101e77a31b
- AudioPolicyManager: allow platform specific choice for opening a direct output. Also fixed problems in direct output management. - AudioFliinger: use shorter standby delay and track inactivity grace period for direct output thread to free hardware resources as soon as possible. - AudioSystem: do not use cached output selection in getOutput() when a direct output can be selected. Change-Id: If44b50d29237b8402ffd7a5ba1dc43c56f903e9b
1973 lines
77 KiB
C++
1973 lines
77 KiB
C++
/*
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* Copyright (C) 2009 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#define LOG_TAG "AudioPolicyManagerBase"
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//#define LOG_NDEBUG 0
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#include <utils/Log.h>
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#include <hardware_legacy/AudioPolicyManagerBase.h>
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#include <media/mediarecorder.h>
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namespace android {
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// ----------------------------------------------------------------------------
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// AudioPolicyInterface implementation
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// ----------------------------------------------------------------------------
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status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device,
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AudioSystem::device_connection_state state,
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const char *device_address)
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{
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LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
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// connect/disconnect only 1 device at a time
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if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
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if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
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LOGE("setDeviceConnectionState() invalid address: %s", device_address);
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return BAD_VALUE;
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}
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// handle output devices
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if (AudioSystem::isOutputDevice(device)) {
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#ifndef WITH_A2DP
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if (AudioSystem::isA2dpDevice(device)) {
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LOGE("setDeviceConnectionState() invalid device: %x", device);
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return BAD_VALUE;
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}
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#endif
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switch (state)
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{
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// handle output device connection
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case AudioSystem::DEVICE_STATE_AVAILABLE:
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if (mAvailableOutputDevices & device) {
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LOGW("setDeviceConnectionState() device already connected: %x", device);
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return INVALID_OPERATION;
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}
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LOGV("setDeviceConnectionState() connecting device %x", device);
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// register new device as available
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mAvailableOutputDevices |= device;
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#ifdef WITH_A2DP
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// handle A2DP device connection
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if (AudioSystem::isA2dpDevice(device)) {
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status_t status = handleA2dpConnection(device, device_address);
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if (status != NO_ERROR) {
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mAvailableOutputDevices &= ~device;
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return status;
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}
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} else
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#endif
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{
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if (AudioSystem::isBluetoothScoDevice(device)) {
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LOGV("setDeviceConnectionState() BT SCO device, address %s", device_address);
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// keep track of SCO device address
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mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
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#ifdef WITH_A2DP
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if (mA2dpOutput != 0 &&
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mPhoneState != AudioSystem::MODE_NORMAL) {
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mpClientInterface->suspendOutput(mA2dpOutput);
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}
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#endif
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}
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}
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break;
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// handle output device disconnection
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case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
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if (!(mAvailableOutputDevices & device)) {
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LOGW("setDeviceConnectionState() device not connected: %x", device);
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return INVALID_OPERATION;
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}
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LOGV("setDeviceConnectionState() disconnecting device %x", device);
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// remove device from available output devices
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mAvailableOutputDevices &= ~device;
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#ifdef WITH_A2DP
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// handle A2DP device disconnection
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if (AudioSystem::isA2dpDevice(device)) {
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status_t status = handleA2dpDisconnection(device, device_address);
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if (status != NO_ERROR) {
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mAvailableOutputDevices |= device;
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return status;
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}
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} else
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#endif
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{
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if (AudioSystem::isBluetoothScoDevice(device)) {
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mScoDeviceAddress = "";
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#ifdef WITH_A2DP
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if (mA2dpOutput != 0 &&
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mPhoneState != AudioSystem::MODE_NORMAL) {
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mpClientInterface->restoreOutput(mA2dpOutput);
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}
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#endif
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}
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}
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} break;
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default:
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LOGE("setDeviceConnectionState() invalid state: %x", state);
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return BAD_VALUE;
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}
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// request routing change if necessary
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uint32_t newDevice = getNewDevice(mHardwareOutput, false);
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#ifdef WITH_A2DP
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checkOutputForAllStrategies(newDevice);
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// A2DP outputs must be closed after checkOutputForAllStrategies() is executed
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if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && AudioSystem::isA2dpDevice(device)) {
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closeA2dpOutputs();
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}
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#endif
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updateDeviceForStrategy();
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setOutputDevice(mHardwareOutput, newDevice);
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if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
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device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
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} else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO ||
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device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
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device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
