08b014d9e5
Use definition from audio.h for A2DP sink address parameter. Change-Id: I2d7905b8e3dd71fab2efc68ae16682e09c3f872e
2557 lines
99 KiB
C++
2557 lines
99 KiB
C++
/*
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* Copyright (C) 2009 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#define LOG_TAG "AudioPolicyManagerBase"
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//#define LOG_NDEBUG 0
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#include <utils/Log.h>
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#include <hardware_legacy/AudioPolicyManagerBase.h>
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#include <hardware/audio_effect.h>
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#include <hardware/audio.h>
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#include <math.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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////////////////////
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// TODO: the following static configuration will be read from a configuration file
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// sHasA2dp is true on platforms with support for bluetooth A2DP
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bool sHasA2dp = true;
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// devices that are always available on the platform
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audio_devices_t sAttachedOutputDevices =
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(audio_devices_t)(AUDIO_DEVICE_OUT_EARPIECE | AUDIO_DEVICE_OUT_SPEAKER);
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// device selected by default at boot time must be in sAttachedOutputDevices
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audio_devices_t sDefaultOutputDevice = AUDIO_DEVICE_OUT_SPEAKER;
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uint32_t sSamplingRates[] = {44100, 0};
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audio_channel_mask_t sChannels[] = {AUDIO_CHANNEL_OUT_STEREO, (audio_channel_mask_t)0};
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audio_format_t sFormats[] = {AUDIO_FORMAT_PCM_16_BIT, (audio_format_t)0};
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// the primary output (identified by AUDIO_POLICY_OUTPUT_FLAG_PRIMARY in its profile) must exist and
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// is unique on a platform. It is the output receiving the routing and volume commands for telephony
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// use cases. It is normally exposed by the primary audio hw module and opened at boot time by
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// the audio policy manager.
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const output_profile_t sPrimaryOutput = {
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sSamplingRates,
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sChannels,
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sFormats,
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(audio_devices_t)(AUDIO_DEVICE_OUT_EARPIECE |
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AUDIO_DEVICE_OUT_SPEAKER |
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AUDIO_DEVICE_OUT_WIRED_HEADSET |
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AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
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AUDIO_DEVICE_OUT_ALL_SCO |
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AUDIO_DEVICE_OUT_AUX_DIGITAL |
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AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
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AUDIO_POLICY_OUTPUT_FLAG_PRIMARY
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};
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const output_profile_t sA2dpOutput = {
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sSamplingRates,
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sChannels,
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sFormats,
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AUDIO_DEVICE_OUT_ALL_A2DP,
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(audio_policy_output_flags_t)0
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};
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const output_profile_t *sAvailableOutputs[] = {
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&sPrimaryOutput,
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&sA2dpOutput,
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NULL
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};
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///////////// end of temporary static configuration data
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status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device,
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AudioSystem::device_connection_state state,
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const char *device_address)
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{
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audio_io_handle_t output = 0;
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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 (AudioSystem::popCount(device) != 1) 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 (AudioSystem::isOutputDevice(device)) {
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if (!sHasA2dp && AudioSystem::isA2dpDevice(device)) {
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ALOGE("setDeviceConnectionState() invalid device: %x", device);
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return BAD_VALUE;
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}
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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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// register new device as available
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mAvailableOutputDevices |= device;
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output = checkOutputForDevice(device, state);
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if (output == 0) {
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mAvailableOutputDevices &= ~device;
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return INVALID_OPERATION;
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}
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// handle A2DP device connection
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if (sHasA2dp && AudioSystem::isA2dpDevice(device)) {
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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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mpClientInterface->setParameters(output, 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 (AudioSystem::isBluetoothScoDevice(device)) {
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ALOGV("setDeviceConnectionState() BT SCO device, address %s", device_address);
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// keep track of SCO device address
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mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
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}
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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 &= ~device;
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output = checkOutputForDevice(device, state);
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// handle A2DP device disconnection
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if (sHasA2dp && AudioSystem::isA2dpDevice(device)) {
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mA2dpDeviceAddress = "";
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mA2dpSuspended = false;
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} else if (AudioSystem::isBluetoothScoDevice(device)) {
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mScoDeviceAddress = "";
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}
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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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// request routing change if necessary
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uint32_t newDevice = getNewDevice(mPrimaryOutput, false);
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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 (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && output != 0) {
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closeOutput(output);
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}
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updateDeviceForStrategy();
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setOutputDevice(mPrimaryOutput, newDevice);
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if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
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device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
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} else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO ||
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device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
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device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
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device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
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} else {
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return NO_ERROR;
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}
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}
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// handle input devices
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if (AudioSystem::isInputDevice(device)) {
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switch (state)
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{
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// handle input device connection
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case AudioSystem::DEVICE_STATE_AVAILABLE: {
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if (mAvailableInputDevices & device) {
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ALOGW("setDeviceConnectionState() device already connected: %d", device);
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return INVALID_OPERATION;
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}
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mAvailableInputDevices |= device;
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}
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break;
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// handle input device disconnection
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case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
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if (!(mAvailableInputDevices & device)) {
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ALOGW("setDeviceConnectionState() device not connected: %d", device);
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return INVALID_OPERATION;
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}
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mAvailableInputDevices &= ~device;
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} break;
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default:
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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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uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
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if (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(AudioSystem::audio_devices device,
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const char *device_address)
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{
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AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
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String8 address = String8(device_address);
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if (AudioSystem::isOutputDevice(device)) {
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if (device & mAvailableOutputDevices) {
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if (AudioSystem::isA2dpDevice(device) &&
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(!sHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
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return state;
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}
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if (AudioSystem::isBluetoothScoDevice(device) &&
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address != "" && mScoDeviceAddress != address) {
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return state;
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}
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state = AudioSystem::DEVICE_STATE_AVAILABLE;
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}
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} else if (AudioSystem::isInputDevice(device)) {
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if (device & mAvailableInputDevices) {
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state = AudioSystem::DEVICE_STATE_AVAILABLE;
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}
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}
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return state;
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}
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void AudioPolicyManagerBase::setPhoneState(int state)
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{
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ALOGV("setPhoneState() state %d", state);
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uint32_t newDevice = 0;
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if (state < 0 || state >= AudioSystem::NUM_MODES) {
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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);
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checkA2dpSuspend();
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checkOutputForAllStrategies();
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updateDeviceForStrategy();
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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 == 0) {
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newDevice = hwOutputDesc->device();
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}
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// when changing from ring tone to in call mode, mute the ringing tone
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// immediately and delay the route change to avoid sending the ring tone
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// tail into the earpiece or headset.
