virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ virtual bool isSourceActive(audio_source_t source) const;
+
+ virtual status_t dump(int fd);
+
+ virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+protected:
+
+ enum routing_strategy {
+ STRATEGY_MEDIA,
+ STRATEGY_PHONE,
+ STRATEGY_SONIFICATION,
+ STRATEGY_SONIFICATION_RESPECTFUL,
+ STRATEGY_DTMF,
+ STRATEGY_ENFORCED_AUDIBLE,
+ NUM_STRATEGIES
+ };
+
+ // 4 points to define the volume attenuation curve, each characterized by the volume
+ // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+ // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+
+ enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
+
+ class VolumeCurvePoint
+ {
+ public:
+ int mIndex;
+ float mDBAttenuation;
+ };
+
+ // device categories used for volume curve management.
+ enum device_category {
+ DEVICE_CATEGORY_HEADSET,
+ DEVICE_CATEGORY_SPEAKER,
+ DEVICE_CATEGORY_EARPIECE,
+ DEVICE_CATEGORY_CNT
+ };
+
+ class IOProfile;
+
+ class HwModule {
+ public:
+ HwModule(const char *name);
+ ~HwModule();
+
+ void dump(int fd);
+
+ const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+ audio_module_handle_t mHandle;
+ Vector <IOProfile *> mOutputProfiles; // output profiles exposed by this module
+ Vector <IOProfile *> mInputProfiles; // input profiles exposed by this module
+ };
+
+ // the IOProfile class describes the capabilities of an output or input stream.
+ // It is currently assumed that all combination of listed parameters are supported.
+ // It is used by the policy manager to determine if an output or input is suitable for
+ // a given use case, open/close it accordingly and connect/disconnect audio tracks
+ // to/from it.
+ class IOProfile
+ {
+ public:
+ IOProfile(HwModule *module);
+ ~IOProfile();
+
+ bool isCompatibleProfile(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags) const;
+
+ void dump(int fd);
+
+ // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+ // indicates the supported parameters should be read from the output stream
+ // after it is opened for the first time
+ Vector <uint32_t> mSamplingRates; // supported sampling rates
+ Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+ Vector <audio_format_t> mFormats; // supported audio formats
+ audio_devices_t mSupportedDevices; // supported devices (devices this output can be
+ // routed to)
+ audio_output_flags_t mFlags; // attribute flags (e.g primary output,
+ // direct output...). For outputs only.
+ HwModule *mModule; // audio HW module exposing this I/O stream
+ };
+
+ // default volume curve
+ static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT];
+ // default volume curve for media strategy
+ static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT];
+ // volume curve for media strategy on speakers
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT];
+ // volume curve for sonification strategy on speakers
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT];
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT];
+ static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT];
+ static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
+ static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
+ // default volume curves per stream and device category. See initializeVolumeCurves()
+ static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
+
+ // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+ // and keep track of the usage of this output by each audio stream type.
+ class AudioOutputDescriptor
+ {
+ public:
+ AudioOutputDescriptor(const IOProfile *profile);
+
+ status_t dump(int fd);
+
+ audio_devices_t device() const;
+ void changeRefCount(audio_stream_type_t stream, int delta);
+
+ bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+ audio_devices_t supportedDevices();
+ uint32_t latency();
+ bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc);
+ bool isActive(uint32_t inPastMs = 0) const;
+ bool isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
+ bool isStrategyActive(routing_strategy strategy,
+ uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
+
+ audio_io_handle_t mId; // output handle
+ uint32_t mSamplingRate; //
+ audio_format_t mFormat; //
+ audio_channel_mask_t mChannelMask; // output configuration
+ uint32_t mLatency; //
+ audio_output_flags_t mFlags; //
+ audio_devices_t mDevice; // current device this output is routed to
+ uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
+ nsecs_t mStopTime[AUDIO_STREAM_CNT];
+ AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output
+ AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output
+ float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
+ int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
+ const IOProfile *mProfile; // I/O profile this output derives from
+ bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+ // device selection. See checkDeviceMuteStrategies()
+ uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+ };
+
+ // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+ // and keep track of the usage of this input.
+ class AudioInputDescriptor
+ {
+ public:
+ AudioInputDescriptor(const IOProfile *profile);
+
+ status_t dump(int fd);
+
+ uint32_t mSamplingRate; //
+ audio_format_t mFormat; // input configuration
+ audio_channel_mask_t mChannelMask; //
+ audio_devices_t mDevice; // current device this input is routed to
+ uint32_t mRefCount; // number of AudioRecord clients using this output
+ audio_source_t mInputSource; // input source selected by application (mediarecorder.h)
+ const IOProfile *mProfile; // I/O profile this output derives from
+ };
+
+ // stream descriptor used for volume control
+ class StreamDescriptor
+ {
+ public:
+ StreamDescriptor();
+
+ int getVolumeIndex(audio_devices_t device);
+ void dump(int fd);
+
+ int mIndexMin; // min volume index
+ int mIndexMax; // max volume index
+ KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
+ bool mCanBeMuted; // true is the stream can be muted
+
+ const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
+ };
+
+ // stream descriptor used for volume control
+ class EffectDescriptor
+ {
+ public:
+
+ status_t dump(int fd);
+
+ int mIo; // io the effect is attached to
+ routing_strategy mStrategy; // routing strategy the effect is associated to
+ int mSession; // audio session the effect is on
+ effect_descriptor_t mDesc; // effect descriptor
+ bool mEnabled; // enabled state: CPU load being used or not
+ };
+
+ void addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc);
+
+ // return the strategy corresponding to a given stream type
+ static routing_strategy getStrategy(audio_stream_type_t stream);
+
+ // return appropriate device for streams handled by the specified strategy according to current
+ // phone state, connected devices...
+ // if fromCache is true, the device is returned from mDeviceForStrategy[],
+ // otherwise it is determine by current state
+ // (device connected,phone state, force use, a2dp output...)
+ // This allows to:
+ // 1 speed up process when the state is stable (when starting or stopping an output)
+ // 2 access to either current device selection (fromCache == true) or
+ // "future" device selection (fromCache == false) when called from a context
+ // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+ // before updateDevicesAndOutputs() is called.
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache);
+
+ // change the route of the specified output. Returns the number of ms we have slept to
+ // allow new routing to take effect in certain cases.
+ uint32_t setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force = false,
+ int delayMs = 0);
+
+ // select input device corresponding to requested audio source
+ virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+
+ // return io handle of active input or 0 if no input is active
+ // Only considers inputs from physical devices (e.g. main mic, headset mic) when
+ // ignoreVirtualInputs is true.
+ audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+ // initialize volume curves for each strategy and device category
+ void initializeVolumeCurves();
+
+ // compute the actual volume for a given stream according to the requested index and a particular
+ // device
+ virtual float computeVolume(audio_stream_type_t stream, int index,
+ audio_io_handle_t output, audio_devices_t device);
+
+ // check that volume change is permitted, compute and send new volume to audio hardware
+ status_t checkAndSetVolume(audio_stream_type_t stream, int index, audio_io_handle_t output,
+ audio_devices_t device, int delayMs = 0, bool force = false);
+
+ // apply all stream volumes to the specified output and device
+ void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+ // Mute or unmute all streams handled by the specified strategy on the specified output
+ void setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // Mute or unmute the stream on the specified output
+ void setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // handle special cases for sonification strategy while in call: mute streams or replace by
+ // a special tone in the device used for communication
+ void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
+
+ // true if device is in a telephony or VoIP call
+ virtual bool isInCall();
+
+ // true if given state represents a device in a telephony or VoIP call
+ virtual bool isStateInCall(int state);
+
+ // when a device is connected, checks if an open output can be routed
+ // to this device. If none is open, tries to open one of the available outputs.
