msm8660: use policy manager from ALSA
Change-Id: I94db1a64072e94f3d85dd0fcf15565f3dd673104
diff --git a/msm8660/AudioPolicyManager.cpp b/msm8660/AudioPolicyManager.cpp
index c7e605d..f6076da 100644
--- a/msm8660/AudioPolicyManager.cpp
+++ b/msm8660/AudioPolicyManager.cpp
@@ -1,8 +1,6 @@
/*
* Copyright (C) 2009 The Android Open Source Project
- * Copyright (c) 2009-2013, The Linux Foundation. All rights reserved.
- * Not a Contribution, Apache license notifications and license are retained
- * for attribution purposes only.
+ * Copyright (c) 2012-2013, The Linux Foundation. All rights reserved.
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -17,7 +15,8 @@
* limitations under the License.
*/
-#define LOG_TAG "AudioPolicyManager8660"
+#define LOG_TAG "AudioPolicyManager"
+//#define LOG_NDEBUG 0
//#define LOG_NDDEBUG 0
//#define VERY_VERBOSE_LOGGING
@@ -30,245 +29,202 @@
// A device mask for all audio input devices that are considered "virtual" when evaluating
// active inputs in getActiveInput()
#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL AUDIO_DEVICE_IN_REMOTE_SUBMIX
+// A device mask for all audio output devices that are considered "remote" when evaluating
+// active output devices in isStreamActiveRemotely()
+#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
#include <utils/Log.h>
+
#include "AudioPolicyManager.h"
#include <hardware/audio_effect.h>
-#include <media/mediarecorder.h>
#include <hardware/audio.h>
-#include <math.h>
#include <hardware_legacy/audio_policy_conf.h>
+#include <math.h>
+#include <media/mediarecorder.h>
+#include <stdio.h>
+#include <cutils/properties.h>
namespace android_audio_legacy {
-
-
// ----------------------------------------------------------------------------
-// AudioPolicyManager for msm8660 platform
+// AudioPolicyManager
// ----------------------------------------------------------------------------
-audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
+
+AudioParameter param;
+
+void AudioPolicyManager::setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
{
- uint32_t device = 0;
-
- if (fromCache) {
- ALOGV("getDeviceForStrategy() from cache strategy %d, device %x",
- strategy, mDeviceForStrategy[strategy]);
- return mDeviceForStrategy[strategy];
+ ALOGVV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
+ setStreamMute(stream, on, output, delayMs, device);
+ }
}
-
- switch (strategy) {
-
- case STRATEGY_SONIFICATION_RESPECTFUL:
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
- // while media is playing (or has recently played), use the same device
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- } else {
- // when media is not playing anymore, fall back on the sonification behavior
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
- }
-
- break;
-
- case STRATEGY_DTMF:
- if (!isInCall()) {
- // when off call, DTMF strategy follows the same rules as MEDIA strategy
- device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
- break;
- }
- // when in call, DTMF and PHONE strategies follow the same rules
- // FALL THROUGH
-
- case STRATEGY_PHONE:
- // for phone strategy, we first consider the forced use and then the available devices by order
- // of priority
- switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
- case AudioSystem::FORCE_BT_SCO:
- if (!isInCall() || strategy != STRATEGY_DTMF) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
- if (device) break;
- }
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
- if (device) break;
- // if SCO device is requested but no SCO device is available, fall back to default case
- // FALL THROUGH
-
- default: // FORCE_NONE
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
- if (mHasA2dp && !isInCall() &&
- (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- if (device) break;
- }
-
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
- if (device) break;
- device = mDefaultOutputDevice;
- if (device == 0) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
- }
- break;
-
- case AudioSystem::FORCE_SPEAKER:
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
- // A2DP speaker when forcing to speaker output
- if (mHasA2dp && !isInCall() &&
- (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- if (device) break;
- }
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
- if (device) break;
- device = mDefaultOutputDevice;
- if (device == 0) {
- ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
- }
- break;
- }
-#ifdef QCOM_FM_ENABLED
- if (mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM) {
- device |= AUDIO_DEVICE_OUT_FM;
- if (mForceUse[AudioSystem::FOR_MEDIA] == AudioSystem::FORCE_SPEAKER) {
- device &= ~(AUDIO_DEVICE_OUT_WIRED_HEADSET);
- device |= AUDIO_DEVICE_OUT_SPEAKER;
- }
- }
-#endif
- break;
-
- case STRATEGY_SONIFICATION:
-
- // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
- // handleIncallSonification().
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
- break;
- }
- // FALL THROUGH
-
- case STRATEGY_ENFORCED_AUDIBLE:
- // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
- // except:
- // - when in call where it doesn't default to STRATEGY_PHONE behavior
- // - in countries where not enforced in which case it follows STRATEGY_MEDIA
-
- if (strategy == STRATEGY_SONIFICATION ||
- !mStreams[AUDIO_STREAM_ENFORCED_AUDIBLE].mCanBeMuted) {
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
- if (device == 0) {
- ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
- }
- }
- // The second device used for sonification is the same as the device used by media strategy
- // FALL THROUGH
-
- case STRATEGY_MEDIA: {
- //To route FM stream to speaker when headset is connected, a new switch case is added.
- //case AudioSystem::FORCE_SPEAKER for STRATEGY_MEDIA will come only when we need to route
- //FM stream to speaker.
- switch (mForceUse[AudioSystem::FOR_MEDIA]) {
- default:{
- uint32_t device2 = 0;
- if ((mHasA2dp && (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
- (getA2dpOutput() != 0) && !mA2dpSuspended )) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- }
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- }
-#ifdef QCOM_FM_ENABLED
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM_TX;
- }
-#endif
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
- }
-
- // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
- // STRATEGY_ENFORCED_AUDIBLE, 0 otherwise
- device |= device2;
- if (device) break;
- device = mDefaultOutputDevice;
- }
- break;
-
- case AudioSystem::FORCE_SPEAKER:
- device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
- break;
- }
-#ifdef QCOM_FM_ENABLED
- if (mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM) {
- device |= AUDIO_DEVICE_OUT_FM;
- }
-#endif
- // Do not play media stream if in call and the requested device would change the hardware
- // output routing
- if (mPhoneState == AudioSystem::MODE_IN_CALL &&
- !AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device) &&
- device != getDeviceForStrategy(STRATEGY_PHONE)) {
- device = getDeviceForStrategy(STRATEGY_PHONE);
- ALOGV("getDeviceForStrategy() incompatible media and phone devices");
- }
- } break;
-
- default:
- ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
- break;
- }
-
- ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
-
- return (audio_devices_t)device;
}
+void AudioPolicyManager::releaseOutput(audio_io_handle_t output)
+{
+ ALOGV("releaseOutput() %d", output);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("releaseOutput() releasing unknown output %d", output);
+ return;
+ }
+
+#ifdef AUDIO_POLICY_TEST
+ int testIndex = testOutputIndex(output);
+ if (testIndex != 0) {
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->isActive()) {
+ mpClientInterface->closeOutput(output);
+ delete mOutputs.valueAt(index);
+ mOutputs.removeItem(output);
+ mTestOutputs[testIndex] = 0;
+ }
+ return;
+ }
+#endif //AUDIO_POLICY_TEST
+
+ AudioOutputDescriptor *desc = mOutputs.valueAt(index);
+ if (desc->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
+ if ((desc->mDirectOpenCount <= 0) && !(desc->mFlags & AUDIO_OUTPUT_FLAG_LPA || desc->mFlags & AUDIO_OUTPUT_FLAG_TUNNEL ||
+ desc->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX)) {
+ ALOGW("releaseOutput() invalid open count %d for output %d",
+ desc->mDirectOpenCount, output);
+ return;
+ }
+ if ((--desc->mDirectOpenCount == 0) || ((desc->mFlags & AUDIO_OUTPUT_FLAG_LPA || desc->mFlags & AUDIO_OUTPUT_FLAG_TUNNEL ||
+ desc->mFlags & AUDIO_OUTPUT_FLAG_VOIP_RX))) {
+ ALOGV("releaseOutput() closing output");
+ closeOutput(output);
+ }
+ }
+
+}
+
+uint32_t AudioPolicyManager::checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs)
+{
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ if (outputDesc->isDuplicated()) {
+ return 0;
+ }
+
+ uint32_t muteWaitMs = 0;
+ audio_devices_t device = outputDesc->device();
+ bool shouldMute = (outputDesc->isActive()) &&
+ (AudioSystem::popCount(device) >= 2);
+ // temporary mute output if device selection changes to avoid volume bursts due to
+ // different per device volumes
+ bool tempMute = (outputDesc->isActive()) && (device != prevDevice);
+
+ for (size_t i = 0; i < NUM_STRATEGIES; i++) {
+ audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+ bool doMute = false;
+
+ if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = true;
+ } else if (!mute && outputDesc->mStrategyMutedByDevice[i]){
+ doMute = true;
+ outputDesc->mStrategyMutedByDevice[i] = false;
+ }
+ if (doMute || tempMute) {
+ for (size_t j = 0; j < mOutputs.size(); j++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(j);
+ if ((desc->supportedDevices() & outputDesc->supportedDevices())
+ == AUDIO_DEVICE_NONE) {
+ continue;
+ }
+ audio_io_handle_t curOutput = mOutputs.keyAt(j);
+ ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d",
+ mute ? "muting" : "unmuting", i, curDevice, curOutput);
+ setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs);
+ if (desc->isStrategyActive((routing_strategy)i)) {
+ // do tempMute only for current output
+ if (tempMute && (desc == outputDesc)) {
+ setStrategyMute((routing_strategy)i, true, curOutput);
+ setStrategyMute((routing_strategy)i, false, curOutput,
+ desc->latency() * ((desc->mFlags & AUDIO_OUTPUT_FLAG_LPA) ? 4 : 2),
+ device);
+ }
+ if ((tempMute && (desc == outputDesc)) || mute) {
+ if (muteWaitMs < desc->latency()) {
+ muteWaitMs = desc->latency();
+ }
+ }
+ }
+ }
+ }
+ }
+
+ // FIXME: should not need to double latency if volume could be applied immediately by the
+ // audioflinger mixer. We must account for the delay between now and the next time
+ // the audioflinger thread for this output will process a buffer (which corresponds to
+ // one buffer size, usually 1/2 or 1/4 of the latency).