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device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
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} else {
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return NO_ERROR;
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}
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}
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// handle input devices
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if (AudioSystem::isInputDevice(device)) {
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switch (state)
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{
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// handle input device connection
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case AudioSystem::DEVICE_STATE_AVAILABLE: {
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if (mAvailableInputDevices & device) {
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LOGW("setDeviceConnectionState() device already connected: %d", device);
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return INVALID_OPERATION;
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}
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mAvailableInputDevices |= device;
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}
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break;
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// handle input device disconnection
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case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
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if (!(mAvailableInputDevices & device)) {
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LOGW("setDeviceConnectionState() device not connected: %d", device);
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return INVALID_OPERATION;
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}
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mAvailableInputDevices &= ~device;
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} break;
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default:
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LOGE("setDeviceConnectionState() invalid state: %x", state);
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return BAD_VALUE;
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}
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audio_io_handle_t activeInput = getActiveInput();
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if (activeInput != 0) {
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AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
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uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
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if (newDevice != inputDesc->mDevice) {
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LOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
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inputDesc->mDevice, newDevice, activeInput);
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inputDesc->mDevice = newDevice;
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AudioParameter param = AudioParameter();
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param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
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mpClientInterface->setParameters(activeInput, param.toString());
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}
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}
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return NO_ERROR;
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}
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LOGW("setDeviceConnectionState() invalid device: %x", device);
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return BAD_VALUE;
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}
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AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(AudioSystem::audio_devices device,
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const char *device_address)
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{
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AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
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String8 address = String8(device_address);
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if (AudioSystem::isOutputDevice(device)) {
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if (device & mAvailableOutputDevices) {
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#ifdef WITH_A2DP
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if (AudioSystem::isA2dpDevice(device) &&
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address != "" && mA2dpDeviceAddress != address) {
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return state;
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}
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#endif
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if (AudioSystem::isBluetoothScoDevice(device) &&
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address != "" && mScoDeviceAddress != address) {
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return state;
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}
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state = AudioSystem::DEVICE_STATE_AVAILABLE;
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}
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} else if (AudioSystem::isInputDevice(device)) {
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if (device & mAvailableInputDevices) {
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state = AudioSystem::DEVICE_STATE_AVAILABLE;
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}
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}
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return state;
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}
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void AudioPolicyManagerBase::setPhoneState(int state)
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{
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LOGV("setPhoneState() state %d", state);
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uint32_t newDevice = 0;
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if (state < 0 || state >= AudioSystem::NUM_MODES) {
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LOGW("setPhoneState() invalid state %d", state);
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return;
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}
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if (state == mPhoneState ) {
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LOGW("setPhoneState() setting same state %d", state);
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return;
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}
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// if leaving call state, handle special case of active streams
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// pertaining to sonification strategy see handleIncallSonification()
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if (mPhoneState == AudioSystem::MODE_IN_CALL) {
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LOGV("setPhoneState() in call state management: new state is %d", state);
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for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
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handleIncallSonification(stream, false, true);
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}
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}
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// store previous phone state for management of sonification strategy below
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int oldState = mPhoneState;
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mPhoneState = state;
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bool force = false;
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// are we entering or starting a call
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if ((oldState != AudioSystem::MODE_IN_CALL) && (state == AudioSystem::MODE_IN_CALL)) {
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LOGV(" Entering call in setPhoneState()");
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// force routing command to audio hardware when starting a call
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// even if no device change is needed
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force = true;
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} else if ((oldState == AudioSystem::MODE_IN_CALL) && (state != AudioSystem::MODE_IN_CALL)) {
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LOGV(" Exiting call in setPhoneState()");
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// force routing command to audio hardware when exiting a call
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// even if no device change is needed
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force = true;
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}
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// check for device and output changes triggered by new phone state
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newDevice = getNewDevice(mHardwareOutput, false);
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#ifdef WITH_A2DP
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checkOutputForAllStrategies(newDevice);
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// suspend A2DP output if a SCO device is present.
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if (mA2dpOutput != 0 && mScoDeviceAddress != "") {
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if (oldState == AudioSystem::MODE_NORMAL) {
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mpClientInterface->suspendOutput(mA2dpOutput);
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} else if (state == AudioSystem::MODE_NORMAL) {
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mpClientInterface->restoreOutput(mA2dpOutput);