+ // Returns an output suitable to this device or 0.
+ // when a device is disconnected, checks if an output is not used any more and
+ // returns its handle if any.
+ // transfers the audio tracks and effects from one output thread to another accordingly.
+ status_t checkOutputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 paramStr);
+
+ // close an output and its companion duplicating output.
+ void closeOutput(audio_io_handle_t output);
+
+ // checks and if necessary changes outputs used for all strategies.
+ // must be called every time a condition that affects the output choice for a given strategy
+ // changes: connected device, phone state, force use...
+ // Must be called before updateDevicesAndOutputs()
+ void checkOutputForStrategy(routing_strategy strategy);
+
+ // Same as checkOutputForStrategy() but for a all strategies in order of priority
+ void checkOutputForAllStrategies();
+
+ // manages A2DP output suspend/restore according to phone state and BT SCO usage
+ void checkA2dpSuspend();
+
+ // returns the A2DP output handle if it is open or 0 otherwise
+ audio_io_handle_t getA2dpOutput();
+
+ // selects the most appropriate device on output for current state
+ // must be called every time a condition that affects the device choice for a given output is
+ // changed: connected device, phone state, force use, output start, output stop..
+ // see getDeviceForStrategy() for the use of fromCache parameter
+
+ audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache);
+ // updates cache of device used by all strategies (mDeviceForStrategy[])
+ // must be called every time a condition that affects the device choice for a given strategy is
+ // changed: connected device, phone state, force use...
+ // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+ // Must be called after checkOutputForAllStrategies()
+
+ void updateDevicesAndOutputs();
+
+ virtual uint32_t getMaxEffectsCpuLoad();
+ virtual uint32_t getMaxEffectsMemory();
+#ifdef AUDIO_POLICY_TEST
+ virtual bool threadLoop();
+ void exit();
+ int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+ status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled);
+
+ // returns the category the device belongs to with regard to volume curve management
+ static device_category getDeviceCategory(audio_devices_t device);
+
+ // extract one device relevant for volume control from multiple device selection
+ static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+ SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+ DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs);
+ bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2);
+
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ // Returns the number of ms waited
+ uint32_t checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs);
+
+ audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags);
+ IOProfile *getInputProfile(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask);
+ IOProfile *getProfileForDirectOutput(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags);
+
+ audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+ bool isNonOffloadableEffectEnabled();
+
+ //
+ // Audio policy configuration file parsing (audio_policy.conf)
+ //
+ static uint32_t stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name);
+ static bool stringToBool(const char *value);
+ static audio_output_flags_t parseFlagNames(char *name);
+ static audio_devices_t parseDeviceNames(char *name);
+ void loadSamplingRates(char *name, IOProfile *profile);
+ void loadFormats(char *name, IOProfile *profile);
+ void loadOutChannels(char *name, IOProfile *profile);
+ void loadInChannels(char *name, IOProfile *profile);
+ status_t loadOutput(cnode *root, HwModule *module);
+ status_t loadInput(cnode *root, HwModule *module);
+ void loadHwModule(cnode *root);
+ void loadHwModules(cnode *root);
+ void loadGlobalConfig(cnode *root);
+ status_t loadAudioPolicyConfig(const char *path);
+ void defaultAudioPolicyConfig(void);
+
+
+ AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
+ audio_io_handle_t mPrimaryOutput; // primary output handle
+ // list of descriptors for outputs currently opened
+ DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;
+ // copy of mOutputs before setDeviceConnectionState() opens new outputs
+ // reset to mOutputs when updateDevicesAndOutputs() is called.
+ DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
+ DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors
+ audio_devices_t mAvailableOutputDevices; // bit field of all available output devices
+ audio_devices_t mAvailableInputDevices; // bit field of all available input devices
+ // without AUDIO_DEVICE_BIT_IN to allow direct bit
+ // field comparisons
+ int mPhoneState; // current phone state
+ audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration
+
+ StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control
+ String8 mA2dpDeviceAddress; // A2DP device MAC address
+ String8 mScoDeviceAddress; // SCO device MAC address
+ String8 mUsbCardAndDevice; // USB audio ALSA card and device numbers:
+ // card=<card_number>;device=<><device_number>
+ bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
+ audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+ float mLastVoiceVolume; // last voice volume value sent to audio HAL
+
+ // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+ static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+ // Maximum memory allocated to audio effects in KB
+ static const uint32_t MAX_EFFECTS_MEMORY = 512;
+ uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+ uint32_t mTotalEffectsMemory; // current memory used by effects
+ KeyedVector<int, EffectDescriptor *> mEffects; // list of registered audio effects
+ bool mA2dpSuspended; // true if A2DP output is suspended
+ bool mHasA2dp; // true on platforms with support for bluetooth A2DP
+ bool mHasUsb; // true on platforms with support for USB audio
+ bool mHasRemoteSubmix; // true on platforms with support for remote presentation of a submix
+ audio_devices_t mAttachedOutputDevices; // output devices always available on the platform
+ audio_devices_t mDefaultOutputDevice; // output device selected by default at boot time
+ // (must be in mAttachedOutputDevices)
+ bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+ // to boost soft sounds, used to adjust volume curves accordingly
+
+ Vector <HwModule *> mHwModules;
+
+#ifdef AUDIO_POLICY_TEST
+ Mutex mLock;
+ Condition mWaitWorkCV;
+
+ int mCurOutput;
+ bool mDirectOutput;
+ audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+ int mTestInput;
+ uint32_t mTestDevice;
+ uint32_t mTestSamplingRate;
+ uint32_t mTestFormat;
+ uint32_t mTestChannels;
+ uint32_t mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+private:
+ static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi);
+ // updates device caching and output for streams that can influence the
+ // routing of notifications
+ void handleNotificationRoutingForStream(audio_stream_type_t stream);
+ static bool isVirtualInputDevice(audio_devices_t device);
+};
+
+};
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp
new file mode 100644
index 0000000..a052e04
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyService.cpp
@@ -0,0 +1,1085 @@
+/*
+ * Copyright (C) 2009 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 "AudioPolicyService"
+//#define LOG_NDEBUG 0
+
+#include "Configuration.h"
+#undef __STRICT_ANSI__
+#define __STDINT_LIMITS
+#define __STDC_LIMIT_MACROS
+#include <stdint.h>
+
+#include <sys/time.h>
+#include <binder/IServiceManager.h>
+#include <utils/Log.h>
+#include <cutils/properties.h>
