2246 lines
101 KiB
C++
2246 lines
101 KiB
C++
/*
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* Copyright (c) 2013-2017 The Linux Foundation. All rights reserved.
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* Not a contribution.
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*
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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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* This file was modified by Dolby Laboratories, Inc. The portions of the
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* code that are surrounded by "DOLBY..." are copyrighted and
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* licensed separately, as follows:
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*
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* (C) 2015 Dolby Laboratories, Inc.
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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 "AudioPolicyManagerCustom"
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//#define LOG_NDEBUG 0
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//#define VERY_VERBOSE_LOGGING
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#ifdef VERY_VERBOSE_LOGGING
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#define ALOGVV ALOGV
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#else
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#define ALOGVV(a...) do { } while(0)
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#endif
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// A device mask for all audio output devices that are considered "remote" when evaluating
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// active output devices in isStreamActiveRemotely()
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#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
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// A device mask for all audio input and output devices where matching inputs/outputs on device
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// type alone is not enough: the address must match too
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#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \
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AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
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#define SAMPLE_RATE_8000 8000
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#include <inttypes.h>
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#include <math.h>
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#include <cutils/properties.h>
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#include <utils/Log.h>
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#include <hardware/audio.h>
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#include <hardware/audio_effect.h>
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#include <media/AudioParameter.h>
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#include <soundtrigger/SoundTrigger.h>
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#include "AudioPolicyManager.h"
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#include <policy.h>
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#ifdef DOLBY_ENABLE
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#include "DolbyAudioPolicy_impl.h"
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#endif // DOLBY_END
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#ifndef AUDIO_OUTPUT_FLAG_VOIP_RX
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#define AUDIO_OUTPUT_FLAG_VOIP_RX 0x800
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#endif
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namespace android {
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/*audio policy: workaround for truncated touch sounds*/
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//FIXME: workaround for truncated touch sounds
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// to be removed when the problem is handled by system UI
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#define TOUCH_SOUND_FIXED_DELAY_MS 100
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#ifdef VOICE_CONCURRENCY
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audio_output_flags_t AudioPolicyManagerCustom::getFallBackPath()
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{
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audio_output_flags_t flag = AUDIO_OUTPUT_FLAG_FAST;
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char propValue[PROPERTY_VALUE_MAX];
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if (property_get("voice.conc.fallbackpath", propValue, NULL)) {
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if (!strncmp(propValue, "deep-buffer", 11)) {
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flag = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
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}
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else if (!strncmp(propValue, "fast", 4)) {
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flag = AUDIO_OUTPUT_FLAG_FAST;
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}
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else {
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ALOGD("voice_conc:not a recognised path(%s) in prop voice.conc.fallbackpath",
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propValue);
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}
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}
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else {
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ALOGD("voice_conc:prop voice.conc.fallbackpath not set");
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}
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ALOGD("voice_conc:picked up flag(0x%x) from prop voice.conc.fallbackpath",
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flag);
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return flag;
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}
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#endif /*VOICE_CONCURRENCY*/
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void AudioPolicyManagerCustom::moveGlobalEffect()
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{
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audio_io_handle_t dstOutput = getOutputForEffect();
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if (hasPrimaryOutput() && dstOutput != mPrimaryOutput->mIoHandle) {
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#ifdef DOLBY_ENABLE
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status_t status =
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mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
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mPrimaryOutput->mIoHandle,
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dstOutput);
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if (status == NO_ERROR) {
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for (size_t i = 0; i < mEffects.size(); i++) {
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sp<EffectDescriptor> desc = mEffects.valueAt(i);
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if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) {
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// update the mIo member of EffectDescriptor
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// for the global effect
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ALOGV("%s updating mIo", __FUNCTION__);
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desc->mIo = dstOutput;
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}
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}
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} else {
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ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__,
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mPrimaryOutput->mIoHandle, dstOutput);
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}
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#else // DOLBY_END
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mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX,
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mPrimaryOutput->mIoHandle, dstOutput);
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#endif
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}
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}
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// ----------------------------------------------------------------------------
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// AudioPolicyInterface implementation
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// ----------------------------------------------------------------------------
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extern "C" AudioPolicyInterface* createAudioPolicyManager(
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AudioPolicyClientInterface *clientInterface)
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{
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return new AudioPolicyManagerCustom(clientInterface);
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}
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extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
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{
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delete interface;
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}
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status_t AudioPolicyManagerCustom::setDeviceConnectionStateInt(audio_devices_t device,
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audio_policy_dev_state_t state,
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const char *device_address,
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const char *device_name)
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{
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ALOGD("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
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device, state, device_address, device_name);
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// connect/disconnect only 1 device at a time
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if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
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sp<DeviceDescriptor> devDesc =
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mHwModules.getDeviceDescriptor(device, device_address, device_name);
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// handle output devices
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if (audio_is_output_device(device)) {
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SortedVector <audio_io_handle_t> outputs;
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ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
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// save a copy of the opened output descriptors before any output is opened or closed
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// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
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mPreviousOutputs = mOutputs;
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switch (state)
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{
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// handle output device connection
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case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
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if (index >= 0) {
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = false;
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} else {
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mHdmiAudioEvent = true;
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}
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}
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#endif
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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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index = mAvailableOutputDevices.add(devDesc);
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = false;
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} else {
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mHdmiAudioEvent = true;
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}
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if (mHdmiAudioDisabled || !mHdmiAudioEvent) {
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mAvailableOutputDevices.remove(devDesc);
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ALOGW("HDMI sink not connected, do not route audio to HDMI out");
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return INVALID_OPERATION;
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}
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}
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#endif
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if (index >= 0) {
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sp<HwModule> module = mHwModules.getModuleForDevice(device);
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if (module == 0) {
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ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
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device);
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mAvailableOutputDevices.remove(devDesc);
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return INVALID_OPERATION;
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}
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mAvailableOutputDevices[index]->attach(module);
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} else {
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return NO_MEMORY;
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}
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// Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
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// parameters on newly connected devices (instead of opening the outputs...)
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broadcastDeviceConnectionState(device, state, devDesc->mAddress);
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if (checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress) != NO_ERROR) {
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mAvailableOutputDevices.remove(devDesc);
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broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
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devDesc->mAddress);
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return INVALID_OPERATION;
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}
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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// outputs should never be empty here
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ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
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"checkOutputsForDevice() returned no outputs but status OK");
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ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
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outputs.size());
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} break;
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// handle output device disconnection
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case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
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if (index < 0) {
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = true;
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} else {
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mHdmiAudioEvent = false;
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}
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}
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#endif
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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 output device %x", device);
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// Send Disconnect to HALs
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broadcastDeviceConnectionState(device, state, devDesc->mAddress);
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// remove device from available output devices
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mAvailableOutputDevices.remove(devDesc);
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#ifdef AUDIO_EXTN_HDMI_SPK_ENABLED
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if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) {
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if (!strncmp(device_address, "hdmi_spkr", 9)) {
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mHdmiAudioDisabled = true;
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} else {
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mHdmiAudioEvent = false;
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}
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}
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#endif
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checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress);
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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} break;
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default:
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ALOGE("setDeviceConnectionState() invalid state: %x", state);
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return BAD_VALUE;
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}
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// checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
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// output is suspended before any tracks are moved to it
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checkA2dpSuspend();
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checkOutputForAllStrategies();
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// outputs must be closed after checkOutputForAllStrategies() is executed
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if (!outputs.isEmpty()) {
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for (size_t i = 0; i < outputs.size(); i++) {
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sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]);
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// close unused outputs after device disconnection or direct outputs that have been
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// opened by checkOutputsForDevice() to query dynamic parameters
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if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
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(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
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(desc->mDirectOpenCount == 0))) {
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closeOutput(outputs[i]);
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}
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}
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// check again after closing A2DP output to reset mA2dpSuspended if needed
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checkA2dpSuspend();
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}
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#ifdef FM_POWER_OPT
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// handle FM device connection state to trigger FM AFE loopback
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if (device == AUDIO_DEVICE_OUT_FM && hasPrimaryOutput()) {
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audio_devices_t newDevice = AUDIO_DEVICE_NONE;
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if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
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mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, 1);
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newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false)|AUDIO_DEVICE_OUT_FM);
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mFMIsActive = true;
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mPrimaryOutput->mDevice = newDevice & ~AUDIO_DEVICE_OUT_FM;
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} else {
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newDevice = (audio_devices_t)(getNewOutputDevice(mPrimaryOutput, false));
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mFMIsActive = false;
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mPrimaryOutput->changeRefCount(AUDIO_STREAM_MUSIC, -1);
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}
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AudioParameter param = AudioParameter();
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param.addInt(String8("handle_fm"), (int)newDevice);
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mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString());
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}
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#endif /* FM_POWER_OPT end */
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updateDevicesAndOutputs();
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#ifdef DOLBY_ENABLE
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// Before closing the opened outputs, update endpoint property with device capabilities
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audio_devices_t audioOutputDevice = getDeviceForStrategy(getStrategy(AUDIO_STREAM_MUSIC), true);
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mDolbyAudioPolicy.setEndpointSystemProperty(audioOutputDevice, mHwModules);
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#endif // DOLBY_END
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if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
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audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
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updateCallRouting(newDevice);
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}
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for (size_t i = 0; i < mOutputs.size(); i++) {
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sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
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if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
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audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
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// do not force device change on duplicated output because if device is 0, it will
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// also force a device 0 for the two outputs it is duplicated to which may override
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// a valid device selection on those outputs.
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bool force = !desc->isDuplicated()
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&& (!device_distinguishes_on_address(device)
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// always force when disconnecting (a non-duplicated device)
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|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
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setOutputDevice(desc, newDevice, force, 0);
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}
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}
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if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
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cleanUpForDevice(devDesc);
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}
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mpClientInterface->onAudioPortListUpdate();
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return NO_ERROR;
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} // end if is output device
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// handle input devices
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if (audio_is_input_device(device)) {
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SortedVector <audio_io_handle_t> inputs;
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ssize_t index = mAvailableInputDevices.indexOf(devDesc);
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switch (state)
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{
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// handle input device connection
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case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
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if (index >= 0) {
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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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sp<HwModule> module = mHwModules.getModuleForDevice(device);
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if (module == NULL) {
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ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
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device);
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return INVALID_OPERATION;
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}
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// Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
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// parameters on newly connected devices (instead of opening the inputs...)