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}
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}
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#endif
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updateDeviceForStrategy();
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AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
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// force routing command to audio hardware when ending call
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// even if no device change is needed
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if (oldState == AudioSystem::MODE_IN_CALL && newDevice == 0) {
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newDevice = hwOutputDesc->device();
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}
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// when changing from ring tone to in call mode, mute the ringing tone
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// immediately and delay the route change to avoid sending the ring tone
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// tail into the earpiece or headset.
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int delayMs = 0;
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if (state == AudioSystem::MODE_IN_CALL && oldState == AudioSystem::MODE_RINGTONE) {
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// delay the device change command by twice the output latency to have some margin
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// and be sure that audio buffers not yet affected by the mute are out when
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// we actually apply the route change
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delayMs = hwOutputDesc->mLatency*2;
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setStreamMute(AudioSystem::RING, true, mHardwareOutput);
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}
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// change routing is necessary
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setOutputDevice(mHardwareOutput, newDevice, force, delayMs);
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// if entering in call state, handle special case of active streams
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// pertaining to sonification strategy see handleIncallSonification()
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if (state == AudioSystem::MODE_IN_CALL) {
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LOGV("setPhoneState() in call state management: new state is %d", state);
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// unmute the ringing tone after a sufficient delay if it was muted before
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// setting output device above
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if (oldState == AudioSystem::MODE_RINGTONE) {
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setStreamMute(AudioSystem::RING, false, mHardwareOutput, MUTE_TIME_MS);
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}
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for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
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handleIncallSonification(stream, true, true);
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}
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}
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// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
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if (state == AudioSystem::MODE_RINGTONE &&
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(hwOutputDesc->mRefCount[AudioSystem::MUSIC] ||
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(systemTime() - mMusicStopTime) < seconds(SONIFICATION_HEADSET_MUSIC_DELAY))) {
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mLimitRingtoneVolume = true;
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} else {
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mLimitRingtoneVolume = false;
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}
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}
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void AudioPolicyManagerBase::setRingerMode(uint32_t mode, uint32_t mask)
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{
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LOGV("setRingerMode() mode %x, mask %x", mode, mask);
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mRingerMode = mode;
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}
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void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
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{
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LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
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bool forceVolumeReeval = false;
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switch(usage) {
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case AudioSystem::FOR_COMMUNICATION:
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if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
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config != AudioSystem::FORCE_NONE) {
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LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
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return;
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}
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mForceUse[usage] = config;
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break;
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case AudioSystem::FOR_MEDIA:
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if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
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config != AudioSystem::FORCE_WIRED_ACCESSORY && config != AudioSystem::FORCE_NONE) {
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LOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
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return;
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}
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mForceUse[usage] = config;
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break;
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case AudioSystem::FOR_RECORD:
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if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
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config != AudioSystem::FORCE_NONE) {
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LOGW("setForceUse() invalid config %d for FOR_RECORD", config);
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return;
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}
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mForceUse[usage] = config;
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break;
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case AudioSystem::FOR_DOCK:
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if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK &&
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config != AudioSystem::FORCE_BT_DESK_DOCK && config != AudioSystem::FORCE_WIRED_ACCESSORY) {
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LOGW("setForceUse() invalid config %d for FOR_DOCK", config);
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}
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forceVolumeReeval = true;
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mForceUse[usage] = config;
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break;
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default:
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LOGW("setForceUse() invalid usage %d", usage);
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break;
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}
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// check for device and output changes triggered by new phone state
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uint32_t newDevice = getNewDevice(mHardwareOutput, false);
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#ifdef WITH_A2DP
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checkOutputForAllStrategies(newDevice);
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#endif
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updateDeviceForStrategy();
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setOutputDevice(mHardwareOutput, newDevice);
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if (forceVolumeReeval) {
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applyStreamVolumes(mHardwareOutput, newDevice);
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}
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}
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AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
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{
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return mForceUse[usage];
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}
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void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
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{
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LOGV("setSystemProperty() property %s, value %s", property, value);
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if (strcmp(property, "ro.camera.sound.forced") == 0) {
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if (atoi(value)) {
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LOGV("ENFORCED_AUDIBLE cannot be muted");
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mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
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} else {
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LOGV("ENFORCED_AUDIBLE can be muted");
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mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
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}
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}
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}
