41b150b659
Do not enable offloading of audio decompression when playing video. Change-Id: I6ee8b59370219f5ef810a99a27ba4a1cf222b55b
3946 lines
157 KiB
C++
3946 lines
157 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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//#define VERY_VERBOSE_LOGGING
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#ifdef VERY_VERBOSE_LOGGING
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#define ALOGVV ALOGV
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#else
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#define ALOGVV(a...) do { } while(0)
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#endif
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// A device mask for all audio input devices that are considered "virtual" when evaluating
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// active inputs in getActiveInput()
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#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX
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// A device mask for all audio output devices that are considered "remote" when evaluating
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// active output devices in isStreamActiveRemotely()
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#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
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#include <utils/Log.h>
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#include <hardware_legacy/AudioPolicyManagerBase.h>
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#include <hardware/audio_effect.h>
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#include <hardware/audio.h>
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#include <math.h>
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#include <hardware_legacy/audio_policy_conf.h>
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#include <cutils/properties.h>
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namespace android_audio_legacy {
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// ----------------------------------------------------------------------------
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// AudioPolicyInterface implementation
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// ----------------------------------------------------------------------------
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status_t AudioPolicyManagerBase::setDeviceConnectionState(audio_devices_t 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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SortedVector <audio_io_handle_t> outputs;
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ALOGV("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 (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
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if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
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ALOGE("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 (audio_is_output_device(device)) {
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if (!mHasA2dp && audio_is_a2dp_device(device)) {
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ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
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return BAD_VALUE;
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}
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if (!mHasUsb && audio_is_usb_device(device)) {
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ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
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return BAD_VALUE;
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}
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if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
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ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
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return BAD_VALUE;
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}
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// save a copy of the opened output descriptors before any output is opened or closed
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// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
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mPreviousOutputs = mOutputs;
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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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ALOGW("setDeviceConnectionState() device already connected: %x", device);
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return INVALID_OPERATION;
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}
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ALOGV("setDeviceConnectionState() connecting device %x", device);
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if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
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return INVALID_OPERATION;
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}
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ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %d outputs",
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outputs.size());
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// register new device as available
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mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
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if (!outputs.isEmpty()) {
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String8 paramStr;
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if (mHasA2dp && audio_is_a2dp_device(device)) {
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// handle A2DP device connection
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AudioParameter param;
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param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
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paramStr = param.toString();
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mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
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mA2dpSuspended = false;
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} else if (audio_is_bluetooth_sco_device(device)) {
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// handle SCO device connection
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mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
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} else if (mHasUsb && audio_is_usb_device(device)) {
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// handle USB device connection
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mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
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paramStr = mUsbCardAndDevice;
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}
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// not currently handling multiple simultaneous submixes: ignoring remote submix
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// case and address
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if (!paramStr.isEmpty()) {
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for (size_t i = 0; i < outputs.size(); i++) {
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mpClientInterface->setParameters(outputs[i], paramStr);
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}
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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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ALOGW("setDeviceConnectionState() device not connected: %x", device);
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return INVALID_OPERATION;
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}
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ALOGV("setDeviceConnectionState() disconnecting device %x", device);
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// remove device from available output devices
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mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
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checkOutputsForDevice(device, state, outputs);
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if (mHasA2dp && audio_is_a2dp_device(device)) {
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// handle A2DP device disconnection
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mA2dpDeviceAddress = "";
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mA2dpSuspended = false;
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} else if (audio_is_bluetooth_sco_device(device)) {
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// handle SCO device disconnection
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mScoDeviceAddress = "";
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} else if (mHasUsb && audio_is_usb_device(device)) {
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// handle USB device disconnection
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mUsbCardAndDevice = "";
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}
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// not currently handling multiple simultaneous submixes: ignoring remote submix
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// case and address
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} break;
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default:
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ALOGE("setDeviceConnectionState() invalid state: %x", state);
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return BAD_VALUE;
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}
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checkA2dpSuspend();
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checkOutputForAllStrategies();
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// outputs must be closed after checkOutputForAllStrategies() is executed
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if (!outputs.isEmpty()) {
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for (size_t i = 0; i < outputs.size(); i++) {
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AudioOutputDescriptor *desc = mOutputs.valueFor(outputs[i]);
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// close unused outputs after device disconnection or direct outputs that have been
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// opened by checkOutputsForDevice() to query dynamic parameters
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if ((state == AudioSystem::DEVICE_STATE_UNAVAILABLE) ||
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(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
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(desc->mDirectOpenCount == 0))) {
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closeOutput(outputs[i]);
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}
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}
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}
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updateDevicesAndOutputs();
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for (size_t i = 0; i < mOutputs.size(); i++) {
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// do not force device change on duplicated output because if device is 0, it will
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// also force a device 0 for the two outputs it is duplicated to which may override
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// a valid device selection on those outputs.
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setOutputDevice(mOutputs.keyAt(i),
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getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
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!mOutputs.valueAt(i)->isDuplicated(),
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0);
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}
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if (device == AUDIO_DEVICE_OUT_WIRED_HEADSET) {
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device = AUDIO_DEVICE_IN_WIRED_HEADSET;
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} else if (device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO ||
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device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
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device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
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device = AUDIO_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 (audio_is_input_device(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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ALOGW("setDeviceConnectionState() device already connected: %d", device);
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return INVALID_OPERATION;
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}
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mAvailableInputDevices = mAvailableInputDevices | (device & ~AUDIO_DEVICE_BIT_IN);
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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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ALOGW("setDeviceConnectionState() device not connected: %d", device);
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return INVALID_OPERATION;
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}
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mAvailableInputDevices = (audio_devices_t) (mAvailableInputDevices & ~device);
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} break;
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default:
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ALOGE("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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audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
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if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
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ALOGV("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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ALOGW("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(audio_devices_t 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 (audio_is_output_device(device)) {
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if (device & mAvailableOutputDevices) {
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if (audio_is_a2dp_device(device) &&
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(!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
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return state;
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}
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if (audio_is_bluetooth_sco_device(device) &&
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address != "" && mScoDeviceAddress != address) {
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return state;
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}
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if (audio_is_usb_device(device) &&
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(!mHasUsb || (address != "" && mUsbCardAndDevice != address))) {
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ALOGE("getDeviceConnectionState() invalid device: %x", device);
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return state;
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}
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if (audio_is_remote_submix_device((audio_devices_t)device) && !mHasRemoteSubmix) {
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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 (audio_is_input_device(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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ALOGV("setPhoneState() state %d", state);
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audio_devices_t newDevice = AUDIO_DEVICE_NONE;
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if (state < 0 || state >= AudioSystem::NUM_MODES) {
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ALOGW("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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ALOGW("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 (isInCall()) {
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ALOGV("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 (!isStateInCall(oldState) && isStateInCall(state)) {
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ALOGV(" 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 (isStateInCall(oldState) && !isStateInCall(state)) {
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ALOGV(" 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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} else if (isStateInCall(state) && (state != oldState)) {
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ALOGV(" Switching between telephony and VoIP in setPhoneState()");
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// force routing command to audio hardware when switching between telephony and VoIP
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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(mPrimaryOutput, false /*fromCache*/);
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checkA2dpSuspend();
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checkOutputForAllStrategies();
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updateDevicesAndOutputs();
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AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
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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 (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
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newDevice = hwOutputDesc->device();
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}
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int delayMs = 0;
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if (isStateInCall(state)) {
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nsecs_t sysTime = systemTime();
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for (size_t i = 0; i < mOutputs.size(); i++) {
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AudioOutputDescriptor *desc = mOutputs.valueAt(i);
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// mute media and sonification strategies and delay device switch by the largest
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// latency of any output where either strategy is active.
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// This avoid sending the ring tone or music tail into the earpiece or headset.
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if ((desc->isStrategyActive(STRATEGY_MEDIA,
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SONIFICATION_HEADSET_MUSIC_DELAY,
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sysTime) ||
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desc->isStrategyActive(STRATEGY_SONIFICATION,
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SONIFICATION_HEADSET_MUSIC_DELAY,
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sysTime)) &&
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(delayMs < (int)desc->mLatency*2)) {
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delayMs = desc->mLatency*2;
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}
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setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
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setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
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getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
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setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
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setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
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getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
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}
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}
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// change routing is necessary
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setOutputDevice(mPrimaryOutput, 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 (isStateInCall(state)) {
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ALOGV("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, 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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isStreamActive(AudioSystem::MUSIC, 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::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
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{
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ALOGV("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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ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
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return;
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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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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 &&
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config != AudioSystem::FORCE_ANALOG_DOCK &&
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config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE &&
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config != AudioSystem::FORCE_NO_BT_A2DP) {
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ALOGW("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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ALOGW("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 &&
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config != AudioSystem::FORCE_WIRED_ACCESSORY &&
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config != AudioSystem::FORCE_ANALOG_DOCK &&
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config != AudioSystem::FORCE_DIGITAL_DOCK) {
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ALOGW("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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case AudioSystem::FOR_SYSTEM:
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if (config != AudioSystem::FORCE_NONE &&
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config != AudioSystem::FORCE_SYSTEM_ENFORCED) {
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ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", 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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ALOGW("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 force usage
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checkA2dpSuspend();
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checkOutputForAllStrategies();
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updateDevicesAndOutputs();
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for (size_t i = 0; i < mOutputs.size(); i++) {
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audio_io_handle_t output = mOutputs.keyAt(i);
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audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
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setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
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if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
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applyStreamVolumes(output, newDevice, 0, true);
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}
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}
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audio_io_handle_t activeInput = getActiveInput();
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if (activeInput != 0) {
|
|
AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
|
|
audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
|
|
if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
|
|
ALOGV("setForceUse() changing device from %x to %x for input %d",
|
|
inputDesc->mDevice, newDevice, activeInput);
|
|
inputDesc->mDevice = newDevice;
|
|
AudioParameter param = AudioParameter();
|
|
param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
|
|
mpClientInterface->setParameters(activeInput, param.toString());
|
|
}
|
|
}
|
|
|
|
}
|
|
|
|
AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
|
|
{
|
|
return mForceUse[usage];
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
|
|
{
|
|
ALOGV("setSystemProperty() property %s, value %s", property, value);
|
|
}
|
|
|
|
AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getProfileForDirectOutput(
|
|
audio_devices_t device,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channelMask,
|
|
audio_output_flags_t flags)
|
|
{
|
|
for (size_t i = 0; i < mHwModules.size(); i++) {
|
|
if (mHwModules[i]->mHandle == 0) {
|
|
continue;
|
|
}
|
|
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
|
|
IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
|
|
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
|
|
if (profile->isCompatibleProfile(device, samplingRate, format,
|
|
channelMask,
|
|
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
|
|
if (mAvailableOutputDevices & profile->mSupportedDevices) {
|
|
return mHwModules[i]->mOutputProfiles[j];
|
|
}
|
|
}
|
|
} else if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
|
|
if (profile->isCompatibleProfile(device, samplingRate, format,
|
|
channelMask,
|
|
AUDIO_OUTPUT_FLAG_DIRECT)) {
|
|
if (mAvailableOutputDevices & profile->mSupportedDevices) {
|
|
return mHwModules[i]->mOutputProfiles[j];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channelMask,
|
|
AudioSystem::output_flags flags,
|
|
const audio_offload_info_t *offloadInfo)
|
|
{
|
|
audio_io_handle_t output = 0;
|
|
uint32_t latency = 0;
|
|
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
|
|
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
|
|
ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
|
|
device, stream, samplingRate, format, channelMask, flags);
|
|
|
|
#ifdef AUDIO_POLICY_TEST
|
|
if (mCurOutput != 0) {
|
|
ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
|
|
mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
|
|
|
|
if (mTestOutputs[mCurOutput] == 0) {
|
|
ALOGV("getOutput() opening test output");
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
|
|
outputDesc->mDevice = mTestDevice;
|
|
outputDesc->mSamplingRate = mTestSamplingRate;
|
|
outputDesc->mFormat = mTestFormat;
|
|
outputDesc->mChannelMask = mTestChannels;
|
|
outputDesc->mLatency = mTestLatencyMs;
|
|
outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
|
|
outputDesc->mRefCount[stream] = 0;
|
|
mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannelMask,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags,
|
|
offloadInfo);
|
|
if (mTestOutputs[mCurOutput]) {
|
|
AudioParameter outputCmd = AudioParameter();
|
|
outputCmd.addInt(String8("set_id"),mCurOutput);
|
|
mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
|
|
addOutput(mTestOutputs[mCurOutput], outputDesc);
|
|
}
|
|
}
|
|
return mTestOutputs[mCurOutput];
|
|
}
|
|
#endif //AUDIO_POLICY_TEST
|
|
|
|
// open a direct output if required by specified parameters
|
|
//force direct flag if offload flag is set: offloading implies a direct output stream
|
|
// and all common behaviors are driven by checking only the direct flag
|
|
// this should normally be set appropriately in the policy configuration file
|
|
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
|
|
flags = (AudioSystem::output_flags)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
|
|
}
|
|
|
|
IOProfile *profile = getProfileForDirectOutput(device,
|
|
samplingRate,
|
|
format,
|
|
channelMask,
|
|
(audio_output_flags_t)flags);
|
|
if (profile != NULL) {
|
|
AudioOutputDescriptor *outputDesc = NULL;
|
|
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
AudioOutputDescriptor *desc = mOutputs.valueAt(i);
|
|
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
|
|
outputDesc = desc;
|
|
// reuse direct output if currently open and configured with same parameters
|
|
if ((samplingRate == outputDesc->mSamplingRate) &&
|
|
(format == outputDesc->mFormat) &&
|
|
(channelMask == outputDesc->mChannelMask)) {
|
|
outputDesc->mDirectOpenCount++;
|
|
ALOGV("getOutput() reusing direct output %d", mOutputs.keyAt(i));
|
|
return mOutputs.keyAt(i);
|
|
}
|
|
}
|
|
}
|
|
// close direct output if currently open and configured with different parameters
|
|
if (outputDesc != NULL) {
|
|
closeOutput(outputDesc->mId);
|
|
}
|
|
outputDesc = new AudioOutputDescriptor(profile);
|
|
outputDesc->mDevice = device;
|
|
outputDesc->mSamplingRate = samplingRate;
|
|
outputDesc->mFormat = (audio_format_t)format;
|
|
outputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
|
|
outputDesc->mLatency = 0;
|
|
outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags);
|
|
outputDesc->mRefCount[stream] = 0;
|
|
outputDesc->mStopTime[stream] = 0;
|
|
outputDesc->mDirectOpenCount = 1;
|
|
output = mpClientInterface->openOutput(profile->mModule->mHandle,
|
|
&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannelMask,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags,
|
|
offloadInfo);
|
|
|
|
// only accept an output with the requested parameters
|
|
if (output == 0 ||
|
|
(samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
|
|
(format != 0 && format != outputDesc->mFormat) ||
|
|
(channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
|
|
ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
|
|
"format %d %d, channelMask %04x %04x", output, samplingRate,
|
|
outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
|
|
outputDesc->mChannelMask);
|
|
if (output != 0) {
|
|
mpClientInterface->closeOutput(output);
|
|
}
|
|
delete outputDesc;
|
|
return 0;
|
|
}
|
|
addOutput(output, outputDesc);
|
|
mPreviousOutputs = mOutputs;
|
|
ALOGV("getOutput() returns new direct output %d", output);
|
|
return output;
|
|
}
|
|
|
|
// ignoring channel mask due to downmix capability in mixer
|
|
|
|
// open a non direct output
|
|
|
|
// for non direct outputs, only PCM is supported
|
|
if (audio_is_linear_pcm((audio_format_t)format)) {
|
|
// get which output is suitable for the specified stream. The actual
|
|
// routing change will happen when startOutput() will be called
|
|
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
|
|
|
|
output = selectOutput(outputs, flags);
|
|
}
|
|
ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
|
|
"format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
|
|
|
|
ALOGV("getOutput() returns output %d", output);
|
|
|
|
return output;
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
|
|
AudioSystem::output_flags flags)
|
|
{
|
|
// select one output among several that provide a path to a particular device or set of
|
|
// devices (the list was previously build by getOutputsForDevice()).