+ muteWaitMs *= 2;
+ // wait for the PCM output buffers to empty before proceeding with the rest of the command
+ if (muteWaitMs > delayMs) {
+ muteWaitMs -= delayMs;
+ usleep(muteWaitMs * 1000);
+ return muteWaitMs;
+ }
+ return 0;
+}
+
+void AudioPolicyManager::setStreamMute(int stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs,
+ audio_devices_t device)
+{
+ StreamDescriptor &streamDesc = mStreams[stream];
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ if (device == AUDIO_DEVICE_NONE) {
+ device = outputDesc->device();
+ }
+
+ ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x",
+ stream, on, output, outputDesc->mMuteCount[stream], device);
+
+ if (on) {
+ if (outputDesc->mMuteCount[stream] > 0) {
+ ALOGV("setStreamMute() muting already muted stream!");
+ return;
+ }
+ if (outputDesc->mMuteCount[stream] == 0) {
+ if (streamDesc.mCanBeMuted &&
+ ((stream != AudioSystem::ENFORCED_AUDIBLE) ||
+ (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_NONE))) {
+ checkAndSetVolume(stream, 0, output, device, delayMs);
+ }
+ }
+ // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
+ outputDesc->mMuteCount[stream]++;
+ } else {
+ if (outputDesc->mMuteCount[stream] == 0) {
+ ALOGV("setStreamMute() unmuting non muted stream!");
+ return;
+ }
+ if (--outputDesc->mMuteCount[stream] == 0) {
+ checkAndSetVolume(stream,
+ streamDesc.getVolumeIndex(device),
+ output,
+ device,
+ delayMs);
+ }
+ }
+}
+
+
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
AudioSystem::device_connection_state state,
const char *device_address)
@@ -288,14 +244,15 @@
// handle output devices
if (audio_is_output_device(device)) {
- if (!mHasA2dp && audio_is_a2dp_device(device)) {
+ //Use QCOM's a2dp and usb audio solution, no need to check here
+ /*if (!mHasA2dp && audio_is_a2dp_device(device)) {
ALOGE("setDeviceConnectionState() invalid A2DP device: %x", device);
return BAD_VALUE;
}
if (!mHasUsb && audio_is_usb_device(device)) {
ALOGE("setDeviceConnectionState() invalid USB audio device: %x", device);
return BAD_VALUE;
- }
+ }*/
if (!mHasRemoteSubmix && audio_is_remote_submix_device((audio_devices_t)device)) {
ALOGE("setDeviceConnectionState() invalid remote submix audio device: %x", device);
return BAD_VALUE;
@@ -308,7 +265,6 @@
{
// handle output device connection
case AudioSystem::DEVICE_STATE_AVAILABLE:
-
if (mAvailableOutputDevices & device) {
ALOGW("setDeviceConnectionState() device already connected: %x", device);
return INVALID_OPERATION;
@@ -322,11 +278,20 @@
outputs.size());
// register new device as available
mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices | device);
- ALOGV("setDeviceConnectionState() connecting device %x", mAvailableOutputDevices);
+ if (audio_is_a2dp_device(device)) {
+ AudioParameter param;
+ param.add(String8("a2dp_connected"), String8("true"));
+ mpClientInterface->setParameters(0, param.toString());
+ }
+ if ( audio_is_usb_device(device)) {
+ AudioParameter param;
+ param.add(String8("usb_connected"), String8("true"));
+ mpClientInterface->setParameters(0, param.toString());
+ }
if (!outputs.isEmpty()) {
String8 paramStr;
- if (mHasA2dp && audio_is_a2dp_device(device)) {
+ if (audio_is_a2dp_device(device)) {
// handle A2DP device connection
AudioParameter param;
param.add(String8(AUDIO_PARAMETER_A2DP_SINK_ADDRESS), String8(device_address));
@@ -336,11 +301,13 @@
} else if (audio_is_bluetooth_sco_device(device)) {
// handle SCO device connection
mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
- } else if (mHasUsb && audio_is_usb_device(device)) {
+ } else if (audio_is_usb_device(device)) {
// handle USB device connection
mUsbCardAndDevice = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
paramStr = mUsbCardAndDevice;
}
+ // not currently handling multiple simultaneous submixes: ignoring remote submix
+ // case and address
if (!paramStr.isEmpty()) {
for (size_t i = 0; i < outputs.size(); i++) {
mpClientInterface->setParameters(outputs[i], paramStr);
@@ -360,49 +327,87 @@
mAvailableOutputDevices = (audio_devices_t)(mAvailableOutputDevices & ~device);
checkOutputsForDevice(device, state, outputs);
- if (mHasA2dp && audio_is_a2dp_device(device)) {
+ if (audio_is_a2dp_device(device)) {
// handle A2DP device disconnection
mA2dpDeviceAddress = "";
mA2dpSuspended = false;
+
+ AudioParameter param;
+ param.add(String8("a2dp_connected"), String8("false"));
+ mpClientInterface->setParameters(0, param.toString());
+
} else if (audio_is_bluetooth_sco_device(device)) {
// handle SCO device disconnection
mScoDeviceAddress = "";
- } else if (mHasUsb && audio_is_usb_device(device)) {
+ } else if (audio_is_usb_device(device)) {
// handle USB device disconnection
mUsbCardAndDevice = "";
+
+ AudioParameter param;
+ param.add(String8("usb_connected"), String8("false"));
+ mpClientInterface->setParameters(0, param.toString());
}
+ // not currently handling multiple simultaneous submixes: ignoring remote submix
+ // case and address
} break;
default:
ALOGE("setDeviceConnectionState() invalid state: %x", state);
return BAD_VALUE;
}
-#ifdef QCOM_FM_ENABLED
- audio_devices_t NewDevice = AudioPolicyManagerBase::getNewDevice(mPrimaryOutput, false /*fromCache*/);
- if (device == AUDIO_DEVICE_OUT_FM) {
- if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
- mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::FM, 1);
- }
- else {
- mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::FM, -1);
- }
- }
-#endif
checkA2dpSuspend();
- AudioPolicyManagerBase::checkOutputForAllStrategies();
+ checkOutputForAllStrategies();
// outputs must be closed after checkOutputForAllStrategies() is executed
if (!outputs.isEmpty()) {
for (size_t i = 0; i < outputs.size(); i++) {
// close unused outputs after device disconnection or direct outputs that have been
- if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE){
+ // opened by checkOutputsForDevice() to query dynamic parameters
+ if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE) {
closeOutput(outputs[i]);
}
}
}
updateDevicesAndOutputs();
- for (size_t i = 0; i < mOutputs.size(); i++) {
+#ifdef QCOM_PROXY_DEVICE_ENABLED
+ if (state == AudioSystem::DEVICE_STATE_AVAILABLE &&
+ audio_is_a2dp_device(device) &&
+ (mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY)) {
+ ALOGV("Delay the proxy device open");
+ return NO_ERROR;
+ }
+#endif
+
+ audio_devices_t newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+#ifdef QCOM_FM_ENABLED
+ if(device == AUDIO_DEVICE_OUT_FM) {
+ if (state == AudioSystem::DEVICE_STATE_AVAILABLE) {
+ ALOGV("setDeviceConnectionState() changeRefCount Inc");
+ mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::FM, 1);
+ newDevice = (audio_devices_t)(AudioPolicyManagerBase::getNewDevice(mPrimaryOutput, false) | AUDIO_DEVICE_OUT_FM);
+ }
+ else {
+ ALOGV("setDeviceConnectionState() changeRefCount Dec");
+ mOutputs.valueFor(mPrimaryOutput)->changeRefCount(AudioSystem::FM, -1);
+ }
+
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyHandleFm), (int)newDevice);
+ ALOGV("setDeviceConnectionState() setParameters handle_fm");
+ mpClientInterface->setParameters(mPrimaryOutput, param.toString());
+ }
+#endif
+ for (int i = mOutputs.size() -1; i >= 0; i--) {
+ audio_devices_t newDevice = getNewDevice(mOutputs.keyAt(i), true /*fromCache*/);
+#ifdef QCOM_ANC_HEADSET_ENABLED
+ if(device == AUDIO_DEVICE_OUT_ANC_HEADPHONE ||
+ device == AUDIO_DEVICE_OUT_ANC_HEADSET) {
+ if(newDevice == 0){
+ newDevice = getDeviceForStrategy(STRATEGY_MEDIA, false);
+ }
+ }
+#endif
setOutputDevice(mOutputs.keyAt(i),
getNewDevice(mOutputs.keyAt(i), true /*fromCache*/),
true,
@@ -415,6 +420,12 @@
device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
device == AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if(device == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET){
+ device = AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET;
+#ifdef QCOM_ANC_HEADSET_ENABLED
+ } else if(device == AUDIO_DEVICE_OUT_ANC_HEADSET){
+ device = AUDIO_DEVICE_IN_ANC_HEADSET; //wait for actual ANC device
+#endif
} else {
return NO_ERROR;
}
@@ -477,7 +488,7 @@
if (audio_is_output_device(device)) {
if (device & mAvailableOutputDevices) {
if (audio_is_a2dp_device(device) &&
- (!mHasA2dp || (address != "" && mA2dpDeviceAddress != address))) {
+ ((address != "" && mA2dpDeviceAddress != address))) {
return state;
}
if (audio_is_bluetooth_sco_device(device) &&
@@ -485,7 +496,7 @@
return state;
}
if (audio_is_usb_device(device) &&
- (!mHasUsb || (address != "" && mUsbCardAndDevice != address))) {
+ ((address != "" && mUsbCardAndDevice != address))) {
ALOGE("getDeviceConnectionState() invalid device: %x", device);
return state;
}
@@ -503,6 +514,131 @@
return state;
}
+void AudioPolicyManager::setPhoneState(int state)
+{
+ ALOGV("setPhoneState() state %d", state);
+ audio_devices_t newDevice = AUDIO_DEVICE_NONE;
+ if (state < 0 || state >= AudioSystem::NUM_MODES) {
+ ALOGW("setPhoneState() invalid state %d", state);
+ return;
+ }
+
+ if (state == mPhoneState ) {
+ ALOGW("setPhoneState() setting same state %d", state);
+ return;
+ }
+
+ // if leaving call state, handle special case of active streams
+ // pertaining to sonification strategy see handleIncallSonification()
+ if (isInCall()) {
+ ALOGV("setPhoneState() in call state management: new state is %d", state);
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, false, true);
+ }
+ }
+
+ // store previous phone state for management of sonification strategy below
+ int oldState = mPhoneState;
+ mPhoneState = state;
+ bool force = false;
+
+ // are we entering or starting a call
+ if (!isStateInCall(oldState) && isStateInCall(state)) {
+ ALOGV(" Entering call in setPhoneState()");
+ // force routing command to audio hardware when starting a call
+ // even if no device change is needed
+ force = true;
+ } else if (isStateInCall(oldState) && !isStateInCall(state)) {
+ ALOGV(" Exiting call in setPhoneState()");
+ // force routing command to audio hardware when exiting a call
+ // even if no device change is needed
+ force = true;
+ } else if (isStateInCall(state) && (state != oldState)) {
+ ALOGV(" Switching between telephony and VoIP in setPhoneState()");
+ // force routing command to audio hardware when switching between telephony and VoIP
+ // even if no device change is needed
+ force = true;
+ }
+
+ // check for device and output changes triggered by new phone state
+ // Need to update A2DP suspend first then getNewDevice(from cache)
+ checkA2dpSuspend();
+ checkOutputForAllStrategies();
+ updateDevicesAndOutputs();
+ newDevice = getNewDevice(mPrimaryOutput, false /*fromCache*/);
+
+ AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+
+ // force routing command to audio hardware when ending call
+ // even if no device change is needed
+ if (isStateInCall(oldState) && newDevice == AUDIO_DEVICE_NONE) {
+ newDevice = hwOutputDesc->device();
+ }
+
+ int delayMs = 0;
+ if (state == AUDIO_MODE_IN_CALL &&
+#ifdef QCOM_CSDCLIENT_ENABLED
+ platform_is_Fusion3() &&
+#endif
+ oldState == AUDIO_MODE_RINGTONE) {
+ delayMs = 40;
+ }
+
+ if (isStateInCall(state)) {
+ nsecs_t sysTime = systemTime();
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ // mute media and sonification strategies and delay device switch by the largest
+ // latency of any output where either strategy is active.