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audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
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uint32_t samplingRate,
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uint32_t format,
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uint32_t channels,
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AudioSystem::output_flags flags)
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{
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audio_io_handle_t output = 0;
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uint32_t latency = 0;
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routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
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uint32_t device = getDeviceForStrategy(strategy);
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LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
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#ifdef AUDIO_POLICY_TEST
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if (mCurOutput != 0) {
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LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
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mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
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if (mTestOutputs[mCurOutput] == 0) {
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LOGV("getOutput() opening test output");
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AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
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outputDesc->mDevice = mTestDevice;
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outputDesc->mSamplingRate = mTestSamplingRate;
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outputDesc->mFormat = mTestFormat;
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outputDesc->mChannels = mTestChannels;
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outputDesc->mLatency = mTestLatencyMs;
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outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
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outputDesc->mRefCount[stream] = 0;
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mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
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&outputDesc->mSamplingRate,
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&outputDesc->mFormat,
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&outputDesc->mChannels,
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&outputDesc->mLatency,
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outputDesc->mFlags);
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if (mTestOutputs[mCurOutput]) {
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AudioParameter outputCmd = AudioParameter();
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outputCmd.addInt(String8("set_id"),mCurOutput);
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mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
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addOutput(mTestOutputs[mCurOutput], outputDesc);
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}
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}
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return mTestOutputs[mCurOutput];
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}
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#endif //AUDIO_POLICY_TEST
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// open a direct output if required by specified parameters
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if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) {
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LOGV("getOutput() opening direct output device %x", device);
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AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
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outputDesc->mDevice = device;
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outputDesc->mSamplingRate = samplingRate;
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outputDesc->mFormat = format;
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outputDesc->mChannels = channels;
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outputDesc->mLatency = 0;
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outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
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outputDesc->mRefCount[stream] = 0;
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output = mpClientInterface->openOutput(&outputDesc->mDevice,
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&outputDesc->mSamplingRate,
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&outputDesc->mFormat,
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&outputDesc->mChannels,
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&outputDesc->mLatency,
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outputDesc->mFlags);
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// only accept an output with the requeted parameters
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if (output == 0 ||
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(samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
|
|
(format != 0 && format != outputDesc->mFormat) ||
|
|
(channels != 0 && channels != outputDesc->mChannels)) {
|
|
LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
|
|
samplingRate, format, channels);
|
|
if (output != 0) {
|
|
mpClientInterface->closeOutput(output);
|
|
}
|
|
delete outputDesc;
|
|
return 0;
|
|
}
|
|
addOutput(output, outputDesc);
|
|
return output;
|
|
}
|
|
|
|
if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
|
|
channels != AudioSystem::CHANNEL_OUT_STEREO) {
|
|
return 0;
|
|
}
|
|
// open a non direct output
|
|
|
|
// get which output is suitable for the specified stream. The actual routing change will happen
|
|
// when startOutput() will be called
|
|
uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP;
|
|
if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
|
|
#ifdef WITH_A2DP
|
|
if (a2dpUsedForSonification() && a2dpDevice != 0) {
|
|
// if playing on 2 devices among which one is A2DP, use duplicated output
|
|
LOGV("getOutput() using duplicated output");
|
|
LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device);
|
|
output = mDuplicatedOutput;
|
|
} else
|
|
#endif
|
|
{
|
|
// if playing on 2 devices among which none is A2DP, use hardware output
|
|
output = mHardwareOutput;
|
|
}
|
|
LOGV("getOutput() using output %d for 2 devices %x", output, device);
|
|
} else {
|
|
#ifdef WITH_A2DP
|
|
if (a2dpDevice != 0) {
|
|
// if playing on A2DP device, use a2dp output
|
|
LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device);
|
|
output = mA2dpOutput;
|
|
} else
|
|
#endif
|
|
{
|
|
// if playing on not A2DP device, use hardware output
|
|
output = mHardwareOutput;
|
|
}
|
|
}
|
|
|
|
|
|
LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
|
|
stream, samplingRate, format, channels, flags);
|
|
|
|
return output;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
|
|
{
|
|
LOGV("startOutput() output %d, stream %d", output, stream);
|
|
ssize_t index = mOutputs.indexOfKey(output);
|
|
if (index < 0) {
|
|
LOGW("startOutput() unknow output %d", output);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
|
|
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
|
|
|
|
#ifdef WITH_A2DP
|
|
if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
|
|
setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
|
|
}
|
|
#endif
|
|
|
|
// incremenent 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
|
|
// necassary for a correct control of hardware output routing by startOutput() and stopOutput()
|
|
outputDesc->changeRefCount(stream, 1);
|
|
|
|
setOutputDevice(output, getNewDevice(output));
|
|
|
|
// handle special case for sonification while in call
|
|
if (mPhoneState == AudioSystem::MODE_IN_CALL) {
|
|
handleIncallSonification(stream, true, false);
|
|
}
|
|
|
|
// apply volume rules for current stream and device if necessary
|
|
checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device());
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream)
|
|
{
|
|
LOGV("stopOutput() output %d, stream %d", output, stream);
|
|
ssize_t index = mOutputs.indexOfKey(output);
|
|
if (index < 0) {
|
|
LOGW("stopOutput() unknow output %d", output);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
|
|
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
|
|
|
|
// handle special case for sonification while in call
|
|
if (mPhoneState == AudioSystem::MODE_IN_CALL) {
|
|
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 last music track was stopped - see computeVolume()
|
|
if (stream == AudioSystem::MUSIC) {
|
|
mMusicStopTime = systemTime();
|
|
}
|
|
|
|
setOutputDevice(output, getNewDevice(output));
|
|
|
|
#ifdef WITH_A2DP
|
|
if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
|
|
setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput, mOutputs.valueFor(mHardwareOutput)->mLatency*2);
|
|
}
|
|
#endif
|
|
if (output != mHardwareOutput) {
|
|
setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true);
|
|
}
|
|
return NO_ERROR;
|
|
} else {
|
|
LOGW("stopOutput() refcount is already 0 for output %d", output);
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
|
|
{
|
|
LOGV("releaseOutput() %d", output);
|
|
ssize_t index = mOutputs.indexOfKey(output);
|
|
if (index < 0) {
|
|
LOGW("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->refCount() == 0) {
|
|
mpClientInterface->closeOutput(output);
|
|
delete mOutputs.valueAt(index);
|
|
mOutputs.removeItem(output);
|
|
mTestOutputs[testIndex] = 0;
|
|
}
|
|
return;
|
|
}
|
|
#endif //AUDIO_POLICY_TEST
|
|
|
|
if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
|
|
mpClientInterface->closeOutput(output);
|
|
delete mOutputs.valueAt(index);
|
|
mOutputs.removeItem(output);
|
|
}
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channels,
|
|
AudioSystem::audio_in_acoustics acoustics)
|
|
{
|
|
audio_io_handle_t input = 0;
|
|
uint32_t device = getDeviceForInputSource(inputSource);