|
|
// The priority is as follows:
|
|
// 1: the output with the highest number of requested policy flags
|
|
// 2: the primary output
|
|
// 3: the first output in the list
|
|
|
|
if (outputs.size() == 0) {
|
|
return 0;
|
|
}
|
|
if (outputs.size() == 1) {
|
|
return outputs[0];
|
|
}
|
|
|
|
int maxCommonFlags = 0;
|
|
audio_io_handle_t outputFlags = 0;
|
|
audio_io_handle_t outputPrimary = 0;
|
|
|
|
for (size_t i = 0; i < outputs.size(); i++) {
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(outputs[i]);
|
|
if (!outputDesc->isDuplicated()) {
|
|
int commonFlags = (int)AudioSystem::popCount(outputDesc->mProfile->mFlags & flags);
|
|
if (commonFlags > maxCommonFlags) {
|
|
outputFlags = outputs[i];
|
|
maxCommonFlags = commonFlags;
|
|
ALOGV("selectOutput() commonFlags for output %d, %04x", outputs[i], commonFlags);
|
|
}
|
|
if (outputDesc->mProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
|
|
outputPrimary = outputs[i];
|
|
}
|
|
}
|
|
}
|
|
|
|
if (outputFlags != 0) {
|
|
return outputFlags;
|
|
}
|
|
if (outputPrimary != 0) {
|
|
return outputPrimary;
|
|
}
|
|
|
|
return outputs[0];
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
|
|
AudioSystem::stream_type stream,
|
|
int session)
|
|
{
|
|
ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
|
|
ssize_t index = mOutputs.indexOfKey(output);
|
|
if (index < 0) {
|
|
ALOGW("startOutput() unknow output %d", output);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
|
|
|
|
// increment usage count for this stream on the requested output:
|
|
// NOTE that the usage count is the same for duplicated output and hardware output which is
|
|
// necessary for a correct control of hardware output routing by startOutput() and stopOutput()
|
|
outputDesc->changeRefCount(stream, 1);
|
|
|
|
if (outputDesc->mRefCount[stream] == 1) {
|
|
audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
|
|
routing_strategy strategy = getStrategy(stream);
|
|
bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
|
|
(strategy == STRATEGY_SONIFICATION_RESPECTFUL);
|
|
uint32_t waitMs = 0;
|
|
bool force = false;
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
AudioOutputDescriptor *desc = mOutputs.valueAt(i);
|
|
if (desc != outputDesc) {
|
|
// force a device change if any other output is managed by the same hw
|
|
// module and has a current device selection that differs from selected device.
|
|
// In this case, the audio HAL must receive the new device selection so that it can
|
|
// change the device currently selected by the other active output.
|
|
if (outputDesc->sharesHwModuleWith(desc) &&
|
|
desc->device() != newDevice) {
|
|
force = true;
|
|
}
|
|
// wait for audio on other active outputs to be presented when starting
|
|
// a notification so that audio focus effect can propagate.
|
|
uint32_t latency = desc->latency();
|
|
if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
|
|
waitMs = latency;
|
|
}
|
|
}
|
|
}
|
|
uint32_t muteWaitMs = setOutputDevice(output, newDevice, force);
|
|
|
|
// handle special case for sonification while in call
|
|
if (isInCall()) {
|
|
handleIncallSonification(stream, true, false);
|
|
}
|
|
|
|
// apply volume rules for current stream and device if necessary
|
|
checkAndSetVolume(stream,
|
|
mStreams[stream].getVolumeIndex(newDevice),
|
|
output,
|
|
newDevice);
|
|
|
|
// update the outputs if starting an output with a stream that can affect notification
|
|
// routing
|
|
handleNotificationRoutingForStream(stream);
|
|
if (waitMs > muteWaitMs) {
|
|
usleep((waitMs - muteWaitMs) * 2 * 1000);
|
|
}
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
|
|
status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
|
|
AudioSystem::stream_type stream,
|
|
int session)
|
|
{
|
|
ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
|
|
ssize_t index = mOutputs.indexOfKey(output);
|
|
if (index < 0) {
|
|
ALOGW("stopOutput() unknow output %d", output);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
|
|
|
|
// handle special case for sonification while in call
|
|
if (isInCall()) {
|
|
handleIncallSonification(stream, false, false);
|
|
}
|
|
|
|
if (outputDesc->mRefCount[stream] > 0) {
|
|
// decrement usage count of this stream on the output
|
|
outputDesc->changeRefCount(stream, -1);
|
|
// store time at which the stream was stopped - see isStreamActive()
|
|
if (outputDesc->mRefCount[stream] == 0) {
|
|
outputDesc->mStopTime[stream] = systemTime();
|
|
audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
|
|
// delay the device switch by twice the latency because stopOutput() is executed when
|
|
// the track stop() command is received and at that time the audio track buffer can
|
|
// still contain data that needs to be drained. The latency only covers the audio HAL
|
|
// and kernel buffers. Also the latency does not always include additional delay in the
|
|
// audio path (audio DSP, CODEC ...)
|
|
setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
|
|
|
|
// force restoring the device selection on other active outputs if it differs from the
|
|
// one being selected for this output
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(i);
|
|
AudioOutputDescriptor *desc = mOutputs.valueAt(i);
|
|
if (curOutput != output &&
|
|
desc->isActive() &&
|
|
outputDesc->sharesHwModuleWith(desc) &&
|
|
(newDevice != desc->device())) {
|
|
setOutputDevice(curOutput,
|
|
getNewDevice(curOutput, false /*fromCache*/),
|
|
true,
|
|
outputDesc->mLatency*2);
|
|
}
|
|
}
|
|
// update the outputs if stopping one with a stream that can affect notification routing
|
|
handleNotificationRoutingForStream(stream);
|
|
}
|
|
return NO_ERROR;
|
|
} else {
|
|
ALOGW("stopOutput() refcount is already 0 for output %d", output);
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
|
|
{
|
|
ALOGV("releaseOutput() %d", output);
|
|
ssize_t index = mOutputs.indexOfKey(output);
|
|
if (index < 0) {
|
|
ALOGW("releaseOutput() releasing unknown output %d", output);
|
|
return;
|
|
}
|
|
|
|
#ifdef AUDIO_POLICY_TEST
|
|
int testIndex = testOutputIndex(output);
|
|
if (testIndex != 0) {
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
|
|
if (outputDesc->isActive()) {
|
|
mpClientInterface->closeOutput(output);
|
|
delete mOutputs.valueAt(index);
|
|
mOutputs.removeItem(output);
|
|
mTestOutputs[testIndex] = 0;
|
|
}
|
|
return;
|
|
}
|
|
#endif //AUDIO_POLICY_TEST
|
|
|
|
AudioOutputDescriptor *desc = mOutputs.valueAt(index);
|
|
if (desc->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
|
|
if (desc->mDirectOpenCount <= 0) {
|
|
ALOGW("releaseOutput() invalid open count %d for output %d",
|
|
desc->mDirectOpenCount, output);
|
|
return;
|
|
}
|
|
if (--desc->mDirectOpenCount == 0) {
|
|
closeOutput(output);
|
|
}
|
|
}
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channelMask,
|
|
AudioSystem::audio_in_acoustics acoustics)
|
|
{
|
|
audio_io_handle_t input = 0;
|
|
audio_devices_t device = getDeviceForInputSource(inputSource);
|
|
|
|
ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
|
|
inputSource, samplingRate, format, channelMask, acoustics);
|
|
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
ALOGW("getInput() could not find device for inputSource %d", inputSource);
|
|
return 0;
|
|
}
|
|
|
|
// adapt channel selection to input source
|
|
switch(inputSource) {
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
channelMask = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
channelMask = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
channelMask = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
IOProfile *profile = getInputProfile(device,
|
|
samplingRate,
|
|
format,
|
|
channelMask);
|
|
if (profile == NULL) {
|
|
ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d,"
|
|
"channelMask %04x",
|
|
device, samplingRate, format, channelMask);
|
|
return 0;
|
|
}
|
|
|
|
if (profile->mModule->mHandle == 0) {
|
|
ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
|
|
return 0;
|
|
}
|
|
|
|
AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
|
|
|
|
inputDesc->mInputSource = inputSource;
|
|
inputDesc->mDevice = device;
|
|
inputDesc->mSamplingRate = samplingRate;
|
|
inputDesc->mFormat = (audio_format_t)format;
|
|
inputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
|
|
inputDesc->mRefCount = 0;
|
|
input = mpClientInterface->openInput(profile->mModule->mHandle,
|
|
&inputDesc->mDevice,
|
|
&inputDesc->mSamplingRate,
|
|
&inputDesc->mFormat,
|
|
&inputDesc->mChannelMask);
|
|
|
|
// only accept input with the exact requested set of parameters
|
|
if (input == 0 ||
|
|
(samplingRate != inputDesc->mSamplingRate) ||
|
|
(format != inputDesc->mFormat) ||
|
|
(channelMask != inputDesc->mChannelMask)) {
|
|
ALOGV("getInput() failed opening input: samplingRate %d, format %d, channelMask %d",
|
|
samplingRate, format, channelMask);
|
|
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)
|
|
{
|
|
ALOGV("startInput() input %d", input);
|
|
ssize_t index = mInputs.indexOfKey(input);
|
|
if (index < 0) {
|
|
ALOGW("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 (!isVirtualInputDevice(inputDesc->mDevice) && getActiveInput(true) != 0) {
|
|
ALOGW("startInput() input %d failed: other input already started", input);
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
|
|
audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
|
|
if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
|
|
inputDesc->mDevice = newDevice;
|
|
}
|
|
|
|
// automatically enable the remote submix output when input is started
|
|
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
|
|
setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
|
|
AudioSystem::DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
|
|
}
|
|
|
|
AudioParameter param = AudioParameter();
|
|
param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
|
|
|
|
param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource);
|
|
ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
|
|
|
|
mpClientInterface->setParameters(input, param.toString());
|
|
|
|
inputDesc->mRefCount = 1;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
|
|
{
|
|
ALOGV("stopInput() input %d", input);
|
|
ssize_t index = mInputs.indexOfKey(input);
|
|
if (index < 0) {
|
|
ALOGW("stopInput() unknow input %d", input);
|
|
return BAD_VALUE;
|
|
}
|
|
AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
|
|
|
|
if (inputDesc->mRefCount == 0) {
|
|
ALOGW("stopInput() input %d already stopped", input);
|
|
return INVALID_OPERATION;
|
|
} else {
|
|
// automatically disable the remote submix output when input is stopped
|
|
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
|
|
setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
|
|
AudioSystem::DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
|
|
}
|
|
|
|
AudioParameter param = AudioParameter();
|
|
param.addInt(String8(AudioParameter::keyRouting), 0);
|
|
mpClientInterface->setParameters(input, param.toString());
|
|
inputDesc->mRefCount = 0;
|
|
return NO_ERROR;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
|
|
{
|
|
ALOGV("releaseInput() %d", input);
|
|
ssize_t index = mInputs.indexOfKey(input);
|
|
if (index < 0) {
|
|
ALOGW("releaseInput() releasing unknown input %d", input);
|
|
return;
|
|
}
|
|
mpClientInterface->closeInput(input);
|
|
delete mInputs.valueAt(index);
|
|
mInputs.removeItem(input);
|
|
ALOGV("releaseInput() exit");
|
|
}
|
|
|
|
void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
|
|
int indexMin,
|
|
int indexMax)
|
|
{
|
|
ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
|
|
if (indexMin < 0 || indexMin >= indexMax) {
|
|
ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
|
|
return;
|
|
}
|
|
mStreams[stream].mIndexMin = indexMin;
|
|
mStreams[stream].mIndexMax = indexMax;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream,
|
|
int index,
|
|
audio_devices_t device)
|
|
{
|
|
|
|
if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
|
|
return BAD_VALUE;
|
|
}
|
|
if (!audio_is_output_device(device)) {
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
// Force max volume if stream cannot be muted
|
|
if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
|
|
|
|
ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
|
|
stream, device, index);
|
|
|
|
// if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
|
|
// clear all device specific values
|
|
if (device == AUDIO_DEVICE_OUT_DEFAULT) {
|
|
mStreams[stream].mIndexCur.clear();
|
|
}
|
|
mStreams[stream].mIndexCur.add(device, index);
|
|
|
|
// compute and apply stream volume on all outputs according to connected device
|
|
status_t status = NO_ERROR;
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_devices_t curDevice =
|
|
getDeviceForVolume(mOutputs.valueAt(i)->device());
|
|
if ((device == AUDIO_DEVICE_OUT_DEFAULT) || (device == curDevice)) {
|
|
status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
|
|
if (volStatus != NO_ERROR) {
|
|
status = volStatus;
|
|
}
|
|
}
|
|
}
|
|
return status;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream,
|
|
int *index,
|
|
audio_devices_t device)
|
|
{
|
|
if (index == NULL) {
|
|
return BAD_VALUE;
|
|
}
|
|
if (!audio_is_output_device(device)) {
|
|
return BAD_VALUE;
|
|
}
|
|
// if device is AUDIO_DEVICE_OUT_DEFAULT, return volume for device corresponding to
|
|
// the strategy the stream belongs to.