+#include <binder/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
+#include <hardware_legacy/power.h>
+#include <media/AudioEffect.h>
+#include <media/EffectsFactoryApi.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <hardware/audio_policy.h>
+#include <audio_effects/audio_effects_conf.h>
+#include <media/AudioParameter.h>
+
+namespace android {
+
+static const char kDeadlockedString[] = "AudioPolicyService may be deadlocked\n";
+static const char kCmdDeadlockedString[] = "AudioPolicyService command thread may be deadlocked\n";
+
+static const int kDumpLockRetries = 50;
+static const int kDumpLockSleepUs = 20000;
+
+static const nsecs_t kAudioCommandTimeout = 3000000000LL; // 3 seconds
+
+namespace {
+ extern struct audio_policy_service_ops aps_ops;
+};
+
+// ----------------------------------------------------------------------------
+
+AudioPolicyService::AudioPolicyService()
+ : BnAudioPolicyService(), mpAudioPolicyDev(NULL), mpAudioPolicy(NULL)
+{
+ char value[PROPERTY_VALUE_MAX];
+ const struct hw_module_t *module;
+ int forced_val;
+ int rc;
+
+ Mutex::Autolock _l(mLock);
+
+ // start tone playback thread
+ mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this);
+ // start audio commands thread
+ mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this);
+ // start output activity command thread
+ mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this);
+ /* instantiate the audio policy manager */
+ rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
+ if (rc) {
+ return;
+ }
+
+ rc = audio_policy_dev_open(module, &mpAudioPolicyDev);
+ ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc));
+ if (rc) {
+ return;
+ }
+
+ rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this,
+ &mpAudioPolicy);
+ ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc));
+ if (rc) {
+ return;
+ }
+
+ rc = mpAudioPolicy->init_check(mpAudioPolicy);
+ ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc));
+ if (rc) {
+ return;
+ }
+
+ ALOGI("Loaded audio policy from %s (%s)", module->name, module->id);
+
+ // load audio pre processing modules
+ if (access(AUDIO_EFFECT_VENDOR_CONFIG_FILE, R_OK) == 0) {
+ loadPreProcessorConfig(AUDIO_EFFECT_VENDOR_CONFIG_FILE);
+ } else if (access(AUDIO_EFFECT_DEFAULT_CONFIG_FILE, R_OK) == 0) {
+ loadPreProcessorConfig(AUDIO_EFFECT_DEFAULT_CONFIG_FILE);
+ }
+}
+
+AudioPolicyService::~AudioPolicyService()
+{
+ mTonePlaybackThread->exit();
+ mTonePlaybackThread.clear();
+ mAudioCommandThread->exit();
+ mAudioCommandThread.clear();
+
+
+ // release audio pre processing resources
+ for (size_t i = 0; i < mInputSources.size(); i++) {
+ delete mInputSources.valueAt(i);
+ }
+ mInputSources.clear();
+
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ mInputs.valueAt(i)->mEffects.clear();
+ delete mInputs.valueAt(i);
+ }
+ mInputs.clear();
+
+ if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) {
+ mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy);
+ }
+ if (mpAudioPolicyDev != NULL) {
+ audio_policy_dev_close(mpAudioPolicyDev);
+ }
+}
+
+
+void AudioPolicyService::binderDied(const wp<IBinder>& who) {
+ ALOGW("binderDied() %p, calling pid %d", who.unsafe_get(),
+ IPCThreadState::self()->getCallingPid());
+}
+
+static bool tryLock(Mutex& mutex)
+{
+ bool locked = false;
+ for (int i = 0; i < kDumpLockRetries; ++i) {
+ if (mutex.tryLock() == NO_ERROR) {
+ locked = true;
+ break;
+ }
+ usleep(kDumpLockSleepUs);
+ }
+ return locked;
+}
+
+status_t AudioPolicyService::dumpInternals(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "PolicyManager Interface: %p\n", mpAudioPolicy);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Command Thread: %p\n", mAudioCommandThread.get());
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Tones Thread: %p\n", mTonePlaybackThread.get());
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::dump(int fd, const Vector<String16>& args __unused)
+{
+ if (!dumpAllowed()) {
+ dumpPermissionDenial(fd);
+ } else {
+ bool locked = tryLock(mLock);
+ if (!locked) {
+ String8 result(kDeadlockedString);
+ write(fd, result.string(), result.size());
+ }
+
+ dumpInternals(fd);
+ if (mAudioCommandThread != 0) {
+ mAudioCommandThread->dump(fd);
+ }
+ if (mTonePlaybackThread != 0) {
+ mTonePlaybackThread->dump(fd);
+ }
+
+ if (mpAudioPolicy) {
+ mpAudioPolicy->dump(mpAudioPolicy, fd);
+ }
+
+ if (locked) mLock.unlock();
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::dumpPermissionDenial(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+ snprintf(buffer, SIZE, "Permission Denial: "
+ "can't dump AudioPolicyService from pid=%d, uid=%d\n",
+ IPCThreadState::self()->getCallingPid(),
+ IPCThreadState::self()->getCallingUid());
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+void AudioPolicyService::setPreProcessorEnabled(const InputDesc *inputDesc, bool enabled)
+{
+ const Vector<sp<AudioEffect> > &fxVector = inputDesc->mEffects;
+ for (size_t i = 0; i < fxVector.size(); i++) {
+ fxVector.itemAt(i)->setEnabled(enabled);
+ }
+}
+
+status_t AudioPolicyService::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioPolicyService::onTransact(code, data, reply, flags);
+}
+
+
+// ----------- AudioPolicyService::AudioCommandThread implementation ----------
+
+AudioPolicyService::AudioCommandThread::AudioCommandThread(String8 name,
+ const wp<AudioPolicyService>& service)
+ : Thread(false), mName(name), mService(service)
+{
+ mpToneGenerator = NULL;
+}
+
+
+AudioPolicyService::AudioCommandThread::~AudioCommandThread()
+{
+ if (!mAudioCommands.isEmpty()) {
+ release_wake_lock(mName.string());
+ }
+ mAudioCommands.clear();
+ delete mpToneGenerator;
+}
+
+void AudioPolicyService::AudioCommandThread::onFirstRef()
+{
+ run(mName.string(), ANDROID_PRIORITY_AUDIO);
+}
+
+bool AudioPolicyService::AudioCommandThread::threadLoop()
+{
+ nsecs_t waitTime = INT64_MAX;
+
+ mLock.lock();
+ while (!exitPending())
+ {
+ while (!mAudioCommands.isEmpty()) {
+ nsecs_t curTime = systemTime();
+ // commands are sorted by increasing time stamp: execute them from index 0 and up
+ if (mAudioCommands[0]->mTime <= curTime) {
+ AudioCommand *command = mAudioCommands[0];
+ mAudioCommands.removeAt(0);
+ mLastCommand = *command;
+
+ switch (command->mCommand) {
+ case START_TONE: {
+ mLock.unlock();
+ ToneData *data = (ToneData *)command->mParam;
+ ALOGV("AudioCommandThread() processing start tone %d on stream %d",
+ data->mType, data->mStream);
+ delete mpToneGenerator;
+ mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
+ mpToneGenerator->startTone(data->mType);
+ delete data;
+ mLock.lock();
+ }break;
+ case STOP_TONE: {
+ mLock.unlock();
+ ALOGV("AudioCommandThread() processing stop tone");
+ if (mpToneGenerator != NULL) {
+ mpToneGenerator->stopTone();
+ delete mpToneGenerator;
+ mpToneGenerator = NULL;
+ }
+ mLock.lock();
+ }break;
+ case SET_VOLUME: {
+ VolumeData *data = (VolumeData *)command->mParam;
+ ALOGV("AudioCommandThread() processing set volume stream %d, \
+ volume %f, output %d", data->mStream, data->mVolume, data->mIO);
+ command->mStatus = AudioSystem::setStreamVolume(data->mStream,
+ data->mVolume,
+ data->mIO);
+ if (command->mWaitStatus) {
+ command->mCond.signal();
+ command->mCond.waitRelative(mLock, kAudioCommandTimeout);
+ }
+ delete data;
+ }break;
+ case SET_PARAMETERS: {
+ ParametersData *data = (ParametersData *)command->mParam;
+ ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
+ data->mKeyValuePairs.string(), data->mIO);
+ command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
+ if (command->mWaitStatus) {
+ command->mCond.signal();
+ command->mCond.waitRelative(mLock, kAudioCommandTimeout);
+ }
+ delete data;
+ }break;