|
|
|
|
LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
|
|
|
|
if (device == 0) {
|
|
return 0;
|
|
}
|
|
|
|
// adapt channel selection to input source
|
|
switch(inputSource) {
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
|
|
|
|
inputDesc->mInputSource = inputSource;
|
|
inputDesc->mDevice = device;
|
|
inputDesc->mSamplingRate = samplingRate;
|
|
inputDesc->mFormat = format;
|
|
inputDesc->mChannels = channels;
|
|
inputDesc->mAcoustics = acoustics;
|
|
inputDesc->mRefCount = 0;
|
|
input = mpClientInterface->openInput(&inputDesc->mDevice,
|
|
&inputDesc->mSamplingRate,
|
|
&inputDesc->mFormat,
|
|
&inputDesc->mChannels,
|
|
inputDesc->mAcoustics);
|
|
|
|
// only accept input with the exact requested set of parameters
|
|
if (input == 0 ||
|
|
(samplingRate != inputDesc->mSamplingRate) ||
|
|
(format != inputDesc->mFormat) ||
|
|
(channels != inputDesc->mChannels)) {
|
|
LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d",
|
|
samplingRate, format, channels);
|
|
if (input != 0) {
|
|
mpClientInterface->closeInput(input);
|
|
}
|
|
delete inputDesc;
|
|
return 0;
|
|
}
|
|
mInputs.add(input, inputDesc);
|
|
return input;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
|
|
{
|
|
LOGV("startInput() input %d", input);
|
|
ssize_t index = mInputs.indexOfKey(input);
|
|
if (index < 0) {
|
|
LOGW("startInput() unknow 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
|
|
if (getActiveInput() != 0) {
|
|
LOGW("startInput() input %d failed: other input already started", input);
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
|
|
AudioParameter param = AudioParameter();
|
|
param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
|
|
|
|
// use Voice Recognition mode or not for this input based on input source
|
|
int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0;
|
|
param.addInt(String8("vr_mode"), vr_enabled);
|
|
LOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled);
|
|
|
|
mpClientInterface->setParameters(input, param.toString());
|
|
|
|
inputDesc->mRefCount = 1;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
|
|
{
|
|
LOGV("stopInput() input %d", input);
|
|
ssize_t index = mInputs.indexOfKey(input);
|
|
if (index < 0) {
|
|
LOGW("stopInput() unknow input %d", input);
|
|
return BAD_VALUE;
|
|
}
|
|
AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
|
|
|
|
if (inputDesc->mRefCount == 0) {
|
|
LOGW("stopInput() input %d already stopped", input);
|
|
return INVALID_OPERATION;
|
|
} else {
|
|
AudioParameter param = AudioParameter();
|
|
param.addInt(String8(AudioParameter::keyRouting), 0);
|
|
mpClientInterface->setParameters(input, param.toString());
|
|
inputDesc->mRefCount = 0;
|
|
return NO_ERROR;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
|
|
{
|
|
LOGV("releaseInput() %d", input);
|
|
ssize_t index = mInputs.indexOfKey(input);
|
|
if (index < 0) {
|
|
LOGW("releaseInput() releasing unknown input %d", input);
|
|
return;
|
|
}
|
|
mpClientInterface->closeInput(input);
|
|
delete mInputs.valueAt(index);
|
|
mInputs.removeItem(input);
|
|
LOGV("releaseInput() exit");
|
|
}
|
|
|
|
void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
|
|
int indexMin,
|
|
int indexMax)
|
|
{
|
|
LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
|
|
if (indexMin < 0 || indexMin >= indexMax) {
|
|
LOGW("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 AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
|
|
{
|
|
|
|
if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
// Force max volume if stream cannot be muted
|
|
if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
|
|
|
|
LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
|
|
mStreams[stream].mIndexCur = 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++) {
|
|
status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device());
|
|
if (volStatus != NO_ERROR) {
|
|
status = volStatus;
|
|
}
|
|
}
|
|
return status;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
|
|
{
|
|
if (index == 0) {
|
|
return BAD_VALUE;
|
|
}
|
|
LOGV("getStreamVolumeIndex() stream %d", stream);
|
|
*index = mStreams[stream].mIndexCur;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::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, " Hardware Output: %d\n", mHardwareOutput);
|
|
result.append(buffer);
|
|
#ifdef WITH_A2DP
|
|
snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
|
|
result.append(buffer);
|
|
#endif
|
|
snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.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, " Ringer mode: %d\n", mRingerMode);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]);
|
|
result.append(buffer);
|
|
write(fd, result.string(), result.size());
|
|
|
|
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 Index Min Index Max Index Cur Can be muted\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
snprintf(buffer, SIZE, " %02d", i);
|
|
mStreams[i].dump(buffer + 3, SIZE);
|
|
write(fd, buffer, strlen(buffer));
|
|
}
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
// AudioPolicyManagerBase
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
|
|
:
|
|
#ifdef AUDIO_POLICY_TEST
|
|
Thread(false),
|
|
#endif //AUDIO_POLICY_TEST
|
|
mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0), mMusicStopTime(0), mLimitRingtoneVolume(false)
|
|
{
|
|
mpClientInterface = clientInterface;
|
|
|
|
for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
|
|
mForceUse[i] = AudioSystem::FORCE_NONE;
|
|
}
|
|
|
|
// devices available by default are speaker, ear piece and microphone
|
|
mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE |
|
|
AudioSystem::DEVICE_OUT_SPEAKER;
|
|
mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
|
|
|
|
#ifdef WITH_A2DP
|
|
mA2dpOutput = 0;
|
|
mDuplicatedOutput = 0;
|
|
mA2dpDeviceAddress = String8("");
|
|
#endif
|
|
mScoDeviceAddress = String8("");
|
|
|
|
// open hardware output
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
|
|
outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
|
|
mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
|
|
if (mHardwareOutput == 0) {
|
|
LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
|
|
outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
|
|
} else {
|
|
addOutput(mHardwareOutput, outputDesc);
|
|
setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true);
|
|
}
|
|
|
|
updateDeviceForStrategy();
|
|
#ifdef AUDIO_POLICY_TEST
|
|
AudioParameter outputCmd = AudioParameter();
|
|
outputCmd.addInt(String8("set_id"), 0);
|
|
mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
|
|
|
|
mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
|
|
mTestSamplingRate = 44100;
|
|
mTestFormat = AudioSystem::PCM_16_BIT;
|
|
mTestChannels = AudioSystem::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
|
|
}
|
|
|
|
AudioPolicyManagerBase::~AudioPolicyManagerBase()
|
|
{
|
|
#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);
|
|
}
|
|
mOutputs.clear();
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
mpClientInterface->closeInput(mInputs.keyAt(i));
|
|
delete mInputs.valueAt(i);
|
|
}
|
|
mInputs.clear();
|
|
}
|
|
|
|
#ifdef AUDIO_POLICY_TEST
|
|
bool AudioPolicyManagerBase::threadLoop()
|
|
{
|
|
LOGV("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) {
|
|
LOGV("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 = AudioSystem::INVALID_FORMAT;
|
|
if (value == "PCM 16 bits") {
|
|
format = AudioSystem::PCM_16_BIT;
|
|
} else if (value == "PCM 8 bits") {
|
|
format = AudioSystem::PCM_8_BIT;
|
|
} else if (value == "Compressed MP3") {
|
|
format = AudioSystem::MP3;
|
|
}
|
|
if (format != AudioSystem::INVALID_FORMAT) {
|
|
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 = AudioSystem::CHANNEL_OUT_STEREO;
|
|
} else if (value == "Channels Mono") {
|
|
channels = AudioSystem::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"));
|
|
|
|
mpClientInterface->closeOutput(mHardwareOutput);
|
|
delete mOutputs.valueFor(mHardwareOutput);
|
|
mOutputs.removeItem(mHardwareOutput);
|
|
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
|
|
outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
|
|
mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
if (mHardwareOutput == 0) {
|
|
LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
|
|
outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
|
|
} else {
|
|
AudioParameter outputCmd = AudioParameter();
|
|
outputCmd.addInt(String8("set_id"), 0);
|
|
mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
|
|
addOutput(mHardwareOutput, outputDesc);
|
|
}
|
|
}
|
|
|
|
|
|
mpClientInterface->setParameters(0, String8("test_cmd_policy="));
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::exit()
|
|
{
|
|
{
|
|
AutoMutex _l(mLock);
|
|
requestExit();
|
|
mWaitWorkCV.signal();
|
|
}
|
|
requestExitAndWait();
|
|
}
|
|
|
|
int AudioPolicyManagerBase::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 AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
|
|
{
|
|
outputDesc->mId = id;
|
|
mOutputs.add(id, outputDesc);
|
|
}
|
|
|
|
|
|
#ifdef WITH_A2DP
|
|
status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device,
|
|
const char *device_address)
|
|
{
|
|
// when an A2DP device is connected, open an A2DP and a duplicated output
|
|
LOGV("opening A2DP output for device %s", device_address);
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
|
|
outputDesc->mDevice = device;
|
|
mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
if (mA2dpOutput) {
|
|
// add A2DP output descriptor
|
|
addOutput(mA2dpOutput, outputDesc);
|
|
// set initial stream volume for A2DP device
|
|
applyStreamVolumes(mA2dpOutput, device);
|
|
if (a2dpUsedForSonification()) {
|
|
mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput);
|
|
}
|
|
if (mDuplicatedOutput != 0 ||
|
|
!a2dpUsedForSonification()) {
|
|
// If both A2DP and duplicated outputs are open, send device address to A2DP hardware
|
|
// interface
|
|
AudioParameter param;
|
|
param.add(String8("a2dp_sink_address"), String8(device_address));
|
|
mpClientInterface->setParameters(mA2dpOutput, param.toString());
|
|
mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
|
|
|
|
if (a2dpUsedForSonification()) {
|
|
// add duplicated output descriptor
|
|
AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor();
|
|
dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput);