|
|
if (device == AUDIO_DEVICE_OUT_DEFAULT) {
|
|
device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
|
|
}
|
|
device = getDeviceForVolume(device);
|
|
|
|
*index = mStreams[stream].getVolumeIndex(device);
|
|
ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(const effect_descriptor_t *desc)
|
|
{
|
|
ALOGV("getOutputForEffect()");
|
|
// apply simple rule where global effects are attached to the same output as MUSIC streams
|
|
|
|
routing_strategy strategy = getStrategy(AudioSystem::MUSIC);
|
|
audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
|
|
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(device, mOutputs);
|
|
int outIdx = 0;
|
|
for (size_t i = 0; i < dstOutputs.size(); i++) {
|
|
AudioOutputDescriptor *desc = mOutputs.valueFor(dstOutputs[i]);
|
|
if (desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
|
|
outIdx = i;
|
|
}
|
|
}
|
|
return dstOutputs[outIdx];
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::registerEffect(const effect_descriptor_t *desc,
|
|
audio_io_handle_t io,
|
|
uint32_t strategy,
|
|
int session,
|
|
int id)
|
|
{
|
|
ssize_t index = mOutputs.indexOfKey(io);
|
|
if (index < 0) {
|
|
index = mInputs.indexOfKey(io);
|
|
if (index < 0) {
|
|
ALOGW("registerEffect() unknown io %d", io);
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
|
|
if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
|
|
ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
|
|
desc->name, desc->memoryUsage);
|
|
return INVALID_OPERATION;
|
|
}
|
|
mTotalEffectsMemory += desc->memoryUsage;
|
|
ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d",
|
|
desc->name, io, strategy, session, id);
|
|
ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory);
|
|
|
|
EffectDescriptor *pDesc = new EffectDescriptor();
|
|
memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
|
|
pDesc->mIo = io;
|
|
pDesc->mStrategy = (routing_strategy)strategy;
|
|
pDesc->mSession = session;
|
|
pDesc->mEnabled = false;
|
|
|
|
mEffects.add(id, pDesc);
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::unregisterEffect(int id)
|
|
{
|
|
ssize_t index = mEffects.indexOfKey(id);
|
|
if (index < 0) {
|
|
ALOGW("unregisterEffect() unknown effect ID %d", id);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
EffectDescriptor *pDesc = mEffects.valueAt(index);
|
|
|
|
setEffectEnabled(pDesc, false);
|
|
|
|
if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
|
|
ALOGW("unregisterEffect() memory %d too big for total %d",
|
|
pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
|
|
pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
|
|
}
|
|
mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
|
|
ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d",
|
|
pDesc->mDesc.name, id, pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
|
|
|
|
mEffects.removeItem(id);
|
|
delete pDesc;
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::setEffectEnabled(int id, bool enabled)
|
|
{
|
|
ssize_t index = mEffects.indexOfKey(id);
|
|
if (index < 0) {
|
|
ALOGW("unregisterEffect() unknown effect ID %d", id);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
return setEffectEnabled(mEffects.valueAt(index), enabled);
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::setEffectEnabled(EffectDescriptor *pDesc, bool enabled)
|
|
{
|
|
if (enabled == pDesc->mEnabled) {
|
|
ALOGV("setEffectEnabled(%s) effect already %s",
|
|
enabled?"true":"false", enabled?"enabled":"disabled");
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
if (enabled) {
|
|
if (mTotalEffectsCpuLoad + pDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) {
|
|
ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS",
|
|
pDesc->mDesc.name, (float)pDesc->mDesc.cpuLoad/10);
|
|
return INVALID_OPERATION;
|
|
}
|
|
mTotalEffectsCpuLoad += pDesc->mDesc.cpuLoad;
|
|
ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad);
|
|
} else {
|
|
if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
|
|
ALOGW("setEffectEnabled(false) CPU load %d too high for total %d",
|
|
pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
|
|
pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
|
|
}
|
|
mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
|
|
ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad);
|
|
}
|
|
pDesc->mEnabled = enabled;
|
|
return NO_ERROR;
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const
|
|
{
|
|
nsecs_t sysTime = systemTime();
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
|
|
if (outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::isStreamActiveRemotely(int stream, uint32_t inPastMs) const
|
|
{
|
|
nsecs_t sysTime = systemTime();
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
const AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
|
|
if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
|
|
outputDesc->isStreamActive((AudioSystem::stream_type)stream, inPastMs, sysTime)) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::isSourceActive(audio_source_t source) const
|
|
{
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
const AudioInputDescriptor * inputDescriptor = mInputs.valueAt(i);
|
|
if ((inputDescriptor->mInputSource == (int) source)
|
|
&& (inputDescriptor->mRefCount > 0)) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
|
|
|
|
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, " Primary Output: %d\n", mPrimaryOutput);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " USB audio ALSA %s\n", mUsbCardAndDevice.string());
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[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);
|
|
snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AudioSystem::FOR_SYSTEM]);
|
|
result.append(buffer);
|
|
write(fd, result.string(), result.size());
|
|
|
|
|
|
snprintf(buffer, SIZE, "\nHW Modules dump:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < mHwModules.size(); i++) {
|
|
snprintf(buffer, SIZE, "- HW Module %d:\n", i + 1);
|
|
write(fd, buffer, strlen(buffer));
|
|
mHwModules[i]->dump(fd);
|
|
}
|
|
|
|
snprintf(buffer, SIZE, "\nOutputs dump:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
|
|
write(fd, buffer, strlen(buffer));
|
|
mOutputs.valueAt(i)->dump(fd);
|
|
}
|
|
|
|
snprintf(buffer, SIZE, "\nInputs dump:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
|
|
write(fd, buffer, strlen(buffer));
|
|
mInputs.valueAt(i)->dump(fd);
|
|
}
|
|
|
|
snprintf(buffer, SIZE, "\nStreams dump:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
snprintf(buffer, SIZE,
|
|
" Stream Can be muted Index Min Index Max Index Cur [device : index]...\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
snprintf(buffer, SIZE, " %02d ", i);
|
|
write(fd, buffer, strlen(buffer));
|
|
mStreams[i].dump(fd);
|
|
}
|
|
|
|
snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
|
|
(float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
|
|
write(fd, buffer, strlen(buffer));
|
|
|
|
snprintf(buffer, SIZE, "Registered effects:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < mEffects.size(); i++) {
|
|
snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
|
|
write(fd, buffer, strlen(buffer));
|
|
mEffects.valueAt(i)->dump(fd);
|
|
}
|
|
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// This function checks for the parameters which can be offloaded.
|
|
// This can be enhanced depending on the capability of the DSP and policy
|
|
// of the system.
|
|
bool AudioPolicyManagerBase::isOffloadSupported(const audio_offload_info_t& offloadInfo)
|
|
{
|
|
ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
|
|
" BitRate=%u, duration=%lld us, has_video=%d",
|
|
offloadInfo.sample_rate, offloadInfo.channel_mask,
|
|
offloadInfo.format,
|
|
offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
|
|
offloadInfo.has_video);
|
|
|
|
// Check if offload has been disabled
|
|
char propValue[PROPERTY_VALUE_MAX];
|
|
if (property_get("audio.offload.disable", propValue, "0")) {
|
|
if (atoi(propValue) != 0) {
|
|
ALOGV("offload disabled by audio.offload.disable=%s", propValue );
|
|
return false;
|
|
}
|
|
}
|
|
|
|
// Check if stream type is music, then only allow offload as of now.
|
|
if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
|
|
{
|
|
ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
|
|
return false;
|
|
}
|
|
|
|
//TODO: enable audio offloading with video when ready
|
|
if (offloadInfo.has_video)
|
|
{
|
|
ALOGV("isOffloadSupported: has_video == true, returning false");
|
|
return false;
|
|
}
|
|
|
|
//If duration is less than minimum value defined in property, return false
|
|
if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
|
|
if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
|
|
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
|
|
return false;
|
|
}
|
|
} else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
|
|
ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
|
|
return false;
|
|
}
|
|
|
|
// See if there is a profile to support this.
|
|
// AUDIO_DEVICE_NONE
|
|
IOProfile *profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
|
|
offloadInfo.sample_rate,
|
|
offloadInfo.format,
|
|
offloadInfo.channel_mask,
|
|
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
|
|
ALOGV("isOffloadSupported() profile %sfound", profile != NULL ? "" : "NOT ");
|
|
return (profile != NULL);
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
// AudioPolicyManagerBase
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
|
|
:
|
|
#ifdef AUDIO_POLICY_TEST
|
|
Thread(false),
|
|
#endif //AUDIO_POLICY_TEST
|
|
mPrimaryOutput((audio_io_handle_t)0),
|
|
mAvailableOutputDevices(AUDIO_DEVICE_NONE),
|
|
mPhoneState(AudioSystem::MODE_NORMAL),
|
|
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
|
|
mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
|
|
mA2dpSuspended(false), mHasA2dp(false), mHasUsb(false), mHasRemoteSubmix(false)
|
|
{
|
|
mpClientInterface = clientInterface;
|
|
|
|
for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
|
|
mForceUse[i] = AudioSystem::FORCE_NONE;
|
|
}
|
|
|
|
initializeVolumeCurves();
|
|
|
|
mA2dpDeviceAddress = String8("");
|
|
mScoDeviceAddress = String8("");
|
|
mUsbCardAndDevice = String8("");
|
|
|
|
if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
|
|
if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
|
|
ALOGE("could not load audio policy configuration file, setting defaults");
|
|
defaultAudioPolicyConfig();
|
|
}
|
|
}
|
|
|
|
// open all output streams needed to access attached devices
|
|
for (size_t i = 0; i < mHwModules.size(); i++) {
|
|
mHwModules[i]->mHandle = mpClientInterface->loadHwModule(mHwModules[i]->mName);
|
|
if (mHwModules[i]->mHandle == 0) {
|
|
ALOGW("could not open HW module %s", mHwModules[i]->mName);
|
|
continue;
|
|
}
|
|
// open all output streams needed to access attached devices
|
|
// except for direct output streams that are only opened when they are actually
|
|
// required by an app.