+ case SET_VOICE_VOLUME: {
+ VoiceVolumeData *data = (VoiceVolumeData *)command->mParam;
+ ALOGV("AudioCommandThread() processing set voice volume volume %f",
+ data->mVolume);
+ command->mStatus = AudioSystem::setVoiceVolume(data->mVolume);
+ if (command->mWaitStatus) {
+ command->mCond.signal();
+ command->mCond.waitRelative(mLock, kAudioCommandTimeout);
+ }
+ delete data;
+ }break;
+ case STOP_OUTPUT: {
+ StopOutputData *data = (StopOutputData *)command->mParam;
+ ALOGV("AudioCommandThread() processing stop output %d",
+ data->mIO);
+ sp<AudioPolicyService> svc = mService.promote();
+ if (svc == 0) {
+ break;
+ }
+ mLock.unlock();
+ svc->doStopOutput(data->mIO, data->mStream, data->mSession);
+ mLock.lock();
+ delete data;
+ }break;
+ case RELEASE_OUTPUT: {
+ ReleaseOutputData *data = (ReleaseOutputData *)command->mParam;
+ ALOGV("AudioCommandThread() processing release output %d",
+ data->mIO);
+ sp<AudioPolicyService> svc = mService.promote();
+ if (svc == 0) {
+ break;
+ }
+ mLock.unlock();
+ svc->doReleaseOutput(data->mIO);
+ mLock.lock();
+ delete data;
+ }break;
+ default:
+ ALOGW("AudioCommandThread() unknown command %d", command->mCommand);
+ }
+ delete command;
+ waitTime = INT64_MAX;
+ } else {
+ waitTime = mAudioCommands[0]->mTime - curTime;
+ break;
+ }
+ }
+ // release delayed commands wake lock
+ if (mAudioCommands.isEmpty()) {
+ release_wake_lock(mName.string());
+ }
+ ALOGV("AudioCommandThread() going to sleep");
+ mWaitWorkCV.waitRelative(mLock, waitTime);
+ ALOGV("AudioCommandThread() waking up");
+ }
+ mLock.unlock();
+ return false;
+}
+
+status_t AudioPolicyService::AudioCommandThread::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "AudioCommandThread %p Dump\n", this);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+
+ bool locked = tryLock(mLock);
+ if (!locked) {
+ String8 result2(kCmdDeadlockedString);
+ write(fd, result2.string(), result2.size());
+ }
+
+ snprintf(buffer, SIZE, "- Commands:\n");
+ result = String8(buffer);
+ result.append(" Command Time Wait pParam\n");
+ for (size_t i = 0; i < mAudioCommands.size(); i++) {
+ mAudioCommands[i]->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ result.append(" Last Command\n");
+ mLastCommand.dump(buffer, SIZE);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+
+ if (locked) mLock.unlock();
+
+ return NO_ERROR;
+}
+
+void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream)
+{
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = START_TONE;
+ ToneData *data = new ToneData();
+ data->mType = type;
+ data->mStream = stream;
+ command->mParam = (void *)data;
+ Mutex::Autolock _l(mLock);
+ insertCommand_l(command);
+ ALOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream);
+ mWaitWorkCV.signal();
+}
+
+void AudioPolicyService::AudioCommandThread::stopToneCommand()
+{
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = STOP_TONE;
+ command->mParam = NULL;
+ Mutex::Autolock _l(mLock);
+ insertCommand_l(command);
+ ALOGV("AudioCommandThread() adding tone stop");
+ mWaitWorkCV.signal();
+}
+
+status_t AudioPolicyService::AudioCommandThread::volumeCommand(audio_stream_type_t stream,
+ float volume,
+ audio_io_handle_t output,
+ int delayMs)
+{
+ status_t status = NO_ERROR;
+
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = SET_VOLUME;
+ VolumeData *data = new VolumeData();
+ data->mStream = stream;
+ data->mVolume = volume;
+ data->mIO = output;
+ command->mParam = data;
+ Mutex::Autolock _l(mLock);
+ insertCommand_l(command, delayMs);
+ ALOGV("AudioCommandThread() adding set volume stream %d, volume %f, output %d",
+ stream, volume, output);
+ mWaitWorkCV.signal();
+ if (command->mWaitStatus) {
+ command->mCond.wait(mLock);
+ status = command->mStatus;
+ command->mCond.signal();
+ }
+ return status;
+}
+
+status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle,
+ const char *keyValuePairs,
+ int delayMs)
+{
+ status_t status = NO_ERROR;
+
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = SET_PARAMETERS;
+ ParametersData *data = new ParametersData();
+ data->mIO = ioHandle;
+ data->mKeyValuePairs = String8(keyValuePairs);
+ command->mParam = data;
+ Mutex::Autolock _l(mLock);
+ insertCommand_l(command, delayMs);
+ ALOGV("AudioCommandThread() adding set parameter string %s, io %d ,delay %d",
+ keyValuePairs, ioHandle, delayMs);
+ mWaitWorkCV.signal();
+ if (command->mWaitStatus) {
+ command->mCond.wait(mLock);
+ status = command->mStatus;
+ command->mCond.signal();
+ }
+ return status;
+}
+
+status_t AudioPolicyService::AudioCommandThread::voiceVolumeCommand(float volume, int delayMs)
+{
+ status_t status = NO_ERROR;
+
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = SET_VOICE_VOLUME;
+ VoiceVolumeData *data = new VoiceVolumeData();
+ data->mVolume = volume;
+ command->mParam = data;
+ Mutex::Autolock _l(mLock);
+ insertCommand_l(command, delayMs);
+ ALOGV("AudioCommandThread() adding set voice volume volume %f", volume);
+ mWaitWorkCV.signal();
+ if (command->mWaitStatus) {
+ command->mCond.wait(mLock);
+ status = command->mStatus;
+ command->mCond.signal();
+ }
+ return status;
+}
+
+void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session)
+{
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = STOP_OUTPUT;
+ StopOutputData *data = new StopOutputData();
+ data->mIO = output;
+ data->mStream = stream;
+ data->mSession = session;
+ command->mParam = (void *)data;
+ Mutex::Autolock _l(mLock);
+ insertCommand_l(command);
+ ALOGV("AudioCommandThread() adding stop output %d", output);
+ mWaitWorkCV.signal();
+}
+
+void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_io_handle_t output)
+{
+ AudioCommand *command = new AudioCommand();
+ command->mCommand = RELEASE_OUTPUT;
+ ReleaseOutputData *data = new ReleaseOutputData();
+ data->mIO = output;
+ command->mParam = (void *)data;
+ Mutex::Autolock _l(mLock);
+ insertCommand_l(command);
+ ALOGV("AudioCommandThread() adding release output %d", output);
+ mWaitWorkCV.signal();
+}
+
+// insertCommand_l() must be called with mLock held
+void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *command, int delayMs)
+{
+ ssize_t i; // not size_t because i will count down to -1
+ Vector <AudioCommand *> removedCommands;
+ command->mTime = systemTime() + milliseconds(delayMs);
+
+ // acquire wake lock to make sure delayed commands are processed
+ if (mAudioCommands.isEmpty()) {
+ acquire_wake_lock(PARTIAL_WAKE_LOCK, mName.string());
+ }
+
+ // check same pending commands with later time stamps and eliminate them
+ for (i = mAudioCommands.size()-1; i >= 0; i--) {
+ AudioCommand *command2 = mAudioCommands[i];
+ // commands are sorted by increasing time stamp: no need to scan the rest of mAudioCommands
+ if (command2->mTime <= command->mTime) break;
+ if (command2->mCommand != command->mCommand) continue;
+
+ switch (command->mCommand) {
+ case SET_PARAMETERS: {
+ ParametersData *data = (ParametersData *)command->mParam;
+ ParametersData *data2 = (ParametersData *)command2->mParam;
+ if (data->mIO != data2->mIO) break;
+ ALOGV("Comparing parameter command %s to new command %s",
+ data2->mKeyValuePairs.string(), data->mKeyValuePairs.string());
+ AudioParameter param = AudioParameter(data->mKeyValuePairs);
+ AudioParameter param2 = AudioParameter(data2->mKeyValuePairs);
+ for (size_t j = 0; j < param.size(); j++) {
+ String8 key;
+ String8 value;
+ param.getAt(j, key, value);
+ for (size_t k = 0; k < param2.size(); k++) {
+ String8 key2;
+ String8 value2;
+ param2.getAt(k, key2, value2);
+ if (key2 == key) {
+ param2.remove(key2);
+ ALOGV("Filtering out parameter %s", key2.string());
+ break;
+ }
+ }
+ }
+ // if all keys have been filtered out, remove the command.