|
|
dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput);
|
|
dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate;
|
|
dupOutputDesc->mFormat = outputDesc->mFormat;
|
|
dupOutputDesc->mChannels = outputDesc->mChannels;
|
|
dupOutputDesc->mLatency = outputDesc->mLatency;
|
|
addOutput(mDuplicatedOutput, dupOutputDesc);
|
|
applyStreamVolumes(mDuplicatedOutput, device);
|
|
}
|
|
} else {
|
|
LOGW("getOutput() could not open duplicated output for %d and %d",
|
|
mHardwareOutput, mA2dpOutput);
|
|
mpClientInterface->closeOutput(mA2dpOutput);
|
|
mOutputs.removeItem(mA2dpOutput);
|
|
mA2dpOutput = 0;
|
|
delete outputDesc;
|
|
return NO_INIT;
|
|
}
|
|
} else {
|
|
LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
|
|
delete outputDesc;
|
|
return NO_INIT;
|
|
}
|
|
AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
|
|
|
|
if (mScoDeviceAddress != "") {
|
|
// It is normal to suspend twice if we are both in call,
|
|
// and have the hardware audio output routed to BT SCO
|
|
if (mPhoneState != AudioSystem::MODE_NORMAL) {
|
|
mpClientInterface->suspendOutput(mA2dpOutput);
|
|
}
|
|
if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)hwOutputDesc->device())) {
|
|
mpClientInterface->suspendOutput(mA2dpOutput);
|
|
}
|
|
}
|
|
|
|
if (!a2dpUsedForSonification()) {
|
|
// mute music on A2DP output if a notification or ringtone is playing
|
|
uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION);
|
|
for (uint32_t i = 0; i < refCount; i++) {
|
|
setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
|
|
}
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::handleA2dpDisconnection(AudioSystem::audio_devices device,
|
|
const char *device_address)
|
|
{
|
|
if (mA2dpOutput == 0) {
|
|
LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!");
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
if (mA2dpDeviceAddress != device_address) {
|
|
LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
// mute media strategy to avoid outputting sound on hardware output while music stream
|
|
// is switched from A2DP output and before music is paused by music application
|
|
setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput);
|
|
setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, MUTE_TIME_MS);
|
|
|
|
if (!a2dpUsedForSonification()) {
|
|
// unmute music on A2DP output if a notification or ringtone is playing
|
|
uint32_t refCount = mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_SONIFICATION);
|
|
for (uint32_t i = 0; i < refCount; i++) {
|
|
setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput);
|
|
}
|
|
}
|
|
mA2dpDeviceAddress = "";
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::closeA2dpOutputs()
|
|
{
|
|
LOGV("setDeviceConnectionState() closing A2DP and duplicated output!");
|
|
|
|
if (mDuplicatedOutput != 0) {
|
|
mpClientInterface->closeOutput(mDuplicatedOutput);
|
|
delete mOutputs.valueFor(mDuplicatedOutput);
|
|
mOutputs.removeItem(mDuplicatedOutput);
|
|
mDuplicatedOutput = 0;
|
|
}
|
|
if (mA2dpOutput != 0) {
|
|
AudioParameter param;
|
|
param.add(String8("closing"), String8("true"));
|
|
mpClientInterface->setParameters(mA2dpOutput, param.toString());
|
|
mpClientInterface->closeOutput(mA2dpOutput);
|
|
delete mOutputs.valueFor(mA2dpOutput);
|
|
mOutputs.removeItem(mA2dpOutput);
|
|
mA2dpOutput = 0;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy, uint32_t &newDevice)
|
|
{
|
|
uint32_t prevDevice = getDeviceForStrategy(strategy);
|
|
uint32_t curDevice = getDeviceForStrategy(strategy, false);
|
|
bool a2dpWasUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(prevDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
|
|
bool a2dpIsUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(curDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
|
|
AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
|
|
AudioOutputDescriptor *a2dpOutputDesc;
|
|
|
|
if (a2dpWasUsed && !a2dpIsUsed) {
|
|
bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2);
|
|
|
|
if (dupUsed) {
|
|
LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy);
|
|
a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
|
|
} else {
|
|
LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy);
|
|
a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput);
|
|
}
|
|
|
|
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
|
|
mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
|
|
int refCount = a2dpOutputDesc->mRefCount[i];
|
|
// in the case of duplicated output, the ref count is first incremented
|
|
// and then decremented on hardware output tus keeping its value
|
|
hwOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
|
|
a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
|
|
}
|
|
}
|
|
// do not change newDevice if it was already set before this call by a previous call to
|
|
// getNewDevice() or checkOutputForStrategy() for a strategy with higher priority
|
|
if (newDevice == 0 && hwOutputDesc->isUsedByStrategy(strategy)) {
|
|
newDevice = getDeviceForStrategy(strategy, false);
|
|
}
|
|
}
|
|
if (a2dpIsUsed && !a2dpWasUsed) {
|
|
bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2);
|
|
audio_io_handle_t a2dpOutput;
|
|
|
|
if (dupUsed) {
|
|
LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy);
|
|
a2dpOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
|
|
a2dpOutput = mDuplicatedOutput;
|
|
} else {
|
|
LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy);
|
|
a2dpOutputDesc = mOutputs.valueFor(mA2dpOutput);
|
|
a2dpOutput = mA2dpOutput;
|
|
}
|
|
|
|
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
|
|
mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, a2dpOutput);
|
|
int refCount = hwOutputDesc->mRefCount[i];
|
|
// in the case of duplicated output, the ref count is first incremented
|
|
// and then decremented on hardware output tus keeping its value
|
|
a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
|
|
hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::checkOutputForAllStrategies(uint32_t &newDevice)
|
|
{
|
|
// Check strategies in order of priority so that once newDevice is set
|
|
// for a given strategy it is not modified by subsequent calls to
|
|
// checkOutputForStrategy()
|
|
checkOutputForStrategy(STRATEGY_PHONE, newDevice);
|
|
checkOutputForStrategy(STRATEGY_SONIFICATION, newDevice);
|
|
checkOutputForStrategy(STRATEGY_MEDIA, newDevice);
|
|
checkOutputForStrategy(STRATEGY_DTMF, newDevice);
|
|
}
|
|
|
|
#endif
|
|
|
|
uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
|
|
{
|
|
uint32_t device = 0;
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
// check the following by order of priority to request a routing change if necessary:
|
|
// 1: we are in call or the strategy phone is active on the hardware output:
|
|
// use device for strategy phone
|
|
// 2: the strategy sonification is active on the hardware output:
|
|
// use device for strategy sonification
|
|
// 3: the strategy media is active on the hardware output:
|
|
// use device for strategy media
|
|
// 4: the strategy DTMF is active on the hardware output:
|
|
// use device for strategy DTMF
|
|
if (mPhoneState == AudioSystem::MODE_IN_CALL ||
|
|
outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
|
|
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
|
|
} else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
|
|
} else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
|
|
} else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
|
|
device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
|
|
}
|
|
|
|
LOGV("getNewDevice() selected device %x", device);
|
|
return device;
|
|
}
|
|
|
|
AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(AudioSystem::stream_type stream)
|
|
{
|
|
// stream to strategy mapping
|
|
switch (stream) {
|
|
case AudioSystem::VOICE_CALL:
|
|
case AudioSystem::BLUETOOTH_SCO:
|
|
return STRATEGY_PHONE;
|
|
case AudioSystem::RING:
|
|
case AudioSystem::NOTIFICATION:
|
|
case AudioSystem::ALARM:
|
|
case AudioSystem::ENFORCED_AUDIBLE:
|
|
return STRATEGY_SONIFICATION;
|
|
case AudioSystem::DTMF:
|
|
return STRATEGY_DTMF;
|
|
default:
|
|
LOGE("unknown stream type");
|
|
case AudioSystem::SYSTEM:
|
|
// NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
|
|
// while key clicks are played produces a poor result
|
|
case AudioSystem::TTS:
|
|
case AudioSystem::MUSIC:
|
|
return STRATEGY_MEDIA;
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
|
|
{
|
|
uint32_t device = 0;
|
|
|
|
if (fromCache) {
|
|
LOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]);
|
|
return mDeviceForStrategy[strategy];
|
|
}
|
|
|
|
switch (strategy) {
|
|
case STRATEGY_DTMF:
|
|
if (mPhoneState != AudioSystem::MODE_IN_CALL) {
|
|
// when off call, DTMF strategy follows the same rules as MEDIA strategy
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, false);
|
|
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[AudioSystem::FOR_COMMUNICATION]) {
|
|
case AudioSystem::FORCE_BT_SCO:
|
|
if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::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
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
|
|
if (device) break;
|
|
#ifdef WITH_A2DP
|
|
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
|
|
if (mPhoneState != AudioSystem::MODE_IN_CALL) {
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
|
|
if (device) break;
|
|
}
|
|
#endif
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
|
|
if (device == 0) {
|
|
LOGE("getDeviceForStrategy() earpiece device not found");
|
|
}
|
|
break;
|
|
|
|
case AudioSystem::FORCE_SPEAKER:
|
|
if (mPhoneState != AudioSystem::MODE_IN_CALL || strategy != STRATEGY_DTMF) {
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
|
|
if (device) break;
|
|
}
|
|
#ifdef WITH_A2DP
|
|
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
|
|
// A2DP speaker when forcing to speaker output
|
|
if (mPhoneState != AudioSystem::MODE_IN_CALL) {
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
|
|
if (device) break;
|
|
}
|
|
#endif
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
|
|
if (device == 0) {
|
|
LOGE("getDeviceForStrategy() speaker device not found");
|
|
}
|
|
break;
|
|
}
|
|
break;
|
|
|
|
case STRATEGY_SONIFICATION:
|
|
|
|
// If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
|
|
// handleIncallSonification().