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broadcastDeviceConnectionState(device, state, devDesc->mAddress);
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if (checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress) != NO_ERROR) {
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broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
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devDesc->mAddress);
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return INVALID_OPERATION;
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}
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index = mAvailableInputDevices.add(devDesc);
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if (index >= 0) {
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mAvailableInputDevices[index]->attach(module);
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} else {
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return NO_MEMORY;
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}
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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} break;
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// handle input device disconnection
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case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
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if (index < 0) {
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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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ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
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// Set Disconnect to HALs
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broadcastDeviceConnectionState(device, state, devDesc->mAddress);
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checkInputsForDevice(devDesc, state, inputs, devDesc->mAddress);
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mAvailableInputDevices.remove(devDesc);
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// Propagate device availability to Engine
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mEngine->setDeviceConnectionState(devDesc, state);
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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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closeAllInputs();
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/*audio policy: fix call volume over USB*/
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// As the input device list can impact the output device selection, update
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// getDeviceForStrategy() cache
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updateDevicesAndOutputs();
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if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
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audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
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updateCallRouting(newDevice);
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}
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if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
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cleanUpForDevice(devDesc);
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}
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mpClientInterface->onAudioPortListUpdate();
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return NO_ERROR;
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} // end if is input device
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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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bool AudioPolicyManagerCustom::isInvalidationOfMusicStreamNeeded(routing_strategy strategy)
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{
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if (strategy == STRATEGY_MEDIA) {
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for (size_t i = 0; i < mOutputs.size(); i++) {
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sp<SwAudioOutputDescriptor> newOutputDesc = mOutputs.valueAt(i);
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if (newOutputDesc->mFormat == AUDIO_FORMAT_DSD)
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return false;
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}
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}
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return true;
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}
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void AudioPolicyManagerCustom::checkOutputForStrategy(routing_strategy strategy)
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{
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audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
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audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
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SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mOutputs);
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SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
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// also take into account external policy-related changes: add all outputs which are
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// associated with policies in the "before" and "after" output vectors
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ALOGV("checkOutputForStrategy(): policy related outputs");
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for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
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const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
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if (desc != 0 && desc->mPolicyMix != NULL) {
|
|
srcOutputs.add(desc->mIoHandle);
|
|
ALOGV(" previous outputs: adding %d", desc->mIoHandle);
|
|
}
|
|
}
|
|
for (size_t i = 0 ; i < mOutputs.size() ; i++) {
|
|
const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if (desc != 0 && desc->mPolicyMix != NULL) {
|
|
dstOutputs.add(desc->mIoHandle);
|
|
ALOGV(" new outputs: adding %d", desc->mIoHandle);
|
|
}
|
|
}
|
|
|
|
if (!vectorsEqual(srcOutputs,dstOutputs) && isInvalidationOfMusicStreamNeeded(strategy)) {
|
|
AudioPolicyManager::checkOutputForStrategy(strategy);
|
|
}
|
|
}
|
|
|
|
// This function checks for the parameters which can be offloaded.
|
|
// This can be enhanced depending on the capability of the DSP and policy
|
|
// of the system.
|
|
bool AudioPolicyManagerCustom::isOffloadSupported(const audio_offload_info_t& offloadInfo)
|
|
{
|
|
ALOGV("isOffloadSupported: SR=%u, CM=0x%x, Format=0x%x, StreamType=%d,"
|
|
" BitRate=%u, duration=%" PRId64 " us, has_video=%d",
|
|
offloadInfo.sample_rate, offloadInfo.channel_mask,
|
|
offloadInfo.format,
|
|
offloadInfo.stream_type, offloadInfo.bit_rate, offloadInfo.duration_us,
|
|
offloadInfo.has_video);
|
|
|
|
if (mMasterMono) {
|
|
return false; // no offloading if mono is set.
|
|
}
|
|
|
|
#ifdef VOICE_CONCURRENCY
|
|
char concpropValue[PROPERTY_VALUE_MAX];
|
|
if (property_get("voice.playback.conc.disabled", concpropValue, NULL)) {
|
|
bool propenabled = atoi(concpropValue) || !strncmp("true", concpropValue, 4);
|
|
if (propenabled) {
|
|
if (isInCall())
|
|
{
|
|
ALOGD("\n copl: blocking compress offload on call mode\n");
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
|
|
isInCall() && (offloadInfo.format == AUDIO_FORMAT_DSD)) {
|
|
ALOGD("blocking DSD compress offload on call mode");
|
|
return false;
|
|
}
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
char recConcPropValue[PROPERTY_VALUE_MAX];
|
|
bool prop_rec_play_enabled = false;
|
|
|
|
if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
|
|
prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
|
|
}
|
|
|
|
if ((prop_rec_play_enabled) &&
|
|
((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCountOnDevices() > 0))) {
|
|
ALOGD("copl: blocking compress offload for record concurrency");
|
|
return false;
|
|
}
|
|
#endif
|
|
// Check if stream type is music, then only allow offload as of now.
|
|
if (offloadInfo.stream_type != AUDIO_STREAM_MUSIC)
|
|
{
|
|
ALOGV("isOffloadSupported: stream_type != MUSIC, returning false");
|
|
return false;
|
|
}
|
|
|
|
// Check if offload has been disabled
|
|
bool offloadDisabled = property_get_bool("audio.offload.disable", false);
|
|
if (offloadDisabled) {
|
|
ALOGI("offload disabled by audio.offload.disable=%d", offloadDisabled);
|
|
return false;
|
|
}
|
|
|
|
//check if it's multi-channel AAC (includes sub formats) and FLAC format
|
|
if ((popcount(offloadInfo.channel_mask) > 2) &&
|
|
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))) {
|
|
ALOGD("offload disabled for multi-channel AAC,FLAC and VORBIS format");
|
|
return false;
|
|
}
|
|
|
|
#ifdef AUDIO_EXTN_FORMATS_ENABLED
|
|
//check if it's multi-channel FLAC/ALAC/WMA format with sample rate > 48k
|
|
if ((popcount(offloadInfo.channel_mask) > 2) &&
|
|
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
|
|
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) && (offloadInfo.sample_rate > 48000)) ||
|
|
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) && (offloadInfo.sample_rate > 48000)) ||
|
|
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.sample_rate > 48000)) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))) {
|
|
ALOGD("offload disabled for multi-channel FLAC/ALAC/WMA/AAC_ADTS clips with sample rate > 48kHz");
|
|
return false;
|
|
}
|
|
|
|
// check against wma std bit rate restriction
|
|
if ((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) {
|
|
int32_t sr_id = -1;
|
|
uint32_t min_bitrate, max_bitrate;
|
|
for (int i = 0; i < WMA_STD_NUM_FREQ; i++) {
|
|
if (offloadInfo.sample_rate == wmaStdSampleRateTbl[i]) {
|
|
sr_id = i;
|
|
break;
|
|
}
|
|
}
|
|
if ((sr_id < 0) || (popcount(offloadInfo.channel_mask) > 2)
|
|
|| (popcount(offloadInfo.channel_mask) <= 0)) {
|
|
ALOGE("invalid sample rate or channel count");
|
|
return false;
|
|
}
|
|
|
|
min_bitrate = wmaStdMinAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
|
|
max_bitrate = wmaStdMaxAvgByteRateTbl[sr_id][popcount(offloadInfo.channel_mask) - 1];
|
|
if ((offloadInfo.bit_rate > max_bitrate) || (offloadInfo.bit_rate < min_bitrate)) {
|
|
ALOGD("offload disabled for WMA clips with unsupported bit rate");
|
|
ALOGD("bit_rate %d, max_bitrate %d, min_bitrate %d", offloadInfo.bit_rate, max_bitrate, min_bitrate);
|
|
return false;
|
|
}
|
|
}
|
|
|
|
// Safely choose the min bitrate as threshold and leave the restriction to NT decoder as we can't distinguish wma pro and wma lossless here.
|
|
if ((((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_PRO)) ||
|
|
(((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) && (offloadInfo.bit_rate > MAX_BITRATE_WMA_LOSSLESS))) {
|
|
ALOGD("offload disabled for WMA_PRO/WMA_LOSSLESS clips with bit rate over maximum supported value");
|
|
return false;
|
|
}
|
|
#endif
|
|
//TODO: enable audio offloading with video when ready
|
|
const bool allowOffloadWithVideo =
|
|
property_get_bool("audio.offload.video", false /* default_value */);
|
|
if (offloadInfo.has_video && !allowOffloadWithVideo) {
|
|
ALOGV("isOffloadSupported: has_video == true, returning false");
|
|
return false;
|
|
}
|
|
|
|
const bool allowOffloadStreamingWithVideo = property_get_bool("av.streaming.offload.enable",
|
|
false /*default value*/);
|
|
if (offloadInfo.has_video && offloadInfo.is_streaming && !allowOffloadStreamingWithVideo) {
|
|
ALOGW("offload disabled by av.streaming.offload.enable %d",allowOffloadStreamingWithVideo);
|
|
return false;
|
|
}
|
|
|
|
//If duration is less than minimum value defined in property, return false
|
|
char propValue[PROPERTY_VALUE_MAX];
|
|
if (property_get("audio.offload.min.duration.secs", propValue, NULL)) {
|
|
if (offloadInfo.duration_us < (atoi(propValue) * 1000000 )) {
|
|
ALOGV("Offload denied by duration < audio.offload.min.duration.secs(=%s)", propValue);
|
|
return false;
|
|
}
|
|
} else if (offloadInfo.duration_us < OFFLOAD_DEFAULT_MIN_DURATION_SECS * 1000000) {
|
|
ALOGV("Offload denied by duration < default min(=%u)", OFFLOAD_DEFAULT_MIN_DURATION_SECS);
|
|
//duration checks only valid for MP3/AAC/ formats,
|
|
//do not check duration for other audio formats, e.g. dolby AAC/AC3 and amrwb+ formats
|
|
if ((offloadInfo.format == AUDIO_FORMAT_MP3) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_FLAC) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_VORBIS))
|
|
return false;
|
|
|
|
#ifdef AUDIO_EXTN_FORMATS_ENABLED
|
|
if (((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_WMA_PRO) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_ALAC) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_APE) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_DSD) ||
|
|
((offloadInfo.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS))
|
|
return false;
|
|
#endif
|
|
}
|
|
|
|
// Do not allow offloading if one non offloadable effect is enabled. This prevents from
|
|
// creating an offloaded track and tearing it down immediately after start when audioflinger
|
|
// detects there is an active non offloadable effect.
|
|
// FIXME: We should check the audio session here but we do not have it in this context.
|
|
// This may prevent offloading in rare situations where effects are left active by apps
|
|
// in the background.
|
|
if (mEffects.isNonOffloadableEffectEnabled()) {
|
|
return false;
|
|
}
|
|
|
|
// See if there is a profile to support this.