+ // This avoid sending the ring tone or music tail into the earpiece or headset.
+ if ((desc->isStrategyActive(STRATEGY_MEDIA,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime) ||
+ desc->isStrategyActive(STRATEGY_SONIFICATION,
+ SONIFICATION_HEADSET_MUSIC_DELAY,
+ sysTime)) &&
+ (delayMs < (int)desc->mLatency*2)) {
+ delayMs = desc->mLatency*2;
+ }
+ setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/));
+ setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i));
+ setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS,
+ getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/));
+ }
+ }
+
+ // change routing is necessary
+ setOutputDevice(mPrimaryOutput, newDevice, force, delayMs);
+
+ //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) {
+ newDevice = getNewDevice(output, false /*fromCache*/);
+ setOutputDevice(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 (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ handleIncallSonification(stream, true, true);
+ }
+ }
+
+ // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
+ if (state == AudioSystem::MODE_RINGTONE &&
+ isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
+ mLimitRingtoneVolume = true;
+ } else {
+ mLimitRingtoneVolume = false;
+ }
+}
+
void AudioPolicyManager::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
{
ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
@@ -523,7 +659,7 @@
config != AudioSystem::FORCE_WIRED_ACCESSORY &&
config != AudioSystem::FORCE_ANALOG_DOCK &&
config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE &&
- config != AudioSystem::FORCE_NO_BT_A2DP) {
+ config != AudioSystem::FORCE_NO_BT_A2DP && config != AudioSystem::FORCE_SPEAKER) {
ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
return;
}
@@ -565,7 +701,7 @@
checkA2dpSuspend();
checkOutputForAllStrategies();
updateDevicesAndOutputs();
- for (size_t i = 0; i < mOutputs.size(); i++) {
+ for (int i = mOutputs.size() -1; i >= 0; i--) {
audio_io_handle_t output = mOutputs.keyAt(i);
audio_devices_t newDevice = getNewDevice(output, true /*fromCache*/);
setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE));
@@ -590,89 +726,476 @@
}
-
-AudioPolicyManagerBase::IOProfile *AudioPolicyManager::getProfileForDirectOutput(
- audio_devices_t device,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channelMask,
- audio_output_flags_t flags)
+audio_io_handle_t AudioPolicyManager::getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ AudioSystem::output_flags flags)
{
+ audio_io_handle_t output = 0;
+ uint32_t latency = 0;
+ routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
+ audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ ALOGV("getOutput() stream %d, samplingRate %d, format %d, channelMask %x, flags %x",
+ stream, samplingRate, format, channelMask, flags);
- for (size_t i = 0; i < mHwModules.size(); i++) {
- if (mHwModules[i]->mHandle == 0) {
- continue;
+#ifdef AUDIO_POLICY_TEST
+ if (mCurOutput != 0) {
+ ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
+ mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
+
+ if (mTestOutputs[mCurOutput] == 0) {
+ ALOGV("getOutput() opening test output");
+ AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(NULL);
+ outputDesc->mDevice = mTestDevice;
+ outputDesc->mSamplingRate = mTestSamplingRate;
+ outputDesc->mFormat = mTestFormat;
+ outputDesc->mChannelMask = mTestChannels;
+ outputDesc->mLatency = mTestLatencyMs;
+ outputDesc->mFlags = (audio_output_flags_t)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
+ outputDesc->mRefCount[stream] = 0;
+ mTestOutputs[mCurOutput] = mpClientInterface->openOutput(0, &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+ if (mTestOutputs[mCurOutput]) {
+ AudioParameter outputCmd = AudioParameter();
+ outputCmd.addInt(String8("set_id"),mCurOutput);
+ mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
+ addOutput(mTestOutputs[mCurOutput], outputDesc);
+ }
}
- for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
- AudioPolicyManagerBase::IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
- if (isCompatibleProfile(profile, device, samplingRate, format,
- channelMask,
- (audio_output_flags_t)(flags|AUDIO_OUTPUT_FLAG_DIRECT))) {
- if (mAvailableOutputDevices & profile->mSupportedDevices) {
- return mHwModules[i]->mOutputProfiles[j];
- }
- }
+ return mTestOutputs[mCurOutput];
+ }
+#endif //AUDIO_POLICY_TEST
+
+ // open a direct output if required by specified parameters
+ IOProfile *profile = getProfileForDirectOutput(device,
+ samplingRate,
+ format,
+ channelMask,
+ (audio_output_flags_t)flags);
+ if (profile != NULL) {
+ AudioOutputDescriptor *outputDesc = NULL;
+
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (!desc->isDuplicated() && (profile == desc->mProfile)) {
+ outputDesc = desc;
+ // reuse direct output if currently open and configured with same parameters
+ if ((samplingRate == outputDesc->mSamplingRate) &&
+ (format == outputDesc->mFormat) &&
+ (channelMask == outputDesc->mChannelMask)) {
+ outputDesc->mDirectOpenCount++;
+ ALOGV("getOutput() reusing direct output %d", output);
+ return mOutputs.keyAt(i);
+ }
+ }
+ }
+
+ // close direct output if currently open and configured with different parameters
+ if (outputDesc != NULL) {
+ closeOutput(outputDesc->mId);
+ }
+ outputDesc = new AudioOutputDescriptor(profile);
+ outputDesc->mDevice = device;
+ outputDesc->mSamplingRate = samplingRate;
+ outputDesc->mFormat = (audio_format_t)format;
+ outputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
+ outputDesc->mLatency = 0;
+ outputDesc->mFlags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);;
+ outputDesc->mRefCount[stream] = 0;
+ outputDesc->mStopTime[stream] = 0;
+ outputDesc->mDirectOpenCount = 1;
+ output = mpClientInterface->openOutput(profile->mModule->mHandle,
+ &outputDesc->mDevice,
+ &outputDesc->mSamplingRate,
+ &outputDesc->mFormat,
+ &outputDesc->mChannelMask,
+ &outputDesc->mLatency,
+ outputDesc->mFlags);
+
+ // only accept an output with the requested parameters
+ if (output == 0 ||
+ (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
+ (format != 0 && format != outputDesc->mFormat) ||
+ (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+ ALOGV("getOutput() failed opening direct output: output %d samplingRate %d %d,"
+ "format %d %d, channelMask %04x %04x", output, samplingRate,
+ outputDesc->mSamplingRate, format, outputDesc->mFormat, channelMask,
+ outputDesc->mChannelMask);
+ if (output != 0) {
+ mpClientInterface->closeOutput(output);
+ }
+ delete outputDesc;
+ return 0;
+ }
+ addOutput(output, outputDesc);
+ mPreviousOutputs = mOutputs;
+ ALOGV("getOutput() returns new direct output %d", output);
+ return output;
+ }
+
+ // ignoring channel mask due to downmix capability in mixer
+
+ // open a non direct output
+
+ // get which output is suitable for the specified stream. The actual routing change will happen
+ // when startOutput() will be called
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+
+ output = selectOutput(outputs, flags);
+
+ ALOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d,"
+ "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags);
+
+ ALOGV("getOutput() returns output %d", output);
+
+ return output;
+}
+status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session)
+{
+ ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("startOutput() unknow output %d", output);
+ return BAD_VALUE;
+ }
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // increment usage count for this stream on the requested output:
+ // NOTE that the usage count is the same for duplicated output and hardware output which is
+ // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
+ outputDesc->changeRefCount(stream, 1);
+
+ if (outputDesc->mRefCount[stream] == 1) {
+ audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+ routing_strategy strategy = getStrategy(stream);
+ bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
+ (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
+ uint32_t waitMs = 0;
+ uint32_t muteWaitMs = 0;
+ bool force = false;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (desc != outputDesc) {
+ // force a device change if any other output is managed by the same hw
+ // module and has a current device selection that differs from selected device.
+ // In this case, the audio HAL must receive the new device selection so that it can
+ // change the device currently selected by the other active output.
+ if (outputDesc->sharesHwModuleWith(desc) &&
+ desc->device() != newDevice) {
+ force = true;
+ }
+ // wait for audio on other active outputs to be presented when starting
+ // a notification so that audio focus effect can propagate.