|
|
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
|
|
{
|
|
const IOProfile *outProfile = mHwModules[i]->mOutputProfiles[j];
|
|
|
|
if ((outProfile->mSupportedDevices & mAttachedOutputDevices) &&
|
|
((outProfile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0)) {
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(outProfile);
|
|
outputDesc->mDevice = (audio_devices_t)(mDefaultOutputDevice &
|
|
outProfile->mSupportedDevices);
|
|
audio_io_handle_t output = mpClientInterface->openOutput(
|
|
outProfile->mModule->mHandle,
|
|
&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannelMask,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
if (output == 0) {
|
|
delete outputDesc;
|
|
} else {
|
|
mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices |
|
|
(outProfile->mSupportedDevices & mAttachedOutputDevices));
|
|
if (mPrimaryOutput == 0 &&
|
|
outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) {
|
|
mPrimaryOutput = output;
|
|
}
|
|
addOutput(output, outputDesc);
|
|
setOutputDevice(output,
|
|
(audio_devices_t)(mDefaultOutputDevice &
|
|
outProfile->mSupportedDevices),
|
|
true);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
ALOGE_IF((mAttachedOutputDevices & ~mAvailableOutputDevices),
|
|
"Not output found for attached devices %08x",
|
|
(mAttachedOutputDevices & ~mAvailableOutputDevices));
|
|
|
|
ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
|
|
|
|
updateDevicesAndOutputs();
|
|
|
|
#ifdef AUDIO_POLICY_TEST
|
|
if (mPrimaryOutput != 0) {
|
|
AudioParameter outputCmd = AudioParameter();
|
|
outputCmd.addInt(String8("set_id"), 0);
|
|
mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
|
|
|
|
mTestDevice = AUDIO_DEVICE_OUT_SPEAKER;
|
|
mTestSamplingRate = 44100;
|
|
mTestFormat = 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);
|
|
}
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
mpClientInterface->closeInput(mInputs.keyAt(i));
|
|
delete mInputs.valueAt(i);
|
|
}
|
|
for (size_t i = 0; i < mHwModules.size(); i++) {
|
|
delete mHwModules[i];
|
|
}
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::initCheck()
|
|
{
|
|
return (mPrimaryOutput == 0) ? NO_INIT : NO_ERROR;
|
|
}
|
|
|
|
#ifdef AUDIO_POLICY_TEST
|
|
bool AudioPolicyManagerBase::threadLoop()
|
|
{
|
|
ALOGV("entering threadLoop()");
|
|
while (!exitPending())
|
|
{
|
|
String8 command;
|
|
int valueInt;
|
|
String8 value;
|
|
|
|
Mutex::Autolock _l(mLock);
|
|
mWaitWorkCV.waitRelative(mLock, milliseconds(50));
|
|
|
|
command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
|
|
AudioParameter param = AudioParameter(command);
|
|
|
|
if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
|
|
valueInt != 0) {
|
|
ALOGV("Test command %s received", command.string());
|
|
String8 target;
|
|
if (param.get(String8("target"), target) != NO_ERROR) {
|
|
target = "Manager";
|
|
}
|
|
if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
|
|
param.remove(String8("test_cmd_policy_output"));
|
|
mCurOutput = valueInt;
|
|
}
|
|
if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
|
|
param.remove(String8("test_cmd_policy_direct"));
|
|
if (value == "false") {
|
|
mDirectOutput = false;
|
|
} else if (value == "true") {
|
|
mDirectOutput = true;
|
|
}
|
|
}
|
|
if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
|
|
param.remove(String8("test_cmd_policy_input"));
|
|
mTestInput = valueInt;
|
|
}
|
|
|
|
if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
|
|
param.remove(String8("test_cmd_policy_format"));
|
|
int format = 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"));
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
|
|
mpClientInterface->closeOutput(mPrimaryOutput);
|
|
|
|
audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle;
|
|
|
|
delete mOutputs.valueFor(mPrimaryOutput);
|
|
mOutputs.removeItem(mPrimaryOutput);
|
|
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
|
|
outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER;
|
|
mPrimaryOutput = mpClientInterface->openOutput(moduleHandle,
|
|
&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannelMask,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
if (mPrimaryOutput == 0) {
|
|
ALOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
|
|
outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannelMask);
|
|
} else {
|
|
AudioParameter outputCmd = AudioParameter();
|
|
outputCmd.addInt(String8("set_id"), 0);
|
|
mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
|
|
addOutput(mPrimaryOutput, outputDesc);
|
|
}
|
|
}
|
|
|
|
|
|
mpClientInterface->setParameters(0, String8("test_cmd_policy="));
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
void 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);
|
|
}
|
|
|
|
|
|
status_t AudioPolicyManagerBase::checkOutputsForDevice(audio_devices_t device,
|
|
AudioSystem::device_connection_state state,
|
|
SortedVector<audio_io_handle_t>& outputs)
|
|
{
|
|
AudioOutputDescriptor *desc;
|
|
|
|
if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
|
|
// first list already open outputs that can be routed to this device
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
desc = mOutputs.valueAt(i);
|
|
if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices & device)) {
|
|
ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
|
|
outputs.add(mOutputs.keyAt(i));
|
|
}
|
|
}
|
|
// then look for output profiles that can be routed to this device
|
|
SortedVector<IOProfile *> profiles;
|
|
for (size_t i = 0; i < mHwModules.size(); i++)
|
|
{
|
|
if (mHwModules[i]->mHandle == 0) {
|
|
continue;
|
|
}
|
|
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
|
|
{
|
|
if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices & device) {
|
|
ALOGV("checkOutputsForDevice(): adding profile %d from module %d", j, i);
|
|
profiles.add(mHwModules[i]->mOutputProfiles[j]);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (profiles.isEmpty() && outputs.isEmpty()) {
|
|
ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
// open outputs for matching profiles if needed. Direct outputs are also opened to
|
|
// query for dynamic parameters and will be closed later by setDeviceConnectionState()
|
|
for (ssize_t profile_index = 0; profile_index < (ssize_t)profiles.size(); profile_index++) {
|
|
IOProfile *profile = profiles[profile_index];
|
|
|
|
// nothing to do if one output is already opened for this profile
|
|
size_t j;
|
|
for (j = 0; j < mOutputs.size(); j++) {
|
|
desc = mOutputs.valueAt(j);
|
|
if (!desc->isDuplicated() && desc->mProfile == profile) {
|
|
break;
|
|
}
|
|
}
|
|
if (j != mOutputs.size()) {
|
|
continue;
|
|
}
|
|
|
|
ALOGV("opening output for device %08x", device);
|
|
desc = new AudioOutputDescriptor(profile);
|
|
desc->mDevice = device;
|
|
audio_io_handle_t output = mpClientInterface->openOutput(profile->mModule->mHandle,
|
|
&desc->mDevice,
|
|
&desc->mSamplingRate,
|
|
&desc->mFormat,
|
|
&desc->mChannelMask,
|
|
&desc->mLatency,
|
|
desc->mFlags);
|
|
if (output != 0) {
|
|
if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
|
|
String8 reply;
|
|
char *value;
|
|
if (profile->mSamplingRates[0] == 0) {
|
|
reply = mpClientInterface->getParameters(output,
|
|
String8(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES));
|
|
ALOGV("checkOutputsForDevice() direct output sup sampling rates %s",
|
|
reply.string());
|
|
value = strpbrk((char *)reply.string(), "=");
|
|
if (value != NULL) {
|
|
loadSamplingRates(value + 1, profile);
|
|
}
|
|
}
|
|
if (profile->mFormats[0] == 0) {
|
|
reply = mpClientInterface->getParameters(output,
|
|
String8(AUDIO_PARAMETER_STREAM_SUP_FORMATS));
|
|
ALOGV("checkOutputsForDevice() direct output sup formats %s",
|
|
reply.string());
|
|
value = strpbrk((char *)reply.string(), "=");
|
|
if (value != NULL) {
|
|
loadFormats(value + 1, profile);
|
|
}
|
|
}
|
|
if (profile->mChannelMasks[0] == 0) {
|
|
reply = mpClientInterface->getParameters(output,
|
|
String8(AUDIO_PARAMETER_STREAM_SUP_CHANNELS));
|
|
ALOGV("checkOutputsForDevice() direct output sup channel masks %s",
|
|
reply.string());
|
|
value = strpbrk((char *)reply.string(), "=");
|
|
if (value != NULL) {
|
|
loadOutChannels(value + 1, profile);
|
|
}
|
|
}
|
|
if (((profile->mSamplingRates[0] == 0) &&
|
|
(profile->mSamplingRates.size() < 2)) ||
|
|
((profile->mFormats[0] == 0) &&
|
|
(profile->mFormats.size() < 2)) ||
|
|
((profile->mFormats[0] == 0) &&
|
|
(profile->mChannelMasks.size() < 2))) {
|
|
ALOGW("checkOutputsForDevice() direct output missing param");
|
|
mpClientInterface->closeOutput(output);
|
|
output = 0;
|
|
} else {
|
|
addOutput(output, desc);
|
|
}
|
|
} else {
|
|
audio_io_handle_t duplicatedOutput = 0;
|
|
// add output descriptor
|
|
addOutput(output, desc);
|
|
// set initial stream volume for device
|
|
applyStreamVolumes(output, device, 0, true);
|
|
|
|
//TODO: configure audio effect output stage here
|
|
|
|
// open a duplicating output thread for the new output and the primary output
|
|
duplicatedOutput = mpClientInterface->openDuplicateOutput(output,
|
|
mPrimaryOutput);
|
|
if (duplicatedOutput != 0) {
|
|
// add duplicated output descriptor
|
|
AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(NULL);
|
|
dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput);
|
|
dupOutputDesc->mOutput2 = mOutputs.valueFor(output);
|
|
dupOutputDesc->mSamplingRate = desc->mSamplingRate;
|
|
dupOutputDesc->mFormat = desc->mFormat;
|
|
dupOutputDesc->mChannelMask = desc->mChannelMask;
|
|
dupOutputDesc->mLatency = desc->mLatency;
|
|
addOutput(duplicatedOutput, dupOutputDesc);
|
|
applyStreamVolumes(duplicatedOutput, device, 0, true);
|
|
} else {
|
|
ALOGW("checkOutputsForDevice() could not open dup output for %d and %d",
|
|
mPrimaryOutput, output);
|
|
mpClientInterface->closeOutput(output);
|
|
mOutputs.removeItem(output);
|
|
output = 0;
|
|
}
|
|
}
|
|
}
|
|
if (output == 0) {
|
|
ALOGW("checkOutputsForDevice() could not open output for device %x", device);
|
|
delete desc;
|
|
profiles.removeAt(profile_index);
|
|
profile_index--;
|
|
} else {
|
|
outputs.add(output);
|
|
ALOGV("checkOutputsForDevice(): adding output %d", output);
|
|
}
|
|
}
|
|
|
|
if (profiles.isEmpty()) {
|
|
ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
|
|
return BAD_VALUE;
|
|
}
|
|
} else {
|
|
// check if one opened output is not needed any more after disconnecting one device
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
desc = mOutputs.valueAt(i);
|
|
if (!desc->isDuplicated() &&
|
|
!(desc->mProfile->mSupportedDevices & mAvailableOutputDevices)) {
|
|
ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i));
|
|
outputs.add(mOutputs.keyAt(i));
|
|
}
|
|
}
|
|
for (size_t i = 0; i < mHwModules.size(); i++)
|
|
{
|
|
if (mHwModules[i]->mHandle == 0) {
|
|
continue;
|
|
}
|
|
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
|
|
{
|
|
IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
|
|
if ((profile->mSupportedDevices & device) &&
|
|
(profile->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
|
|
ALOGV("checkOutputsForDevice(): clearing direct output profile %d on module %d",
|
|
j, i);
|
|
if (profile->mSamplingRates[0] == 0) {
|
|
profile->mSamplingRates.clear();
|
|
profile->mSamplingRates.add(0);
|
|
}
|
|
if (profile->mFormats[0] == 0) {
|
|
profile->mFormats.clear();
|
|
profile->mFormats.add((audio_format_t)0);
|
|
}
|
|
if (profile->mChannelMasks[0] == 0) {
|
|
profile->mChannelMasks.clear();
|
|
profile->mChannelMasks.add((audio_channel_mask_t)0);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::closeOutput(audio_io_handle_t output)
|
|
{
|
|
ALOGV("closeOutput(%d)", output);
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
if (outputDesc == NULL) {
|
|
ALOGW("closeOutput() unknown output %d", output);
|
|
return;
|
|
}
|
|
|
|
// look for duplicated outputs connected to the output being removed.
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
AudioOutputDescriptor *dupOutputDesc = mOutputs.valueAt(i);
|
|
if (dupOutputDesc->isDuplicated() &&
|
|
(dupOutputDesc->mOutput1 == outputDesc ||
|
|
dupOutputDesc->mOutput2 == outputDesc)) {
|
|
AudioOutputDescriptor *outputDesc2;
|
|
if (dupOutputDesc->mOutput1 == outputDesc) {
|
|
outputDesc2 = dupOutputDesc->mOutput2;
|
|
} else {
|
|
outputDesc2 = dupOutputDesc->mOutput1;
|
|
}
|
|
// As all active tracks on duplicated output will be deleted,
|
|
// and as they were also referenced on the other output, the reference
|
|
// count for their stream type must be adjusted accordingly on
|
|
// the other output.