|
|
// AUDIO_DEVICE_NONE
|
|
sp<IOProfile> profile = getProfileForDirectOutput(AUDIO_DEVICE_NONE /*ignore device */,
|
|
offloadInfo.sample_rate,
|
|
offloadInfo.format,
|
|
offloadInfo.channel_mask,
|
|
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
|
|
ALOGV("isOffloadSupported() profile %sfound", profile != 0 ? "" : "NOT ");
|
|
return (profile != 0);
|
|
}
|
|
|
|
void AudioPolicyManagerCustom::setPhoneState(audio_mode_t state)
|
|
{
|
|
ALOGD("setPhoneState() state %d", state);
|
|
// store previous phone state for management of sonification strategy below
|
|
audio_devices_t newDevice = AUDIO_DEVICE_NONE;
|
|
int oldState = mEngine->getPhoneState();
|
|
|
|
if (mEngine->setPhoneState(state) != NO_ERROR) {
|
|
ALOGW("setPhoneState() invalid or same state %d", state);
|
|
return;
|
|
}
|
|
/// Opens: can these line be executed after the switch of volume curves???
|
|
// if leaving call state, handle special case of active streams
|
|
// pertaining to sonification strategy see handleIncallSonification()
|
|
if (isStateInCall(oldState)) {
|
|
ALOGV("setPhoneState() in call state management: new state is %d", state);
|
|
for (size_t j = 0; j < mOutputs.size(); j++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(j);
|
|
for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
|
|
handleIncallSonification((audio_stream_type_t)stream, false, true, curOutput);
|
|
}
|
|
}
|
|
|
|
// force reevaluating accessibility routing when call stops
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
|
|
}
|
|
|
|
/**
|
|
* Switching to or from incall state or switching between telephony and VoIP lead to force
|
|
* routing command.
|
|
*/
|
|
bool force = ((is_state_in_call(oldState) != is_state_in_call(state))
|
|
|| (is_state_in_call(state) && (state != oldState)));
|
|
|
|
// check for device and output changes triggered by new phone state
|
|
checkA2dpSuspend();
|
|
checkOutputForAllStrategies();
|
|
updateDevicesAndOutputs();
|
|
|
|
sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput;
|
|
#ifdef VOICE_CONCURRENCY
|
|
char propValue[PROPERTY_VALUE_MAX];
|
|
bool prop_playback_enabled = false, prop_rec_enabled=false, prop_voip_enabled = false;
|
|
|
|
if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
|
|
prop_playback_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
if(property_get("voice.record.conc.disabled", propValue, NULL)) {
|
|
prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
|
|
prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
if ((AUDIO_MODE_IN_CALL != oldState) && (AUDIO_MODE_IN_CALL == state)) {
|
|
ALOGD("voice_conc:Entering to call mode oldState :: %d state::%d ",
|
|
oldState, state);
|
|
mvoice_call_state = state;
|
|
if (prop_rec_enabled) {
|
|
//Close all active inputs
|
|
Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
|
|
if (activeInputs.size() != 0) {
|
|
for (size_t i = 0; i < activeInputs.size(); i++) {
|
|
sp<AudioInputDescriptor> activeInput = activeInputs[i];
|
|
switch(activeInput->inputSource()) {
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
ALOGD("voice_conc:FOUND active input during call active: %d",activeInput->inputSource());
|
|
break;
|
|
|
|
case AUDIO_SOURCE_VOICE_COMMUNICATION:
|
|
if(prop_voip_enabled) {
|
|
ALOGD("voice_conc:CLOSING VoIP input source on call setup :%d ",activeInput->inputSource());
|
|
AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
|
|
audio_session_t activeSession = activeSessions.keyAt(0);
|
|
stopInput(activeInput->mIoHandle, activeSession);
|
|
releaseInput(activeInput->mIoHandle, activeSession);
|
|
}
|
|
break;
|
|
|
|
default:
|
|
ALOGD("voice_conc:CLOSING input on call setup for inputSource: %d",activeInput->inputSource());
|
|
AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
|
|
audio_session_t activeSession = activeSessions.keyAt(0);
|
|
stopInput(activeInput->mIoHandle, activeSession);
|
|
releaseInput(activeInput->mIoHandle, activeSession);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
} else if (prop_voip_enabled) {
|
|
Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
|
|
if (activeInputs.size() != 0) {
|
|
for (size_t i = 0; i < activeInputs.size(); i++) {
|
|
sp<AudioInputDescriptor> activeInput = activeInputs[i];
|
|
if (AUDIO_SOURCE_VOICE_COMMUNICATION == activeInput->inputSource()) {
|
|
ALOGD("voice_conc:CLOSING VoIP on call setup : %d",activeInput->inputSource());
|
|
AudioSessionCollection activeSessions = activeInput->getAudioSessions(true);
|
|
audio_session_t activeSession = activeSessions.keyAt(0);
|
|
stopInput(activeInput->mIoHandle, activeSession);
|
|
releaseInput(activeInput->mIoHandle, activeSession);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (prop_playback_enabled) {
|
|
// Move tracks associated to this strategy from previous output to new output
|
|
for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
|
|
ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
|
|
if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
|
|
if ((AUDIO_STREAM_MUSIC == i) ||
|
|
(AUDIO_STREAM_VOICE_CALL == i) ) {
|
|
ALOGD("voice_conc:Invalidate stream type %d", i);
|
|
mpClientInterface->invalidateStream((audio_stream_type_t)i);
|
|
}
|
|
} else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
|
|
ALOGD("voice_conc:Invalidate stream type %d", i);
|
|
mpClientInterface->invalidateStream((audio_stream_type_t)i);
|
|
}
|
|
}
|
|
}
|
|
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
|
|
if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
|
|
ALOGD("voice_conc:ouput desc / profile is NULL");
|
|
continue;
|
|
}
|
|
|
|
bool isFastFallBackNeeded =
|
|
((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & outputDesc->mProfile->getFlags());
|
|
|
|
if ((AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) && isFastFallBackNeeded) {
|
|
if (((!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY))
|
|
&& prop_playback_enabled) {
|
|
ALOGD("voice_conc:calling suspendOutput on call mode for primary output");
|
|
mpClientInterface->suspendOutput(mOutputs.keyAt(i));
|
|
} //Close compress all sessions
|
|
else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
|
|
&& prop_playback_enabled) {
|
|
ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
|
|
closeOutput(mOutputs.keyAt(i));
|
|
}
|
|
else if ((outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX)
|
|
&& prop_voip_enabled) {
|
|
ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output");
|
|
closeOutput(mOutputs.keyAt(i));
|
|
}
|
|
} else if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
|
|
if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_VOIP_RX) {
|
|
if (prop_voip_enabled) {
|
|
ALOGD("voice_conc:calling closeOutput on call mode for DIRECT output");
|
|
closeOutput(mOutputs.keyAt(i));
|
|
}
|
|
}
|
|
else if (prop_playback_enabled
|
|
&& (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
|
|
ALOGD("voice_conc:calling closeOutput on call mode for COMPRESS output");
|
|
closeOutput(mOutputs.keyAt(i));
|
|
}
|
|
}
|
|
}
|
|
// If effects where present on any of the above closed outputs,
|
|
// audioflinger moved them to the primary output by default
|
|
// move them back to the appropriate output.
|
|
moveGlobalEffect();
|
|
}
|
|
|
|
if ((AUDIO_MODE_IN_CALL == oldState || AUDIO_MODE_IN_COMMUNICATION == oldState) &&
|
|
(AUDIO_MODE_NORMAL == state) && prop_playback_enabled && mvoice_call_state) {
|
|
ALOGD("voice_conc:EXITING from call mode oldState :: %d state::%d \n",oldState, state);
|
|
mvoice_call_state = 0;
|
|
if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
|
|
//restore PCM (deep-buffer) output after call termination
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
|
|
if ( (outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
|
|
ALOGD("voice_conc:ouput desc / profile is NULL");
|
|
continue;
|
|
}
|
|
if (!outputDesc->isDuplicated() && outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
|
|
ALOGD("voice_conc:calling restoreOutput after call mode for primary output");
|
|
mpClientInterface->restoreOutput(mOutputs.keyAt(i));
|
|
}
|
|
}
|
|
}
|
|
//call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
|
|
for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
|
|
ALOGV("voice_conc:Invalidate on call mode for stream :: %d ", i);
|
|
if (AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag) {
|
|
if ((AUDIO_STREAM_MUSIC == i) ||
|
|
(AUDIO_STREAM_VOICE_CALL == i) ) {
|
|
mpClientInterface->invalidateStream((audio_stream_type_t)i);
|
|
}
|
|
} else if (AUDIO_OUTPUT_FLAG_FAST == mFallBackflag) {
|
|
mpClientInterface->invalidateStream((audio_stream_type_t)i);
|
|
}
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
sp<SwAudioOutputDescriptor> outputDesc = NULL;
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
outputDesc = mOutputs.valueAt(i);
|
|
if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
|
|
ALOGD("voice_conc:ouput desc / profile is NULL");
|
|
continue;
|
|
}
|
|
|
|
if (property_get_bool("voice.dsd.playback.conc.disabled", true) &&
|
|
(outputDesc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
|
|
(outputDesc->mFormat == AUDIO_FORMAT_DSD)) {
|
|
ALOGD("voice_conc:calling closeOutput on call mode for DSD COMPRESS output");
|
|
closeOutput(mOutputs.keyAt(i));
|
|
// call invalidate for music, so that DSD compress will fallback to deep-buffer.
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
|
|
}
|
|
|
|
}
|
|
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
char recConcPropValue[PROPERTY_VALUE_MAX];
|
|
bool prop_rec_play_enabled = false;
|
|
|
|
if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
|
|
prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
|
|
}
|
|
if (prop_rec_play_enabled) {
|
|
if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
|
|
ALOGD("phone state changed to MODE_IN_COMM invlaidating music and voice streams");
|
|
// call invalidate for voice streams, so that it can use deepbuffer with VoIP out device from HAL
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_VOICE_CALL);
|
|
// call invalidate for music, so that compress will fallback to deep-buffer with VoIP out device
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
|
|
|
|
// close compress output to make sure session will be closed before timeout(60sec)
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
|
|
if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
|
|
ALOGD("ouput desc / profile is NULL");
|
|
continue;
|
|
}
|
|
|
|
if (outputDesc->mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
|
|
ALOGD("calling closeOutput on call mode for COMPRESS output");
|
|
closeOutput(mOutputs.keyAt(i));
|
|
}
|
|
}
|
|
// If effects where present on any of the above closed outputs,
|
|
// audioflinger moved them to the primary output by default
|
|
// move them back to the appropriate output.
|
|
moveGlobalEffect();
|
|
} else if ((oldState == AUDIO_MODE_IN_COMMUNICATION) &&
|
|
(mEngine->getPhoneState() == AUDIO_MODE_NORMAL)) {
|
|
// call invalidate for music so that music can fallback to compress
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_MUSIC);
|
|
}
|
|
}
|
|
#endif
|
|
mPrevPhoneState = oldState;
|
|
int delayMs = 0;
|
|
if (isStateInCall(state)) {
|
|
nsecs_t sysTime = systemTime();
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
// mute media and sonification strategies and delay device switch by the largest
|
|
// latency of any output where either strategy is active.