+ // otherwise, update the key value pairs
+ if (param2.size() == 0) {
+ removedCommands.add(command2);
+ } else {
+ data2->mKeyValuePairs = param2.toString();
+ }
+ command->mTime = command2->mTime;
+ // force delayMs to non 0 so that code below does not request to wait for
+ // command status as the command is now delayed
+ delayMs = 1;
+ } break;
+
+ case SET_VOLUME: {
+ VolumeData *data = (VolumeData *)command->mParam;
+ VolumeData *data2 = (VolumeData *)command2->mParam;
+ if (data->mIO != data2->mIO) break;
+ if (data->mStream != data2->mStream) break;
+ ALOGV("Filtering out volume command on output %d for stream %d",
+ data->mIO, data->mStream);
+ removedCommands.add(command2);
+ command->mTime = command2->mTime;
+ // force delayMs to non 0 so that code below does not request to wait for
+ // command status as the command is now delayed
+ delayMs = 1;
+ } break;
+ case START_TONE:
+ case STOP_TONE:
+ default:
+ break;
+ }
+ }
+
+ // remove filtered commands
+ for (size_t j = 0; j < removedCommands.size(); j++) {
+ // removed commands always have time stamps greater than current command
+ for (size_t k = i + 1; k < mAudioCommands.size(); k++) {
+ if (mAudioCommands[k] == removedCommands[j]) {
+ ALOGV("suppressing command: %d", mAudioCommands[k]->mCommand);
+ mAudioCommands.removeAt(k);
+ break;
+ }
+ }
+ }
+ removedCommands.clear();
+
+ // wait for status only if delay is 0
+ if (delayMs == 0) {
+ command->mWaitStatus = true;
+ } else {
+ command->mWaitStatus = false;
+ }
+
+ // insert command at the right place according to its time stamp
+ ALOGV("inserting command: %d at index %d, num commands %d",
+ command->mCommand, (int)i+1, mAudioCommands.size());
+ mAudioCommands.insertAt(command, i + 1);
+}
+
+void AudioPolicyService::AudioCommandThread::exit()
+{
+ ALOGV("AudioCommandThread::exit");
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+void AudioPolicyService::AudioCommandThread::AudioCommand::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %02d %06d.%03d %01u %p\n",
+ mCommand,
+ (int)ns2s(mTime),
+ (int)ns2ms(mTime)%1000,
+ mWaitStatus,
+ mParam);
+}
+
+/******* helpers for the service_ops callbacks defined below *********/
+void AudioPolicyService::setParameters(audio_io_handle_t ioHandle,
+ const char *keyValuePairs,
+ int delayMs)
+{
+ mAudioCommandThread->parametersCommand(ioHandle, keyValuePairs,
+ delayMs);
+}
+
+int AudioPolicyService::setStreamVolume(audio_stream_type_t stream,
+ float volume,
+ audio_io_handle_t output,
+ int delayMs)
+{
+ return (int)mAudioCommandThread->volumeCommand(stream, volume,
+ output, delayMs);
+}
+
+int AudioPolicyService::startTone(audio_policy_tone_t tone,
+ audio_stream_type_t stream)
+{
+ if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) {
+ ALOGE("startTone: illegal tone requested (%d)", tone);
+ }
+ if (stream != AUDIO_STREAM_VOICE_CALL) {
+ ALOGE("startTone: illegal stream (%d) requested for tone %d", stream,
+ tone);
+ }
+ mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING,
+ AUDIO_STREAM_VOICE_CALL);
+ return 0;
+}
+
+int AudioPolicyService::stopTone()
+{
+ mTonePlaybackThread->stopToneCommand();
+ return 0;
+}
+
+int AudioPolicyService::setVoiceVolume(float volume, int delayMs)
+{
+ return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs);
+}
+
+// ----------------------------------------------------------------------------
+// Audio pre-processing configuration
+// ----------------------------------------------------------------------------
+
+/*static*/ const char * const AudioPolicyService::kInputSourceNames[AUDIO_SOURCE_CNT -1] = {
+ MIC_SRC_TAG,
+ VOICE_UL_SRC_TAG,
+ VOICE_DL_SRC_TAG,
+ VOICE_CALL_SRC_TAG,
+ CAMCORDER_SRC_TAG,
+ VOICE_REC_SRC_TAG,
+ VOICE_COMM_SRC_TAG
+};
+
+// returns the audio_source_t enum corresponding to the input source name or
+// AUDIO_SOURCE_CNT is no match found
+audio_source_t AudioPolicyService::inputSourceNameToEnum(const char *name)
+{
+ int i;
+ for (i = AUDIO_SOURCE_MIC; i < AUDIO_SOURCE_CNT; i++) {
+ if (strcmp(name, kInputSourceNames[i - AUDIO_SOURCE_MIC]) == 0) {
+ ALOGV("inputSourceNameToEnum found source %s %d", name, i);
+ break;
+ }
+ }
+ return (audio_source_t)i;
+}
+
+size_t AudioPolicyService::growParamSize(char *param,
+ size_t size,
+ size_t *curSize,
+ size_t *totSize)
+{
+ // *curSize is at least sizeof(effect_param_t) + 2 * sizeof(int)
+ size_t pos = ((*curSize - 1 ) / size + 1) * size;
+
+ if (pos + size > *totSize) {
+ while (pos + size > *totSize) {
+ *totSize += ((*totSize + 7) / 8) * 4;
+ }
+ param = (char *)realloc(param, *totSize);
+ }
+ *curSize = pos + size;
+ return pos;
+}
+
+size_t AudioPolicyService::readParamValue(cnode *node,
+ char *param,
+ size_t *curSize,
+ size_t *totSize)
+{
+ if (strncmp(node->name, SHORT_TAG, sizeof(SHORT_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(short), curSize, totSize);
+ *(short *)((char *)param + pos) = (short)atoi(node->value);
+ ALOGV("readParamValue() reading short %d", *(short *)((char *)param + pos));
+ return sizeof(short);
+ } else if (strncmp(node->name, INT_TAG, sizeof(INT_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(int), curSize, totSize);
+ *(int *)((char *)param + pos) = atoi(node->value);
+ ALOGV("readParamValue() reading int %d", *(int *)((char *)param + pos));
+ return sizeof(int);
+ } else if (strncmp(node->name, FLOAT_TAG, sizeof(FLOAT_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(float), curSize, totSize);
+ *(float *)((char *)param + pos) = (float)atof(node->value);
+ ALOGV("readParamValue() reading float %f",*(float *)((char *)param + pos));
+ return sizeof(float);
+ } else if (strncmp(node->name, BOOL_TAG, sizeof(BOOL_TAG) + 1) == 0) {
+ size_t pos = growParamSize(param, sizeof(bool), curSize, totSize);
+ if (strncmp(node->value, "false", strlen("false") + 1) == 0) {
+ *(bool *)((char *)param + pos) = false;
+ } else {
+ *(bool *)((char *)param + pos) = true;