+ uint32_t latency = desc->latency();
+ if (shouldWait && desc->isActive(latency * 2) && (waitMs < latency)) {
+ waitMs = latency;
+ }
+ }
+ }
+
+#ifdef QCOM_FM_ENABLED
+ if(stream == AudioSystem::FM && output == getA2dpOutput()) {
+ muteWaitMs = setOutputDevice(output, newDevice, true);
+ } else
+#endif
+ {
+ muteWaitMs = setOutputDevice(output, newDevice, force);
+ }
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false);
+ }
+
+ // apply volume rules for current stream and device if necessary
+ checkAndSetVolume(stream,
+ mStreams[stream].getVolumeIndex(newDevice),
+ output,
+ newDevice);
+
+ // update the outputs if starting an output with a stream that can affect notification
+ // routing
+ handleNotificationRoutingForStream(stream);
+ if (waitMs > muteWaitMs) {
+ usleep((waitMs - muteWaitMs) * 2 * 1000);
+ }
+ } else {
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, true, false);
}
}
- return 0;
+ return NO_ERROR;
}
-bool AudioPolicyManager::isCompatibleProfile(AudioPolicyManagerBase::IOProfile *profile,
- audio_devices_t device,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channelMask,
- audio_output_flags_t flags)
+status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session)
{
- if ((profile->mSupportedDevices & device) != device) {
- return false;
+ ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index < 0) {
+ ALOGW("stopOutput() unknow output %d", output);
+ return BAD_VALUE;
}
- if (profile->mFlags != flags) {
- return false;
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+
+ // handle special case for sonification while in call
+ if (isInCall()) {
+ handleIncallSonification(stream, false, false);
}
- if (samplingRate != 0) {
- size_t i;
- for (i = 0; i < profile->mSamplingRates.size(); i++)
- {
- if (profile->mSamplingRates[i] == samplingRate) {
- break;
+
+ if (outputDesc->mRefCount[stream] > 0) {
+ // decrement usage count of this stream on the output
+ outputDesc->changeRefCount(stream, -1);
+ // store time at which the stream was stopped - see isStreamActive()
+ if (outputDesc->mRefCount[stream] == 0) {
+ outputDesc->mStopTime[stream] = systemTime();
+
+ if ((outputDesc->mRefCount[AUDIO_STREAM_RING]!= 0) && (stream == AUDIO_STREAM_VOICE_CALL)) {
+ // When AUDIO_STREAM_RING is present, Send Mute on RING
+ // if it gets stopOutput on AUDIO_STREAM_VOICE_CALL
+ setStreamMute(AudioSystem::RING, true, mPrimaryOutput);
}
- }
- if (i == profile->mSamplingRates.size()) {
- return false;
- }
- }
- if (format != 0) {
- size_t i;
- for (i = 0; i < profile->mFormats.size(); i++)
- {
- if (profile->mFormats[i] == format) {
- break;
+
+ audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
+ // delay the device switch by twice the latency because stopOutput() is executed when
+ // the track stop() command is received and at that time the audio track buffer can
+ // still contain data that needs to be drained. The latency only covers the audio HAL
+ // and kernel buffers. Also the latency does not always include additional delay in the
+ // audio path (audio DSP, CODEC ...)
+ setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+
+ // force restoring the device selection on other active outputs if it differs from the
+ // one being selected for this output
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if (curOutput != output &&
+ desc->isActive() &&
+ outputDesc->sharesHwModuleWith(desc) &&
+ newDevice != desc->device()) {
+ setOutputDevice(curOutput,
+ getNewDevice(curOutput, false /*fromCache*/),
+ true,
+ outputDesc->mLatency*2);
+ }
}
+ // update the outputs if stopping one with a stream that can affect notification routing
+ handleNotificationRoutingForStream(stream);
}
- if (i == profile->mFormats.size()) {
- return false;
- }
+ return NO_ERROR;
+ } else {
+ ALOGW("stopOutput() refcount is already 0 for output %d", output);
+ return INVALID_OPERATION;
}
- if (channelMask != 0) {
- size_t i;
- for (i = 0; i < profile->mChannelMasks.size(); i++)
- {
- if (profile->mChannelMasks[i] == channelMask) {
- break;
- }
- }
- if (i == profile->mChannelMasks.size()) {
- return false;
- }
- }
- ALOGD(" profile found: device %x, flags %x, samplingrate %d,\
- format %x, channelMask %d",
- device, flags, samplingRate, format, channelMask);
- return true;
}
+
+audio_io_handle_t AudioPolicyManager::getInput(int inputSource,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ AudioSystem::audio_in_acoustics acoustics)
+{
+ audio_io_handle_t input = 0;
+ audio_devices_t device = getDeviceForInputSource(inputSource);
+
+ ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
+ inputSource, samplingRate, format, channelMask, acoustics);
+
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGW("getInput() could not find device for inputSource %d", inputSource);
+ return 0;
+ }
+
+ // adapt channel selection to input source
+ switch(inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ channelMask |= AudioSystem::CHANNEL_IN_VOICE_UPLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ channelMask |= AudioSystem::CHANNEL_IN_VOICE_DNLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ channelMask |= (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
+ break;
+ default:
+ break;
+ }
+
+ IOProfile *profile = getInputProfile(device,
+ samplingRate,
+ format,
+ channelMask);
+ if (profile == NULL) {
+ ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d,"
+ "channelMask %04x",
+ device, samplingRate, format, channelMask);
+ return 0;
+ }
+
+ if (profile->mModule->mHandle == 0) {
+ ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
+ return 0;
+ }
+
+ AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
+
+ inputDesc->mInputSource = inputSource;
+ inputDesc->mDevice = device;
+ inputDesc->mSamplingRate = samplingRate;
+ inputDesc->mFormat = (audio_format_t)format;
+ inputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
+ inputDesc->mRefCount = 0;
+ input = mpClientInterface->openInput(profile->mModule->mHandle,
+ &inputDesc->mDevice,
+ &inputDesc->mSamplingRate,
+ &inputDesc->mFormat,
+ &inputDesc->mChannelMask);
+
+ // only accept input with the exact requested set of parameters
+ if (input == 0 ||
+ (samplingRate != inputDesc->mSamplingRate) ||
+ (format != inputDesc->mFormat) ||
+ (channelMask != inputDesc->mChannelMask)) {
+ ALOGV("getInput() failed opening input: samplingRate %d, format %d, channelMask %d",
+ samplingRate, format, channelMask);
+ if (input != 0) {
+ mpClientInterface->closeInput(input);
+ }
+ delete inputDesc;
+ return 0;
+ }
+ mInputs.add(input, inputDesc);
+ return input;
+}
+
+/*
+Overwriting this function from base class to allow 2 acitve AudioRecord clients in case of FM.
+One for FM A2DP playbck and other for FM recording.
+*/
+status_t AudioPolicyManager::startInput(audio_io_handle_t input)
+{
+ ALOGV("startInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("startInput() unknow input %d", input);
+ return BAD_VALUE;
+ }
+ AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+/*
+#ifdef AUDIO_POLICY_TEST
+ if (mTestInput == 0)
+#endif //AUDIO_POLICY_TEST
+ {
+ // refuse 2 active AudioRecord clients at the same time
+ if (getActiveInput() != 0) {
+ ALOGW("startInput() input %d failed: other input already started", input);
+ return INVALID_OPERATION;
+ }
+ }
+*/
+ audio_devices_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
+ if ((newDevice != AUDIO_DEVICE_NONE) && (newDevice != inputDesc->mDevice)) {
+ inputDesc->mDevice = newDevice;
+ }
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
+
+ param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource);
+
+ // use Voice Recognition mode or not for this input based on input source
+ int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0;
+ param.addInt(String8("vr_mode"), vr_enabled);
+ ALOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled);
+
+ //to pass on if camcorder mode is enabled to HAL
+ int camcorder_enabled = inputDesc->mInputSource == AUDIO_SOURCE_CAMCORDER ? 1 : 0;
+ param.addInt(String8("camcorder_mode"), camcorder_enabled);
+
+ mpClientInterface->setParameters(input, param.toString());
+
+ inputDesc->mRefCount = 1;
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::stopInput(audio_io_handle_t input)
+{
+ ALOGV("stopInput() input %d", input);
+ ssize_t index = mInputs.indexOfKey(input);
+ if (index < 0) {
+ ALOGW("stopInput() unknow input %d", input);
+ return BAD_VALUE;
+ }
+ AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
+
+ if (inputDesc->mRefCount == 0) {
+ ALOGW("stopInput() input %d already stopped", input);
+ return INVALID_OPERATION;
+ } else {
+ AudioParameter param = AudioParameter();
+ param.addInt(String8(AudioParameter::keyRouting), 0);
+ mpClientInterface->setParameters(input, param.toString());
+ setOutputDevice(mPrimaryOutput, getNewDevice(mPrimaryOutput, true), true);
+ inputDesc->mRefCount = 0;
+ return NO_ERROR;
+ }
+}
+
+
+status_t AudioPolicyManager::setStreamVolumeIndex(AudioSystem::stream_type stream,
+ int index,
+ audio_devices_t device)
+{
+
+ if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
+ return BAD_VALUE;
+ }
+ if (!audio_is_output_device(device)) {
+ return BAD_VALUE;
+ }
+
+ // Force max volume if stream cannot be muted
+ if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
+
+ ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d",
+ stream, device, index);
+
+ // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and
+ // clear all device specific values
+ if ((device == AUDIO_DEVICE_OUT_DEFAULT) && (AUDIO_DEVICE_OUT_DEFAULT != AUDIO_DEVICE_OUT_SPEAKER)) {
+ mStreams[stream].mIndexCur.clear();
+ }
+ mStreams[stream].mIndexCur.add(device, index);
+
+ // compute and apply stream volume on all outputs according to connected device
+ status_t status = NO_ERROR;
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_devices_t curDevice =
+ getDeviceForVolume(mOutputs.valueAt(i)->device());
+ if (device == curDevice) {
+ status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
+ if (volStatus != NO_ERROR) {
+ status = volStatus;
+ }
+ }
+ }
+ return status;
+}
+
+
status_t AudioPolicyManager::checkOutputsForDevice(audio_devices_t device,
AudioSystem::device_connection_state state,
SortedVector<audio_io_handle_t>& outputs)
@@ -781,6 +1304,7 @@
((profile->mFormats[0] == 0) &&
(profile->mChannelMasks.size() < 2))) {
ALOGW("checkOutputsForDevice() direct output missing param");
+ mpClientInterface->closeOutput(output);
output = 0;
} else {
addOutput(output, desc);
@@ -872,157 +1396,341 @@
}
return NO_ERROR;
}
-status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
- int session)
+
+audio_devices_t AudioPolicyManager::getNewDevice(audio_io_handle_t output, bool fromCache)
{
- ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("startOutput() unknow output %d", output);
- return BAD_VALUE;
+ audio_devices_t device = AUDIO_DEVICE_NONE;
+
+ AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
+ AudioOutputDescriptor *primaryOutputDesc = mOutputs.valueFor(mPrimaryOutput);
+ // check the following by order of priority to request a routing change if necessary:
+ // 1: the strategy enforced audible is active on the output:
+ // use device for strategy enforced audible
+ // 2: we are in call or the strategy phone is active on the output:
+ // use device for strategy phone
+ // 3: the strategy sonification is active on the output:
+ // use device for strategy sonification
+ // 4: the strategy "respectful" sonification is active on the output:
+ // use device for strategy "respectful" sonification
+ // 5: the strategy media is active on the output:
+ // use device for strategy media
+ // 6: the strategy DTMF is active on the output:
+ // use device for strategy DTMF
+ if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) {
+ device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ } else if (isInCall() ||
+ outputDesc->isStrategyActive(STRATEGY_PHONE)) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) {
+ device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) {
+ device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
}
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ ALOGV("getNewDevice() selected device %x", device);
+ return device;
+}
- // increment usage count for this stream on the requested output:
- // NOTE that the usage count is the same for duplicated output and hardware output which is
- // necessary for a correct control of hardware output routing by startOutput() and stopOutput()
- outputDesc->changeRefCount(stream, 1);
-
- if (outputDesc->mRefCount[stream] == 1) {
- audio_devices_t prevDevice = outputDesc->device();
- audio_devices_t newDevice = AudioPolicyManagerBase::getNewDevice(output, false /*fromCache*/);
- routing_strategy strategy = AudioPolicyManagerBase::getStrategy(stream);
- bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
- (strategy == STRATEGY_SONIFICATION_RESPECTFUL);
- uint32_t waitMs = 0;
- bool force = false;
- uint32_t muteWaitMs;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (desc != outputDesc) {
- // force a device change if any other output is managed by the same hw
- // module and has a current device selection that differs from selected device.
