audio policy: add support for custom mixes
Add support for custom mixes in AudioPolicyManager.
Two methods are added to register or unregister a list of custom mixes
with their attributes and format.
getOutputForAttr() and getInputForAttr() first look for a match in
registered mixes before defaulting to normal output/input selection
Remote submix device connection disconnection now takes address into
account to identify the correspnoding custom mix.
Bug: 16009464.
Change-Id: I3f1c2a485a0fb71b1f984ed0adc9b68aa971e408
diff --git a/include/media/AudioPolicy.h b/include/media/AudioPolicy.h
new file mode 100644
index 0000000..a755e1e
--- /dev/null
+++ b/include/media/AudioPolicy.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#ifndef ANDROID_AUDIO_POLICY_H
+#define ANDROID_AUDIO_POLICY_H
+
+#include <system/audio.h>
+#include <system/audio_policy.h>
+#include <binder/Parcel.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+// Keep in sync with AudioMix.java, AudioMixingRule.java, AudioPolicyConfig.java
+#define RULE_EXCLUSION_MASK 0x8000
+#define RULE_MATCH_ATTRIBUTE_USAGE 0x1
+#define RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET (0x1 << 1)
+#define RULE_EXCLUDE_ATTRIBUTE_USAGE (RULE_EXCLUSION_MASK|RULE_MATCH_ATTRIBUTE_USAGE)
+#define RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET \
+ (RULE_EXCLUSION_MASK|RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET)
+
+#define MIX_TYPE_INVALID -1
+#define MIX_TYPE_PLAYERS 0
+#define MIX_TYPE_RECORDERS 1
+
+#define ROUTE_FLAG_RENDER 0x1
+#define ROUTE_FLAG_LOOP_BACK (0x1 << 1)
+
+#define MAX_MIXES_PER_POLICY 10
+#define MAX_CRITERIA_PER_MIX 20
+
+class AttributeMatchCriterion {
+public:
+ AttributeMatchCriterion() {}
+ AttributeMatchCriterion(audio_usage_t usage, audio_source_t source, uint32_t rule);
+
+ status_t readFromParcel(Parcel *parcel);
+ status_t writeToParcel(Parcel *parcel) const;
+
+ union {
+ audio_usage_t mUsage;
+ audio_source_t mSource;
+ } mAttr;
+ uint32_t mRule;
+};
+
+class AudioMix {
+public:
+ AudioMix() {}
+ AudioMix(Vector<AttributeMatchCriterion> criteria, uint32_t mixType, audio_config_t format,
+ uint32_t routeFlags, String8 registrationId) :
+ mCriteria(criteria), mMixType(mixType), mFormat(format),
+ mRouteFlags(routeFlags), mRegistrationId(registrationId) {}
+
+ status_t readFromParcel(Parcel *parcel);
+ status_t writeToParcel(Parcel *parcel) const;
+
+ Vector<AttributeMatchCriterion> mCriteria;
+ uint32_t mMixType;
+ audio_config_t mFormat;
+ uint32_t mRouteFlags;
+ String8 mRegistrationId;
+};
+
+}; // namespace android
+
+#endif // ANDROID_AUDIO_POLICY_H
diff --git a/include/media/AudioPolicyHelper.h b/include/media/AudioPolicyHelper.h
index 3ed0b74..79231be 100644
--- a/include/media/AudioPolicyHelper.h
+++ b/include/media/AudioPolicyHelper.h
@@ -63,7 +63,7 @@
static void stream_type_to_audio_attributes(audio_stream_type_t streamType,
audio_attributes_t *attr) {
- attr->flags = 0x0;
+ memset(attr, 0, sizeof(audio_attributes_t));
switch (streamType) {
case AUDIO_STREAM_DEFAULT:
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index e012116..a2e0909 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -59,7 +59,8 @@
MemoryLeakTrackUtil.cpp \
SoundPool.cpp \
SoundPoolThread.cpp \
- StringArray.cpp
+ StringArray.cpp \
+ AudioPolicy.cpp
LOCAL_SRC_FILES += ../libnbaio/roundup.c
diff --git a/media/libmedia/AudioPolicy.cpp b/media/libmedia/AudioPolicy.cpp
new file mode 100644
index 0000000..d2d0971
--- /dev/null
+++ b/media/libmedia/AudioPolicy.cpp
@@ -0,0 +1,115 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicy"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include <media/AudioPolicy.h>
+
+namespace android {
+
+//
+// AttributeMatchCriterion implementation
+//
+AttributeMatchCriterion::AttributeMatchCriterion(audio_usage_t usage,
+ audio_source_t source,
+ uint32_t rule)
+: mRule(rule)
+{
+ if (mRule == RULE_MATCH_ATTRIBUTE_USAGE ||
+ mRule == RULE_EXCLUDE_ATTRIBUTE_USAGE) {
+ mAttr.mUsage = usage;
+ } else {
+ mAttr.mSource = source;
+ }
+}
+
+status_t AttributeMatchCriterion::readFromParcel(Parcel *parcel)
+{
+ mRule = parcel->readInt32();
+ if (mRule == RULE_MATCH_ATTRIBUTE_USAGE ||
+ mRule == RULE_EXCLUDE_ATTRIBUTE_USAGE) {
+ mAttr.mUsage = (audio_usage_t)parcel->readInt32();
+ } else {
+ mAttr.mSource = (audio_source_t)parcel->readInt32();
+ }
+ return NO_ERROR;
+}
+
+status_t AttributeMatchCriterion::writeToParcel(Parcel *parcel) const
+{
+ parcel->writeInt32(mRule);
+ parcel->writeInt32(mAttr.mUsage);
+ return NO_ERROR;
+}
+
+//
+// AudioMix implementation
+//
+
+status_t AudioMix::readFromParcel(Parcel *parcel)
+{
+ mMixType = parcel->readInt32();
+ mFormat.sample_rate = (uint32_t)parcel->readInt32();
+ mFormat.channel_mask = (audio_channel_mask_t)parcel->readInt32();
+ mFormat.format = (audio_format_t)parcel->readInt32();
+ mRouteFlags = parcel->readInt32();
+ mRegistrationId = parcel->readString8();
+ size_t size = (size_t)parcel->readInt32();
+ if (size > MAX_CRITERIA_PER_MIX) {
+ size = MAX_CRITERIA_PER_MIX;
+ }
+ for (size_t i = 0; i < size; i++) {