+ }
+ ALOGV("readParamValue() reading bool %s",*(bool *)((char *)param + pos) ? "true" : "false");
+ return sizeof(bool);
+ } else if (strncmp(node->name, STRING_TAG, sizeof(STRING_TAG) + 1) == 0) {
+ size_t len = strnlen(node->value, EFFECT_STRING_LEN_MAX);
+ if (*curSize + len + 1 > *totSize) {
+ *totSize = *curSize + len + 1;
+ param = (char *)realloc(param, *totSize);
+ }
+ strncpy(param + *curSize, node->value, len);
+ *curSize += len;
+ param[*curSize] = '\0';
+ ALOGV("readParamValue() reading string %s", param + *curSize - len);
+ return len;
+ }
+ ALOGW("readParamValue() unknown param type %s", node->name);
+ return 0;
+}
+
+effect_param_t *AudioPolicyService::loadEffectParameter(cnode *root)
+{
+ cnode *param;
+ cnode *value;
+ size_t curSize = sizeof(effect_param_t);
+ size_t totSize = sizeof(effect_param_t) + 2 * sizeof(int);
+ effect_param_t *fx_param = (effect_param_t *)malloc(totSize);
+
+ param = config_find(root, PARAM_TAG);
+ value = config_find(root, VALUE_TAG);
+ if (param == NULL && value == NULL) {
+ // try to parse simple parameter form {int int}
+ param = root->first_child;
+ if (param != NULL) {
+ // Note: that a pair of random strings is read as 0 0
+ int *ptr = (int *)fx_param->data;
+ int *ptr2 = (int *)((char *)param + sizeof(effect_param_t));
+ ALOGW("loadEffectParameter() ptr %p ptr2 %p", ptr, ptr2);
+ *ptr++ = atoi(param->name);
+ *ptr = atoi(param->value);
+ fx_param->psize = sizeof(int);
+ fx_param->vsize = sizeof(int);
+ return fx_param;
+ }
+ }
+ if (param == NULL || value == NULL) {
+ ALOGW("loadEffectParameter() invalid parameter description %s", root->name);
+ goto error;
+ }
+
+ fx_param->psize = 0;
+ param = param->first_child;
+ while (param) {
+ ALOGV("loadEffectParameter() reading param of type %s", param->name);
+ size_t size = readParamValue(param, (char *)fx_param, &curSize, &totSize);
+ if (size == 0) {
+ goto error;
+ }
+ fx_param->psize += size;
+ param = param->next;
+ }
+
+ // align start of value field on 32 bit boundary
+ curSize = ((curSize - 1 ) / sizeof(int) + 1) * sizeof(int);
+
+ fx_param->vsize = 0;
+ value = value->first_child;
+ while (value) {
+ ALOGV("loadEffectParameter() reading value of type %s", value->name);
+ size_t size = readParamValue(value, (char *)fx_param, &curSize, &totSize);
+ if (size == 0) {
+ goto error;
+ }
+ fx_param->vsize += size;
+ value = value->next;
+ }
+
+ return fx_param;
+
+error:
+ free(fx_param);
+ return NULL;
+}
+
+void AudioPolicyService::loadEffectParameters(cnode *root, Vector <effect_param_t *>& params)
+{
+ cnode *node = root->first_child;
+ while (node) {
+ ALOGV("loadEffectParameters() loading param %s", node->name);
+ effect_param_t *param = loadEffectParameter(node);
+ if (param == NULL) {
+ node = node->next;
+ continue;
+ }
+ params.add(param);
+ node = node->next;
+ }
+}
+
+AudioPolicyService::InputSourceDesc *AudioPolicyService::loadInputSource(
+ cnode *root,
+ const Vector <EffectDesc *>& effects)
+{
+ cnode *node = root->first_child;
+ if (node == NULL) {
+ ALOGW("loadInputSource() empty element %s", root->name);
+ return NULL;
+ }
+ InputSourceDesc *source = new InputSourceDesc();
+ while (node) {
+ size_t i;
+ for (i = 0; i < effects.size(); i++) {
+ if (strncmp(effects[i]->mName, node->name, EFFECT_STRING_LEN_MAX) == 0) {
+ ALOGV("loadInputSource() found effect %s in list", node->name);
+ break;
+ }
+ }
+ if (i == effects.size()) {
+ ALOGV("loadInputSource() effect %s not in list", node->name);
+ node = node->next;
+ continue;
+ }
+ EffectDesc *effect = new EffectDesc(*effects[i]); // deep copy
+ loadEffectParameters(node, effect->mParams);
+ ALOGV("loadInputSource() adding effect %s uuid %08x", effect->mName, effect->mUuid.timeLow);
+ source->mEffects.add(effect);
+ node = node->next;
+ }
+ if (source->mEffects.size() == 0) {
+ ALOGW("loadInputSource() no valid effects found in source %s", root->name);
+ delete source;
+ return NULL;
+ }
+ return source;
+}
+
+status_t AudioPolicyService::loadInputSources(cnode *root, const Vector <EffectDesc *>& effects)
+{
+ cnode *node = config_find(root, PREPROCESSING_TAG);
+ if (node == NULL) {
+ return -ENOENT;
+ }
+ node = node->first_child;
+ while (node) {
+ audio_source_t source = inputSourceNameToEnum(node->name);
+ if (source == AUDIO_SOURCE_CNT) {
+ ALOGW("loadInputSources() invalid input source %s", node->name);
+ node = node->next;
+ continue;
+ }
+ ALOGV("loadInputSources() loading input source %s", node->name);
+ InputSourceDesc *desc = loadInputSource(node, effects);
+ if (desc == NULL) {
+ node = node->next;
+ continue;
+ }
+ mInputSources.add(source, desc);
+ node = node->next;
+ }
+ return NO_ERROR;
+}
+
+AudioPolicyService::EffectDesc *AudioPolicyService::loadEffect(cnode *root)
+{
+ cnode *node = config_find(root, UUID_TAG);
+ if (node == NULL) {
+ return NULL;
+ }
+ effect_uuid_t uuid;
+ if (AudioEffect::stringToGuid(node->value, &uuid) != NO_ERROR) {
+ ALOGW("loadEffect() invalid uuid %s", node->value);
+ return NULL;
+ }
+ return new EffectDesc(root->name, uuid);
+}
+
+status_t AudioPolicyService::loadEffects(cnode *root, Vector <EffectDesc *>& effects)
+{
+ cnode *node = config_find(root, EFFECTS_TAG);
+ if (node == NULL) {
+ return -ENOENT;
+ }
+ node = node->first_child;
+ while (node) {
+ ALOGV("loadEffects() loading effect %s", node->name);
+ EffectDesc *effect = loadEffect(node);
+ if (effect == NULL) {
+ node = node->next;
+ continue;
+ }
+ effects.add(effect);
+ node = node->next;
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyService::loadPreProcessorConfig(const char *path)
+{
+ cnode *root;
+ char *data;
+
+ data = (char *)load_file(path, NULL);
+ if (data == NULL) {
+ return -ENODEV;
+ }
+ root = config_node("", "");
+ config_load(root, data);
+
+ Vector <EffectDesc *> effects;
+ loadEffects(root, effects);
+ loadInputSources(root, effects);
+
+ // delete effects to fix memory leak.