|
|
// This avoid sending the ring tone or music tail into the earpiece or headset.
|
|
if ((isStrategyActive(desc, STRATEGY_MEDIA,
|
|
SONIFICATION_HEADSET_MUSIC_DELAY,
|
|
sysTime) ||
|
|
isStrategyActive(desc, STRATEGY_SONIFICATION,
|
|
SONIFICATION_HEADSET_MUSIC_DELAY,
|
|
sysTime)) &&
|
|
(delayMs < (int)desc->latency()*2)) {
|
|
delayMs = desc->latency()*2;
|
|
}
|
|
setStrategyMute(STRATEGY_MEDIA, true, desc);
|
|
setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS,
|
|
getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
|
|
setStrategyMute(STRATEGY_SONIFICATION, true, desc);
|
|
setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS,
|
|
getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
|
|
}
|
|
}
|
|
|
|
if (hasPrimaryOutput()) {
|
|
// Note that despite the fact that getNewOutputDevice() is called on the primary output,
|
|
// the device returned is not necessarily reachable via this output
|
|
audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
|
|
// force routing command to audio hardware when ending call
|
|
// even if no device change is needed
|
|
if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
|
|
rxDevice = mPrimaryOutput->device();
|
|
}
|
|
|
|
if (state == AUDIO_MODE_IN_CALL) {
|
|
updateCallRouting(rxDevice, delayMs);
|
|
} else if (oldState == AUDIO_MODE_IN_CALL) {
|
|
if (mCallRxPatch != 0) {
|
|
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
|
|
mCallRxPatch.clear();
|
|
}
|
|
if (mCallTxPatch != 0) {
|
|
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
|
|
mCallTxPatch.clear();
|
|
}
|
|
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
|
|
} else {
|
|
setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
|
|
}
|
|
}
|
|
//update device for all non-primary outputs
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_io_handle_t output = mOutputs.keyAt(i);
|
|
if (output != mPrimaryOutput->mIoHandle) {
|
|
newDevice = getNewOutputDevice(mOutputs.valueFor(output), false /*fromCache*/);
|
|
setOutputDevice(mOutputs.valueFor(output), newDevice, (newDevice != AUDIO_DEVICE_NONE));
|
|
}
|
|
}
|
|
// if entering in call state, handle special case of active streams
|
|
// pertaining to sonification strategy see handleIncallSonification()
|
|
if (isStateInCall(state)) {
|
|
ALOGV("setPhoneState() in call state management: new state is %d", state);
|
|
for (size_t j = 0; j < mOutputs.size(); j++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(j);
|
|
for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
|
|
handleIncallSonification((audio_stream_type_t)stream, true, true, curOutput);
|
|
}
|
|
}
|
|
|
|
// force reevaluating accessibility routing when call starts
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
|
|
}
|
|
|
|
// Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
|
|
if (state == AUDIO_MODE_RINGTONE &&
|
|
isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
|
|
mLimitRingtoneVolume = true;
|
|
} else {
|
|
mLimitRingtoneVolume = false;
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerCustom::setForceUse(audio_policy_force_use_t usage,
|
|
audio_policy_forced_cfg_t config)
|
|
{
|
|
ALOGD("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
|
|
|
|
if (mEngine->setForceUse(usage, config) != NO_ERROR) {
|
|
ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage);
|
|
return;
|
|
}
|
|
bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) ||
|
|
(usage == AUDIO_POLICY_FORCE_FOR_DOCK) ||
|
|
(usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
|
|
|
|
// check for device and output changes triggered by new force usage
|
|
checkA2dpSuspend();
|
|
checkOutputForAllStrategies();
|
|
updateDevicesAndOutputs();
|
|
|
|
/*audio policy: workaround for truncated touch sounds*/
|
|
//FIXME: workaround for truncated touch sounds
|
|
// to be removed when the problem is handled by system UI
|
|
uint32_t delayMs = 0;
|
|
uint32_t waitMs = 0;
|
|
if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) {
|
|
delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
|
|
}
|
|
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
|
|
audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
|
|
waitMs = updateCallRouting(newDevice, delayMs);
|
|
}
|
|
// Use reverse loop to make sure any low latency usecases (generally tones)
|
|
// are not routed before non LL usecases (generally music).
|
|
// We can safely assume that LL output would always have lower index,
|
|
// and use this work-around to avoid routing of output with music stream
|
|
// from the context of short lived LL output.
|
|
// Note: in case output's share backend(HAL sharing is implicit) all outputs
|
|
// gets routing update while processing first output itself.
|
|
for (size_t i = mOutputs.size(); i > 0; i--) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i-1);
|
|
audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
|
|
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
|
|
waitMs = setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE),
|
|
delayMs);
|
|
}
|
|
if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
|
|
applyStreamVolumes(outputDesc, newDevice, waitMs, true);
|
|
}
|
|
}
|
|
|
|
Vector<sp <AudioInputDescriptor> > activeInputs = mInputs.getActiveInputs();
|
|
for (size_t i = 0; i < activeInputs.size(); i++) {
|
|
sp<AudioInputDescriptor> activeDesc = activeInputs[i];
|
|
audio_devices_t newDevice = getNewInputDevice(activeDesc);
|
|
// Force new input selection if the new device can not be reached via current input
|
|
if (activeDesc->mProfile->getSupportedDevices().types() &
|
|
(newDevice & ~AUDIO_DEVICE_BIT_IN)) {
|
|
setInputDevice(activeDesc->mIoHandle, newDevice);
|
|
} else {
|
|
closeInput(activeDesc->mIoHandle);
|
|
}
|
|
}
|
|
}
|
|
|
|
status_t AudioPolicyManagerCustom::stopSource(const sp<AudioOutputDescriptor>& outputDesc,
|
|
audio_stream_type_t stream,
|
|
bool forceDeviceUpdate)
|
|
{
|
|
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
|
|
ALOGW("stopSource() invalid stream %d", stream);
|
|
return INVALID_OPERATION;
|
|
}
|
|
// always handle stream stop, check which stream type is stopping
|
|
handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
|
|
|
|
// handle special case for sonification while in call
|
|
if (isInCall()) {
|
|
if (outputDesc->isDuplicated()) {
|
|
handleIncallSonification(stream, false, false, outputDesc->subOutput1()->mIoHandle);
|
|
handleIncallSonification(stream, false, false, outputDesc->subOutput2()->mIoHandle);
|
|
}
|
|
handleIncallSonification(stream, false, false, outputDesc->mIoHandle);
|
|
}
|
|
|
|
if (outputDesc->mRefCount[stream] > 0) {
|
|
// decrement usage count of this stream on the output
|
|
outputDesc->changeRefCount(stream, -1);
|
|
|
|
// store time at which the stream was stopped - see isStreamActive()
|
|
if (outputDesc->mRefCount[stream] == 0 || forceDeviceUpdate) {
|
|
outputDesc->mStopTime[stream] = systemTime();
|
|
audio_devices_t prevDevice = outputDesc->device();
|
|
audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
|
|
// delay the device switch by twice the latency because stopOutput() is executed when
|
|
// the track stop() command is received and at that time the audio track buffer can
|
|
// still contain data that needs to be drained. The latency only covers the audio HAL
|
|
// and kernel buffers. Also the latency does not always include additional delay in the
|
|
// audio path (audio DSP, CODEC ...)
|
|
setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2);
|
|
|
|
// force restoring the device selection on other active outputs if it differs from the
|
|
// one being selected for this output
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(i);
|
|
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if (desc != outputDesc &&
|
|
desc->isActive() &&
|
|
outputDesc->sharesHwModuleWith(desc) &&
|
|
(newDevice != desc->device())) {
|
|
audio_devices_t dev = getNewOutputDevice(mOutputs.valueFor(curOutput), false /*fromCache*/);
|
|
bool force = desc->device() != dev;
|
|
uint32_t delayMs;
|
|
if (dev == prevDevice) {
|
|
delayMs = 0;
|
|
} else {
|
|
delayMs = outputDesc->latency()*2;
|
|
}
|
|
setOutputDevice(desc,
|
|
dev,
|
|
force,
|
|
delayMs);
|
|
/*audio policy: fix media volume after ringtone*/
|
|
// re-apply device specific volume if not done by setOutputDevice()
|
|
if (!force) {
|
|
applyStreamVolumes(desc, dev, delayMs);
|
|
}
|
|
}
|
|
}
|
|
// update the outputs if stopping one with a stream that can affect notification routing
|
|
handleNotificationRoutingForStream(stream);
|
|
}
|
|
return NO_ERROR;
|
|
} else {
|
|
ALOGW("stopOutput() refcount is already 0");
|
|
return INVALID_OPERATION;
|
|
}
|
|
}
|
|
|
|
status_t AudioPolicyManagerCustom::startSource(const sp<AudioOutputDescriptor>& outputDesc,
|
|
audio_stream_type_t stream,
|
|
audio_devices_t device,
|
|
const char *address,
|
|
uint32_t *delayMs)
|
|
{
|
|
// cannot start playback of STREAM_TTS if any other output is being used
|
|
uint32_t beaconMuteLatency = 0;
|
|
|
|
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
|
|
ALOGW("startSource() invalid stream %d", stream);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
*delayMs = 0;
|
|
if (stream == AUDIO_STREAM_TTS) {
|
|
ALOGV("\t found BEACON stream");
|
|
if (!mTtsOutputAvailable && mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) {
|
|
return INVALID_OPERATION;
|
|
} else {
|
|
beaconMuteLatency = handleEventForBeacon(STARTING_BEACON);
|
|
}
|
|
} else {
|
|
// some playback other than beacon starts
|
|
beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT);
|
|
}
|
|
|
|
// force device change if the output is inactive and no audio patch is already present.
|
|
// check active before incrementing usage count
|
|
bool force = !outputDesc->isActive() &&
|
|
(outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE);
|
|
|
|
// increment usage count for this stream on the requested output:
|
|
// NOTE that the usage count is the same for duplicated output and hardware output which is
|
|
// necessary for a correct control of hardware output routing by startOutput() and stopOutput()
|
|
outputDesc->changeRefCount(stream, 1);
|
|
|
|
if (outputDesc->mRefCount[stream] == 1 || device != AUDIO_DEVICE_NONE) {
|
|
// starting an output being rerouted?