- // In this case, the audio HAL must receive the new device selection so that it can
- // change the device currently selected by the other active output.
- if (outputDesc->sharesHwModuleWith(desc) &&
- desc->device() != newDevice) {
- force = true;
- }
- // wait for audio on other active outputs to be presented when starting
- // a notification so that audio focus effect can propagate.
- if (shouldWait && (desc->refCount() != 0) && (waitMs < desc->latency())) {
- waitMs = desc->latency();
- }
- }
- }
+AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
+ AudioSystem::stream_type stream) {
+ // stream to strategy mapping
+ switch (stream) {
+ case AudioSystem::VOICE_CALL:
+ case AudioSystem::BLUETOOTH_SCO:
+ return STRATEGY_PHONE;
+ case AudioSystem::RING:
+ case AudioSystem::ALARM:
+ return STRATEGY_SONIFICATION;
+ case AudioSystem::NOTIFICATION:
+ return STRATEGY_SONIFICATION_RESPECTFUL;
+ case AudioSystem::DTMF:
+ return STRATEGY_DTMF;
+ default:
+ ALOGE("unknown stream type");
+ case AudioSystem::SYSTEM:
+ // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
+ // while key clicks are played produces a poor result
+ case AudioSystem::TTS:
+ case AudioSystem::MUSIC:
#ifdef QCOM_FM_ENABLED
- if(stream == AudioSystem::FM && output == getA2dpOutput()) {
- setOutputDevice(output, AudioPolicyManagerBase::getNewDevice((output), true));
- } else
+ case AudioSystem::FM:
#endif
- {
- setOutputDevice(output, AudioPolicyManagerBase::getNewDevice((output), true));
+ return STRATEGY_MEDIA;
+ case AudioSystem::ENFORCED_AUDIBLE:
+ return STRATEGY_ENFORCED_AUDIBLE;
}
-
- // handle special case for sonification while in call
- if (isInCall()) {
- AudioPolicyManagerBase::handleIncallSonification(stream, true, false);
- }
-
- // apply volume rules for current stream and device if necessary
- checkAndSetVolume(stream,
- mStreams[stream].getVolumeIndex(newDevice),
- output,
- newDevice);
-
- // update the outputs if starting an output with a stream that can affect notification
- // routing
- handleNotificationRoutingForStream(stream);
- if (waitMs > muteWaitMs) {
- usleep((waitMs - muteWaitMs) * 2 * 1000);
- }
- }
- return NO_ERROR;
}
-status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session)
+audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache)
{
- ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- ALOGW("stopOutput() unknow output %d", output);
- return BAD_VALUE;
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ if (fromCache) {
+ ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
+ strategy, mDeviceForStrategy[strategy]);
+ return mDeviceForStrategy[strategy];
}
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
+ switch (strategy) {
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, false, false);
- }
+ case STRATEGY_SONIFICATION_RESPECTFUL:
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActiveRemotely(AudioSystem::MUSIC,
+ SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing on a remote device, use the the sonification behavior.
+ // Note that we test this usecase before testing if media is playing because
+ // the isStreamActive() method only informs about the activity of a stream, not
+ // if it's for local playback. Note also that we use the same delay between both tests
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ } else if (isStreamActive(AudioSystem::MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
+ // while media is playing (or has recently played), use the same device
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ } else {
+ // when media is not playing anymore, fall back on the sonification behavior
+ device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/);
+ }
- 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) {
- if (stream == AudioSystem::MUSIC) {
- outputDesc->mStopTime[stream] = systemTime();
+ break;
+
+ case STRATEGY_DTMF:
+ if (!isInCall()) {
+ // when off call, DTMF strategy follows the same rules as MEDIA strategy
+ device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ break;
+ }
+ // when in call, DTMF and PHONE strategies follow the same rules
+ // FALL THROUGH
+
+ case STRATEGY_PHONE:
+ // for phone strategy, we first consider the forced use and then the available devices by order
+ // of priority
+ switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
+ case AudioSystem::FORCE_BT_SCO:
+ if (!isInCall() || strategy != STRATEGY_DTMF) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
+ if (device) break;
}
- audio_devices_t newDevice = getNewDevice(output, false /*fromCache*/);
- // delay the device switch by twice the latency because stopOutput() is executed when
- // the track stop() command is received and at that time the audio track buffer can
- // still contain data that needs to be drained. The latency only covers the audio HAL
- // and kernel buffers. Also the latency does not always include additional delay in the
- // audio path (audio DSP, CODEC ...)
- setOutputDevice(output, newDevice, false, outputDesc->mLatency*2);
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
+ if (device) break;
+ // if SCO device is requested but no SCO device is available, fall back to default case
+ // FALL THROUGH
- // force restoring the device selection on other active outputs if it differs from the
- // one being selected for this output
- for (size_t i = 0; i < mOutputs.size(); i++) {
- audio_io_handle_t curOutput = mOutputs.keyAt(i);
- AudioOutputDescriptor *desc = mOutputs.valueAt(i);
- if (curOutput != output &&
- desc->refCount() != 0 &&
- outputDesc->sharesHwModuleWith(desc) &&
- newDevice != desc->device()) {
- setOutputDevice(curOutput,
- getNewDevice(curOutput, false /*fromCache*/),
- true,
- outputDesc->mLatency*2);
+ default: // FORCE_NONE
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
+ if (!isInCall())
+ {
+ if ((mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+ !mA2dpSuspended) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ if (device) break;
}
}
- // update the outputs if stopping one with a stream that can affect notification routing
- handleNotificationRoutingForStream(stream);
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ if (device) break;
+#ifdef QCOM_ANC_HEADSET_ENABLED
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADPHONE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADSET;
+ if (device) break;
+#endif
+ if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_EARPIECE;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE");
+ }
+ break;
+
+ case AudioSystem::FORCE_SPEAKER:
+ // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
+ // A2DP speaker when forcing to speaker output
+ if (!isInCall() &&
+ (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+ !mA2dpSuspended) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ if (device) break;
+ }
+ if (mPhoneState != AudioSystem::MODE_IN_CALL) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ if (device) break;
+ }
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ if (device) break;
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device) break;
+ device = mDefaultOutputDevice;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER");
+ }
+ break;
}
- return NO_ERROR;
- } else {
- ALOGW("stopOutput() refcount is already 0 for output %d", output);
- return INVALID_OPERATION;
+#if defined(QCOM_FM_ENABLED) || defined(STE_FM)
+ if (mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM) {
+ if (mForceUse[AudioSystem::FOR_MEDIA] == AudioSystem::FORCE_SPEAKER) {
+ device &= ~(AUDIO_DEVICE_OUT_WIRED_HEADSET);
+ device &= ~(AUDIO_DEVICE_OUT_WIRED_HEADPHONE);
+ device &= ~(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET);
+ device |= AUDIO_DEVICE_OUT_SPEAKER;
+ }
+ }
+#endif
+ break;
+
+ case STRATEGY_SONIFICATION:
+
+ // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
+ // handleIncallSonification().
+ if (isInCall()) {
+ device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+ break;
+ }
+ // FALL THROUGH
+
+ case STRATEGY_ENFORCED_AUDIBLE:
+ // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION
+ // except:
+ // - when in call where it doesn't default to STRATEGY_PHONE behavior
+ // - in countries where not enforced in which case it follows STRATEGY_MEDIA
+
+ if ((strategy == STRATEGY_SONIFICATION) ||
+ (mForceUse[AudioSystem::FOR_SYSTEM] == AudioSystem::FORCE_SYSTEM_ENFORCED)) {
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION");
+ }
+ }
+ // The second device used for sonification is the same as the device used by media strategy
+ // FALL THROUGH
+
+ case STRATEGY_MEDIA: {
+ uint32_t device2 = AUDIO_DEVICE_NONE;
+ if (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_SPEAKER) {
+ if (strategy != STRATEGY_SONIFICATION) {
+#ifdef QCOM_PROXY_DEVICE_ENABLED
+ // no sonification on WFD sink
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_PROXY;
+#else
+ // no sonification on remote submix (e.g. WFD)
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+#endif
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) &&
+ (mForceUse[AudioSystem::FOR_MEDIA] != AudioSystem::FORCE_NO_BT_A2DP) &&
+ !mA2dpSuspended) {
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
+ }
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ }
+#ifdef QCOM_ANC_HEADSET_ENABLED
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADPHONE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANC_HEADSET;
+ }
+#endif
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_ACCESSORY;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_USB_DEVICE;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
+ }
+ if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+ // no sonification on aux digital (e.g. HDMI)
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ }
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ }
+#if defined(QCOM_FM_ENABLED) || defined(STE_FM)
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_FM_TX;
+ }
+#endif
+ if (device2 == AUDIO_DEVICE_NONE) {
+ device2 = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
+ // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
+ device |= device2;
+ if (!device) {
+ device = mDefaultOutputDevice;
+ }
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA");
+ }
+
+ } else {
+ //AudioSystem::FORCE_SPEAKER for STRATEGY_MEDIA
+ device = mAvailableOutputDevices & AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ if (isInCall()) {
+ // when in call, get the device for Phone strategy
+ device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/);
+ }
+
+ } break;
+
+ default:
+ ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ break;
}
+
+ ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
+ return device;
}
-// ----------------------------------------------------------------------------
-// AudioPolicyManager
-// ----------------------------------------------------------------------------
+uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force,
+ int delayMs)
-// --- class factory
-
-
-extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
-{
- return new AudioPolicyManager(clientInterface);
-}
-
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
- delete interface;
-}
-
-// ---
-
-uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, audio_devices_t device, bool force, int delayMs)
{
ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs);
AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
@@ -1034,6 +1742,13 @@
muteWaitMs += setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
return muteWaitMs;
}
+ // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+ // output profile
+ if ((device != AUDIO_DEVICE_NONE) &&
+ ((device & outputDesc->mProfile->mSupportedDevices) == 0)) {
+ return 0;
+ }
+
// filter devices according to output selected
device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices);
@@ -1041,21 +1756,24 @@
ALOGV("setOutputDevice() prevDevice %04x", prevDevice);
- if (device != 0) {
+ if (device != AUDIO_DEVICE_NONE) {
outputDesc->mDevice = device;
}
muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
// Do not change the routing if:
- // - the requested device is 0
+ // - the requested device is AUDIO_DEVICE_NONE
// - the requested device is the same as current device and force is not specified.