+ // as effects is local var and valgrind would treat this as memory leak
+ // and although it only did in mediaserver init, but free it in case mediaserver reboot
+ size_t i;
+ for (i = 0; i < effects.size(); i++) {
+ delete effects[i];
+ }
+
+ config_free(root);
+ free(root);
+ free(data);
+
+ return NO_ERROR;
+}
+
+extern "C" {
+audio_module_handle_t aps_load_hw_module(void *service __unused,
+ const char *name);
+audio_io_handle_t aps_open_output(void *service __unused,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ uint32_t *pLatencyMs,
+ audio_output_flags_t flags);
+
+audio_io_handle_t aps_open_output_on_module(void *service __unused,
+ audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ uint32_t *pLatencyMs,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+audio_io_handle_t aps_open_dup_output(void *service __unused,
+ audio_io_handle_t output1,
+ audio_io_handle_t output2);
+int aps_close_output(void *service __unused, audio_io_handle_t output);
+int aps_suspend_output(void *service __unused, audio_io_handle_t output);
+int aps_restore_output(void *service __unused, audio_io_handle_t output);
+audio_io_handle_t aps_open_input(void *service __unused,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask,
+ audio_in_acoustics_t acoustics __unused);
+audio_io_handle_t aps_open_input_on_module(void *service __unused,
+ audio_module_handle_t module,
+ audio_devices_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ audio_channel_mask_t *pChannelMask);
+int aps_close_input(void *service __unused, audio_io_handle_t input);
+int aps_invalidate_stream(void *service __unused, audio_stream_type_t stream);
+int aps_move_effects(void *service __unused, int session,
+ audio_io_handle_t src_output,
+ audio_io_handle_t dst_output);
+char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
+ const char *keys);
+void aps_set_parameters(void *service, audio_io_handle_t io_handle,
+ const char *kv_pairs, int delay_ms);
+int aps_set_stream_volume(void *service, audio_stream_type_t stream,
+ float volume, audio_io_handle_t output,
+ int delay_ms);
+int aps_start_tone(void *service, audio_policy_tone_t tone,
+ audio_stream_type_t stream);
+int aps_stop_tone(void *service);
+int aps_set_voice_volume(void *service, float volume, int delay_ms);
+};
+
+namespace {
+ struct audio_policy_service_ops aps_ops = {
+ .open_output = aps_open_output,
+ .open_duplicate_output = aps_open_dup_output,
+ .close_output = aps_close_output,
+ .suspend_output = aps_suspend_output,
+ .restore_output = aps_restore_output,
+ .open_input = aps_open_input,
+ .close_input = aps_close_input,
+ .set_stream_volume = aps_set_stream_volume,
+ .invalidate_stream = aps_invalidate_stream,
+ .set_parameters = aps_set_parameters,
+ .get_parameters = aps_get_parameters,
+ .start_tone = aps_start_tone,
+ .stop_tone = aps_stop_tone,
+ .set_voice_volume = aps_set_voice_volume,
+ .move_effects = aps_move_effects,
+ .load_hw_module = aps_load_hw_module,
+ .open_output_on_module = aps_open_output_on_module,
+ .open_input_on_module = aps_open_input_on_module,
+ };
+}; // namespace <unnamed>
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/AudioPolicyService.h
new file mode 100644
index 0000000..ae053a9
--- /dev/null
+++ b/services/audiopolicy/AudioPolicyService.h
@@ -0,0 +1,353 @@
+/*
+ * Copyright (C) 2009 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.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYSERVICE_H
+#define ANDROID_AUDIOPOLICYSERVICE_H
+
+#include <cutils/misc.h>
+#include <cutils/config_utils.h>
+#include <cutils/compiler.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+#include <utils/SortedVector.h>
+#include <binder/BinderService.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <hardware/audio_policy.h>
+#include <media/IAudioPolicyService.h>
+#include <media/ToneGenerator.h>
+#include <media/AudioEffect.h>
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioPolicyService :
+ public BinderService<AudioPolicyService>,
+ public BnAudioPolicyService,
+// public AudioPolicyClientInterface,
+ public IBinder::DeathRecipient
+{
+ friend class BinderService<AudioPolicyService>;
+
+public:
+ // for BinderService
+ static const char *getServiceName() ANDROID_API { return "media.audio_policy"; }
+
+ virtual status_t dump(int fd, const Vector<String16>& args);
+
+ //
+ // BnAudioPolicyService (see AudioPolicyInterface for method descriptions)
+ //
+
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address);
+ virtual audio_policy_dev_state_t getDeviceConnectionState(
+ audio_devices_t device,
+ const char *device_address);
+ virtual status_t setPhoneState(audio_mode_t state);
+ virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+ virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ audio_channel_mask_t channelMask = 0,
+ audio_output_flags_t flags =
+ AUDIO_OUTPUT_FLAG_NONE,
+ const audio_offload_info_t *offloadInfo = NULL);
+ virtual status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session = 0);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session = 0);
+ virtual void releaseOutput(audio_io_handle_t output);
+ virtual audio_io_handle_t getInput(audio_source_t inputSource,
+ uint32_t samplingRate = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ audio_channel_mask_t channelMask = 0,
+ int audioSession = 0);
+ virtual status_t startInput(audio_io_handle_t input);
+ virtual status_t stopInput(audio_io_handle_t input);
+ virtual void releaseInput(audio_io_handle_t input);
+ virtual status_t initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax);
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
+
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id);
+ virtual status_t unregisterEffect(int id);
+ virtual status_t setEffectEnabled(int id, bool enabled);
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ virtual bool isSourceActive(audio_source_t source) const;
+
+ virtual status_t queryDefaultPreProcessing(int audioSession,
+ effect_descriptor_t *descriptors,
+ uint32_t *count);
+ virtual status_t onTransact(
+ uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags);
+
+ // IBinder::DeathRecipient
+ virtual void binderDied(const wp<IBinder>& who);
+
+ //
+ // Helpers for the struct audio_policy_service_ops implementation.
+ // This is used by the audio policy manager for certain operations that
+ // are implemented by the policy service.