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
device = getNewOutputDevice(outputDesc, false /*fromCache*/);
|
|
}
|
|
routing_strategy strategy = getStrategy(stream);
|
|
bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
|
|
(strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
|
|
(beaconMuteLatency > 0);
|
|
uint32_t waitMs = beaconMuteLatency;
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if (desc != outputDesc) {
|
|
// force a device change if any other output is:
|
|
// - managed by the same hw module
|
|
// - has a current device selection that differs from selected device.
|
|
// - supports currently selected device
|
|
// - has an active audio patch
|
|
// In this case, the audio HAL must receive the new device selection so that it can
|
|
// change the device currently selected by the other active output.
|
|
if (outputDesc->sharesHwModuleWith(desc) &&
|
|
desc->device() != device &&
|
|
desc->supportedDevices() & device &&
|
|
desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) {
|
|
force = true;
|
|
}
|
|
// wait for audio on other active outputs to be presented when starting
|
|
// a notification so that audio focus effect can propagate, or that a mute/unmute
|
|
// event occurred for beacon
|
|
uint32_t latency = desc->latency();
|
|
if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
|
|
waitMs = latency;
|
|
}
|
|
}
|
|
}
|
|
uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force, 0, NULL, address);
|
|
|
|
// handle special case for sonification while in call
|
|
if (isInCall()) {
|
|
handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
|
|
}
|
|
|
|
// apply volume rules for current stream and device if necessary
|
|
checkAndSetVolume(stream,
|
|
mVolumeCurves->getVolumeIndex(stream, device),
|
|
outputDesc,
|
|
device);
|
|
|
|
// update the outputs if starting an output with a stream that can affect notification
|
|
// routing
|
|
handleNotificationRoutingForStream(stream);
|
|
|
|
// force reevaluating accessibility routing when ringtone or alarm starts
|
|
if (strategy == STRATEGY_SONIFICATION) {
|
|
mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
|
|
}
|
|
if (waitMs > muteWaitMs) {
|
|
*delayMs = waitMs - muteWaitMs;
|
|
}
|
|
|
|
} else {
|
|
// handle special case for sonification while in call
|
|
if (isInCall()) {
|
|
handleIncallSonification(stream, true, false, outputDesc->mIoHandle);
|
|
}
|
|
}
|
|
return NO_ERROR;
|
|
}
|
|
|
|
void AudioPolicyManagerCustom::handleIncallSonification(audio_stream_type_t stream,
|
|
bool starting, bool stateChange,
|
|
audio_io_handle_t output)
|
|
{
|
|
if(!hasPrimaryOutput()) {
|
|
return;
|
|
}
|
|
// no action needed for AUDIO_STREAM_PATCH stream type, it's for internal flinger tracks
|
|
if (stream == AUDIO_STREAM_PATCH) {
|
|
return;
|
|
}
|
|
// if the stream pertains to sonification strategy and we are in call we must
|
|
// mute the stream if it is low visibility. If it is high visibility, we must play a tone
|
|
// in the device used for phone strategy and play the tone if the selected device does not
|
|
// interfere with the device used for phone strategy
|
|
// if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
|
|
// many times as there are active tracks on the output
|
|
const routing_strategy stream_strategy = getStrategy(stream);
|
|
if ((stream_strategy == STRATEGY_SONIFICATION) ||
|
|
((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
|
|
ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
|
|
stream, starting, outputDesc->mDevice, stateChange);
|
|
if (outputDesc->mRefCount[stream]) {
|
|
int muteCount = 1;
|
|
if (stateChange) {
|
|
muteCount = outputDesc->mRefCount[stream];
|
|
}
|
|
if (audio_is_low_visibility(stream)) {
|
|
ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
|
|
for (int i = 0; i < muteCount; i++) {
|
|
setStreamMute(stream, starting, outputDesc);
|
|
}
|
|
} else {
|
|
ALOGV("handleIncallSonification() high visibility");
|
|
if (outputDesc->device() &
|
|
getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
|
|
ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
|
|
for (int i = 0; i < muteCount; i++) {
|
|
setStreamMute(stream, starting, outputDesc);
|
|
}
|
|
}
|
|
if (starting) {
|
|
mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
|
|
AUDIO_STREAM_VOICE_CALL);
|
|
} else {
|
|
mpClientInterface->stopTone();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void AudioPolicyManagerCustom::handleNotificationRoutingForStream(audio_stream_type_t stream) {
|
|
switch(stream) {
|
|
case AUDIO_STREAM_MUSIC:
|
|
checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL);
|
|
updateDevicesAndOutputs();
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
status_t AudioPolicyManagerCustom::checkAndSetVolume(audio_stream_type_t stream,
|
|
int index,
|
|
const sp<AudioOutputDescriptor>& outputDesc,
|
|
audio_devices_t device,
|
|
int delayMs,
|
|
bool force)
|
|
{
|
|
if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
|
|
ALOGW("checkAndSetVolume() invalid stream %d", stream);
|
|
return INVALID_OPERATION;
|
|
}
|
|
// do not change actual stream volume if the stream is muted
|
|
if (outputDesc->mMuteCount[stream] != 0) {
|
|
ALOGVV("checkAndSetVolume() stream %d muted count %d",
|
|
stream, outputDesc->mMuteCount[stream]);
|
|
return NO_ERROR;
|
|
}
|
|
audio_policy_forced_cfg_t forceUseForComm =
|
|
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION);
|
|
// do not change in call volume if bluetooth is connected and vice versa
|
|
if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) ||
|
|
(stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) {
|
|
ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
|
|
stream, forceUseForComm);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
if (device == AUDIO_DEVICE_NONE) {
|
|
device = outputDesc->device();
|
|
}
|
|
|
|
float volumeDb = computeVolume(stream, index, device);
|
|
if (outputDesc->isFixedVolume(device)) {
|
|
volumeDb = 0.0f;
|
|
}
|
|
|
|
outputDesc->setVolume(volumeDb, stream, device, delayMs, force);
|
|
|
|
if (stream == AUDIO_STREAM_VOICE_CALL ||
|
|
stream == AUDIO_STREAM_BLUETOOTH_SCO) {
|
|
float voiceVolume;
|
|
// Force voice volume to max for bluetooth SCO as volume is managed by the headset
|
|
if (stream == AUDIO_STREAM_VOICE_CALL) {
|
|
voiceVolume = (float)index/(float)mVolumeCurves->getVolumeIndexMax(stream);
|
|
} else {
|
|
voiceVolume = 1.0;
|
|
}
|
|
|
|
if (voiceVolume != mLastVoiceVolume) {
|
|
mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
|
|
mLastVoiceVolume = voiceVolume;
|
|
}
|
|
#ifdef FM_POWER_OPT
|
|
} else if (stream == AUDIO_STREAM_MUSIC && hasPrimaryOutput() &&
|
|
outputDesc == mPrimaryOutput && mFMIsActive) {
|
|
/* Avoid unnecessary set_parameter calls as it puts the primary
|
|
outputs FastMixer in HOT_IDLE leading to breaks in audio */
|
|
if (volumeDb != mPrevFMVolumeDb) {
|
|
mPrevFMVolumeDb = volumeDb;
|
|
AudioParameter param = AudioParameter();
|
|
param.addFloat(String8("fm_volume"), Volume::DbToAmpl(volumeDb));
|
|
//Double delayMs to avoid sound burst while device switch.
|
|
mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, param.toString(), delayMs*2);
|
|
}
|
|
#endif /* FM_POWER_OPT end */
|
|
}
|
|
|
|
return NO_ERROR;
|
|
}
|
|
|
|
bool AudioPolicyManagerCustom::isDirectOutput(audio_io_handle_t output) {
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
audio_io_handle_t curOutput = mOutputs.keyAt(i);
|
|
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool static tryForDirectPCM(audio_output_flags_t flags)
|
|
{
|
|
bool trackDirectPCM = false; // Output request for track created by other apps
|
|
|
|
if (flags == AUDIO_OUTPUT_FLAG_NONE) {
|
|
trackDirectPCM = property_get_bool("audio.offload.track.enable", true);
|
|
}
|
|
return trackDirectPCM;
|
|
}
|
|
|
|
status_t AudioPolicyManagerCustom::getOutputForAttr(const audio_attributes_t *attr,
|
|
audio_io_handle_t *output,
|
|
audio_session_t session,
|
|
audio_stream_type_t *stream,
|
|
uid_t uid,
|
|
const audio_config_t *config,
|
|
audio_output_flags_t flags,
|
|
audio_port_handle_t selectedDeviceId,
|
|
audio_port_handle_t *portId)
|
|
{
|
|
audio_offload_info_t tOffloadInfo = AUDIO_INFO_INITIALIZER;
|
|
audio_config_t tConfig;
|
|
|
|
uint32_t bitWidth = (audio_bytes_per_sample(config->format) * 8);
|
|
|
|
memcpy(&tConfig, config, sizeof(audio_config_t));
|
|
if ((flags == AUDIO_OUTPUT_FLAG_DIRECT || tryForDirectPCM(flags)) &&
|
|
(!memcmp(&config->offload_info, &tOffloadInfo, sizeof(audio_offload_info_t)))) {
|
|
tConfig.offload_info.sample_rate = config->sample_rate;
|
|
tConfig.offload_info.channel_mask = config->channel_mask;
|
|
tConfig.offload_info.format = config->format;
|
|
tConfig.offload_info.stream_type = *stream;
|
|
tConfig.offload_info.bit_width = bitWidth;
|
|
if (attr != NULL) {
|
|
ALOGV("found attribute .. setting usage %d ", attr->usage);
|
|
tConfig.offload_info.usage = attr->usage;
|
|
} else {
|
|
ALOGI("%s:: attribute is NULL .. no usage set", __func__);
|
|
}
|
|
}
|
|
|
|
return AudioPolicyManager::getOutputForAttr(attr, output, session, stream,
|
|
(uid_t)uid, &tConfig,
|
|
flags, (audio_port_handle_t)selectedDeviceId,
|
|
portId);
|
|
}
|
|
|
|
audio_io_handle_t AudioPolicyManagerCustom::getOutputForDevice(
|
|
audio_devices_t device,
|
|
audio_session_t session,
|
|
audio_stream_type_t stream,
|
|
uint32_t samplingRate,
|
|
audio_format_t format,
|
|
audio_channel_mask_t channelMask,
|
|
audio_output_flags_t flags,
|
|
const audio_offload_info_t *offloadInfo)
|
|
{
|
|
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
|
|
status_t status;
|
|
|
|
#ifdef AUDIO_POLICY_TEST
|
|
if (mCurOutput != 0) {
|
|
ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
|
|
mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
|
|
|
|
if (mTestOutputs[mCurOutput] == 0) {
|
|
ALOGV("getOutput() opening test output");
|
|
sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL,
|
|
mpClientInterface);
|
|
outputDesc->mDevice = mTestDevice;
|
|
outputDesc->mLatency = mTestLatencyMs;
|
|
outputDesc->mFlags =
|
|
(audio_output_flags_t)(mDirectOutput ? AUDIO_OUTPUT_FLAG_DIRECT : 0);
|
|
outputDesc->mRefCount[stream] = 0;
|
|
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
|
|
config.sample_rate = mTestSamplingRate;
|
|
config.channel_mask = mTestChannels;