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int delayMs = 0;
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if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) {
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// delay the device change command by twice the output latency to have some margin
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// and be sure that audio buffers not yet affected by the mute are out when
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// we actually apply the route change
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delayMs = hwOutputDesc->mLatency*2;
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setStreamMute(AudioSystem::RING, true, mPrimaryOutput);
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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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// unmute the ringing tone after a sufficient delay if it was muted before
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// setting output device above
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if (oldState == AudioSystem::MODE_RINGTONE) {
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setStreamMute(AudioSystem::RING, false, mPrimaryOutput, MUTE_TIME_MS);
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}
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for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
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handleIncallSonification(stream, true, true);
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}
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}
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// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
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if (state == AudioSystem::MODE_RINGTONE &&
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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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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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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 phone state
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uint32_t newDevice = getNewDevice(mPrimaryOutput, false);
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checkA2dpSuspend();
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checkOutputForAllStrategies();
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updateDeviceForStrategy();
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setOutputDevice(mPrimaryOutput, newDevice);
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if (forceVolumeReeval) {
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applyStreamVolumes(mPrimaryOutput, newDevice, 0, true);
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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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newDevice = getDeviceForInputSource(inputDesc->mInputSource);
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if (newDevice != inputDesc->mDevice) {
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ALOGV("setForceUse() 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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}
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AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
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{
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return mForceUse[usage];
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}
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void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
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{
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ALOGV("setSystemProperty() property %s, value %s", property, value);
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if (strcmp(property, "ro.camera.sound.forced") == 0) {
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if (atoi(value)) {
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ALOGV("ENFORCED_AUDIBLE cannot be muted");
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mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
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} else {
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ALOGV("ENFORCED_AUDIBLE can be muted");
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mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
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}
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}
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}
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audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
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uint32_t samplingRate,
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uint32_t format,
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uint32_t channels,
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AudioSystem::output_flags flags)
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{
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audio_io_handle_t output = 0;
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uint32_t latency = 0;
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routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
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uint32_t device = getDeviceForStrategy(strategy);
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ALOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
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#ifdef AUDIO_POLICY_TEST
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if (mCurOutput != 0) {
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ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
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mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
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if (mTestOutputs[mCurOutput] == 0) {
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ALOGV("getOutput() opening test output");
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AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
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outputDesc->mDevice = mTestDevice;
|
|
outputDesc->mSamplingRate = mTestSamplingRate;
|
|
outputDesc->mFormat = mTestFormat;
|
|
outputDesc->mChannels = mTestChannels;
|
|
outputDesc->mLatency = mTestLatencyMs;
|
|
outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
|
|
outputDesc->mRefCount[stream] = 0;
|
|
mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
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
|
|
if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) {
|
|
|
|
ALOGV("getOutput() opening direct output device %x", device);
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
|
|
outputDesc->mDevice = device;
|
|
outputDesc->mSamplingRate = samplingRate;
|
|
outputDesc->mFormat = format;
|
|
outputDesc->mChannels = channels;
|
|
outputDesc->mLatency = 0;
|
|
outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
|
|
outputDesc->mRefCount[stream] = 0;
|
|
outputDesc->mStopTime[stream] = 0;
|
|
output = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
|
|
// only accept an output with the requeted parameters
|
|
if (output == 0 ||
|
|
(samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
|
|
(format != 0 && format != outputDesc->mFormat) ||
|
|
(channels != 0 && channels != outputDesc->mChannels)) {
|
|
ALOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
|
|
samplingRate, format, channels);
|
|
if (output != 0) {
|
|
mpClientInterface->closeOutput(output);
|
|
}
|
|
delete outputDesc;
|
|
return 0;
|
|
}
|
|
addOutput(output, outputDesc);
|
|
return output;
|
|
}
|
|
|
|
if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
|
|
channels != AudioSystem::CHANNEL_OUT_STEREO) {
|
|
return 0;
|
|
}
|
|
// open a non direct output
|
|
|
|
// get which output is suitable for the specified stream. The actual routing change will happen
|
|
// when startOutput() will be called
|
|
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device);
|
|
// TODO: current implementation assumes that at most one output corresponds to a device.