|
|
for (int j = 0; j < (int)AudioSystem::NUM_STREAM_TYPES; j++) {
|
|
int refCount = dupOutputDesc->mRefCount[j];
|
|
outputDesc2->changeRefCount((AudioSystem::stream_type)j,-refCount);
|
|
}
|
|
audio_io_handle_t duplicatedOutput = mOutputs.keyAt(i);
|
|
ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput);
|
|
|
|
mpClientInterface->closeOutput(duplicatedOutput);
|
|
delete mOutputs.valueFor(duplicatedOutput);
|
|
mOutputs.removeItem(duplicatedOutput);
|
|
}
|
|
}
|
|
|
|
AudioParameter param;
|
|
param.add(String8("closing"), String8("true"));
|
|
mpClientInterface->setParameters(output, param.toString());
|
|
|
|
mpClientInterface->closeOutput(output);
|
|
delete outputDesc;
|
|
mOutputs.removeItem(output);
|
|
mPreviousOutputs = mOutputs;
|
|
}
|
|
|
|
SortedVector<audio_io_handle_t> AudioPolicyManagerBase::getOutputsForDevice(audio_devices_t device,
|
|
DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs)
|
|
{
|
|
SortedVector<audio_io_handle_t> outputs;
|
|
|
|
ALOGVV("getOutputsForDevice() device %04x", device);
|
|
for (size_t i = 0; i < openOutputs.size(); i++) {
|
|
ALOGVV("output %d isDuplicated=%d device=%04x",
|
|
i, openOutputs.valueAt(i)->isDuplicated(), openOutputs.valueAt(i)->supportedDevices());
|
|
if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
|
|
ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
|
|
outputs.add(openOutputs.keyAt(i));
|
|
}
|
|
}
|
|
return outputs;
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
|
|
SortedVector<audio_io_handle_t>& outputs2)
|
|
{
|
|
if (outputs1.size() != outputs2.size()) {
|
|
return false;
|
|
}
|
|
for (size_t i = 0; i < outputs1.size(); i++) {
|
|
if (outputs1[i] != outputs2[i]) {
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy)
|
|
{
|
|
audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
|
|
audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
|
|
SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
|
|
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
|
|
|
|
if (!vectorsEqual(srcOutputs,dstOutputs)) {
|
|
ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
|
|
strategy, srcOutputs[0], dstOutputs[0]);
|
|
// mute strategy while moving tracks from one output to another
|
|
for (size_t i = 0; i < srcOutputs.size(); i++) {
|
|
AudioOutputDescriptor *desc = mOutputs.valueFor(srcOutputs[i]);
|
|
if (desc->isStrategyActive(strategy)) {
|
|
setStrategyMute(strategy, true, srcOutputs[i]);
|
|
setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice);
|
|
}
|
|
}
|
|
|
|
// Move effects associated to this strategy from previous output to new output
|
|
if (strategy == STRATEGY_MEDIA) {
|
|
int outIdx = 0;
|
|
for (size_t i = 0; i < dstOutputs.size(); i++) {
|
|
AudioOutputDescriptor *desc = mOutputs.valueFor(dstOutputs[i]);
|
|
if (desc->mFlags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
|
|
outIdx = i;
|
|
}
|
|
}
|
|
SortedVector<audio_io_handle_t> moved;
|
|
for (size_t i = 0; i < mEffects.size(); i++) {
|
|
EffectDescriptor *desc = mEffects.valueAt(i);
|
|
if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX &&
|
|
desc->mIo != dstOutputs[outIdx]) {
|
|
if (moved.indexOf(desc->mIo) < 0) {
|
|
ALOGV("checkOutputForStrategy() moving effect %d to output %d",
|
|
mEffects.keyAt(i), dstOutputs[outIdx]);
|
|
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, desc->mIo,
|
|
dstOutputs[outIdx]);
|
|
moved.add(desc->mIo);
|
|
}
|
|
desc->mIo = dstOutputs[outIdx];
|
|
}
|
|
}
|
|
}
|
|
// Move tracks associated to this strategy from previous output to new output
|
|
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
|
|
//FIXME see fixme on name change
|
|
mpClientInterface->setStreamOutput((AudioSystem::stream_type)i,
|
|
dstOutputs[0] /* ignored */);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::checkOutputForAllStrategies()
|
|
{
|
|
checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
|
|
checkOutputForStrategy(STRATEGY_PHONE);
|
|
checkOutputForStrategy(STRATEGY_SONIFICATION);
|
|
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
|
|
checkOutputForStrategy(STRATEGY_MEDIA);
|
|
checkOutputForStrategy(STRATEGY_DTMF);
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput()
|
|
{
|
|
if (!mHasA2dp) {
|
|
return 0;
|
|
}
|
|
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
|
|
if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
|
|
return mOutputs.keyAt(i);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::checkA2dpSuspend()
|
|
{
|
|
if (!mHasA2dp) {
|
|
return;
|
|
}
|
|
audio_io_handle_t a2dpOutput = getA2dpOutput();
|
|
if (a2dpOutput == 0) {
|
|
return;
|
|
}
|
|
|
|
// suspend A2DP output if:
|
|
// (NOT already suspended) &&
|
|
// ((SCO device is connected &&
|
|
// (forced usage for communication || for record is SCO))) ||
|
|
// (phone state is ringing || in call)
|
|
//
|
|
// restore A2DP output if:
|
|
// (Already suspended) &&
|
|
// ((SCO device is NOT connected ||
|
|
// (forced usage NOT for communication && NOT for record is SCO))) &&
|
|
// (phone state is NOT ringing && NOT in call)
|
|
//
|
|
if (mA2dpSuspended) {
|
|
if (((mScoDeviceAddress == "") ||
|
|
((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
|
|
(mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) &&
|
|
((mPhoneState != AudioSystem::MODE_IN_CALL) &&
|
|
(mPhoneState != AudioSystem::MODE_RINGTONE))) {
|
|
|
|
mpClientInterface->restoreOutput(a2dpOutput);
|
|
mA2dpSuspended = false;
|
|
}
|
|
} else {
|
|
if (((mScoDeviceAddress != "") &&
|
|
((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
|
|
(mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) ||
|
|
((mPhoneState == AudioSystem::MODE_IN_CALL) ||
|
|
(mPhoneState == AudioSystem::MODE_RINGTONE))) {
|
|
|
|
mpClientInterface->suspendOutput(a2dpOutput);
|
|
mA2dpSuspended = true;
|
|
}
|
|
}
|
|
}
|
|
|
|
audio_devices_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
|
|
{
|
|
audio_devices_t device = AUDIO_DEVICE_NONE;
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
// check the following by order of priority to request a routing change if necessary:
|
|
// 1: the strategy enforced audible is active on the output:
|
|
// use device for strategy enforced audible
|
|
// 2: we are in call or the strategy phone is active on the output:
|
|
// use device for strategy phone
|
|
// 3: the strategy sonification is active on the output:
|
|
// use device for strategy sonification
|
|
// 4: the strategy "respectful" sonification is active on the output:
|
|
// use device for strategy "respectful" sonification
|
|
// 5: the strategy media is active on the output:
|
|
// use device for strategy media
|
|
// 6: the strategy DTMF is active on the output:
|
|
// use device for strategy DTMF
|
|
if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
|
|
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
|
|
} else if (isInCall() ||
|
|
outputDesc->isStrategyActive(STRATEGY_PHONE)) {
|
|
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
|
|
} else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
|
|
} else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
|
|
} else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
|
|
} else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
|
|
device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
|
|
}
|
|
|
|
ALOGV("getNewDevice() selected device %x", device);
|
|
return device;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
|
|
return (uint32_t)getStrategy(stream);
|
|
}
|
|
|
|
audio_devices_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
|
|
audio_devices_t devices;
|
|
// By checking the range of stream before calling getStrategy, we avoid
|
|
// getStrategy's behavior for invalid streams. getStrategy would do a ALOGE
|
|
// and then return STRATEGY_MEDIA, but we want to return the empty set.
|
|
if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
|
|
devices = AUDIO_DEVICE_NONE;
|
|
} else {
|
|
AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
|
|
devices = getDeviceForStrategy(strategy, true /*fromCache*/);
|
|
}
|
|
return devices;
|
|
}
|
|
|
|
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::ALARM:
|
|
return STRATEGY_SONIFICATION;
|
|
case AudioSystem::NOTIFICATION:
|
|
return STRATEGY_SONIFICATION_RESPECTFUL;
|
|
case AudioSystem::DTMF:
|
|
return STRATEGY_DTMF;
|
|
default:
|
|
ALOGE("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;
|
|
case AudioSystem::ENFORCED_AUDIBLE:
|
|
return STRATEGY_ENFORCED_AUDIBLE;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::handleNotificationRoutingForStream(AudioSystem::stream_type stream) {
|
|
switch(stream) {
|
|
case AudioSystem::MUSIC:
|
|
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
|
|
updateDevicesAndOutputs();
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
audio_devices_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy,
|
|
bool fromCache)
|
|
{
|
|
uint32_t device = AUDIO_DEVICE_NONE;
|
|
|
|
if (fromCache) {
|
|
ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
|
|
strategy, mDeviceForStrategy[strategy]);
|
|
return mDeviceForStrategy[strategy];
|
|
}
|
|
|
|
switch (strategy) {
|
|
|
|
case STRATEGY_SONIFICATION_RESPECTFUL:
|
|
if (isInCall()) {
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
|
|
} else if (isStreamActiveRemotely(AudioSystem::MUSIC,
|
|
SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
|
|
// while media is playing on a remote device, use the the sonification behavior.
|
|
// Note that we test this usecase before testing if media is playing because
|
|
// the isStreamActive() method only informs about the activity of a stream, not
|
|
// if it's for local playback. Note also that we use the same delay between both tests
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
|
|
} else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
|
|
// while media is playing (or has recently played), use the same device
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
|
|
} else {
|
|
// when media is not playing anymore, fall back on the sonification behavior
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
|
|
}
|
|
|
|
break;
|
|
|
|
case STRATEGY_DTMF:
|
|
if (!isInCall()) {
|
|
// when off call, DTMF strategy follows the same rules as MEDIA strategy
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
|
|
break;
|
|
}
|
|
// when in call, DTMF and PHONE strategies follow the same rules
|
|
// FALL THROUGH
|
|
|
|
case STRATEGY_PHONE:
|
|
// for phone strategy, we first consider the forced use and then the available devices by order
|
|
// of priority
|
|
switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
|
|
case AudioSystem::FORCE_BT_SCO:
|
|
if (!isInCall() || strategy != STRATEGY_DTMF) {
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
|
|
if (device) break;
|
|
// if SCO device is requested but no SCO device is available, fall back to default case
|
|
// FALL THROUGH
|
|
|
|
default: // FORCE_NONE
|
|
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
|
|
if (mHasA2dp && !isInCall() &&
|
|
(mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
|
|
(getA2dpOutput() != 0) && !mA2dpSuspended) {
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
|
|
if (device) break;
|
|
if (mPhoneState != AudioSystem::MODE_IN_CALL) {
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
|
|
if (device) break;
|
|
device = mDefaultOutputDevice;
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
|
|
}
|
|
break;
|
|
|
|
case AudioSystem::FORCE_SPEAKER:
|
|
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
|
|
// A2DP speaker when forcing to speaker output
|
|
if (mHasA2dp && !isInCall() &&
|
|
(mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
|
|
(getA2dpOutput() != 0) && !mA2dpSuspended) {
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
|
|
if (device) break;
|
|
}
|
|
if (mPhoneState != AudioSystem::MODE_IN_CALL) {
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
|
|
if (device) break;
|
|
device = mDefaultOutputDevice;
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
|
|
}
|
|
break;
|
|
}
|
|
break;
|
|
|
|
case STRATEGY_SONIFICATION:
|
|
|
|
// If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
|
|
// handleIncallSonification().
|
|
if (isInCall()) {
|
|
device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
|
|
break;
|
|
}
|
|
// FALL THROUGH
|
|
|
|
case STRATEGY_ENFORCED_AUDIBLE:
|
|
// strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
|
|
// except:
|
|
// - when in call where it doesn't default to STRATEGY_PHONE behavior
|
|
// - in countries where not enforced in which case it follows STRATEGY_MEDIA
|
|
|
|
if ((strategy == STRATEGY_SONIFICATION) ||
|
|
(mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) {
|
|
device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
|
|
}
|
|
}
|
|
// The second device used for sonification is the same as the device used by media strategy
|
|
// FALL THROUGH
|
|
|
|
case STRATEGY_MEDIA: {
|
|
uint32_t device2 = AUDIO_DEVICE_NONE;
|
|
if (strategy != STRATEGY_SONIFICATION) {
|
|
// no sonification on remote submix (e.g. WFD)
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
|
|
}
|
|
if ((device2 == AUDIO_DEVICE_NONE) &&
|
|
mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
|
|
(getA2dpOutput() != 0) && !mA2dpSuspended) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
|
|
}
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
|
|
}
|
|
}
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
|
|
}
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
|
|
}
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
|
|
}
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
|
|
}
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
|
|
}
|
|
if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
|
|
// no sonification on aux digital (e.g. HDMI)
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
|
|
}
|
|
if ((device2 == AUDIO_DEVICE_NONE) &&
|
|
(mForceUse[AudioSystem::FOR_DOCK] == AudioSystem::FORCE_ANALOG_DOCK)) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
|
|
}
|
|
if (device2 == AUDIO_DEVICE_NONE) {
|
|
device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
|
|
}
|
|
|
|
// device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
|
|
// STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
|
|
device |= device2;
|
|
if (device) break;
|
|
device = mDefaultOutputDevice;
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
|
|
}
|
|
} break;
|
|
|
|
default:
|
|
ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
|
|
break;
|
|
}
|
|
|
|
ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
|
|
return device;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::updateDevicesAndOutputs()
|
|
{
|
|
for (int i = 0; i < NUM_STRATEGIES; i++) {
|
|
mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
|
|
}
|
|
mPreviousOutputs = mOutputs;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
|
|
audio_devices_t prevDevice,
|
|
uint32_t delayMs)
|
|
{
|
|
// mute/unmute strategies using an incompatible device combination
|
|
// if muting, wait for the audio in pcm buffer to be drained before proceeding
|
|
// if unmuting, unmute only after the specified delay
|
|
if (outputDesc->isDuplicated()) {
|
|
return 0;
|
|
}
|
|
|
|
uint32_t muteWaitMs = 0;
|
|
audio_devices_t device = outputDesc->device();
|
|
bool shouldMute = outputDesc->isActive() && (AudioSystem::popCount(device) >= 2);
|
|
// temporary mute output if device selection changes to avoid volume bursts due to
|
|
// different per device volumes
|
|
bool tempMute = outputDesc->isActive() && (device != prevDevice);
|
|
|
|
for (size_t i = 0; i < NUM_STRATEGIES; i++) {
|
|
audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
|
|
bool mute = shouldMute && (curDevice & device) && (curDevice != device);
|
|
bool doMute = false;
|
|
|
|
if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
|
|
doMute = true;
|
|
outputDesc->mStrategyMutedByDevice[i] = true;
|
|
} else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
|
|
doMute = true;
|
|
outputDesc->mStrategyMutedByDevice[i] = false;
|
|
}
|
|
if (doMute || tempMute) {
|
|
for (size_t j = 0; j < mOutputs.size(); j++) {
|
|
AudioOutputDescriptor *desc = mOutputs.valueAt(j);
|
|
// skip output if it does not share any device with current output
|
|
if ((desc->supportedDevices() & outputDesc->supportedDevices())
|
|
== AUDIO_DEVICE_NONE) {
|
|
continue;
|
|
}
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(j);
|
|
ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
|
|
mute ? "muting" : "unmuting", i, curDevice, curOutput);
|
|
setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
|
|
if (desc->isStrategyActive((routing_strategy)i)) {
|
|
// do tempMute only for current output
|
|
if (tempMute && (desc == outputDesc)) {
|
|
setStrategyMute((routing_strategy)i, true, curOutput);
|
|
setStrategyMute((routing_strategy)i, false, curOutput,
|
|
desc->latency() * 2, device);
|
|
}
|
|
if ((tempMute && (desc == outputDesc)) || mute) {
|
|
if (muteWaitMs < desc->latency()) {
|
|
muteWaitMs = desc->latency();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// FIXME: should not need to double latency if volume could be applied immediately by the
|
|
// audioflinger mixer. We must account for the delay between now and the next time
|
|
// the audioflinger thread for this output will process a buffer (which corresponds to
|
|
// one buffer size, usually 1/2 or 1/4 of the latency).