// Doing this check here allows the caller to call setOutputDevice() without conditions
- if ((device == 0 || device == prevDevice) && !force) {
+ if ((device == AUDIO_DEVICE_NONE) || ((device == prevDevice) && !force)) {
ALOGV("setOutputDevice() setting same device %04x or null device for output %d", device, output);
return muteWaitMs;
}
-
- ALOGV("setOutputDevice() changing device");
+ if (device == prevDevice) {
+ ALOGV("setOutputDevice() Call routing with same device with zero delay ");
+ delayMs = 0;
+ }
+ ALOGV("setOutputDevice() changing device:%x",device);
// do the routing
param.addInt(String8(AudioParameter::keyRouting), (int)device);
mpClientInterface->setParameters(output, param.toString(), delayMs);
@@ -1065,7 +1783,160 @@
return muteWaitMs;
}
-status_t AudioPolicyManager::checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs, bool force)
+
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(int inputSource)
+{
+ uint32_t device = AUDIO_DEVICE_NONE;
+
+ switch (inputSource) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ break;
+ }
+ // FALL THROUGH
+
+ case AUDIO_SOURCE_DEFAULT:
+ case AUDIO_SOURCE_MIC:
+ case AUDIO_SOURCE_VOICE_RECOGNITION:
+ if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
+ mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+#ifdef QCOM_ANC_HEADSET_ENABLED
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_ANC_HEADSET) {
+ device = AUDIO_DEVICE_IN_ANC_HEADSET;
+#endif
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET) {
+ device = AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_COMMUNICATION:
+ device = AUDIO_DEVICE_IN_COMMUNICATION;
+ break;
+ case AUDIO_SOURCE_CAMCORDER:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
+ device = AUDIO_DEVICE_IN_BACK_MIC;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
+ device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
+ device = AUDIO_DEVICE_IN_WIRED_HEADSET;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_AUX_DIGITAL) {
+ device = AUDIO_DEVICE_IN_AUX_DIGITAL;
+ } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
+ device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ }
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ case AUDIO_SOURCE_VOICE_CALL:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
+ device = AUDIO_DEVICE_IN_VOICE_CALL;
+ }
+ break;
+ case AUDIO_SOURCE_REMOTE_SUBMIX:
+ if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+#if defined(QCOM_FM_ENABLED) || defined(STE_FM)
+ case AUDIO_SOURCE_FM_RX:
+ device = AUDIO_DEVICE_IN_FM_RX;
+ break;
+ case AUDIO_SOURCE_FM_RX_A2DP:
+ device = AUDIO_DEVICE_IN_FM_RX_A2DP;
+ break;
+#endif
+ default:
+ ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
+ break;
+ }
+ ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ return device;
+}
+
+
+
+
+audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
+{
+ if (device == AUDIO_DEVICE_NONE) {
+ // this happens when forcing a route update and no track is active on an output.
+ // In this case the returned category is not important.
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (AudioSystem::popCount(device) > 1) {
+ // Multiple device selection is either:
+ // - speaker + one other device: give priority to speaker in this case.
+ // - one A2DP device + another device: happens with duplicated output. In this case
+ // retain the device on the A2DP output as the other must not correspond to an active
+ // selection if not the speaker.
+ if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if ((device & AUDIO_DEVICE_OUT_WIRED_HEADSET) != 0) {
+ device = AUDIO_DEVICE_OUT_WIRED_HEADSET;
+ } else if ((device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE) != 0) {
+ device = AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ } else {
+ device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+ }
+ }
+
+ ALOGW_IF(AudioSystem::popCount(device) != 1,
+ "getDeviceForVolume() invalid device combination: %08x",
+ device);
+
+ return device;
+}
+
+AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
+{
+ switch(getDeviceForVolume(device)) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ return DEVICE_CATEGORY_EARPIECE;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_ANC_HEADSET:
+ case AUDIO_DEVICE_OUT_ANC_HEADPHONE:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+#if defined(QCOM_FM_ENABLED) || defined(STE_FM)
+ case AUDIO_DEVICE_OUT_FM:
+#endif
+ return DEVICE_CATEGORY_HEADSET;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+ case AUDIO_DEVICE_OUT_PROXY:
+ default:
+ return DEVICE_CATEGORY_SPEAKER;
+ }
+}
+
+bool AudioPolicyManager::isDirectOutput(audio_io_handle_t output) {
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ audio_io_handle_t curOutput = mOutputs.keyAt(i);
+ AudioOutputDescriptor *desc = mOutputs.valueAt(i);
+ if ((curOutput == output) && (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+status_t AudioPolicyManager::checkAndSetVolume(int stream,
+ int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs,
+ bool force)
{
// do not change actual stream volume if the stream is muted
if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
@@ -1088,26 +1959,15 @@
// - the force flag is set
if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
#ifdef QCOM_FM_ENABLED
- (stream == AudioSystem::FM) ||
+ (stream == AudioSystem::FM) ||
#endif
- force) {
+ force) {
mOutputs.valueFor(output)->mCurVolume[stream] = volume;
ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
- if (stream == AudioSystem::VOICE_CALL ||
- stream == AudioSystem::DTMF ||
- stream == AudioSystem::BLUETOOTH_SCO) {
- float voiceVolume = -1.0;
- // offset value to reflect actual hardware volume that never reaches 0
- // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
- volume = 0.01 + 0.99 * volume;
- if (stream == AudioSystem::VOICE_CALL) {
- voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
- } else if (stream == AudioSystem::BLUETOOTH_SCO) {
- voiceVolume = 1.0;
- }
- if (voiceVolume >= 0 && output == mPrimaryOutput) {
- mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
- }
+ // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
+ // enabled
+ if (stream == AudioSystem::BLUETOOTH_SCO) {
+ mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
#ifdef QCOM_FM_ENABLED
} else if (stream == AudioSystem::FM) {
float fmVolume = -1.0;
@@ -1123,180 +1983,84 @@
}
mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
}
+
+ if (stream == AudioSystem::VOICE_CALL ||
+ stream == AudioSystem::BLUETOOTH_SCO) {
+ float voiceVolume;
+ // Force voice volume to max for bluetooth SCO as volume is managed by the headset
+ if (stream == AudioSystem::VOICE_CALL) {
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+ } else {
+ voiceVolume = 1.0;
+ }
+
+ voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
+
+ // Force voice volume to max when Vgs is set for bluetooth SCO as volume is managed by the headset
+ if (stream == AudioSystem::BLUETOOTH_SCO) {
+ String8 key ("bt_headset_vgs");
+ mpClientInterface->getParameters(output,key);
+ AudioParameter result(mpClientInterface->getParameters(0,key));
+ int value;
+ if (result.getInt(String8("isVGS"),value) == NO_ERROR) {
+ ALOGV("Use BT-SCO Voice Volume");
+ voiceVolume = 1.0;
+ }
+ }
+
+ if (voiceVolume != mLastVoiceVolume && (output == mPrimaryOutput ||
+ isDirectOutput(output))) {
+ mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
+ mLastVoiceVolume = voiceVolume;
+ }
+ }
+
return NO_ERROR;
}
-audio_io_handle_t AudioPolicyManager::getInput(int inputSource,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channelMask,
- AudioSystem::audio_in_acoustics acoustics)
+
+void AudioPolicyManager::checkA2dpSuspend()
{
- audio_io_handle_t input = 0;
- audio_devices_t device = getDeviceForInputSource(inputSource);
- ALOGV("getInput() inputSource %d, samplingRate %d, format %d, channelMask %x, acoustics %x",
- inputSource, samplingRate, format, channelMask, acoustics);
+ // suspend A2DP output if:
+ // (NOT already suspended) &&
+ // ((SCO device is connected &&
+ // (forced usage for communication || for record is SCO))) ||
+ // (phone state is ringing || in call)
+ //
+ // restore A2DP output if:
+ // (Already suspended) &&
+ // ((SCO device is NOT connected ||
+ // (forced usage NOT for communication && NOT for record is SCO))) &&
+ // (phone state is NOT ringing && NOT in call)
+ //
+ if (mA2dpSuspended) {
+ if (((mScoDeviceAddress == "") ||
+ ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
+ (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) ||
+ ((mPhoneState != AudioSystem::MODE_IN_CALL) &&
+ (mPhoneState != AudioSystem::MODE_RINGTONE))) {
- if (device == AUDIO_DEVICE_NONE) {
- ALOGW("getInput() could not find device for inputSource %d", inputSource);
- return 0;
- }
-
- // adapt channel selection to input source
- switch(inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- channelMask |= AudioSystem::CHANNEL_IN_VOICE_UPLINK;
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- channelMask |= AudioSystem::CHANNEL_IN_VOICE_DNLINK;
- break;
- case AUDIO_SOURCE_VOICE_CALL:
- channelMask |= (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
- break;
- default:
- break;
- }
-
- IOProfile *profile = getInputProfile(device,
- samplingRate,
- format,
- channelMask);
- if (profile == NULL) {
- ALOGW("getInput() could not find profile for device %04x, samplingRate %d, format %d,"
- "channelMask %04x",
- device, samplingRate, format, channelMask);
- return 0;
- }
-
- if (profile->mModule->mHandle == 0) {
- ALOGE("getInput(): HW module %s not opened", profile->mModule->mName);
- return 0;
- }
-
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor(profile);
-
- inputDesc->mInputSource = inputSource;
- inputDesc->mDevice = device;