|
|
// this will change when supporting low power, low latency or tunneled output streams
|
|
ALOG_ASSERT(outputs.size() < 2, "getOutput(): getOutputsForDevice() "
|
|
"returned %d outputs for device %04x", outputs.size(), device);
|
|
|
|
output = outputs[0];
|
|
|
|
ALOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
|
|
stream, samplingRate, format, channels, flags);
|
|
|
|
return output;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
|
|
AudioSystem::stream_type stream,
|
|
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);
|
|
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
|
|
|
|
// incremenent usage count for this stream on the requested output:
|
|
// NOTE that the usage count is the same for duplicated output and hardware output which is
|
|
// necassary for a correct control of hardware output routing by startOutput() and stopOutput()
|
|
outputDesc->changeRefCount(stream, 1);
|
|
|
|
uint32_t prevDevice = outputDesc->device();
|
|
setOutputDevice(output, getNewDevice(output));
|
|
uint32_t newDevice = outputDesc->device();
|
|
|
|
// 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((audio_devices_t)newDevice),
|
|
output,
|
|
newDevice);
|
|
|
|
// FIXME: need a delay to make sure that audio path switches to speaker before sound
|
|
// starts. Should be platform specific?
|
|
if (stream == AudioSystem::ENFORCED_AUDIBLE &&
|
|
prevDevice != newDevice) {
|
|
usleep(outputDesc->mLatency*4*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);
|
|
routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
|
|
|
|
// 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()
|
|
outputDesc->mStopTime[stream] = systemTime();
|
|
|
|
setOutputDevice(output, getNewDevice(output), false, outputDesc->mLatency*2);
|
|
|
|
if (output != mPrimaryOutput) {
|
|
setOutputDevice(mPrimaryOutput, getNewDevice(mPrimaryOutput), true);
|
|
}
|
|
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->refCount() == 0) {
|
|
mpClientInterface->closeOutput(output);
|
|
delete mOutputs.valueAt(index);
|
|
mOutputs.removeItem(output);
|
|
mTestOutputs[testIndex] = 0;
|
|
}
|
|
return;
|
|
}
|
|
#endif //AUDIO_POLICY_TEST
|
|
|
|
if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
|
|
mpClientInterface->closeOutput(output);
|
|
delete mOutputs.valueAt(index);
|
|
mOutputs.removeItem(output);
|
|
}
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channels,
|
|
AudioSystem::audio_in_acoustics acoustics)
|
|
{
|
|
audio_io_handle_t input = 0;
|
|
uint32_t device = getDeviceForInputSource(inputSource);
|
|
|
|
ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
|
|
|
|
if (device == 0) {
|
|
return 0;
|
|
}
|
|
|
|
// adapt channel selection to input source
|
|
switch(inputSource) {
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
|
|
|
|
inputDesc->mInputSource = inputSource;
|
|
inputDesc->mDevice = device;
|
|
inputDesc->mSamplingRate = samplingRate;
|
|
inputDesc->mFormat = format;
|
|
inputDesc->mChannels = channels;
|
|
inputDesc->mAcoustics = acoustics;
|
|
inputDesc->mRefCount = 0;
|
|
input = mpClientInterface->openInput(&inputDesc->mDevice,
|
|
&inputDesc->mSamplingRate,
|
|
&inputDesc->mFormat,
|
|
&inputDesc->mChannels,
|
|
(audio_in_acoustics_t) inputDesc->mAcoustics);
|
|
|
|
// only accept input with the exact requested set of parameters
|
|
if (input == 0 ||
|
|
(samplingRate != inputDesc->mSamplingRate) ||
|
|
(format != inputDesc->mFormat) ||
|
|
(channels != inputDesc->mChannels)) {
|
|
ALOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d",
|
|
samplingRate, format, channels);
|
|
if (input != 0) {
|
|
mpClientInterface->closeInput(input);
|
|
}
|
|
delete inputDesc;
|
|
return 0;
|
|
}
|
|
mInputs.add(input, inputDesc);
|
|
return input;
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
|
|
{
|
|
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 (getActiveInput() != 0) {
|
|
ALOGW("startInput() input %d failed: other input already started", input);
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
|
|
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 {
|
|
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((audio_devices_t)mOutputs.valueAt(i)->device());
|
|
if (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 = (audio_devices_t)getDeviceForStrategy(getStrategy(stream), true);
|
|
}
|
|
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(effect_descriptor_t *desc)
|
|
{
|
|
ALOGV("getOutputForEffect()");
|
|
// apply simple rule where global effects are attached to the same output as MUSIC streams
|
|
return getOutput(AudioSystem::MUSIC);
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::registerEffect(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++) {
|
|
if (mOutputs.valueAt(i)->mRefCount[stream] != 0 ||
|
|
ns2ms(sysTime - mOutputs.valueAt(i)->mStopTime[stream]) < inPastMs) {
|
|
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, " Hardware 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, " 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);
|
|
write(fd, result.string(), result.size());