|
|
muteWaitMs *= 2;
|
|
// wait for the PCM output buffers to empty before proceeding with the rest of the command
|
|
if (muteWaitMs > delayMs) {
|
|
muteWaitMs -= delayMs;
|
|
usleep(muteWaitMs * 1000);
|
|
return muteWaitMs;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output,
|
|
audio_devices_t device,
|
|
bool force,
|
|
int delayMs)
|
|
{
|
|
ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
AudioParameter param;
|
|
uint32_t muteWaitMs;
|
|
|
|
if (outputDesc->isDuplicated()) {
|
|
muteWaitMs = setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
|
|
muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
|
|
return muteWaitMs;
|
|
}
|
|
// no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
|
|
// output profile
|
|
if ((device != AUDIO_DEVICE_NONE) &&
|
|
((device & outputDesc->mProfile->mSupportedDevices) == 0)) {
|
|
return 0;
|
|
}
|
|
|
|
// filter devices according to output selected
|
|
device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices);
|
|
|
|
audio_devices_t prevDevice = outputDesc->mDevice;
|
|
|
|
ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
|
|
|
|
if (device != AUDIO_DEVICE_NONE) {
|
|
outputDesc->mDevice = device;
|
|
}
|
|
muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
|
|
|
|
// Do not change the routing if:
|
|
// - the requested device is AUDIO_DEVICE_NONE
|
|
// - the requested device is the same as current device and force is not specified.
|
|
// Doing this check here allows the caller to call setOutputDevice() without conditions
|
|
if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force) {
|
|
ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
|
|
return muteWaitMs;
|
|
}
|
|
|
|
ALOGV("setOutputDevice() changing device");
|
|
// do the routing
|
|
param.addInt(String8(AudioParameter::keyRouting), (int)device);
|
|
mpClientInterface->setParameters(output, param.toString(), delayMs);
|
|
|
|
// update stream volumes according to new device
|
|
applyStreamVolumes(output, device, delayMs);
|
|
|
|
return muteWaitMs;
|
|
}
|
|
|
|
AudioPolicyManagerBase::IOProfile *AudioPolicyManagerBase::getInputProfile(audio_devices_t device,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channelMask)
|
|
{
|
|
// Choose an input profile based on the requested capture parameters: select the first available
|
|
// profile supporting all requested parameters.
|
|
|
|
for (size_t i = 0; i < mHwModules.size(); i++)
|
|
{
|
|
if (mHwModules[i]->mHandle == 0) {
|
|
continue;
|
|
}
|
|
for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++)
|
|
{
|
|
IOProfile *profile = mHwModules[i]->mInputProfiles[j];
|
|
if (profile->isCompatibleProfile(device, samplingRate, format,
|
|
channelMask,(audio_output_flags_t)0)) {
|
|
return profile;
|
|
}
|
|
}
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
audio_devices_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
|
|
{
|
|
uint32_t device = AUDIO_DEVICE_NONE;
|
|
|
|
switch (inputSource) {
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
|
|
device = AUDIO_DEVICE_IN_VOICE_CALL;
|
|
break;
|
|
}
|
|
// FALL THROUGH
|
|
|
|
case AUDIO_SOURCE_DEFAULT:
|
|
case AUDIO_SOURCE_MIC:
|
|
case AUDIO_SOURCE_VOICE_RECOGNITION:
|
|
case AUDIO_SOURCE_VOICE_COMMUNICATION:
|
|
if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
|
|
mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
|
|
device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
|
|
} else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
|
|
device = AUDIO_DEVICE_IN_WIRED_HEADSET;
|
|
} else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
|
|
device = AUDIO_DEVICE_IN_BUILTIN_MIC;
|
|
}
|
|
break;
|
|
case AUDIO_SOURCE_CAMCORDER:
|
|
if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
|
|
device = AUDIO_DEVICE_IN_BACK_MIC;
|
|
} else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
|
|
device = AUDIO_DEVICE_IN_BUILTIN_MIC;
|
|
}
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
|
|
device = AUDIO_DEVICE_IN_VOICE_CALL;
|
|
}
|
|
break;
|
|
case AUDIO_SOURCE_REMOTE_SUBMIX:
|
|
if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
|
|
device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
|
|
}
|
|
break;
|
|
default:
|
|
ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
|
|
break;
|
|
}
|
|
ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
|
|
return device;
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::isVirtualInputDevice(audio_devices_t device)
|
|
{
|
|
if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
|
|
device &= ~AUDIO_DEVICE_BIT_IN;
|
|
if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0))
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getActiveInput(bool ignoreVirtualInputs)
|
|
{
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
const AudioInputDescriptor * input_descriptor = mInputs.valueAt(i);
|
|
if ((input_descriptor->mRefCount > 0)
|
|
&& (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) {
|
|
return mInputs.keyAt(i);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
audio_devices_t AudioPolicyManagerBase::getDeviceForVolume(audio_devices_t device)
|
|
{
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
// this happens when forcing a route update and no track is active on an output.
|
|
// In this case the returned category is not important.
|
|
device = AUDIO_DEVICE_OUT_SPEAKER;
|
|
} else if (AudioSystem::popCount(device) > 1) {
|
|
// Multiple device selection is either:
|
|
// - speaker + one other device: give priority to speaker in this case.
|
|
// - one A2DP device + another device: happens with duplicated output. In this case
|
|
// retain the device on the A2DP output as the other must not correspond to an active
|
|
// selection if not the speaker.
|
|
if (device & AUDIO_DEVICE_OUT_SPEAKER) {
|
|
device = AUDIO_DEVICE_OUT_SPEAKER;
|
|
} else {
|
|
device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
|
|
}
|
|
}
|
|
|
|
ALOGW_IF(AudioSystem::popCount(device) != 1,
|
|
"getDeviceForVolume() invalid device combination: %08x",
|
|
device);
|
|
|
|
return device;
|
|
}
|
|
|
|
AudioPolicyManagerBase::device_category AudioPolicyManagerBase::getDeviceCategory(audio_devices_t device)
|
|
{
|
|
switch(getDeviceForVolume(device)) {
|
|
case AUDIO_DEVICE_OUT_EARPIECE:
|
|
return DEVICE_CATEGORY_EARPIECE;
|
|
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
|
|
case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
|
|
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
|
|
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
|
|
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
|
|
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
|
|
return DEVICE_CATEGORY_HEADSET;
|
|
case AUDIO_DEVICE_OUT_SPEAKER:
|
|
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
|
|
case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
|
|
case AUDIO_DEVICE_OUT_AUX_DIGITAL:
|
|
case AUDIO_DEVICE_OUT_USB_ACCESSORY:
|
|
case AUDIO_DEVICE_OUT_USB_DEVICE:
|
|
case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
|
|
default:
|
|
return DEVICE_CATEGORY_SPEAKER;
|
|
}
|
|
}
|
|
|
|
float AudioPolicyManagerBase::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
|
|
int indexInUi)
|
|
{
|
|
device_category deviceCategory = getDeviceCategory(device);
|
|
const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
|
|
|
|
// the volume index in the UI is relative to the min and max volume indices for this stream type
|
|
int nbSteps = 1 + curve[VOLMAX].mIndex -
|
|
curve[VOLMIN].mIndex;
|
|
int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
|
|
(streamDesc.mIndexMax - streamDesc.mIndexMin);
|
|
|
|
// find what part of the curve this index volume belongs to, or if it's out of bounds
|
|
int segment = 0;
|
|
if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
|
|
return 0.0f;
|
|
} else if (volIdx < curve[VOLKNEE1].mIndex) {
|
|
segment = 0;
|
|
} else if (volIdx < curve[VOLKNEE2].mIndex) {
|
|
segment = 1;
|
|
} else if (volIdx <= curve[VOLMAX].mIndex) {
|
|
segment = 2;
|
|
} else { // out of bounds
|
|
return 1.0f;
|
|
}
|
|
|
|
// linear interpolation in the attenuation table in dB
|
|
float decibels = curve[segment].mDBAttenuation +
|
|
((float)(volIdx - curve[segment].mIndex)) *
|
|
( (curve[segment+1].mDBAttenuation -
|
|
curve[segment].mDBAttenuation) /
|
|
((float)(curve[segment+1].mIndex -
|
|
curve[segment].mIndex)) );
|
|
|
|
float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
|
|
|
|
ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
|
|
curve[segment].mIndex, volIdx,
|
|
curve[segment+1].mIndex,
|
|
curve[segment].mDBAttenuation,
|
|
decibels,
|
|
curve[segment+1].mDBAttenuation,
|
|
amplification);
|
|
|
|
return amplification;
|
|
}
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
|
|
};
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
|
|
};
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
|
|
};
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
|
|
};
|
|
|
|
// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
|
|
// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets (See AudioService.java).
|
|
// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
|
|
};
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
|
|
};
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
|
|
};
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
AudioPolicyManagerBase::sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT] = {
|
|
{0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
|
|
};
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
*AudioPolicyManagerBase::sVolumeProfiles[AUDIO_STREAM_CNT]
|
|
[AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = {
|
|
{ // AUDIO_STREAM_VOICE_CALL
|
|
sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_SYSTEM
|
|
sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_RING
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_MUSIC
|
|
sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_ALARM
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_NOTIFICATION
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_BLUETOOTH_SCO
|
|
sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVoiceVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_ENFORCED_AUDIBLE
|
|
sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_DTMF
|
|
sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultSystemVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // AUDIO_STREAM_TTS
|
|
sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
};
|
|
|
|
void AudioPolicyManagerBase::initializeVolumeCurves()
|
|
{
|
|
for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
|
|
for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
|
|
mStreams[i].mVolumeCurve[j] =
|
|
sVolumeProfiles[i][j];
|
|
}
|
|
}
|
|
}
|
|
|
|
float AudioPolicyManagerBase::computeVolume(int stream,
|
|
int index,
|
|
audio_io_handle_t output,
|
|
audio_devices_t device)
|
|
{
|
|
float volume = 1.0;
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
StreamDescriptor &streamDesc = mStreams[stream];
|
|
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
device = outputDesc->device();
|
|
}
|
|
|
|
// if volume is not 0 (not muted), force media volume to max on digital output
|
|
if (stream == AudioSystem::MUSIC &&
|
|
index != mStreams[stream].mIndexMin &&
|
|
(device == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
|
|
device == AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET ||
|
|
device == AUDIO_DEVICE_OUT_USB_ACCESSORY ||
|
|
device == AUDIO_DEVICE_OUT_USB_DEVICE)) {
|
|
return 1.0;
|
|
}
|
|
|
|
volume = volIndexToAmpl(device, streamDesc, index);
|
|
|
|
// if a headset is connected, apply the following rules to ring tones and notifications
|
|
// to avoid sound level bursts in user's ears:
|
|
// - always attenuate ring tones and notifications volume by 6dB
|
|
// - if music is playing, always limit the volume to current music volume,
|
|
// with a minimum threshold at -36dB so that notification is always perceived.
|
|
const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream);
|
|
if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
|
|
AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
|
|
AUDIO_DEVICE_OUT_WIRED_HEADSET |
|
|
AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) &&
|
|
((stream_strategy == STRATEGY_SONIFICATION)
|
|
|| (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL)
|
|
|| (stream == AudioSystem::SYSTEM)
|
|
|| ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) &&
|
|
(mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) &&
|
|
streamDesc.mCanBeMuted) {
|
|
volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
|
|
// when the phone is ringing we must consider that music could have been paused just before
|
|
// by the music application and behave as if music was active if the last music track was
|
|
// just stopped
|
|
if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
|
|
mLimitRingtoneVolume) {
|
|
audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/);
|
|
float musicVol = computeVolume(AudioSystem::MUSIC,
|
|
mStreams[AudioSystem::MUSIC].getVolumeIndex(musicDevice),
|
|
output,
|
|
musicDevice);
|
|
float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ?