|
|
config.format = mTestFormat;
|
|
if (offloadInfo != NULL) {
|
|
config.offload_info = *offloadInfo;
|
|
}
|
|
status = mpClientInterface->openOutput(0,
|
|
&mTestOutputs[mCurOutput],
|
|
&config,
|
|
&outputDesc->mDevice,
|
|
String8(""),
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
if (status == NO_ERROR) {
|
|
outputDesc->mSamplingRate = config.sample_rate;
|
|
outputDesc->mFormat = config.format;
|
|
outputDesc->mChannelMask = config.channel_mask;
|
|
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
|
|
if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
|
|
(stream != AUDIO_STREAM_MUSIC)) {
|
|
// compress should not be used for non-music streams
|
|
ALOGE("Offloading only allowed with music stream");
|
|
return 0;
|
|
}
|
|
|
|
if ((stream == AUDIO_STREAM_VOICE_CALL) &&
|
|
(channelMask == 1) &&
|
|
(samplingRate == 8000 || samplingRate == 16000 ||
|
|
samplingRate == 32000 || samplingRate == 48000)) {
|
|
// Allow Voip direct output only if:
|
|
// audio mode is MODE_IN_COMMUNCATION; AND
|
|
// voip output is not opened already; AND
|
|
// requested sample rate matches with that of voip input stream (if opened already)
|
|
int value = 0;
|
|
uint32_t mode = 0, voipOutCount = 1, voipSampleRate = 1;
|
|
bool is_vr_mode_on = false;
|
|
String8 valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("audio_mode"));
|
|
AudioParameter result = AudioParameter(valueStr);
|
|
if (result.getInt(String8("audio_mode"), value) == NO_ERROR) {
|
|
mode = value;
|
|
}
|
|
|
|
valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("voip_out_stream_count"));
|
|
result = AudioParameter(valueStr);
|
|
if (result.getInt(String8("voip_out_stream_count"), value) == NO_ERROR) {
|
|
voipOutCount = value;
|
|
}
|
|
|
|
valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("voip_sample_rate"));
|
|
result = AudioParameter(valueStr);
|
|
if (result.getInt(String8("voip_sample_rate"), value) == NO_ERROR) {
|
|
voipSampleRate = value;
|
|
}
|
|
|
|
if ((mode == AUDIO_MODE_IN_COMMUNICATION) && (voipOutCount == 0) &&
|
|
((voipSampleRate == 0) || (voipSampleRate == samplingRate))) {
|
|
if (audio_is_linear_pcm(format)) {
|
|
char propValue[PROPERTY_VALUE_MAX] = {0};
|
|
property_get("use.voice.path.for.pcm.voip", propValue, "0");
|
|
bool voipPcmSysPropEnabled = !strncmp("true", propValue, sizeof("true"));
|
|
if (voipPcmSysPropEnabled && (format == AUDIO_FORMAT_PCM_16_BIT)) {
|
|
flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_VOIP_RX |
|
|
AUDIO_OUTPUT_FLAG_DIRECT);
|
|
ALOGD("Set VoIP and Direct output flags for PCM format");
|
|
}
|
|
}
|
|
}
|
|
//IF VOIP is going to be started at the same time as when
|
|
//vr is enabled, get VOIP to fallback to low latency
|
|
String8 vr_value;
|
|
valueStr = mpClientInterface->getParameters((audio_io_handle_t)0,
|
|
String8("vr_audio_mode_on"));
|
|
result = AudioParameter(valueStr);
|
|
if (result.get(String8("vr_audio_mode_on"), vr_value) == NO_ERROR) {
|
|
is_vr_mode_on = vr_value.contains("true");
|
|
ALOGI("VR mode is %d, switch to primary output if request is for fast|raw",
|
|
is_vr_mode_on);
|
|
}
|
|
|
|
if (is_vr_mode_on) {
|
|
//check the flags being requested for, and clear FAST|RAW
|
|
flags = (audio_output_flags_t)(flags &
|
|
(~(AUDIO_OUTPUT_FLAG_FAST|AUDIO_OUTPUT_FLAG_RAW)));
|
|
|
|
}
|
|
|
|
}
|
|
|
|
#ifdef VOICE_CONCURRENCY
|
|
char propValue[PROPERTY_VALUE_MAX];
|
|
bool prop_play_enabled=false, prop_voip_enabled = false;
|
|
|
|
if(property_get("voice.playback.conc.disabled", propValue, NULL)) {
|
|
prop_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
|
|
prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
bool isDeepBufferFallBackNeeded =
|
|
((AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & flags);
|
|
bool isFastFallBackNeeded =
|
|
((AUDIO_OUTPUT_FLAG_DEEP_BUFFER | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT_PCM) & flags);
|
|
|
|
if (prop_play_enabled && mvoice_call_state) {
|
|
//check if voice call is active / running in background
|
|
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
|
|
((AUDIO_MODE_IN_CALL == mPrevPhoneState)
|
|
&& (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
|
|
{
|
|
if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
|
|
if(prop_voip_enabled) {
|
|
ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
|
|
flags );
|
|
return 0;
|
|
}
|
|
}
|
|
else {
|
|
if (isFastFallBackNeeded &&
|
|
(AUDIO_OUTPUT_FLAG_FAST == mFallBackflag)) {
|
|
ALOGD("voice_conc:IN call mode adding ULL flags .. flags: %x ", flags );
|
|
flags = AUDIO_OUTPUT_FLAG_FAST;
|
|
} else if (isDeepBufferFallBackNeeded &&
|
|
(AUDIO_OUTPUT_FLAG_DEEP_BUFFER == mFallBackflag)) {
|
|
if (AUDIO_STREAM_MUSIC == stream) {
|
|
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
|
|
ALOGD("voice_conc:IN call mode adding deep-buffer flags %x ", flags );
|
|
}
|
|
else {
|
|
flags = AUDIO_OUTPUT_FLAG_FAST;
|
|
ALOGD("voice_conc:IN call mode adding fast flags %x ", flags );
|
|
}
|
|
}
|
|
}
|
|
}
|
|
} else if (prop_voip_enabled && mvoice_call_state) {
|
|
//check if voice call is active / running in background
|
|
//some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
|
|
//return only ULL ouput
|
|
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
|
|
((AUDIO_MODE_IN_CALL == mPrevPhoneState)
|
|
&& (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
|
|
{
|
|
if(AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
|
|
ALOGD("voice_conc:getoutput:IN call mode return no o/p for VoIP %x",
|
|
flags );
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
char recConcPropValue[PROPERTY_VALUE_MAX];
|
|
bool prop_rec_play_enabled = false;
|
|
|
|
if (property_get("rec.playback.conc.disabled", recConcPropValue, NULL)) {
|
|
prop_rec_play_enabled = atoi(recConcPropValue) || !strncmp("true", recConcPropValue, 4);
|
|
}
|
|
if ((prop_rec_play_enabled) &&
|
|
((true == mIsInputRequestOnProgress) || (mInputs.activeInputsCountOnDevices() > 0))) {
|
|
if (AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState()) {
|
|
if (AUDIO_OUTPUT_FLAG_VOIP_RX & flags) {
|
|
// allow VoIP using voice path
|
|
// Do nothing
|
|
} else if((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
|
|
ALOGD("voice_conc:MODE_IN_COMM is setforcing deep buffer output for non ULL... flags: %x", flags);
|
|
// use deep buffer path for all non ULL outputs
|
|
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
|
|
}
|
|
} else if ((flags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
|
|
ALOGD("voice_conc:Record mode is on forcing deep buffer output for non ULL... flags: %x ", flags);
|
|
// use deep buffer path for all non ULL outputs
|
|
flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
|
|
}
|
|
}
|
|
if (prop_rec_play_enabled &&
|
|
(stream == AUDIO_STREAM_ENFORCED_AUDIBLE)) {
|
|
ALOGD("Record conc is on forcing ULL output for ENFORCED_AUDIBLE");
|
|
flags = AUDIO_OUTPUT_FLAG_FAST;
|
|
}
|
|
#endif
|
|
|
|
#ifdef AUDIO_EXTN_AFE_PROXY_ENABLED
|
|
/*
|
|
* WFD audio routes back to target speaker when starting a ringtone playback.
|
|
* This is because primary output is reused for ringtone, so output device is
|
|
* updated based on SONIFICATION strategy for both ringtone and music playback.
|
|
* The same issue is not seen on remoted_submix HAL based WFD audio because
|
|
* primary output is not reused and a new output is created for ringtone playback.
|
|
* Issue is fixed by updating output flag to AUDIO_OUTPUT_FLAG_FAST when there is
|
|
* a non-music stream playback on WFD, so primary output is not reused for ringtone.
|
|
*/
|
|
audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types();
|
|
if ((availableOutputDeviceTypes & AUDIO_DEVICE_OUT_PROXY)
|
|
&& (stream != AUDIO_STREAM_MUSIC)) {
|
|
ALOGD("WFD audio: use OUTPUT_FLAG_FAST for non music stream. flags:%x", flags );
|
|
//For voip paths
|
|
if(flags & AUDIO_OUTPUT_FLAG_DIRECT)
|
|
flags = AUDIO_OUTPUT_FLAG_DIRECT;
|
|
else //route every thing else to ULL path
|
|
flags = AUDIO_OUTPUT_FLAG_FAST;
|
|
}
|
|
#endif
|
|
|
|
// open a direct output if required by specified parameters
|
|
// force direct flag if offload flag is set: offloading implies a direct output stream
|
|
// and all common behaviors are driven by checking only the direct flag
|
|
// this should normally be set appropriately in the policy configuration file
|
|
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
|
|
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
|
|
}
|
|
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
|
|
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
|
|
}
|
|
|
|
// Do internal direct magic here
|
|
bool offload_disabled = property_get_bool("audio.offload.disable", false);
|
|
if ((flags == AUDIO_OUTPUT_FLAG_NONE) &&
|
|
(stream == AUDIO_STREAM_MUSIC) &&
|
|
(offloadInfo != NULL) && !offload_disabled &&
|
|
((offloadInfo->usage == AUDIO_USAGE_MEDIA) || (offloadInfo->usage == AUDIO_USAGE_GAME))) {
|
|
audio_output_flags_t old_flags = flags;
|
|
flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DIRECT);
|
|
ALOGD("AudioCustomHAL --> Force Direct Flag .. old flags(0x%x)", old_flags);
|
|
} else if (flags == AUDIO_OUTPUT_FLAG_DIRECT && offload_disabled) {
|
|
ALOGD("AudioCustomHAL --> offloading is disabled: Force Remove Direct Flag");
|
|
flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_NONE);
|
|
}
|
|
|
|
bool forced_deep = false;
|
|
// only allow deep buffering for music stream type
|
|
if (stream != AUDIO_STREAM_MUSIC) {
|
|
flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
|
|
} else if (/* stream == AUDIO_STREAM_MUSIC && */
|
|
flags == AUDIO_OUTPUT_FLAG_NONE &&
|
|
property_get_bool("audio.deep_buffer.media", false /* default_value */)) {
|
|
forced_deep = true;
|
|
}
|
|
|
|
if (stream == AUDIO_STREAM_TTS) {
|
|
flags = AUDIO_OUTPUT_FLAG_TTS;
|
|
}
|
|
|
|
sp<IOProfile> profile;
|
|
|
|
// skip direct output selection if the request can obviously be attached to a mixed output
|
|
// and not explicitly requested
|
|
if (((flags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
|
|
audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX &&
|
|
audio_channel_count_from_out_mask(channelMask) <= 2) {
|
|
goto non_direct_output;
|
|
}
|
|
|
|
// Do not allow offloading if one non offloadable effect is enabled or MasterMono is enabled.