|
|
|
|
snprintf(buffer, SIZE, "\nOutputs dump:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
|
|
write(fd, buffer, strlen(buffer));
|
|
mOutputs.valueAt(i)->dump(fd);
|
|
}
|
|
|
|
snprintf(buffer, SIZE, "\nInputs dump:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
|
|
write(fd, buffer, strlen(buffer));
|
|
mInputs.valueAt(i)->dump(fd);
|
|
}
|
|
|
|
snprintf(buffer, SIZE, "\nStreams dump:\n");
|
|
write(fd, buffer, strlen(buffer));
|
|
snprintf(buffer, SIZE,
|
|
" Stream 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;
|
|
}
|
|
|
|
// ----------------------------------------------------------------------------
|
|
// AudioPolicyManagerBase
|
|
// ----------------------------------------------------------------------------
|
|
|
|
AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
|
|
:
|
|
#ifdef AUDIO_POLICY_TEST
|
|
Thread(false),
|
|
#endif //AUDIO_POLICY_TEST
|
|
mAvailableOutputDevices(0),
|
|
mPhoneState(AudioSystem::MODE_NORMAL),
|
|
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
|
|
mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
|
|
mA2dpSuspended(false)
|
|
{
|
|
mpClientInterface = clientInterface;
|
|
|
|
for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
|
|
mForceUse[i] = AudioSystem::FORCE_NONE;
|
|
}
|
|
|
|
initializeVolumeCurves();
|
|
|
|
mA2dpDeviceAddress = String8("");
|
|
mScoDeviceAddress = String8("");
|
|
|
|
// TODO read this from configuration file
|
|
sHasA2dp = true;
|
|
const output_profile_t **outProfile = sAvailableOutputs;
|
|
// open all output streams needed to access attached devices
|
|
while (*outProfile)
|
|
{
|
|
if ((*outProfile)->mSupportedDevices & sAttachedOutputDevices) {
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(*outProfile);
|
|
|
|
outputDesc->mDevice = (uint32_t)sDefaultOutputDevice & (*outProfile)->mSupportedDevices;
|
|
outputDesc->mSamplingRate = (*outProfile)->mSamplingRates[0];
|
|
outputDesc->mFormat = (*outProfile)->mFormats[0];
|
|
outputDesc->mChannels = (*outProfile)->mChannelMasks[0];
|
|
outputDesc->mFlags = (AudioSystem::output_flags)(*outProfile)->mFlags;
|
|
audio_io_handle_t output = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
if (output == 0) {
|
|
delete outputDesc;
|
|
} else {
|
|
mAvailableOutputDevices |= ((*outProfile)->mSupportedDevices & sAttachedOutputDevices);
|
|
if ((*outProfile)->mFlags & AUDIO_POLICY_OUTPUT_FLAG_PRIMARY) {
|
|
mPrimaryOutput = output;
|
|
}
|
|
addOutput(output, outputDesc);
|
|
setOutputDevice(output,
|
|
(uint32_t)(sDefaultOutputDevice & (*outProfile)->mSupportedDevices),
|
|
true);
|
|
}
|
|
}
|
|
outProfile++;
|
|
}
|
|
|
|
ALOGE_IF((sAttachedOutputDevices & ~mAvailableOutputDevices),
|
|
"Not output found for attached devices %08x",
|
|
(sAttachedOutputDevices & ~mAvailableOutputDevices));
|
|
|
|
ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output");
|
|
|
|
mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
|
|
|
|
updateDeviceForStrategy();
|
|
#ifdef AUDIO_POLICY_TEST
|
|
if (mPrimaryOutput != 0) {
|
|
AudioParameter outputCmd = AudioParameter();
|
|
outputCmd.addInt(String8("set_id"), 0);
|
|
mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString());
|
|
|
|
mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
|
|
mTestSamplingRate = 44100;
|
|
mTestFormat = AudioSystem::PCM_16_BIT;
|
|
mTestChannels = AudioSystem::CHANNEL_OUT_STEREO;
|
|
mTestLatencyMs = 0;
|
|
mCurOutput = 0;
|
|
mDirectOutput = false;
|
|
for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
|
|
mTestOutputs[i] = 0;
|
|
}
|
|
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
snprintf(buffer, SIZE, "AudioPolicyManagerTest");
|
|
run(buffer, ANDROID_PRIORITY_AUDIO);
|
|
}
|
|
#endif //AUDIO_POLICY_TEST
|
|
}
|
|
|
|
AudioPolicyManagerBase::~AudioPolicyManagerBase()
|
|
{
|
|
#ifdef AUDIO_POLICY_TEST
|
|
exit();
|
|
#endif //AUDIO_POLICY_TEST
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
mpClientInterface->closeOutput(mOutputs.keyAt(i));
|
|
delete mOutputs.valueAt(i);
|
|
}
|
|
mOutputs.clear();
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
mpClientInterface->closeInput(mInputs.keyAt(i));
|
|
delete mInputs.valueAt(i);
|
|
}
|
|
mInputs.clear();
|
|
}
|
|
|
|
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"));
|
|
|
|
mpClientInterface->closeOutput(mPrimaryOutput);
|
|
delete mOutputs.valueFor(mPrimaryOutput);
|
|
mOutputs.removeItem(mPrimaryOutput);
|
|
|
|
AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
|
|
outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
|
|
mPrimaryOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&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->mChannels);
|
|
} 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);
|
|
}
|
|
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::checkOutputForDevice(
|
|
AudioSystem::audio_devices device,
|
|
AudioSystem::device_connection_state state)
|
|
{
|
|
audio_io_handle_t output = 0;
|
|
AudioOutputDescriptor *outputDesc;
|
|
|
|
// TODO handle multiple outputs supporting overlapping sets of devices.