|
|
if (mPhoneState == AudioSystem::MODE_IN_CALL) {
|
|
device = getDeviceForStrategy(STRATEGY_PHONE, false);
|
|
break;
|
|
}
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
|
|
if (device == 0) {
|
|
LOGE("getDeviceForStrategy() speaker device not found");
|
|
}
|
|
// The second device used for sonification is the same as the device used by media strategy
|
|
// FALL THROUGH
|
|
|
|
case STRATEGY_MEDIA: {
|
|
uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
|
|
}
|
|
#ifdef WITH_A2DP
|
|
if (mA2dpOutput != 0) {
|
|
if (strategy == STRATEGY_SONIFICATION && !a2dpUsedForSonification()) {
|
|
break;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
|
|
}
|
|
}
|
|
#endif
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
|
|
}
|
|
|
|
// device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION, 0 otherwise
|
|
device |= device2;
|
|
if (device == 0) {
|
|
LOGE("getDeviceForStrategy() speaker device not found");
|
|
}
|
|
} break;
|
|
|
|
default:
|
|
LOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
|
|
break;
|
|
}
|
|
|
|
LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
|
|
return device;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::updateDeviceForStrategy()
|
|
{
|
|
for (int i = 0; i < NUM_STRATEGIES; i++) {
|
|
mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false);
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs)
|
|
{
|
|
LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs);
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
|
|
|
|
if (outputDesc->isDuplicated()) {
|
|
setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
|
|
setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
|
|
return;
|
|
}
|
|
#ifdef WITH_A2DP
|
|
// filter devices according to output selected
|
|
if (output == mA2dpOutput) {
|
|
device &= AudioSystem::DEVICE_OUT_ALL_A2DP;
|
|
} else {
|
|
device &= ~AudioSystem::DEVICE_OUT_ALL_A2DP;
|
|
}
|
|
#endif
|
|
|
|
uint32_t prevDevice = (uint32_t)outputDesc->device();
|
|
// Do not change the routing if:
|
|
// - the requestede device is 0
|
|
// - 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 == 0 || device == prevDevice) && !force) {
|
|
LOGV("setOutputDevice() setting same device %x or null device for output %d", device, output);
|
|
return;
|
|
}
|
|
|
|
outputDesc->mDevice = device;
|
|
// mute media streams if both speaker and headset are selected
|
|
if (output == mHardwareOutput && AudioSystem::popCount(device) == 2) {
|
|
setStrategyMute(STRATEGY_MEDIA, true, output);
|
|
// wait for the PCM output buffers to empty before proceeding with the rest of the command
|
|
usleep(outputDesc->mLatency*2*1000);
|
|
}
|
|
#ifdef WITH_A2DP
|
|
// suspend A2DP output if SCO device is selected
|
|
if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)device)) {
|
|
if (mA2dpOutput != 0) {
|
|
mpClientInterface->suspendOutput(mA2dpOutput);
|
|
}
|
|
}
|
|
#endif
|
|
// do the routing
|
|
AudioParameter param = AudioParameter();
|
|
param.addInt(String8(AudioParameter::keyRouting), (int)device);
|
|
mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs);
|
|
// update stream volumes according to new device
|
|
applyStreamVolumes(output, device, delayMs);
|
|
|
|
#ifdef WITH_A2DP
|
|
// if disconnecting SCO device, restore A2DP output
|
|
if (AudioSystem::isBluetoothScoDevice((AudioSystem::audio_devices)prevDevice)) {
|
|
if (mA2dpOutput != 0) {
|
|
LOGV("restore A2DP output");
|
|
mpClientInterface->restoreOutput(mA2dpOutput);
|
|
}
|
|
}
|
|
#endif
|
|
// if changing from a combined headset + speaker route, unmute media streams
|
|
if (output == mHardwareOutput && AudioSystem::popCount(prevDevice) == 2) {
|
|
setStrategyMute(STRATEGY_MEDIA, false, output, delayMs);
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
|
|
{
|
|
uint32_t device;
|
|
|
|
switch(inputSource) {
|
|
case AUDIO_SOURCE_DEFAULT:
|
|
case AUDIO_SOURCE_MIC:
|
|
case AUDIO_SOURCE_VOICE_RECOGNITION:
|
|
if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
|
|
mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
|
|
device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
|
|
} else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
|
|
device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
|
|
} else {
|
|
device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
|
|
}
|
|
break;
|
|
case AUDIO_SOURCE_CAMCORDER:
|
|
if (hasBackMicrophone()) {
|
|
device = AudioSystem::DEVICE_IN_BACK_MIC;
|
|
} else {
|
|
device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
|
|
}
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
device = AudioSystem::DEVICE_IN_VOICE_CALL;
|
|
break;
|
|
default:
|
|
LOGW("getInput() invalid input source %d", inputSource);
|
|
device = 0;
|
|
break;
|
|
}
|
|
LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
|
|
return device;
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getActiveInput()
|
|
{
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
if (mInputs.valueAt(i)->mRefCount > 0) {
|
|
return mInputs.keyAt(i);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device)
|
|
{
|
|
float volume = 1.0;
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
StreamDescriptor &streamDesc = mStreams[stream];
|
|
|
|
if (device == 0) {
|
|
device = outputDesc->device();
|
|
}
|
|
|
|
int volInt = (100 * (index - streamDesc.mIndexMin)) / (streamDesc.mIndexMax - streamDesc.mIndexMin);
|
|
volume = AudioSystem::linearToLog(volInt);
|
|
|
|
// 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.