|
|
// This prevents creating an offloaded track and tearing it down immediately after start
|
|
// when audioflinger detects there is an active non offloadable effect.
|
|
// FIXME: We should check the audio session here but we do not have it in this context.
|
|
// This may prevent offloading in rare situations where effects are left active by apps
|
|
// in the background.
|
|
//
|
|
// Supplementary annotation:
|
|
// For sake of track offload introduced, we need a rollback for both compress offload
|
|
// and track offload use cases.
|
|
if ((flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_DIRECT)) &&
|
|
(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
|
|
ALOGD("non offloadable effect is enabled, try with non direct output");
|
|
goto non_direct_output;
|
|
}
|
|
|
|
profile = getProfileForDirectOutput(device,
|
|
samplingRate,
|
|
format,
|
|
channelMask,
|
|
(audio_output_flags_t)flags);
|
|
|
|
if (profile != 0) {
|
|
|
|
if (!(flags & AUDIO_OUTPUT_FLAG_DIRECT) &&
|
|
(profile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT)) {
|
|
ALOGI("got Direct without requesting ... reject ");
|
|
profile = NULL;
|
|
goto non_direct_output;
|
|
}
|
|
|
|
sp<SwAudioOutputDescriptor> outputDesc = NULL;
|
|
|
|
// if multiple concurrent offload decode is supported
|
|
// do no check for reuse and also don't close previous output if its offload
|
|
// previous output will be closed during track destruction
|
|
if (!(property_get_bool("audio.offload.multiple.enabled", false) &&
|
|
((flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0))) {
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
|
|
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
|
|
outputDesc = desc;
|
|
// reuse direct output if currently open by the same client
|
|
// and configured with same parameters
|
|
if ((samplingRate == outputDesc->mSamplingRate) &&
|
|
audio_formats_match(format, outputDesc->mFormat) &&
|
|
(channelMask == outputDesc->mChannelMask)) {
|
|
if (session == outputDesc->mDirectClientSession) {
|
|
outputDesc->mDirectOpenCount++;
|
|
ALOGV("getOutput() reusing direct output %d for session %d",
|
|
mOutputs.keyAt(i), session);
|
|
return mOutputs.keyAt(i);
|
|
} else {
|
|
ALOGV("getOutput() do not reuse direct output because current client (%d) "
|
|
"is not the same as requesting client (%d)",
|
|
outputDesc->mDirectClientSession, session);
|
|
goto non_direct_output;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
// close direct output if currently open and configured with different parameters
|
|
if (outputDesc != NULL) {
|
|
closeOutput(outputDesc->mIoHandle);
|
|
}
|
|
}
|
|
|
|
// if the selected profile is offloaded and no offload info was specified,
|
|
// create a default one
|
|
audio_offload_info_t defaultOffloadInfo = AUDIO_INFO_INITIALIZER;
|
|
if ((profile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && !offloadInfo) {
|
|
flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
|
|
defaultOffloadInfo.sample_rate = samplingRate;
|
|
defaultOffloadInfo.channel_mask = channelMask;
|
|
defaultOffloadInfo.format = format;
|
|
defaultOffloadInfo.stream_type = stream;
|
|
defaultOffloadInfo.bit_rate = 0;
|
|
defaultOffloadInfo.duration_us = -1;
|
|
defaultOffloadInfo.has_video = true; // conservative
|
|
defaultOffloadInfo.is_streaming = true; // likely
|
|
offloadInfo = &defaultOffloadInfo;
|
|
}
|
|
|
|
outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface);
|
|
outputDesc->mDevice = device;
|
|
outputDesc->mLatency = 0;
|
|
outputDesc->mFlags = (audio_output_flags_t)(outputDesc->mFlags | flags);
|
|
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
|
|
config.sample_rate = samplingRate;
|
|
config.channel_mask = channelMask;
|
|
config.format = format;
|
|
if (offloadInfo != NULL) {
|
|
config.offload_info = *offloadInfo;
|
|
}
|
|
status = mpClientInterface->openOutput(profile->getModuleHandle(),
|
|
&output,
|
|
&config,
|
|
&outputDesc->mDevice,
|
|
String8(""),
|
|
&outputDesc->mLatency,
|
|
outputDesc->mFlags);
|
|
|
|
// only accept an output with the requested parameters
|
|
if (status != NO_ERROR ||
|
|
(samplingRate != 0 && samplingRate != config.sample_rate) ||
|
|
(format != AUDIO_FORMAT_DEFAULT && !audio_formats_match(format, config.format)) ||
|
|
(channelMask != 0 && channelMask != config.channel_mask)) {
|
|
ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
|
|
"format %d %d, channelMask %04x %04x", output, samplingRate,
|
|
outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
|
|
outputDesc->mChannelMask);
|
|
if (output != AUDIO_IO_HANDLE_NONE) {
|
|
mpClientInterface->closeOutput(output);
|
|
}
|
|
// fall back to mixer output if possible when the direct output could not be open
|
|
if (audio_is_linear_pcm(format) && samplingRate <= SAMPLE_RATE_HZ_MAX) {
|
|
goto non_direct_output;
|
|
}
|
|
return AUDIO_IO_HANDLE_NONE;
|
|
}
|
|
outputDesc->mSamplingRate = config.sample_rate;
|
|
outputDesc->mChannelMask = config.channel_mask;
|
|
outputDesc->mFormat = config.format;
|
|
outputDesc->mRefCount[stream] = 0;
|
|
outputDesc->mStopTime[stream] = 0;
|
|
outputDesc->mDirectOpenCount = 1;
|
|
outputDesc->mDirectClientSession = session;
|
|
|
|
audio_io_handle_t srcOutput = getOutputForEffect();
|
|
addOutput(output, outputDesc);
|
|
audio_io_handle_t dstOutput = getOutputForEffect();
|
|
if (dstOutput == output) {
|
|
#ifdef DOLBY_ENABLE
|
|
status_t status = mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
|
|
if (status == NO_ERROR) {
|
|
for (size_t i = 0; i < mEffects.size(); i++) {
|
|
sp<EffectDescriptor> desc = mEffects.valueAt(i);
|
|
if (desc->mSession == AUDIO_SESSION_OUTPUT_MIX) {
|
|
// update the mIo member of EffectDescriptor for the global effect
|
|
ALOGV("%s updating mIo", __FUNCTION__);
|
|
desc->mIo = dstOutput;
|
|
}
|
|
}
|
|
} else {
|
|
ALOGW("%s moveEffects from %d to %d failed", __FUNCTION__, srcOutput, dstOutput);
|
|
}
|
|
#else // DOLBY_END
|
|
mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, srcOutput, dstOutput);
|
|
#endif // LINE_ADDED_BY_DOLBY
|
|
}
|
|
mPreviousOutputs = mOutputs;
|
|
ALOGV("getOutput() returns new direct output %d", output);
|
|
mpClientInterface->onAudioPortListUpdate();
|
|
return output;
|
|
}
|
|
|
|
non_direct_output:
|
|
|
|
// A request for HW A/V sync cannot fallback to a mixed output because time
|
|
// stamps are embedded in audio data
|
|
if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
|
|
return AUDIO_IO_HANDLE_NONE;
|
|
}
|
|
|
|
// ignoring channel mask due to downmix capability in mixer
|
|
|
|
// open a non direct output
|
|
|
|
// for non direct outputs, only PCM is supported
|
|
if (audio_is_linear_pcm(format)) {
|
|
// get which output is suitable for the specified stream. The actual
|
|
// routing change will happen when startOutput() will be called
|
|
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
|
|
|
|
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
|
|
flags = (audio_output_flags_t)(flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
|
|
|
|
if (forced_deep) {
|
|
flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
|
|
ALOGI("setting force DEEP buffer now ");
|
|
} else if(flags == AUDIO_OUTPUT_FLAG_NONE) {
|
|
// no deep buffer playback is requested hence fallback to primary
|
|
flags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_PRIMARY);
|
|
ALOGI("FLAG None hence request for a primary output");
|
|
}
|
|
|
|
output = selectOutput(outputs, flags, format);
|
|
}
|
|
ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d,"
|
|
"format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
|
|
|
|
ALOGV("getOutputForDevice() returns output %d", output);
|
|
|
|
return output;
|
|
}
|
|
|
|
status_t AudioPolicyManagerCustom::getInputForAttr(const audio_attributes_t *attr,
|
|
audio_io_handle_t *input,
|
|
audio_session_t session,
|
|
uid_t uid,
|
|
const audio_config_base_t *config,
|
|
audio_input_flags_t flags,
|
|
audio_port_handle_t selectedDeviceId,
|
|
input_type_t *inputType,
|
|
audio_port_handle_t *portId)
|
|
{
|
|
audio_source_t inputSource;
|
|
inputSource = attr->source;
|
|
#ifdef VOICE_CONCURRENCY
|
|
|
|
char propValue[PROPERTY_VALUE_MAX];
|
|
bool prop_rec_enabled=false, prop_voip_enabled = false;
|
|
|
|
if(property_get("voice.record.conc.disabled", propValue, NULL)) {
|
|
prop_rec_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
if(property_get("voice.voip.conc.disabled", propValue, NULL)) {
|
|
prop_voip_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
if (prop_rec_enabled && mvoice_call_state) {
|
|
//check if voice call is active / running in background
|
|
//some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
|
|
//Need to block input request
|
|
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
|
|
((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
|
|
(AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
|
|
{
|
|
switch(inputSource) {
|
|
case AUDIO_SOURCE_VOICE_UPLINK:
|
|
case AUDIO_SOURCE_VOICE_DOWNLINK:
|
|
case AUDIO_SOURCE_VOICE_CALL:
|
|
ALOGD("voice_conc:Creating input during incall mode for inputSource: %d",
|
|
inputSource);
|
|
break;
|
|
|
|
case AUDIO_SOURCE_VOICE_COMMUNICATION:
|
|