|
|
|
|
if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
|
|
// first check if one output already open can be routed to this device
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueAt(i);
|
|
if (outputDesc->mProfile && outputDesc->mProfile->mSupportedDevices & device) {
|
|
return mOutputs.keyAt(i);
|
|
}
|
|
}
|
|
// then look for one available output that can be routed to this device
|
|
const output_profile_t **outProfile = sAvailableOutputs;
|
|
while (*outProfile)
|
|
{
|
|
if ((*outProfile)->mSupportedDevices & device) {
|
|
break;
|
|
}
|
|
outProfile++;
|
|
}
|
|
if (*outProfile == NULL) {
|
|
ALOGW("No output available for device %04x", device);
|
|
return output;
|
|
}
|
|
|
|
ALOGV("opening output for device %08x", device);
|
|
outputDesc = new AudioOutputDescriptor(*outProfile);
|
|
outputDesc->mDevice = device;
|
|
output = mpClientInterface->openOutput(&outputDesc->mDevice,
|
|
&outputDesc->mSamplingRate,
|
|
&outputDesc->mFormat,
|
|
&outputDesc->mChannels,
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
|
|
if (output != 0) {
|
|
audio_io_handle_t duplicatedOutput = 0;
|
|
// add output descriptor
|
|
addOutput(output, outputDesc);
|
|
// set initial stream volume for device
|
|
applyStreamVolumes(output, device);
|
|
|
|
//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 = outputDesc->mSamplingRate;
|
|
dupOutputDesc->mFormat = outputDesc->mFormat;
|
|
dupOutputDesc->mChannels = outputDesc->mChannels;
|
|
dupOutputDesc->mLatency = outputDesc->mLatency;
|
|
addOutput(duplicatedOutput, dupOutputDesc);
|
|
applyStreamVolumes(duplicatedOutput, device);
|
|
} else {
|
|
ALOGW("getOutput() could not open duplicated output for %d and %d",
|
|
mPrimaryOutput, output);
|
|
mpClientInterface->closeOutput(output);
|
|
mOutputs.removeItem(output);
|
|
delete outputDesc;
|
|
return 0;
|
|
}
|
|
} else {
|
|
ALOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
|
|
delete outputDesc;
|
|
return 0;
|
|
}
|
|
} else {
|
|
// we assume that one given device is supported by zero or one output
|
|
// check if one opened output is not needed any more after disconnecting one device
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
outputDesc = mOutputs.valueAt(i);
|
|
if (outputDesc->mProfile &&
|
|
!(outputDesc->mProfile->mSupportedDevices & mAvailableOutputDevices)) {
|
|
output = mOutputs.keyAt(i);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
return output;
|
|
}
|
|
|
|
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 mOutputs.valueFor(output);
|
|
mOutputs.removeItem(output);
|
|
}
|
|
|
|
SortedVector<audio_io_handle_t> AudioPolicyManagerBase::getOutputsForDevice(uint32_t device)
|
|
{
|
|
SortedVector<audio_io_handle_t> outputs;
|
|
|
|
ALOGV("getOutputsForDevice() device %04x", device);
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
if ((device & mOutputs.valueAt(i)->supportedDevices()) == device) {
|
|
ALOGV("getOutputsForDevice() found output %d", mOutputs.keyAt(i));
|
|
outputs.add(mOutputs.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)
|
|
{
|
|
SortedVector<audio_io_handle_t> srcOutputs =
|
|
getOutputsForDevice(getDeviceForStrategy(strategy));
|
|
SortedVector<audio_io_handle_t> dstOutputs =
|
|
getOutputsForDevice(getDeviceForStrategy(strategy, false));
|
|
|
|
// TODO: current implementation assumes that at most one output corresponds to a device.
|
|
// this will change when supporting low power, low latency or tunneled output streams
|
|
ALOG_ASSERT(srcOutputs.size() < 2, "checkOutputForStrategy(): "
|
|
"more than one (%d) source output for strategy %d", srcOutputs.size(), strategy);
|
|
ALOG_ASSERT(dstOutputs.size() < 2, "checkOutputForStrategy(): "
|
|
"more than one (%d) destination output for strategy %d", dstOutputs.size(), strategy);
|
|
|
|
if (!vectorsEqual(srcOutputs,dstOutputs)) {
|
|
ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
|
|
strategy, srcOutputs[0], dstOutputs[0]);
|
|
// mute media strategy while moving tracks from one output to another
|
|
setStrategyMute(strategy, true, srcOutputs[0]);
|
|
setStrategyMute(strategy, false, srcOutputs[0], MUTE_TIME_MS);
|
|
|
|
// Move effects associated to this strategy from previous output to new output
|
|
for (size_t i = 0; i < mEffects.size(); i++) {
|
|
EffectDescriptor *desc = mEffects.valueAt(i);
|
|
if (desc->mSession != AudioSystem::SESSION_OUTPUT_STAGE &&
|
|
desc->mStrategy == strategy &&
|
|
desc->mIo == srcOutputs[0]) {
|
|
ALOGV("checkOutputForStrategy() moving effect %d to output %d",
|
|
mEffects.keyAt(i), dstOutputs[0]);
|
|
mpClientInterface->moveEffects(desc->mSession, srcOutputs[0], dstOutputs[0]);
|
|
desc->mIo = dstOutputs[0];
|
|
}
|
|
}
|
|
// 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) {
|
|
mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, dstOutputs[0]);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::checkOutputForAllStrategies()
|
|
{
|
|
checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE);
|
|
checkOutputForStrategy(STRATEGY_PHONE);
|
|
checkOutputForStrategy(STRATEGY_SONIFICATION);
|
|
checkOutputForStrategy(STRATEGY_MEDIA);
|
|
checkOutputForStrategy(STRATEGY_DTMF);
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getA2dpOutput()
|
|
{
|
|
if (!sHasA2dp) {
|
|
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 (!sHasA2dp) {
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
|
|
{
|
|
uint32_t device = 0;
|
|
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
// check the following by order of priority to request a routing change if necessary:
|
|
// 1: 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 media is active on the output:
|
|
// use device for strategy media
|
|
// 5: the strategy DTMF is active on the output:
|
|
// use device for strategy DTMF
|
|
if (outputDesc->isUsedByStrategy(STRATEGY_ENFORCED_AUDIBLE)) {
|
|
device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
|
|
} else if (isInCall() ||
|
|
outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
|
|
device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
|
|
} else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
|
|
device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
|
|
} else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
|
|
} else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
|
|
device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
|
|
}
|
|
|
|
ALOGV("getNewDevice() selected device %x", device);
|
|
return device;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
|
|
return (uint32_t)getStrategy(stream);
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
|
|
uint32_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 = 0;
|
|
} else {
|
|
AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
|
|
devices = getDeviceForStrategy(strategy, true);
|
|
}
|
|
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::NOTIFICATION:
|
|
case AudioSystem::ALARM:
|
|
return STRATEGY_SONIFICATION;
|
|
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;
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
|
|
{
|
|
uint32_t device = 0;
|
|
|
|
if (fromCache) {
|
|
ALOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]);
|
|
return mDeviceForStrategy[strategy];
|
|
}
|
|
|
|
switch (strategy) {
|
|
case STRATEGY_DTMF:
|
|
if (!isInCall()) {
|
|
// when off call, DTMF strategy follows the same rules as MEDIA strategy
|
|
device = getDeviceForStrategy(STRATEGY_MEDIA, false);
|
|
break;
|
|
}
|
|
// when in call, DTMF and PHONE strategies follow the same rules
|
|
// FALL THROUGH
|
|
|
|
case STRATEGY_PHONE:
|
|
// for phone strategy, we first consider the forced use and then the available devices by order
|
|
// of priority
|
|
switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
|
|
case AudioSystem::FORCE_BT_SCO:
|
|
if (!isInCall() || strategy != STRATEGY_DTMF) {
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO;
|
|
if (device) break;
|
|
// if SCO device is requested but no SCO device is available, fall back to default case
|
|
// FALL THROUGH
|
|
|
|
default: // FORCE_NONE
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
|
|
if (device) break;
|
|
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
|
|
if (sHasA2dp && !isInCall() && !mA2dpSuspended) {
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
|
|
if (device == 0) {
|
|
ALOGE("getDeviceForStrategy() earpiece device not found");
|
|
}
|
|
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 (sHasA2dp && !isInCall() && !mA2dpSuspended) {
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
|
|
if (device) break;
|
|
}
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
|
|
if (device) break;
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
|
|
if (device == 0) {
|
|
ALOGE("getDeviceForStrategy() speaker device not found");
|
|
}
|
|
break;
|
|
}
|
|
break;
|
|
|
|
case STRATEGY_SONIFICATION:
|
|
|
|
// If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
|
|
// handleIncallSonification().
|
|
if (isInCall()) {
|
|
device = getDeviceForStrategy(STRATEGY_PHONE, false);
|
|
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
|
|
|
|
device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
|
|
if (device == 0) {
|
|
ALOGE("getDeviceForStrategy() speaker device not found");
|
|
}
|
|
// The second device used for sonification is the same as the device used by media strategy
|
|
// FALL THROUGH
|
|
|
|
case STRATEGY_MEDIA: {
|
|
uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
|
|
}
|
|
if (sHasA2dp && (getA2dpOutput() != 0) && !mA2dpSuspended) {
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
|
|
}
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
|
|
}
|
|
if (device2 == 0) {
|
|
device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
|
|
}
|
|
|
|
// device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
|
|
// STRATEGY_ENFORCED_AUDIBLE, 0 otherwise
|
|
device |= device2;
|
|
if (device == 0) {
|
|
ALOGE("getDeviceForStrategy() speaker device not found");
|
|
}
|
|
} break;
|
|
|
|
default:
|
|
ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
|
|
break;
|
|
}
|
|
|
|
ALOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
|
|
return device;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::updateDeviceForStrategy()
|
|
{
|
|
for (int i = 0; i < NUM_STRATEGIES; i++) {
|
|
mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false);
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs)
|
|
{
|
|
ALOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs);
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
|
|
|
|
if (outputDesc->isDuplicated()) {
|
|
setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
|
|
setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
|
|
return;
|
|
}
|
|
// filter devices according to output selected
|
|
device &= outputDesc->mProfile->mSupportedDevices;
|
|
|
|
uint32_t prevDevice = (uint32_t)outputDesc->device();
|
|
// Do not change the routing if:
|
|
// - the requestede device is 0
|
|
// - the requested device is the same as current device and force is not specified.