|
|
musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
|
|
if (volume > minVol) {
|
|
volume = minVol;
|
|
ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
|
|
}
|
|
}
|
|
}
|
|
|
|
return volume;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::checkAndSetVolume(int stream,
|
|
int index,
|
|
audio_io_handle_t output,
|
|
audio_devices_t device,
|
|
int delayMs,
|
|
bool force)
|
|
{
|
|
|
|
// do not change actual stream volume if the stream is muted
|
|
if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
|
|
ALOGVV("checkAndSetVolume() stream %d muted count %d",
|
|
stream, mOutputs.valueFor(output)->mMuteCount[stream]);
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// do not change in call volume if bluetooth is connected and vice versa
|
|
if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
|
|
(stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
|
|
ALOGV("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);
|
|
// We actually change the volume if:
|
|
// - the float value returned by computeVolume() changed
|
|
// - the force flag is set
|
|
if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
|
|
force) {
|
|
mOutputs.valueFor(output)->mCurVolume[stream] = volume;
|
|
ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
|
|
// Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
|
|
// enabled
|
|
if (stream == AudioSystem::BLUETOOTH_SCO) {
|
|
mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
|
|
}
|
|
mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
|
|
}
|
|
|
|
if (stream == AudioSystem::VOICE_CALL ||
|
|
stream == AudioSystem::BLUETOOTH_SCO) {
|
|
float voiceVolume;
|
|
// Force voice volume to max for bluetooth SCO as volume is managed by the headset
|
|
if (stream == AudioSystem::VOICE_CALL) {
|
|
voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
|
|
} else {
|
|
voiceVolume = 1.0;
|
|
}
|
|
|
|
if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) {
|
|
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
|
|
mLastVoiceVolume = voiceVolume;
|
|
}
|
|
}
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output,
|
|
audio_devices_t device,
|
|
int delayMs,
|
|
bool force)
|
|
{
|
|
ALOGVV("applyStreamVolumes() for output %d and device %x", output, device);
|
|
|
|
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
|
|
checkAndSetVolume(stream,
|
|
mStreams[stream].getVolumeIndex(device),
|
|
output,
|
|
device,
|
|
delayMs,
|
|
force);
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy,
|
|
bool on,
|
|
audio_io_handle_t output,
|
|
int delayMs,
|
|
audio_devices_t device)
|
|
{
|
|
ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
|
|
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
|
|
if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
|
|
setStreamMute(stream, on, output, delayMs, device);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setStreamMute(int stream,
|
|
bool on,
|
|
audio_io_handle_t output,
|
|
int delayMs,
|
|
audio_devices_t device)
|
|
{
|
|
StreamDescriptor &streamDesc = mStreams[stream];
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
device = outputDesc->device();
|
|
}
|
|
|
|
ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
|
|
stream, on, output, outputDesc->mMuteCount[stream], device);
|
|
|
|
if (on) {
|
|
if (outputDesc->mMuteCount[stream] == 0) {
|
|
if (streamDesc.mCanBeMuted &&
|
|
((stream != AudioSystem::ENFORCED_AUDIBLE) ||
|
|
(mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) {
|
|
checkAndSetVolume(stream, 0, output, device, delayMs);
|
|
}
|
|
}
|
|
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
|
|
outputDesc->mMuteCount[stream]++;
|
|
} else {
|
|
if (outputDesc->mMuteCount[stream] == 0) {
|
|
ALOGV("setStreamMute() unmuting non muted stream!");
|
|
return;
|
|
}
|
|
if (--outputDesc->mMuteCount[stream] == 0) {
|
|
checkAndSetVolume(stream,
|
|
streamDesc.getVolumeIndex(device),
|
|
output,
|
|
device,
|
|
delayMs);
|
|
}
|
|
}
|
|
}
|
|
|
|
void 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
|
|
const routing_strategy stream_strategy = getStrategy((AudioSystem::stream_type)stream);
|
|
if ((stream_strategy == STRATEGY_SONIFICATION) ||
|
|
((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mPrimaryOutput);
|
|
ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
|
|
stream, starting, outputDesc->mDevice, stateChange);
|
|
if (outputDesc->mRefCount[stream]) {
|
|
int muteCount = 1;
|
|
if (stateChange) {
|
|
muteCount = outputDesc->mRefCount[stream];
|
|
}
|
|
if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
|
|
ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
|
|
for (int i = 0; i < muteCount; i++) {
|
|
setStreamMute(stream, starting, mPrimaryOutput);
|
|
}
|
|
} else {
|
|
ALOGV("handleIncallSonification() high visibility");
|
|
if (outputDesc->device() &
|
|
getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
|
|
ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
|
|
for (int i = 0; i < muteCount; i++) {
|
|
setStreamMute(stream, starting, mPrimaryOutput);
|
|
}
|
|
}
|
|
if (starting) {
|
|
mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
|
|
} else {
|
|
mpClientInterface->stopTone();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::isInCall()
|
|
{
|
|
return isStateInCall(mPhoneState);
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::isStateInCall(int state) {
|
|
return ((state == AudioSystem::MODE_IN_CALL) ||
|
|
(state == AudioSystem::MODE_IN_COMMUNICATION));
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
|
|
{
|
|
return MAX_EFFECTS_CPU_LOAD;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
|
|
{
|
|
return MAX_EFFECTS_MEMORY;
|
|
}
|
|
|
|
// --- AudioOutputDescriptor class implementation
|
|
|
|
AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor(
|
|
const IOProfile *profile)
|
|
: mId(0), mSamplingRate(0), mFormat((audio_format_t)0),
|
|
mChannelMask((audio_channel_mask_t)0), mLatency(0),
|
|
mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE),
|
|
mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
|
|
{
|
|
// clear usage count for all stream types
|
|
for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
mRefCount[i] = 0;
|
|
mCurVolume[i] = -1.0;
|
|
mMuteCount[i] = 0;
|
|
mStopTime[i] = 0;
|
|
}
|
|
for (int i = 0; i < NUM_STRATEGIES; i++) {
|
|
mStrategyMutedByDevice[i] = false;
|
|
}
|
|
if (profile != NULL) {
|
|
mSamplingRate = profile->mSamplingRates[0];
|
|
mFormat = profile->mFormats[0];
|
|
mChannelMask = profile->mChannelMasks[0];
|
|
mFlags = profile->mFlags;
|
|
}
|
|
}
|
|
|
|
audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::device() const
|
|
{
|
|
if (isDuplicated()) {
|
|
return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
|
|
} else {
|
|
return mDevice;
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::latency()
|
|
{
|
|
if (isDuplicated()) {
|
|
return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
|
|
} else {
|
|
return mLatency;
|
|
}
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::AudioOutputDescriptor::sharesHwModuleWith(
|
|
const AudioOutputDescriptor *outputDesc)
|
|
{
|
|
if (isDuplicated()) {
|
|
return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
|
|
} else if (outputDesc->isDuplicated()){
|
|
return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
|
|
} else {
|
|
return (mProfile->mModule == outputDesc->mProfile->mModule);
|
|
}
|
|
}
|
|
|
|
void 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) {
|
|
ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
|
|
mRefCount[stream] = 0;
|
|
return;
|
|
}
|
|
mRefCount[stream] += delta;
|
|
ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
|
|
}
|
|
|
|
audio_devices_t AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices()
|
|
{
|
|
if (isDuplicated()) {
|
|
return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
|
|
} else {
|
|
return mProfile->mSupportedDevices ;
|
|
}
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
|
|
{
|
|
return isStrategyActive(NUM_STRATEGIES, inPastMs);
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
|
|
uint32_t inPastMs,
|
|
nsecs_t sysTime) const
|
|
{
|
|
if ((sysTime == 0) && (inPastMs != 0)) {
|
|
sysTime = systemTime();
|
|
}
|
|
for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
if (((getStrategy((AudioSystem::stream_type)i) == strategy) ||
|
|
(NUM_STRATEGIES == strategy)) &&
|
|
isStreamActive((AudioSystem::stream_type)i, inPastMs, sysTime)) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::AudioOutputDescriptor::isStreamActive(AudioSystem::stream_type stream,
|
|
uint32_t inPastMs,
|
|
nsecs_t sysTime) const
|
|
{
|
|
if (mRefCount[stream] != 0) {
|
|
return true;
|
|
}
|
|
if (inPastMs == 0) {
|
|
return false;
|
|
}
|
|
if (sysTime == 0) {
|
|
sysTime = systemTime();
|
|
}
|
|
if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
|
|
status_t 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: %08x\n", mFormat);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Devices %08x\n", device());
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
|
|
result.append(buffer);
|
|
for (int i = 0; i < 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(const IOProfile *profile)
|
|
: mSamplingRate(0), mFormat((audio_format_t)0), mChannelMask((audio_channel_mask_t)0),
|
|
mDevice(AUDIO_DEVICE_NONE), mRefCount(0),
|
|
mInputSource(0), mProfile(profile)
|
|
{
|
|
}
|
|
|
|
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", mChannelMask);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
|
|
result.append(buffer);
|
|
write(fd, result.string(), result.size());
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// --- StreamDescriptor class implementation
|
|
|
|
AudioPolicyManagerBase::StreamDescriptor::StreamDescriptor()
|
|
: mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
|
|
{
|
|
mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
|
|
}
|
|
|
|
int AudioPolicyManagerBase::StreamDescriptor::getVolumeIndex(audio_devices_t device)
|
|
{
|
|
device = AudioPolicyManagerBase::getDeviceForVolume(device);
|
|
// there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
|
|
if (mIndexCur.indexOfKey(device) < 0) {
|
|
device = AUDIO_DEVICE_OUT_DEFAULT;
|
|
}
|
|
return mIndexCur.valueFor(device);
|
|
}
|
|
|
|
void AudioPolicyManagerBase::StreamDescriptor::dump(int fd)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, "%s %02d %02d ",
|
|
mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
|
|
result.append(buffer);
|
|
for (size_t i = 0; i < mIndexCur.size(); i++) {
|
|
snprintf(buffer, SIZE, "%04x : %02d, ",
|
|
mIndexCur.keyAt(i),
|
|
mIndexCur.valueAt(i));
|
|
result.append(buffer);
|
|
}
|
|
result.append("\n");
|
|
|
|
write(fd, result.string(), result.size());
|
|
}
|
|
|
|
// --- EffectDescriptor class implementation
|
|
|
|
status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, " I/O: %d\n", mIo);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Session: %d\n", mSession);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled");
|
|
result.append(buffer);
|
|
write(fd, result.string(), result.size());
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// --- IOProfile class implementation
|
|
|
|
AudioPolicyManagerBase::HwModule::HwModule(const char *name)
|
|
: mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)), mHandle(0)
|
|
{
|
|
}
|
|
|
|
AudioPolicyManagerBase::HwModule::~HwModule()
|
|
{
|
|
for (size_t i = 0; i < mOutputProfiles.size(); i++) {
|
|
delete mOutputProfiles[i];
|
|
}
|
|
for (size_t i = 0; i < mInputProfiles.size(); i++) {
|
|
delete mInputProfiles[i];
|
|
}
|
|
free((void *)mName);
|
|
}
|
|
|
|
void AudioPolicyManagerBase::HwModule::dump(int fd)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, " - name: %s\n", mName);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
|
|
result.append(buffer);
|
|
write(fd, result.string(), result.size());
|
|
if (mOutputProfiles.size()) {
|
|
write(fd, " - outputs:\n", sizeof(" - outputs:\n"));
|
|
for (size_t i = 0; i < mOutputProfiles.size(); i++) {
|
|
snprintf(buffer, SIZE, " output %d:\n", i);
|
|
write(fd, buffer, strlen(buffer));
|
|
mOutputProfiles[i]->dump(fd);
|
|
}
|
|
}
|
|
if (mInputProfiles.size()) {
|
|
write(fd, " - inputs:\n", sizeof(" - inputs:\n"));
|
|
for (size_t i = 0; i < mInputProfiles.size(); i++) {
|
|
snprintf(buffer, SIZE, " input %d:\n", i);
|
|
write(fd, buffer, strlen(buffer));
|
|
mInputProfiles[i]->dump(fd);
|
|
}
|
|
}
|
|
}
|
|
|
|
AudioPolicyManagerBase::IOProfile::IOProfile(HwModule *module)
|
|
: mFlags((audio_output_flags_t)0), mModule(module)
|
|
{
|
|
}
|
|
|
|
AudioPolicyManagerBase::IOProfile::~IOProfile()
|
|
{
|
|
}
|
|
|
|
// checks if the IO profile is compatible with specified parameters. By convention a value of 0
|
|
// means a parameter is don't care
|
|
bool AudioPolicyManagerBase::IOProfile::isCompatibleProfile(audio_devices_t device,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channelMask,
|
|
audio_output_flags_t flags) const
|
|
{
|
|
if ((mSupportedDevices & device) != device) {
|
|
return false;
|
|
}
|
|
if ((mFlags & flags) != flags) {
|
|
return false;
|
|
}
|
|
if (samplingRate != 0) {
|
|
size_t i;
|
|
for (i = 0; i < mSamplingRates.size(); i++)
|
|
{
|
|
if (mSamplingRates[i] == samplingRate) {
|
|
break;
|
|
}
|
|
}
|
|
if (i == mSamplingRates.size()) {
|
|
return false;
|
|
}
|
|
}
|
|
if (format != 0) {
|
|
size_t i;
|
|
for (i = 0; i < mFormats.size(); i++)
|
|
{
|
|
if (mFormats[i] == format) {
|
|
break;
|
|
}
|
|
}
|
|
if (i == mFormats.size()) {
|
|
return false;
|
|
}
|
|
}
|
|
if (channelMask != 0) {
|
|
size_t i;
|
|
for (i = 0; i < mChannelMasks.size(); i++)
|
|
{
|
|
if (mChannelMasks[i] == channelMask) {
|
|
break;
|
|
}
|
|
}
|
|
if (i == mChannelMasks.size()) {
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::IOProfile::dump(int fd)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, " - sampling rates: ");
|
|
result.append(buffer);
|
|
for (size_t i = 0; i < mSamplingRates.size(); i++) {
|
|
snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
|
|
result.append(buffer);
|
|
result.append(i == (mSamplingRates.size() - 1) ? "\n" : ", ");
|
|
}
|
|
|
|
snprintf(buffer, SIZE, " - channel masks: ");
|
|
result.append(buffer);
|
|
for (size_t i = 0; i < mChannelMasks.size(); i++) {
|
|
snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