if(prop_voip_enabled) {
|
|
ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
|
|
inputSource);
|
|
return NO_INIT;
|
|
}
|
|
break;
|
|
default:
|
|
ALOGD("voice_conc:BLOCK VoIP requst incall mode for inputSource: %d",
|
|
inputSource);
|
|
return NO_INIT;
|
|
}
|
|
}
|
|
}//check for VoIP flag
|
|
else if(prop_voip_enabled && mvoice_call_state) {
|
|
//check if voice call is active / running in background
|
|
//some of VoIP apps(like SIP2SIP call) supports resume of VoIP call when call in progress
|
|
//Need to block input request
|
|
if((AUDIO_MODE_IN_CALL == mEngine->getPhoneState()) ||
|
|
((AUDIO_MODE_IN_CALL == mPrevPhoneState) &&
|
|
(AUDIO_MODE_IN_COMMUNICATION == mEngine->getPhoneState())))
|
|
{
|
|
if(inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION) {
|
|
ALOGD("BLOCKING VoIP request during incall mode for inputSource: %d ",inputSource);
|
|
return NO_INIT;
|
|
}
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
return AudioPolicyManager::getInputForAttr(attr,
|
|
input,
|
|
session,
|
|
uid,
|
|
config,
|
|
flags,
|
|
selectedDeviceId,
|
|
inputType,
|
|
portId);
|
|
}
|
|
|
|
|
|
status_t AudioPolicyManagerCustom::startInput(audio_io_handle_t input,
|
|
audio_session_t session,
|
|
concurrency_type__mask_t *concurrency)
|
|
{
|
|
ALOGV("startInput() input %d", input);
|
|
*concurrency = API_INPUT_CONCURRENCY_NONE;
|
|
ssize_t index = mInputs.indexOfKey(input);
|
|
if (index < 0) {
|
|
ALOGW("startInput() unknown input %d", input);
|
|
return BAD_VALUE;
|
|
}
|
|
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
|
|
|
|
sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
|
|
if (audioSession == 0) {
|
|
ALOGW("startInput() unknown session %d on input %d", session, input);
|
|
return BAD_VALUE;
|
|
}
|
|
|
|
if (!isConcurentCaptureAllowed(inputDesc, audioSession)) {
|
|
ALOGW("startInput(%d) failed: other input already started", input);
|
|
return INVALID_OPERATION;
|
|
}
|
|
|
|
if (isInCall()) {
|
|
*concurrency |= API_INPUT_CONCURRENCY_CALL;
|
|
}
|
|
|
|
if (mInputs.activeInputsCountOnDevices() != 0) {
|
|
*concurrency |= API_INPUT_CONCURRENCY_CAPTURE;
|
|
}
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
mIsInputRequestOnProgress = true;
|
|
|
|
char getPropValue[PROPERTY_VALUE_MAX];
|
|
bool prop_rec_play_enabled = false;
|
|
|
|
if (property_get("rec.playback.conc.disabled", getPropValue, NULL)) {
|
|
prop_rec_play_enabled = atoi(getPropValue) || !strncmp("true", getPropValue, 4);
|
|
}
|
|
|
|
if ((prop_rec_play_enabled) && (mInputs.activeInputsCountOnDevices() == 0)){
|
|
// send update to HAL on record playback concurrency
|
|
AudioParameter param = AudioParameter();
|
|
param.add(String8("rec_play_conc_on"), String8("true"));
|
|
ALOGD("startInput() setParameters rec_play_conc is setting to ON ");
|
|
mpClientInterface->setParameters(0, param.toString());
|
|
|
|
// Call invalidate to reset all opened non ULL audio tracks
|
|
// Move tracks associated to this strategy from previous output to new output
|
|
for (int i = AUDIO_STREAM_SYSTEM; i < AUDIO_STREAM_FOR_POLICY_CNT; i++) {
|
|
// Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder)
|
|
if (i != AUDIO_STREAM_ENFORCED_AUDIBLE) {
|
|
ALOGD("Invalidate on releaseInput for stream :: %d ", i);
|
|
//FIXME see fixme on name change
|
|
mpClientInterface->invalidateStream((audio_stream_type_t)i);
|
|
}
|
|
}
|
|
// close compress tracks
|
|
for (size_t i = 0; i < mOutputs.size(); i++) {
|
|
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
|
|
if ((outputDesc == NULL) || (outputDesc->mProfile == NULL)) {
|
|
ALOGD("ouput desc / profile is NULL");
|
|
continue;
|
|
}
|
|
if (outputDesc->mProfile->getFlags()
|
|
& AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
|
|
// close compress sessions
|
|
ALOGD("calling closeOutput on record conc for COMPRESS output");
|
|
closeOutput(mOutputs.keyAt(i));
|
|
}
|
|
}
|
|
// If effects where present on any of the above closed outputs,
|
|
// audioflinger moved them to the primary output by default
|
|
// move them back to the appropriate output.
|
|
moveGlobalEffect();
|
|
}
|
|
#endif
|
|
|
|
// increment activity count before calling getNewInputDevice() below as only active sessions
|
|
// are considered for device selection
|
|
audioSession->changeActiveCount(1);
|
|
|
|
// Routing?
|
|
mInputRoutes.incRouteActivity(session);
|
|
|
|
if (audioSession->activeCount() == 1 || mInputRoutes.hasRouteChanged(session)) {
|
|
// indicate active capture to sound trigger service if starting capture from a mic on
|
|
// primary HW module
|
|
audio_devices_t device = getNewInputDevice(inputDesc);
|
|
setInputDevice(input, device, true /* force */);
|
|
|
|
if (inputDesc->getAudioSessionCount(true/*activeOnly*/) == 1) {
|
|
// if input maps to a dynamic policy with an activity listener, notify of state change
|
|
if ((inputDesc->mPolicyMix != NULL)
|
|
&& ((inputDesc->mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0)) {
|
|
mpClientInterface->onDynamicPolicyMixStateUpdate(inputDesc->mPolicyMix->mDeviceAddress,
|
|
MIX_STATE_MIXING);
|
|
}
|
|
|
|
audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
|
|
if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
|
|
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1) {
|
|
SoundTrigger::setCaptureState(true);
|
|
}
|
|
|
|
// automatically enable the remote submix output when input is started if not
|
|
// used by a policy mix of type MIX_TYPE_RECORDERS
|
|
// For remote submix (a virtual device), we open only one input per capture request.
|
|
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
|
|
String8 address = String8("");
|
|
if (inputDesc->mPolicyMix == NULL) {
|
|
address = String8("0");
|
|
} else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
|
|
address = inputDesc->mPolicyMix->mDeviceAddress;
|
|
}
|
|
if (address != "") {
|
|
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
|
|
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
|
|
address, "remote-submix");
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
ALOGV("AudioPolicyManager::startInput() input source = %d", audioSession->inputSource());
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
mIsInputRequestOnProgress = false;
|
|
#endif
|
|
return NO_ERROR;
|
|
}
|
|
|
|
status_t AudioPolicyManagerCustom::stopInput(audio_io_handle_t input,
|
|
audio_session_t session)
|
|
{
|
|
status_t status;
|
|
status = AudioPolicyManager::stopInput(input, session);
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
char propValue[PROPERTY_VALUE_MAX];
|
|
bool prop_rec_play_enabled = false;
|
|
|
|
if (property_get("rec.playback.conc.disabled", propValue, NULL)) {
|
|
prop_rec_play_enabled = atoi(propValue) || !strncmp("true", propValue, 4);
|
|
}
|
|
|
|
if ((prop_rec_play_enabled) && (mInputs.activeInputsCountOnDevices() == 0)) {
|
|
|
|
//send update to HAL on record playback concurrency
|
|
AudioParameter param = AudioParameter();
|
|
param.add(String8("rec_play_conc_on"), String8("false"));
|
|
ALOGD("stopInput() setParameters rec_play_conc is setting to OFF ");
|
|
mpClientInterface->setParameters(0, param.toString());
|
|
|
|
//call invalidate tracks so that any open streams can fall back to deep buffer/compress path from ULL
|
|
for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
|
|
//Do not call invalidate for ENFORCED_AUDIBLE (otherwise pops are seen for camcorder stop tone)
|
|
if ((i != AUDIO_STREAM_ENFORCED_AUDIBLE) && (i != AUDIO_STREAM_PATCH)) {
|
|
ALOGD(" Invalidate on stopInput for stream :: %d ", i);
|
|
//FIXME see fixme on name change
|
|
mpClientInterface->invalidateStream((audio_stream_type_t)i);
|
|
}
|
|
}
|
|
}
|
|
#endif
|
|
return status;
|
|
}
|
|
|
|
void AudioPolicyManagerCustom::closeAllInputs() {
|
|
bool patchRemoved = false;
|
|
|
|
for(size_t input_index = mInputs.size(); input_index > 0; input_index--) {
|
|
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(input_index-1);
|
|
ssize_t patch_index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
|
|
if (patch_index >= 0) {
|
|
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(patch_index);
|
|
(void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
|
|
mAudioPatches.removeItemsAt(patch_index);
|
|
patchRemoved = true;
|
|
}
|
|
mpClientInterface->closeInput(mInputs.keyAt(input_index-1));
|
|
}
|
|
mInputs.clear();
|
|
SoundTrigger::setCaptureState(false);
|
|
nextAudioPortGeneration();
|
|
|
|
if (patchRemoved) {
|
|
mpClientInterface->onAudioPatchListUpdate();
|
|
}
|
|
}
|
|
|
|
AudioPolicyManagerCustom::AudioPolicyManagerCustom(AudioPolicyClientInterface *clientInterface)
|
|
: AudioPolicyManager(clientInterface),
|
|
mHdmiAudioDisabled(false),
|
|
mHdmiAudioEvent(false),
|
|
#ifndef FM_POWER_OPT
|
|
mPrevPhoneState(0)
|
|
#else
|
|
mPrevPhoneState(0),
|
|
mPrevFMVolumeDb(0.0f),
|
|
mFMIsActive(false)
|
|
#endif
|
|
{
|
|
|
|
#ifdef USE_XML_AUDIO_POLICY_CONF
|
|
ALOGD("USE_XML_AUDIO_POLICY_CONF is TRUE");
|
|
#else
|
|
ALOGD("USE_XML_AUDIO_POLICY_CONF is FALSE");
|
|
#endif
|
|
|
|
#ifdef RECORD_PLAY_CONCURRENCY
|
|
mIsInputRequestOnProgress = false;
|
|
#endif
|
|
|
|
|
|
#ifdef VOICE_CONCURRENCY
|
|
mFallBackflag = getFallBackPath();
|
|
#endif
|
|
}
|
|
}
|