|
|
// Doing this check here allows the caller to call setOutputDevice() without conditions
|
|
if ((device == 0 || device == prevDevice) && !force) {
|
|
ALOGV("setOutputDevice() setting same device %x or null device for output %d", device, output);
|
|
return;
|
|
}
|
|
|
|
outputDesc->mDevice = device;
|
|
// mute media streams if both speaker and headset are selected
|
|
if (output == mPrimaryOutput && AudioSystem::popCount(device) == 2) {
|
|
setStrategyMute(STRATEGY_MEDIA, true, output);
|
|
// wait for the PCM output buffers to empty before proceeding with the rest of the command
|
|
// FIXME: increased delay due to larger buffers used for low power audio mode.
|
|
// remove when low power audio is controlled by policy manager.
|
|
usleep(outputDesc->mLatency*8*1000);
|
|
}
|
|
|
|
// do the routing
|
|
AudioParameter param = AudioParameter();
|
|
param.addInt(String8(AudioParameter::keyRouting), (int)device);
|
|
mpClientInterface->setParameters(mPrimaryOutput, param.toString(), delayMs);
|
|
// update stream volumes according to new device
|
|
applyStreamVolumes(output, device, delayMs);
|
|
|
|
// if changing from a combined headset + speaker route, unmute media streams
|
|
if (output == mPrimaryOutput && AudioSystem::popCount(prevDevice) == 2) {
|
|
setStrategyMute(STRATEGY_MEDIA, false, output, delayMs);
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
|
|
{
|
|
uint32_t device;
|
|
|
|
switch(inputSource) {
|
|
case AUDIO_SOURCE_DEFAULT:
|
|
case AUDIO_SOURCE_MIC:
|
|
case AUDIO_SOURCE_VOICE_RECOGNITION:
|
|
case AUDIO_SOURCE_VOICE_COMMUNICATION:
|
|
if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
|
|
mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
|
|
device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
|
|
} else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
|
|
device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
|
|
} else {
|
|
device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
|
|
}
|
|
break;
|
|
case AUDIO_SOURCE_CAMCORDER:
|
|
if (hasBackMicrophone()) {
|
|
device = AudioSystem::DEVICE_IN_BACK_MIC;
|
|
} else {
|
|
device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
|
|
}
|
|
break;
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
device = AudioSystem::DEVICE_IN_VOICE_CALL;
|
|
break;
|
|
default:
|
|
ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
|
|
device = 0;
|
|
break;
|
|
}
|
|
ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
|
|
return device;
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerBase::getActiveInput()
|
|
{
|
|
for (size_t i = 0; i < mInputs.size(); i++) {
|
|
if (mInputs.valueAt(i)->mRefCount > 0) {
|
|
return mInputs.keyAt(i);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
audio_devices_t AudioPolicyManagerBase::getDeviceForVolume(audio_devices_t device)
|
|
{
|
|
if (device == 0) {
|
|
// 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(uint32_t device)
|
|
{
|
|
switch(getDeviceForVolume((audio_devices_t)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:
|
|
default:
|
|
return DEVICE_CATEGORY_SPEAKER;
|
|
}
|
|
}
|
|
|
|
float AudioPolicyManagerBase::volIndexToAmpl(uint32_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 )
|
|
|
|
ALOGV("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}
|
|
};
|
|
|
|
|
|
const AudioPolicyManagerBase::VolumeCurvePoint
|
|
*AudioPolicyManagerBase::sVolumeProfiles[AudioPolicyManagerBase::NUM_STRATEGIES]
|
|
[AudioPolicyManagerBase::DEVICE_CATEGORY_CNT] = {
|
|
{ // STRATEGY_MEDIA
|
|
sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // STRATEGY_PHONE
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // STRATEGY_SONIFICATION
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // STRATEGY_DTMF
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
{ // STRATEGY_ENFORCED_AUDIBLE
|
|
sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
|
|
sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
|
|
sDefaultVolumeCurve // DEVICE_CATEGORY_EARPIECE
|
|
},
|
|
};
|
|
|
|
void AudioPolicyManagerBase::initializeVolumeCurves()
|
|
{
|
|
for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
|
|
mStreams[i].mVolumeCurve[j] =
|
|
sVolumeProfiles[getStrategy((AudioSystem::stream_type)i)][j];
|
|
}
|
|
}
|
|
}
|
|
|
|
float AudioPolicyManagerBase::computeVolume(int stream,
|
|
int index,
|
|
audio_io_handle_t output,
|
|
uint32_t device)
|
|
{
|
|
float volume = 1.0;
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
StreamDescriptor &streamDesc = mStreams[stream];
|
|
|
|
if (device == 0) {
|
|
device = outputDesc->device();
|
|
}
|
|
|
|
// if volume is not 0 (not muted), force media volume to max on digital output
|
|
if (stream == AudioSystem::MUSIC &&
|
|
index != mStreams[stream].mIndexMin &&
|
|
(device == AudioSystem::DEVICE_OUT_AUX_DIGITAL ||
|
|
device == AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET)) {
|
|
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.