|
|
if ((device &
|
|
(AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
|
|
AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
|
|
AudioSystem::DEVICE_OUT_WIRED_HEADSET |
|
|
AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
|
|
(getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) &&
|
|
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 (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) {
|
|
float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device);
|
|
float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
|
|
if (volume > minVol) {
|
|
volume = minVol;
|
|
LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
|
|
}
|
|
}
|
|
}
|
|
|
|
return volume;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force)
|
|
{
|
|
|
|
// do not change actual stream volume if the stream is muted
|
|
if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
|
|
LOGV("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 == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
|
|
(stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
|
|
LOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
|
|
stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
float volume = computeVolume(stream, index, output, device);
|
|
// do not set volume if the float value did not change
|
|
if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || force) {
|
|
mOutputs.valueFor(output)->mCurVolume[stream] = volume;
|
|
LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
|
|
if (stream == AudioSystem::VOICE_CALL ||
|
|
stream == AudioSystem::DTMF ||
|
|
stream == AudioSystem::BLUETOOTH_SCO) {
|
|
float voiceVolume = -1.0;
|
|
// offset value to reflect actual hardware volume that never reaches 0
|
|
// 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
|
|
volume = 0.01 + 0.99 * volume;
|
|
if (stream == AudioSystem::VOICE_CALL) {
|
|
voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
|
|
} else if (stream == AudioSystem::BLUETOOTH_SCO) {
|
|
voiceVolume = 1.0;
|
|
}
|
|
if (voiceVolume >= 0 && output == mHardwareOutput) {
|
|
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
|
|
}
|
|
}
|
|
mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
|
|
}
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs)
|
|
{
|
|
LOGV("applyStreamVolumes() for output %d and device %x", output, device);
|
|
|
|
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
|
|
checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs);
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs)
|
|
{
|
|
LOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
|
|
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
|
|
if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
|
|
setStreamMute(stream, on, output, delayMs);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
|
|
{
|
|
StreamDescriptor &streamDesc = mStreams[stream];
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
|
|
LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]);
|
|
|
|
if (on) {
|
|
if (outputDesc->mMuteCount[stream] == 0) {
|
|
if (streamDesc.mCanBeMuted) {
|
|
checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs);
|
|
}
|
|
}
|
|
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
|
|
outputDesc->mMuteCount[stream]++;
|
|
} else {
|
|
if (outputDesc->mMuteCount[stream] == 0) {
|
|
LOGW("setStreamMute() unmuting non muted stream!");
|
|
return;
|
|
}
|
|
if (--outputDesc->mMuteCount[stream] == 0) {
|
|
checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->device(), delayMs);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::handleIncallSonification(int 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
|
|
|
|
if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
|
|
LOGV("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 (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
|
|
LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
|
|
for (int i = 0; i < muteCount; i++) {
|
|
setStreamMute(stream, starting, mHardwareOutput);
|
|
}
|
|
} else {
|
|
LOGV("handleIncallSonification() high visibility");
|
|
if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) {
|
|
LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
|
|
for (int i = 0; i < muteCount; i++) {
|
|
setStreamMute(stream, starting, mHardwareOutput);
|
|
}
|
|
}
|
|
if (starting) {
|
|
mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
|
|
} else {
|
|
mpClientInterface->stopTone();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channels,
|
|
AudioSystem::output_flags flags,
|
|
uint32_t device)
|
|
{
|
|
return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
|
|
(format !=0 && !AudioSystem::isLinearPCM(format)));
|
|
}
|
|
|
|
// --- AudioOutputDescriptor class implementation
|
|
|
|
AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor()
|
|
: mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
|
|
mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0)
|
|
{
|
|
// clear usage count for all stream types
|
|
for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
mRefCount[i] = 0;
|
|
mCurVolume[i] = -1.0;
|
|
mMuteCount[i] = 0;
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device()
|
|
{
|
|
uint32_t device = 0;
|
|
if (isDuplicated()) {
|
|
device = mOutput1->mDevice | mOutput2->mDevice;
|
|
} else {
|
|
device = mDevice;
|
|
}
|
|
return device;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type 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) {
|
|
LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
|
|
mRefCount[stream] = 0;
|
|
return;
|
|
}
|
|
mRefCount[stream] += delta;
|
|
LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
|
|
{
|
|
uint32_t refcount = 0;
|
|
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
refcount += mRefCount[i];
|
|
}
|
|
return refcount;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy)
|
|
{
|
|
uint32_t refCount = 0;
|
|
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
|
|
refCount += mRefCount[i];
|
|
}
|
|
}
|
|
return refCount;
|
|
}
|
|
|
|
|
|
status_t AudioPolicyManagerBase::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: %d\n", mFormat);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
|
|
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 < AudioSystem::NUM_STREAM_TYPES; 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
|
|
|
|
AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
|
|
: mSamplingRate(0), mFormat(0), mChannels(0),
|
|
mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
|
|
{
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::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", mChannels);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
|
|
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
|
|
|
|
void AudioPolicyManagerBase::StreamDescriptor::dump(char* buffer, size_t size)
|
|
{
|
|
snprintf(buffer, size, " %02d %02d %02d %d\n",
|
|
mIndexMin,
|
|
mIndexMax,
|
|
mIndexCur,
|
|
mCanBeMuted);
|
|
}
|
|
|
|
|
|
}; // namespace android
|