+ //
+ virtual void setParameters(audio_io_handle_t ioHandle,
+ const char *keyValuePairs,
+ int delayMs);
+
+ virtual status_t setStreamVolume(audio_stream_type_t stream,
+ float volume,
+ audio_io_handle_t output,
+ int delayMs = 0);
+ virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream);
+ virtual status_t stopTone();
+ virtual status_t setVoiceVolume(float volume, int delayMs = 0);
+ virtual bool isOffloadSupported(const audio_offload_info_t &config);
+
+ status_t doStopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session = 0);
+ void doReleaseOutput(audio_io_handle_t output);
+
+private:
+ AudioPolicyService() ANDROID_API;
+ virtual ~AudioPolicyService();
+
+ status_t dumpInternals(int fd);
+
+ // Thread used for tone playback and to send audio config commands to audio flinger
+ // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because
+ // startTone() and stopTone() are normally called with mLock locked and requesting a tone start
+ // or stop will cause calls to AudioPolicyService and an attempt to lock mLock.
+ // For audio config commands, it is necessary because audio flinger requires that the calling
+ // process (user) has permission to modify audio settings.
+ class AudioCommandThread : public Thread {
+ class AudioCommand;
+ public:
+
+ // commands for tone AudioCommand
+ enum {
+ START_TONE,
+ STOP_TONE,
+ SET_VOLUME,
+ SET_PARAMETERS,
+ SET_VOICE_VOLUME,
+ STOP_OUTPUT,
+ RELEASE_OUTPUT
+ };
+
+ AudioCommandThread (String8 name, const wp<AudioPolicyService>& service);
+ virtual ~AudioCommandThread();
+
+ status_t dump(int fd);
+
+ // Thread virtuals
+ virtual void onFirstRef();
+ virtual bool threadLoop();
+
+ void exit();
+ void startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream);
+ void stopToneCommand();
+ status_t volumeCommand(audio_stream_type_t stream, float volume,
+ audio_io_handle_t output, int delayMs = 0);
+ status_t parametersCommand(audio_io_handle_t ioHandle,
+ const char *keyValuePairs, int delayMs = 0);
+ status_t voiceVolumeCommand(float volume, int delayMs = 0);
+ void stopOutputCommand(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ int session);
+ void releaseOutputCommand(audio_io_handle_t output);
+
+ void insertCommand_l(AudioCommand *command, int delayMs = 0);
+
+ private:
+ // descriptor for requested tone playback event
+ class AudioCommand {
+
+ public:
+ AudioCommand()
+ : mCommand(-1) {}
+
+ void dump(char* buffer, size_t size);
+
+ int mCommand; // START_TONE, STOP_TONE ...
+ nsecs_t mTime; // time stamp
+ Condition mCond; // condition for status return
+ status_t mStatus; // command status
+ bool mWaitStatus; // true if caller is waiting for status
+ void *mParam; // command parameter (ToneData, VolumeData, ParametersData)
+ };
+
+ class ToneData {
+ public:
+ ToneGenerator::tone_type mType; // tone type (START_TONE only)
+ audio_stream_type_t mStream; // stream type (START_TONE only)
+ };
+
+ class VolumeData {
+ public:
+ audio_stream_type_t mStream;
+ float mVolume;
+ audio_io_handle_t mIO;
+ };
+
+ class ParametersData {
+ public:
+ audio_io_handle_t mIO;
+ String8 mKeyValuePairs;
+ };
+
+ class VoiceVolumeData {
+ public:
+ float mVolume;
+ };
+
+ class StopOutputData {
+ public:
+ audio_io_handle_t mIO;
+ audio_stream_type_t mStream;
+ int mSession;
+ };
+
+ class ReleaseOutputData {
+ public:
+ audio_io_handle_t mIO;
+ };
+
+ Mutex mLock;
+ Condition mWaitWorkCV;
+ Vector <AudioCommand *> mAudioCommands; // list of pending commands
+ ToneGenerator *mpToneGenerator; // the tone generator
+ AudioCommand mLastCommand; // last processed command (used by dump)
+ String8 mName; // string used by wake lock fo delayed commands
+ wp<AudioPolicyService> mService;
+ };
+
+ class EffectDesc {
+ public:
+ EffectDesc(const char *name, const effect_uuid_t& uuid) :
+ mName(strdup(name)),
+ mUuid(uuid) { }
+ EffectDesc(const EffectDesc& orig) :
+ mName(strdup(orig.mName)),
+ mUuid(orig.mUuid) {
+ // deep copy mParams
+ for (size_t k = 0; k < orig.mParams.size(); k++) {
+ effect_param_t *origParam = orig.mParams[k];
+ // psize and vsize are rounded up to an int boundary for allocation
+ size_t origSize = sizeof(effect_param_t) +
+ ((origParam->psize + 3) & ~3) +
+ ((origParam->vsize + 3) & ~3);
+ effect_param_t *dupParam = (effect_param_t *) malloc(origSize);
+ memcpy(dupParam, origParam, origSize);
+ // This works because the param buffer allocation is also done by
+ // multiples of 4 bytes originally. In theory we should memcpy only
+ // the actual param size, that is without rounding vsize.
+ mParams.add(dupParam);
+ }
+ }
+ /*virtual*/ ~EffectDesc() {
+ free(mName);
+ for (size_t k = 0; k < mParams.size(); k++) {
+ free(mParams[k]);
+ }
+ }
+ char *mName;
+ effect_uuid_t mUuid;
+ Vector <effect_param_t *> mParams;
+ };
+
+ class InputSourceDesc {
+ public:
+ InputSourceDesc() {}
+ /*virtual*/ ~InputSourceDesc() {
+ for (size_t j = 0; j < mEffects.size(); j++) {
+ delete mEffects[j];
+ }
+ }
+ Vector <EffectDesc *> mEffects;
+ };
+
+
+ class InputDesc {
+ public:
+ InputDesc(int session) : mSessionId(session) {}
+ /*virtual*/ ~InputDesc() {}
+ const int mSessionId;
+ Vector< sp<AudioEffect> >mEffects;
+ };
+
+ static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1];
+
+ void setPreProcessorEnabled(const InputDesc *inputDesc, bool enabled);
+ status_t loadPreProcessorConfig(const char *path);
+ status_t loadEffects(cnode *root, Vector <EffectDesc *>& effects);
+ EffectDesc *loadEffect(cnode *root);
+ status_t loadInputSources(cnode *root, const Vector <EffectDesc *>& effects);
+ audio_source_t inputSourceNameToEnum(const char *name);
+ InputSourceDesc *loadInputSource(cnode *root, const Vector <EffectDesc *>& effects);
+ void loadEffectParameters(cnode *root, Vector <effect_param_t *>& params);
+ effect_param_t *loadEffectParameter(cnode *root);
+ size_t readParamValue(cnode *node,
+ char *param,
+ size_t *curSize,
+ size_t *totSize);
+ size_t growParamSize(char *param,
+ size_t size,
+ size_t *curSize,
+ size_t *totSize);
+
+ // Internal dump utilities.
+ status_t dumpPermissionDenial(int fd);
+
+
+ mutable Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing
+ // device connection state or routing
+ sp<AudioCommandThread> mAudioCommandThread; // audio commands thread
+ sp<AudioCommandThread> mTonePlaybackThread; // tone playback thread
+ sp<AudioCommandThread> mOutputCommandThread; // process stop and release output
+ struct audio_policy_device *mpAudioPolicyDev;
+ struct audio_policy *mpAudioPolicy;
+ KeyedVector< audio_source_t, InputSourceDesc* > mInputSources;
+ KeyedVector< audio_io_handle_t, InputDesc* > mInputs;
+};
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYSERVICE_H
generated by cgit v1.1 at 2024-05-20 19:21:22 +0000