- inputDesc->mSamplingRate = samplingRate;
- inputDesc->mFormat = (audio_format_t)format;
- inputDesc->mChannelMask = (audio_channel_mask_t)channelMask;
- inputDesc->mRefCount = 0;
- input = mpClientInterface->openInput(profile->mModule->mHandle,
- &inputDesc->mDevice,
- &inputDesc->mSamplingRate,
- &inputDesc->mFormat,
- &inputDesc->mChannelMask);
-
- // only accept input with the exact requested set of parameters
- if (input == 0 ||
- (samplingRate != inputDesc->mSamplingRate) ||
- (format != inputDesc->mFormat) ||
- (channelMask != inputDesc->mChannelMask)) {
- ALOGV("getInput() failed opening input: samplingRate %d, format %d, channelMask %d",
- samplingRate, format, channelMask);
- if (input != 0) {
- mpClientInterface->closeInput(input);
+ mA2dpSuspended = false;
}
- delete inputDesc;
- return 0;
+ } else {
+ if (((mScoDeviceAddress != "") &&
+ ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
+ (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) &&
+ ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
+ (mPhoneState == AudioSystem::MODE_RINGTONE))) {
+
+ mA2dpSuspended = true;
+ }
}
- mInputs.add(input, inputDesc);
- return input;
}
-audio_devices_t AudioPolicyManager::getDeviceForInputSource(int inputSource)
+audio_io_handle_t AudioPolicyManager::getA2dpOutput()
{
- uint32_t device = AUDIO_DEVICE_NONE;
-
- switch(inputSource) {
- case AUDIO_SOURCE_DEFAULT:
- case AUDIO_SOURCE_MIC:
- case AUDIO_SOURCE_VOICE_RECOGNITION:
- if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
- mAvailableInputDevices & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_VOICE_COMMUNICATION:
- device = AUDIO_DEVICE_IN_COMMUNICATION;
- break;
-
- case AUDIO_SOURCE_CAMCORDER:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_BACK_MIC) {
- device = AUDIO_DEVICE_IN_BACK_MIC;
- } else if (mAvailableInputDevices & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_VOICE_UPLINK:
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- case AUDIO_SOURCE_VOICE_CALL:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- }
- break;
- case AUDIO_SOURCE_REMOTE_SUBMIX:
- if (mAvailableInputDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
- device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- }
- break;
-
-#ifdef QCOM_FM_ENABLED
- case AUDIO_SOURCE_FM_RX:
- device = AUDIO_DEVICE_IN_FM_RX;
- break;
- case AUDIO_SOURCE_FM_RX_A2DP:
- device = AUDIO_DEVICE_IN_FM_RX_A2DP;
- break;
-#endif
- default:
- ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
- break;
- }
- ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
- return device;
-}
-/*
-Overwriting this function from base class to allow 2 acitve AudioRecord clients in case of FM.
-One for FM A2DP playbck and other for FM recording.
-*/
-status_t AudioPolicyManager::startInput(audio_io_handle_t input)
-{
- ALOGV("startInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- ALOGW("startInput() unknow input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-#ifdef AUDIO_POLICY_TEST
- if (mTestInput == 0)
-#endif //AUDIO_POLICY_TEST
- {
- // refuse 2 active AudioRecord clients at the same time
- if (getActiveInput() != 0) {
- ALOGW("startInput() input %d failed: other input already started", input);
- return INVALID_OPERATION;
- }
- }
-
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
- // use Voice Recognition mode or not for this input based on input source
- int vr_enabled = inputDesc->mInputSource == AUDIO_SOURCE_VOICE_RECOGNITION ? 1 : 0;
- param.addInt(String8("vr_mode"), vr_enabled);
- ALOGV("AudioPolicyManager::startInput(%d), setting vr_mode to %d", inputDesc->mInputSource, vr_enabled);
- mpClientInterface->setParameters(input, param.toString());
- inputDesc->mRefCount = 1;
- return NO_ERROR;
+ return 0;
}
+//private function, no changes from AudioPolicyManagerBase
void AudioPolicyManager::handleNotificationRoutingForStream(AudioSystem::stream_type stream) {
switch(stream) {
case AudioSystem::MUSIC:
@@ -1308,4 +2072,113 @@
}
}
-}; // namespace android_audio_legacy
+
+AudioPolicyManagerBase::IOProfile *AudioPolicyManager::getProfileForDirectOutput(
+ audio_devices_t device,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ audio_output_flags_t flags)
+{
+ if( !((flags & AUDIO_OUTPUT_FLAG_LPA) ||
+ (flags & AUDIO_OUTPUT_FLAG_TUNNEL)||
+ (flags & AUDIO_OUTPUT_FLAG_VOIP_RX)) )
+ flags = AUDIO_OUTPUT_FLAG_DIRECT;
+
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (mHwModules[i]->mHandle == 0) {
+ continue;
+ }
+ for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
+ AudioPolicyManagerBase::IOProfile *profile = mHwModules[i]->mOutputProfiles[j];
+ if (isCompatibleProfile(profile, device, samplingRate, format,
+ channelMask,
+ flags)) {
+ if (mAvailableOutputDevices & profile->mSupportedDevices) {
+ return mHwModules[i]->mOutputProfiles[j];
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+bool AudioPolicyManager::isCompatibleProfile(AudioPolicyManagerBase::IOProfile *profile,
+ audio_devices_t device,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ audio_output_flags_t flags)
+{
+ if ((profile->mSupportedDevices & device) != device) {
+ return false;
+ }
+ if (profile->mFlags != flags) {
+ return false;
+ }
+ if (samplingRate != 0) {
+ size_t i;
+ for (i = 0; i < profile->mSamplingRates.size(); i++)
+ {
+ if (profile->mSamplingRates[i] == samplingRate) {
+ break;
+ }
+ }
+ if (i == profile->mSamplingRates.size()) {
+ return false;
+ }
+ }
+ if (format != 0) {
+ size_t i;
+ for (i = 0; i < profile->mFormats.size(); i++)
+ {
+ if (profile->mFormats[i] == format) {
+ break;
+ }
+ }
+ if (i == profile->mFormats.size()) {
+ return false;
+ }
+ }
+ if (channelMask != 0) {
+ size_t i;
+ for (i = 0; i < profile->mChannelMasks.size(); i++)
+ {
+ if (profile->mChannelMasks[i] == channelMask) {
+ break;
+ }
+ }
+ if (i == profile->mChannelMasks.size()) {
+ return false;
+ }
+ }
+ ALOGD(" profile found: device %x, flags %x, samplingrate %d,\
+ format %x, channelMask %d",
+ device, flags, samplingRate, format, channelMask);
+ return true;
+}
+
+bool AudioPolicyManager::platform_is_Fusion3()
+{
+ char platform[128], baseband[128], baseband_arch[128];
+ property_get("ro.board.platform", platform, "");
+ property_get("ro.baseband", baseband, "");
+ property_get("ro.baseband.arch", baseband_arch, "");
+ if (!strcmp("msm8960", platform) && (!strcmp("mdm", baseband) || !strcmp("mdm", baseband_arch)))
+ return true;
+ else
+ return false;
+}
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+{
+ return new AudioPolicyManager(clientInterface);
+}
+
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+{
+ delete interface;
+}
+
+
+}; // namespace android
diff --git a/msm8660/AudioPolicyManager.h b/msm8660/AudioPolicyManager.h
index 43316a4..f0f0237 100644
--- a/msm8660/AudioPolicyManager.h
+++ b/msm8660/AudioPolicyManager.h
@@ -1,8 +1,6 @@
/*
* Copyright (C) 2009 The Android Open Source Project
- * Copyright (c) 2009, 2011-2013, The Linux Foundation. All rights reserved.
- * Not a Contribution, Apache license notifications and license are retained
- * for attribution purposes only.
+ * Copyright (c) 2012-2013, The Linux Foundation. All rights reserved.
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -17,24 +15,25 @@
* limitations under the License.
*/
+
#include <stdint.h>
#include <sys/types.h>
-#include <cutils/config_utils.h>
-#include <cutils/misc.h>
#include <utils/Timers.h>
#include <utils/Errors.h>
#include <utils/KeyedVector.h>
#include <hardware_legacy/AudioPolicyManagerBase.h>
+
namespace android_audio_legacy {
+// ----------------------------------------------------------------------------
+
class AudioPolicyManager: public AudioPolicyManagerBase
{
public:
AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
- : AudioPolicyManagerBase(clientInterface) {
- }
+ : AudioPolicyManagerBase(clientInterface) {}
virtual ~AudioPolicyManager() {}
@@ -42,8 +41,36 @@
virtual status_t setDeviceConnectionState(audio_devices_t device,
AudioSystem::device_connection_state state,
const char *device_address);
+ uint32_t checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs);
+
+ void setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+ void setStreamMute(int stream, bool on, audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device,
const char *device_address);
+ virtual void setPhoneState(int state);
+ virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+ virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+ uint32_t samplingRate = 0,
+ uint32_t format = AudioSystem::FORMAT_DEFAULT,
+ uint32_t channels = 0,
+ AudioSystem::output_flags flags =
+ AudioSystem::OUTPUT_FLAG_INDIRECT);
+ virtual status_t startOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session = 0);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ AudioSystem::stream_type stream,
+ int session = 0);
+ virtual void releaseOutput(audio_io_handle_t output);
virtual audio_io_handle_t getInput(int inputSource,
uint32_t samplingRate,
uint32_t format,
@@ -53,25 +80,42 @@
// indicates to the audio policy manager that the input starts being used.
virtual status_t startInput(audio_io_handle_t input);
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input);
+ virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream,
+ int index,
+ audio_devices_t device);
+protected:
+ // return the strategy corresponding to a given stream type
+ static routing_strategy getStrategy(AudioSystem::stream_type stream);
+
// return appropriate device for streams handled by the specified strategy according to current
// phone state, connected devices...
- // if fromCache is true, the device is returned from mDeviceForStrategy[], otherwise it is determined
- // by current state (device connected, phone state, force use, a2dp output...)