|
|
if ((device &
|
|
(AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
|
|
AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
|
|
AudioSystem::DEVICE_OUT_WIRED_HEADSET |
|
|
AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
|
|
((getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) ||
|
|
(stream == AudioSystem::SYSTEM)) &&
|
|
streamDesc.mCanBeMuted) {
|
|
volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
|
|
// when the phone is ringing we must consider that music could have been paused just before
|
|
// by the music application and behave as if music was active if the last music track was
|
|
// just stopped
|
|
if (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) {
|
|
float musicVol = computeVolume(AudioSystem::MUSIC,
|
|
mStreams[AudioSystem::MUSIC].getVolumeIndex((audio_devices_t)device),
|
|
output,
|
|
(uint32_t)device);
|
|
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,
|
|
uint32_t device,
|
|
int delayMs,
|
|
bool force)
|
|
{
|
|
|
|
// do not change actual stream volume if the stream is muted
|
|
if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
|
|
ALOGV("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;
|
|
ALOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
|
|
if (stream == AudioSystem::VOICE_CALL ||
|
|
stream == AudioSystem::DTMF ||
|
|
stream == AudioSystem::BLUETOOTH_SCO) {
|
|
// offset value to reflect actual hardware volume that never reaches 0
|
|
// 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
|
|
volume = 0.01 + 0.99 * volume;
|
|
// 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,
|
|
uint32_t device,
|
|
int delayMs,
|
|
bool force)
|
|
{
|
|
ALOGV("applyStreamVolumes() for output %d and device %x", output, device);
|
|
|
|
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
|
|
checkAndSetVolume(stream,
|
|
mStreams[stream].getVolumeIndex((audio_devices_t)device),
|
|
output,
|
|
device,
|
|
delayMs,
|
|
force);
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs)
|
|
{
|
|
ALOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
|
|
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
|
|
if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
|
|
setStreamMute(stream, on, output, delayMs);
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
|
|
{
|
|
StreamDescriptor &streamDesc = mStreams[stream];
|
|
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
|
|
uint32_t device = outputDesc->device();
|
|
|
|
ALOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]);
|
|
|
|
if (on) {
|
|
if (outputDesc->mMuteCount[stream] == 0) {
|
|
if (streamDesc.mCanBeMuted) {
|
|
checkAndSetVolume(stream, 0, output, device, delayMs);
|
|
}
|
|
}
|
|
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
|
|
outputDesc->mMuteCount[stream]++;
|
|
} else {
|
|
if (outputDesc->mMuteCount[stream] == 0) {
|
|
ALOGW("setStreamMute() unmuting non muted stream!");
|
|
return;
|
|
}
|
|
if (--outputDesc->mMuteCount[stream] == 0) {
|
|
checkAndSetVolume(stream,
|
|
streamDesc.getVolumeIndex((audio_devices_t)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
|
|
|
|
if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
|
|
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)) {
|
|
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));
|
|
}
|
|
|
|
bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
|
|
uint32_t samplingRate,
|
|
uint32_t format,
|
|
uint32_t channels,
|
|
AudioSystem::output_flags flags,
|
|
uint32_t device)
|
|
{
|
|
return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
|
|
(format != 0 && !AudioSystem::isLinearPCM(format)));
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
|
|
{
|
|
return MAX_EFFECTS_CPU_LOAD;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
|
|
{
|
|
return MAX_EFFECTS_MEMORY;
|
|
}
|
|
|
|
// --- AudioOutputDescriptor class implementation
|
|
|
|
AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor(
|
|
const output_profile_t *profile)
|
|
: mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
|
|
mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0), mProfile(profile)
|
|
{
|
|
// 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;
|
|
}
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device()
|
|
{
|
|
uint32_t device = 0;
|
|
if (isDuplicated()) {
|
|
device = mOutput1->mDevice | mOutput2->mDevice;
|
|
} else {
|
|
device = mDevice;
|
|
}
|
|
return device;
|
|
}
|
|
|
|
void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
|
|
{
|
|
// forward usage count change to attached outputs
|
|
if (isDuplicated()) {
|
|
mOutput1->changeRefCount(stream, delta);
|
|
mOutput2->changeRefCount(stream, delta);
|
|
}
|
|
if ((delta + (int)mRefCount[stream]) < 0) {
|
|
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]);
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
|
|
{
|
|
uint32_t refcount = 0;
|
|
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
refcount += mRefCount[i];
|
|
}
|
|
return refcount;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy)
|
|
{
|
|
uint32_t refCount = 0;
|
|
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
|
|
refCount += mRefCount[i];
|
|
}
|
|
}
|
|
return refCount;
|
|
}
|
|
|
|
uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::supportedDevices()
|
|
{
|
|
if (isDuplicated()) {
|
|
return (mOutput1->supportedDevices() | mOutput2->supportedDevices());
|
|
} else {
|
|
return mProfile->mSupportedDevices ;
|
|
}
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Format: %d\n", mFormat);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Devices %08x\n", device());
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
|
|
result.append(buffer);
|
|
for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
|
|
snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
|
|
result.append(buffer);
|
|
}
|
|
write(fd, result.string(), result.size());
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// --- AudioInputDescriptor class implementation
|
|
|
|
AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
|
|
: mSamplingRate(0), mFormat(0), mChannels(0),
|
|
mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0),
|
|
mInputSource(0)
|
|
{
|
|
}
|
|
|
|
status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd)
|
|
{
|
|
const size_t SIZE = 256;
|
|
char buffer[SIZE];
|
|
String8 result;
|
|
|
|
snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Format: %d\n", mFormat);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
|
|
result.append(buffer);
|
|
snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
|
|
result.append(buffer);
|
|
write(fd, result.string(), result.size());
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
// --- StreamDescriptor class implementation
|
|
|
|
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;
|
|
}
|
|
|
|
|
|
|
|
}; // namespace android
|