|
|
result.append(buffer);
|
|
result.append(i == (mChannelMasks.size() - 1) ? "\n" : ", ");
|
|
}
|
|
|
|
snprintf(buffer, SIZE, " - formats: ");
|
|
result.append(buffer);
|
|
for (size_t i = 0; i < mFormats.size(); i++) {
|
|
snprintf(buffer, SIZE, "0x%08x", mFormats[i]);
|
|
result.append(buffer);
|
|
result.append(i == (mFormats.size() - 1) ? "\n" : ", ");
|
|
}
|
|
|
|
snprintf(buffer, SIZE, " - devices: 0x%04x\n", mSupportedDevices);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
|
|
result.append(buffer);
|
|
|
|
write(fd, result.string(), result.size());
|
|
}
|
|
|
|
// --- audio_policy.conf file parsing
|
|
|
|
struct StringToEnum {
|
|
const char *name;
|
|
uint32_t value;
|
|
};
|
|
|
|
#define STRING_TO_ENUM(string) { #string, string }
|
|
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
|
|
|
|
const struct StringToEnum sDeviceNameToEnumTable[] = {
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
|
|
STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
|
|
};
|
|
|
|
const struct StringToEnum sFlagNameToEnumTable[] = {
|
|
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
|
|
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
|
|
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
|
|
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
|
|
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
|
|
STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
|
|
};
|
|
|
|
const struct StringToEnum sFormatNameToEnumTable[] = {
|
|
STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
|
|
STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
|
|
STRING_TO_ENUM(AUDIO_FORMAT_MP3),
|
|
STRING_TO_ENUM(AUDIO_FORMAT_AAC),
|
|
STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
|
|
};
|
|
|
|
const struct StringToEnum sOutChannelsNameToEnumTable[] = {
|
|
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
|
|
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
|
|
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
|
|
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
|
|
};
|
|
|
|
const struct StringToEnum sInChannelsNameToEnumTable[] = {
|
|
STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
|
|
STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
|
|
STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
|
|
};
|
|
|
|
|
|
uint32_t AudioPolicyManagerBase::stringToEnum(const struct StringToEnum *table,
|
|
size_t size,
|
|
const char *name)
|
|
{
|
|
for (size_t i = 0; i < size; i++) {
|
|
if (strcmp(table[i].name, name) == 0) {
|
|
ALOGV("stringToEnum() found %s", table[i].name);
|
|
return table[i].value;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
audio_output_flags_t AudioPolicyManagerBase::parseFlagNames(char *name)
|
|
{
|
|
uint32_t flag = 0;
|
|
|
|
// it is OK to cast name to non const here as we are not going to use it after
|
|
// strtok() modifies it
|
|
char *flagName = strtok(name, "|");
|
|
while (flagName != NULL) {
|
|
if (strlen(flagName) != 0) {
|
|
flag |= stringToEnum(sFlagNameToEnumTable,
|
|
ARRAY_SIZE(sFlagNameToEnumTable),
|
|
flagName);
|
|
}
|
|
flagName = strtok(NULL, "|");
|
|
}
|
|
//force direct flag if offload flag is set: offloading implies a direct output stream
|
|
// and all common behaviors are driven by checking only the direct flag
|
|
// this should normally be set appropriately in the policy configuration file
|
|
if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
|
|
flag |= AUDIO_OUTPUT_FLAG_DIRECT;
|
|
}
|
|
|
|
return (audio_output_flags_t)flag;
|
|
}
|
|
|
|
audio_devices_t AudioPolicyManagerBase::parseDeviceNames(char *name)
|
|
{
|
|
uint32_t device = 0;
|
|
|
|
char *devName = strtok(name, "|");
|
|
while (devName != NULL) {
|
|
if (strlen(devName) != 0) {
|
|
device |= stringToEnum(sDeviceNameToEnumTable,
|
|
ARRAY_SIZE(sDeviceNameToEnumTable),
|
|
devName);
|
|
}
|
|
devName = strtok(NULL, "|");
|
|
}
|
|
return device;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::loadSamplingRates(char *name, IOProfile *profile)
|
|
{
|
|
char *str = strtok(name, "|");
|
|
|
|
// by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
|
|
// rates should be read from the output stream after it is opened for the first time
|
|
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
|
|
profile->mSamplingRates.add(0);
|
|
return;
|
|
}
|
|
|
|
while (str != NULL) {
|
|
uint32_t rate = atoi(str);
|
|
if (rate != 0) {
|
|
ALOGV("loadSamplingRates() adding rate %d", rate);
|
|
profile->mSamplingRates.add(rate);
|
|
}
|
|
str = strtok(NULL, "|");
|
|
}
|
|
return;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::loadFormats(char *name, IOProfile *profile)
|
|
{
|
|
char *str = strtok(name, "|");
|
|
|
|
// by convention, "0' in the first entry in mFormats indicates the supported formats
|
|
// should be read from the output stream after it is opened for the first time
|
|
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
|
|
profile->mFormats.add((audio_format_t)0);
|
|
return;
|
|
}
|
|
|
|
while (str != NULL) {
|
|
audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
|
|
ARRAY_SIZE(sFormatNameToEnumTable),
|
|
str);
|
|
if (format != 0) {
|
|
profile->mFormats.add(format);
|
|
}
|
|
str = strtok(NULL, "|");
|
|
}
|
|
return;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::loadInChannels(char *name, IOProfile *profile)
|
|
{
|
|
const char *str = strtok(name, "|");
|
|
|
|
ALOGV("loadInChannels() %s", name);
|
|
|
|
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
|
|
profile->mChannelMasks.add((audio_channel_mask_t)0);
|
|
return;
|
|
}
|
|
|
|
while (str != NULL) {
|
|
audio_channel_mask_t channelMask =
|
|
(audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
|
|
ARRAY_SIZE(sInChannelsNameToEnumTable),
|
|
str);
|
|
if (channelMask != 0) {
|
|
ALOGV("loadInChannels() adding channelMask %04x", channelMask);
|
|
profile->mChannelMasks.add(channelMask);
|
|
}
|
|
str = strtok(NULL, "|");
|
|
}
|
|
return;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::loadOutChannels(char *name, IOProfile *profile)
|
|
{
|
|
const char *str = strtok(name, "|");
|
|
|
|
ALOGV("loadOutChannels() %s", name);
|
|
|
|
// by convention, "0' in the first entry in mChannelMasks indicates the supported channel
|
|
// masks should be read from the output stream after it is opened for the first time
|
|
if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
|
|
profile->mChannelMasks.add((audio_channel_mask_t)0);
|
|
return;
|
|
}
|
|
|
|
while (str != NULL) {
|
|
audio_channel_mask_t channelMask =
|
|
(audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
|
|
ARRAY_SIZE(sOutChannelsNameToEnumTable),
|
|
str);
|
|
if (channelMask != 0) {
|
|
profile->mChannelMasks.add(channelMask);
|
|
}
|
|
str = strtok(NULL, "|");
|
|
}
|
|
return;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::loadInput(cnode *root, HwModule *module)
|
|
{
|
|
cnode *node = root->first_child;
|
|
|
|
IOProfile *profile = new IOProfile(module);
|
|
|
|
while (node) {
|
|
if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
|
|
loadSamplingRates((char *)node->value, profile);
|
|
} else if (strcmp(node->name, FORMATS_TAG) == 0) {
|
|
loadFormats((char *)node->value, profile);
|
|
} else if (strcmp(node->name, CHANNELS_TAG) == 0) {
|
|
loadInChannels((char *)node->value, profile);
|
|
} else if (strcmp(node->name, DEVICES_TAG) == 0) {
|
|
profile->mSupportedDevices = parseDeviceNames((char *)node->value);
|
|
}
|
|
node = node->next;
|
|
}
|
|
ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
|
|
"loadInput() invalid supported devices");
|
|
ALOGW_IF(profile->mChannelMasks.size() == 0,
|
|
"loadInput() invalid supported channel masks");
|
|
ALOGW_IF(profile->mSamplingRates.size() == 0,
|
|
"loadInput() invalid supported sampling rates");
|
|
ALOGW_IF(profile->mFormats.size() == 0,
|
|
"loadInput() invalid supported formats");
|
|
if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
|
|
(profile->mChannelMasks.size() != 0) &&
|
|
(profile->mSamplingRates.size() != 0) &&
|
|
(profile->mFormats.size() != 0)) {
|
|
|
|
ALOGV("loadInput() adding input mSupportedDevices %04x", profile->mSupportedDevices);
|
|
|
|
module->mInputProfiles.add(profile);
|
|
return NO_ERROR;
|
|
} else {
|
|
delete profile;
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::loadOutput(cnode *root, HwModule *module)
|
|
{
|
|
cnode *node = root->first_child;
|
|
|
|
IOProfile *profile = new IOProfile(module);
|
|
|
|
while (node) {
|
|
if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
|
|
loadSamplingRates((char *)node->value, profile);
|
|
} else if (strcmp(node->name, FORMATS_TAG) == 0) {
|
|
loadFormats((char *)node->value, profile);
|
|
} else if (strcmp(node->name, CHANNELS_TAG) == 0) {
|
|
loadOutChannels((char *)node->value, profile);
|
|
} else if (strcmp(node->name, DEVICES_TAG) == 0) {
|
|
profile->mSupportedDevices = parseDeviceNames((char *)node->value);
|
|
} else if (strcmp(node->name, FLAGS_TAG) == 0) {
|
|
profile->mFlags = parseFlagNames((char *)node->value);
|
|
}
|
|
node = node->next;
|
|
}
|
|
ALOGW_IF(profile->mSupportedDevices == AUDIO_DEVICE_NONE,
|
|
"loadOutput() invalid supported devices");
|
|
ALOGW_IF(profile->mChannelMasks.size() == 0,
|
|
"loadOutput() invalid supported channel masks");
|
|
ALOGW_IF(profile->mSamplingRates.size() == 0,
|
|
"loadOutput() invalid supported sampling rates");
|
|
ALOGW_IF(profile->mFormats.size() == 0,
|
|
"loadOutput() invalid supported formats");
|
|
if ((profile->mSupportedDevices != AUDIO_DEVICE_NONE) &&
|
|
(profile->mChannelMasks.size() != 0) &&
|
|
(profile->mSamplingRates.size() != 0) &&
|
|
(profile->mFormats.size() != 0)) {
|
|
|
|
ALOGV("loadOutput() adding output mSupportedDevices %04x, mFlags %04x",
|
|
profile->mSupportedDevices, profile->mFlags);
|
|
|
|
module->mOutputProfiles.add(profile);
|
|
return NO_ERROR;
|
|
} else {
|
|
delete profile;
|
|
return BAD_VALUE;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::loadHwModule(cnode *root)
|
|
{
|
|
cnode *node = config_find(root, OUTPUTS_TAG);
|
|
status_t status = NAME_NOT_FOUND;
|
|
|
|
HwModule *module = new HwModule(root->name);
|
|
|
|
if (node != NULL) {
|
|
if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
|
|
mHasA2dp = true;
|
|
} else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
|
|
mHasUsb = true;
|
|
} else if (strcmp(root->name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
|
|
mHasRemoteSubmix = true;
|
|
}
|
|
|
|
node = node->first_child;
|
|
while (node) {
|
|
ALOGV("loadHwModule() loading output %s", node->name);
|
|
status_t tmpStatus = loadOutput(node, module);
|
|
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
|
|
status = tmpStatus;
|
|
}
|
|
node = node->next;
|
|
}
|
|
}
|
|
node = config_find(root, INPUTS_TAG);
|
|
if (node != NULL) {
|
|
node = node->first_child;
|
|
while (node) {
|
|
ALOGV("loadHwModule() loading input %s", node->name);
|
|
status_t tmpStatus = loadInput(node, module);
|
|
if (status == NAME_NOT_FOUND || status == NO_ERROR) {
|
|
status = tmpStatus;
|
|
}
|
|
node = node->next;
|
|
}
|
|
}
|
|
if (status == NO_ERROR) {
|
|
mHwModules.add(module);
|
|
} else {
|
|
delete module;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::loadHwModules(cnode *root)
|
|
{
|
|
cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
|
|
if (node == NULL) {
|
|
return;
|
|
}
|
|
|
|
node = node->first_child;
|
|
while (node) {
|
|
ALOGV("loadHwModules() loading module %s", node->name);
|
|
loadHwModule(node);
|
|
node = node->next;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::loadGlobalConfig(cnode *root)
|
|
{
|
|
cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
|
|
if (node == NULL) {
|
|
return;
|
|
}
|
|
node = node->first_child;
|
|
while (node) {
|
|
if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) {
|
|
mAttachedOutputDevices = parseDeviceNames((char *)node->value);
|
|
ALOGW_IF(mAttachedOutputDevices == AUDIO_DEVICE_NONE,
|
|
"loadGlobalConfig() no attached output devices");
|
|
ALOGV("loadGlobalConfig() mAttachedOutputDevices %04x", mAttachedOutputDevices);
|
|
} else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
|
|
mDefaultOutputDevice = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
|
|
ARRAY_SIZE(sDeviceNameToEnumTable),
|
|
(char *)node->value);
|
|
ALOGW_IF(mDefaultOutputDevice == AUDIO_DEVICE_NONE,
|
|
"loadGlobalConfig() default device not specified");
|
|
ALOGV("loadGlobalConfig() mDefaultOutputDevice %04x", mDefaultOutputDevice);
|
|
} else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) {
|
|
mAvailableInputDevices = parseDeviceNames((char *)node->value) & ~AUDIO_DEVICE_BIT_IN;
|
|
ALOGV("loadGlobalConfig() mAvailableInputDevices %04x", mAvailableInputDevices);
|
|
}
|
|
node = node->next;
|
|
}
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::loadAudioPolicyConfig(const char *path)
|
|
{
|
|
cnode *root;
|
|
char *data;
|
|
|
|
data = (char *)load_file(path, NULL);
|
|
if (data == NULL) {
|
|
return -ENODEV;
|
|
}
|
|
root = config_node("", "");
|
|
config_load(root, data);
|
|
|
|
loadGlobalConfig(root);
|
|
loadHwModules(root);
|
|
|
|
config_free(root);
|
|
free(root);
|
|
free(data);
|
|
|
|
ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::defaultAudioPolicyConfig(void)
|
|
{
|
|
HwModule *module;
|
|
IOProfile *profile;
|
|
|
|
mDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER;
|
|
mAttachedOutputDevices = AUDIO_DEVICE_OUT_SPEAKER;
|
|
mAvailableInputDevices = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN;
|
|
|
|
module = new HwModule("primary");
|
|
|
|
profile = new IOProfile(module);
|
|
profile->mSamplingRates.add(44100);
|
|
profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
|
|
profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO);
|
|
profile->mSupportedDevices = AUDIO_DEVICE_OUT_SPEAKER;
|
|
profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY;
|
|
module->mOutputProfiles.add(profile);
|
|
|
|
profile = new IOProfile(module);
|
|
profile->mSamplingRates.add(8000);
|
|
profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT);
|
|
profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO);
|
|
profile->mSupportedDevices = AUDIO_DEVICE_IN_BUILTIN_MIC;
|
|
module->mInputProfiles.add(profile);
|
|
|
|
mHwModules.add(module);
|
|
}
|
|
|
|
}; // namespace android
|