+ // if fromCache is true, the device is returned from mDeviceForStrategy[],
+ // otherwise it is determine by current state
+ // (device connected,phone state, force use, a2dp output...)
// This allows to:
// 1 speed up process when the state is stable (when starting or stopping an output)
// 2 access to either current device selection (fromCache == true) or
// "future" device selection (fromCache == false) when called from a context
// where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
- // before updateDeviceForStrategy() is called.
- virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session = 0);
- virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session = 0);
- virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+ // before updateDevicesAndOutputs() is called.
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache = true);
-protected:
+ // change the route of the specified output. Returns the number of ms we have slept to
+ // allow new routing to take effect in certain cases.
+ uint32_t setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force = false,
+ int delayMs = 0);
- virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, bool fromCache = true);
- // change the route of the specified output
- uint32_t setOutputDevice(audio_io_handle_t output, audio_devices_t device, bool force = false, int delayMs = 0);
+ // select input device corresponding to requested audio source
+ virtual audio_devices_t getDeviceForInputSource(int inputSource);
+
+ // check that volume change is permitted, compute and send new volume to audio hardware
+ status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
// when a device is connected, checks if an open output can be routed
// to this device. If none is open, tries to open one of the available outputs.
@@ -82,6 +126,14 @@
status_t checkOutputsForDevice(audio_devices_t device,
AudioSystem::device_connection_state state,
SortedVector<audio_io_handle_t>& outputs);
+ // manages A2DP output suspend/restore according to phone state and BT SCO usage
+ void checkA2dpSuspend();
+
+ // returns the A2DP output handle if it is open or 0 otherwise
+ audio_io_handle_t getA2dpOutput();
+
+ // returns true if give output is direct output
+ bool isDirectOutput(audio_io_handle_t output);
virtual AudioPolicyManagerBase::IOProfile* getProfileForDirectOutput(
audio_devices_t device,
@@ -89,25 +141,33 @@
uint32_t format,
uint32_t channelMask,
audio_output_flags_t flags);
-
-
bool isCompatibleProfile(AudioPolicyManagerBase::IOProfile *profile,
audio_devices_t device,
uint32_t samplingRate,
uint32_t format,
uint32_t channelMask,
audio_output_flags_t flags);
+ // selects the most appropriate device on output for current state
+ // must be called every time a condition that affects the device choice for a given output is
+ // changed: connected device, phone state, force use, output start, output stop..
+ // see getDeviceForStrategy() for the use of fromCache parameter
- // check that volume change is permitted, compute and send new volume to audio hardware
- status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
- // select input device corresponding to requested audio source
- virtual audio_devices_t getDeviceForInputSource(int inputSource);
+ audio_devices_t getNewDevice(audio_io_handle_t output, bool fromCache);
+ // returns the category the device belongs to with regard to volume curve management
+ static device_category getDeviceCategory(audio_devices_t device);
+
+ // extract one device relevant for volume control from multiple device selection
+ static audio_devices_t getDeviceForVolume(audio_devices_t device);
+ // true is current platform implements a back microphone
+ virtual bool hasBackMicrophone() const { return false; }
+ // true is current platform supports suplication of notifications and ringtones over A2DP output
+ virtual bool a2dpUsedForSonification() const { return true; }
private:
- // updates device caching and output for streams that can influence the
- // routing of notifications
+
void handleNotificationRoutingForStream(AudioSystem::stream_type stream);
+ bool platform_is_Fusion3();
};
-}; //namespace
+};
diff --git a/msm8660/audio_policy_hal.cpp b/msm8660/audio_policy_hal.cpp
index efb6b63..f494d23 100644
--- a/msm8660/audio_policy_hal.cpp
+++ b/msm8660/audio_policy_hal.cpp
@@ -1,8 +1,5 @@
/*
- * Copyright (C) 2011 The Android Open Source Project
- * Copyright (c) 2012, The Linux Foundation. All rights reserved.
- * Not a Contribution, Apache license notifications and license are retained
- * for attribution purposes only.
+ * Copyright (c) 2012-2013, The Linux Foundation. All rights reserved.
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -17,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_TAG "audio_policy.msm8660"
+#define LOG_TAG "qcom_audio_policy_hal"
//#define LOG_NDEBUG 0
#include <stdint.h>
@@ -33,7 +30,6 @@
#include "AudioPolicyCompatClient.h"
namespace android_audio_legacy {
-//using android_audio_legacy::AudioSystem;
extern "C" {
@@ -94,11 +90,13 @@
qap->apm->setPhoneState(state);
}
+ /* indicate a change in ringer mode */
static void ap_set_ringer_mode(struct audio_policy *pol, uint32_t mode,
uint32_t mask)
{
// deprecated, never called
}
+
/* force using a specific device category for the specified usage */
static void ap_set_force_use(struct audio_policy *pol,
audio_policy_force_use_t usage,
@@ -134,58 +132,19 @@
return qap->apm->initCheck();
}
-#ifdef QCOM_TUNNEL_LPA_ENABLED
-static audio_io_handle_t ap_get_session(struct audio_policy *pol,
- audio_stream_type_t stream,
- audio_format_t format,
- audio_policy_output_flags_t flags,
- int sessionId,
- uint32_t samplingRate,
- uint32_t channels)
-{
- struct qcom_audio_policy *qap = to_qap(pol);
-
- ALOGV("%s: tid %d", __func__, gettid());
- return qap->apm->getSession((AudioSystem::stream_type)stream,
- format, (AudioSystem::output_flags)flags,
- sessionId,
- samplingRate,
- channels);
-}
-
-static void ap_pause_session(struct audio_policy *pol, audio_io_handle_t output,
- audio_stream_type_t stream)
-{
- struct qcom_audio_policy *qap = to_qap(pol);
- qap->apm->pauseSession(output, (AudioSystem::stream_type)stream);
-}
-
-static void ap_resume_session(struct audio_policy *pol, audio_io_handle_t output,
- audio_stream_type_t stream)
-{
- struct qcom_audio_policy *qap = to_qap(pol);
- qap->apm->resumeSession(output, (AudioSystem::stream_type)stream);
-}
-
-static void ap_release_session(struct audio_policy *pol, audio_io_handle_t output)
-{
- struct qcom_audio_policy *qap = to_qap(pol);
- qap->apm->releaseSession(output);
-}
-#endif
static audio_io_handle_t ap_get_output(struct audio_policy *pol,
audio_stream_type_t stream,
uint32_t sampling_rate,
audio_format_t format,
- audio_channel_mask_t channelMask,
+ uint32_t channels,
audio_output_flags_t flags)
{
struct qcom_audio_policy *qap = to_qap(pol);
ALOGV("%s: tid %d", __func__, gettid());
return qap->apm->getOutput((AudioSystem::stream_type)stream,
- sampling_rate,(int) format, channelMask,
+ sampling_rate, format, channels,
(AudioSystem::output_flags)flags);
}
@@ -215,11 +174,11 @@
static audio_io_handle_t ap_get_input(struct audio_policy *pol, audio_source_t inputSource,
uint32_t sampling_rate,
audio_format_t format,
- audio_channel_mask_t channelMask,
+ uint32_t channels,
audio_in_acoustics_t acoustics)
{
struct qcom_audio_policy *qap = to_qap(pol);
- return qap->apm->getInput((int) inputSource, sampling_rate,(int) format, channelMask,
+ return qap->apm->getInput(inputSource, sampling_rate, format, channels,
(AudioSystem::audio_in_acoustics)acoustics);
}
@@ -307,7 +266,7 @@
}
static audio_io_handle_t ap_get_output_for_effect(struct audio_policy *pol,
- const struct effect_descriptor_s *desc)
+ const struct effect_descriptor_s *desc)
{
struct qcom_audio_policy *qap = to_qap(pol);
return qap->apm->getOutputForEffect(desc);
@@ -341,11 +300,10 @@
uint32_t in_past_ms)
{
const struct qcom_audio_policy *qap = to_cqap(pol);
- return qap->apm->isStreamActive((int) stream, in_past_ms);
+ return qap->apm->isStreamActive(stream, in_past_ms);
}
-static bool ap_is_stream_active_remotely(const struct audio_policy *pol,
- audio_stream_type_t stream,
+static bool ap_is_stream_active_remotely(const struct audio_policy *pol, audio_stream_type_t stream,
uint32_t in_past_ms)
{
const struct qcom_audio_policy *qap = to_cqap(pol);
@@ -357,6 +315,7 @@
const struct qcom_audio_policy *qap = to_cqap(pol);
return qap->apm->isSourceActive(source);
}
+
static int ap_dump(const struct audio_policy *pol, int fd)
{
const struct qcom_audio_policy *qap = to_cqap(pol);
@@ -381,18 +340,13 @@
qap->policy.set_device_connection_state = ap_set_device_connection_state;
qap->policy.get_device_connection_state = ap_get_device_connection_state;
qap->policy.set_phone_state = ap_set_phone_state;
+ qap->policy.set_ringer_mode = ap_set_ringer_mode;
qap->policy.set_force_use = ap_set_force_use;
qap->policy.get_force_use = ap_get_force_use;
qap->policy.set_can_mute_enforced_audible =
ap_set_can_mute_enforced_audible;
qap->policy.init_check = ap_init_check;
qap->policy.get_output = ap_get_output;
-#ifdef QCOM_TUNNEL_LPA_ENABLED
- qap->policy.get_session = ap_get_session;
- qap->policy.pause_session = ap_pause_session;
- qap->policy.resume_session = ap_resume_session;
- qap->policy.release_session = ap_release_session;
-#endif
qap->policy.start_output = ap_start_output;
qap->policy.stop_output = ap_stop_output;
qap->policy.release_output = ap_release_output;
@@ -412,6 +366,7 @@
qap->policy.unregister_effect = ap_unregister_effect;
qap->policy.set_effect_enabled = ap_set_effect_enabled;
qap->policy.is_stream_active = ap_is_stream_active;
+
qap->policy.is_stream_active_remotely = ap_is_stream_active_remotely;
qap->policy.is_source_active = ap_is_source_active;
qap->policy.dump = ap_dump;
@@ -501,7 +456,7 @@
version_minor: 0,
id: AUDIO_POLICY_HARDWARE_MODULE_ID,
name: "QCOM Audio Policy HAL",
- author: "Code Aurora Forum",
+ author: "The Linux Foundation",
methods: &qcom_ap_module_methods,
dso : NULL,
reserved : {0},