+ AttributeMatchCriterion criterion;
+ if (criterion.readFromParcel(parcel) == NO_ERROR) {
+ mCriteria.add(criterion);
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioMix::writeToParcel(Parcel *parcel) const
+{
+ parcel->writeInt32(mMixType);
+ parcel->writeInt32(mFormat.sample_rate);
+ parcel->writeInt32(mFormat.channel_mask);
+ parcel->writeInt32(mFormat.format);
+ parcel->writeInt32(mRouteFlags);
+ parcel->writeString8(mRegistrationId);
+ size_t size = mCriteria.size();
+ if (size > MAX_CRITERIA_PER_MIX) {
+ size = MAX_CRITERIA_PER_MIX;
+ }
+ size_t sizePosition = parcel->dataPosition();
+ parcel->writeInt32(size);
+ size_t finalSize = size;
+ for (size_t i = 0; i < size; i++) {
+ size_t position = parcel->dataPosition();
+ if (mCriteria[i].writeToParcel(parcel) != NO_ERROR) {
+ parcel->setDataPosition(position);
+ finalSize--;
+ }
+ }
+ if (size != finalSize) {
+ size_t position = parcel->dataPosition();
+ parcel->setDataPosition(sizePosition);
+ parcel->writeInt32(finalSize);
+ parcel->setDataPosition(position);
+ }
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index e4c3c08..d9c3177 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -851,6 +851,10 @@
// due to hardware latency. We leave this behavior for now.
*position = dspFrames;
} else {
+ if (mCblk->mFlags & CBLK_INVALID) {
+ restoreTrack_l("getPosition");
+ }
+
// IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
*position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
0 : updateAndGetPosition_l();
@@ -1946,6 +1950,10 @@
break;
}
+ if (mCblk->mFlags & CBLK_INVALID) {
+ restoreTrack_l("getTimestamp");
+ }
+
// The presented frame count must always lag behind the consumed frame count.
// To avoid a race, read the presented frames first. This ensures that presented <= consumed.
status_t status = mAudioTrack->getTimestamp(timestamp);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index dab6d91..1c3cf5d 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2180,7 +2180,13 @@
void AudioFlinger::PlaybackThread::threadLoop_exit()
{
- // Default implementation has nothing to do
+ {
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mTracks.size(); i++) {
+ sp<Track> track = mTracks[i];
+ track->invalidate();
+ }
+ }
}
/*
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index 8bb35f9..68f3802 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -211,7 +211,6 @@
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
-
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address)
@@ -818,7 +817,7 @@
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) {
sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
- bool found = profile->isCompatibleProfile(device, samplingRate,
+ bool found = profile->isCompatibleProfile(device, String8(""), samplingRate,
NULL /*updatedSamplingRate*/, format, channelMask,
flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD ?
AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD : AUDIO_OUTPUT_FLAG_DIRECT);
@@ -874,6 +873,51 @@
stream_type_to_audio_attributes(*stream, &attributes);
}
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ sp<AudioOutputDescriptor> desc;
+ if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) {
+ for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) ||
+ (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) {
+ desc = mPolicyMixes[i]->mOutput;
+ break;
+ }
+ if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mPolicyMixes[i]->mMix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = mPolicyMixes[i]->mOutput;
+ break;
+ }
+ }
+ } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) {
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE &&
+ strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attributes.tags + strlen("addr="),
+ mPolicyMixes[i]->mMix.mRegistrationId.string(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) {
+ desc = mPolicyMixes[i]->mOutput;
+ break;
+ }
+ }
+ if (desc != 0) {
+ if (!audio_is_linear_pcm(format)) {
+ return BAD_VALUE;
+ }
+ desc->mPolicyMixAddress = mPolicyMixes[i]->mMix.mRegistrationId;
+ *stream = streamTypefromAttributesInt(&attributes);
+ *output = desc->mIoHandle;
+ ALOGV("getOutputForAttr() returns output %d", *output);
+ return NO_ERROR;
+ }
+ }
+ if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+ ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
+ return BAD_VALUE;
+ }
+
ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x",
attributes.usage, attributes.content_type, attributes.tags, attributes.flags);
@@ -1182,7 +1226,14 @@
outputDesc->changeRefCount(stream, 1);
if (outputDesc->mRefCount[stream] == 1) {
- audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ // starting an output being rerouted?
+ audio_devices_t newDevice;
+ if (outputDesc->mPolicyMixAddress != String8("")
+ && outputDesc->mPolicyMixAddress != String8("0")) {
+ newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ } else {
+ newDevice = getNewOutputDevice(output, false /*fromCache*/);
+ }
routing_strategy strategy = getStrategy(stream);
bool shouldWait = (strategy == STRATEGY_SONIFICATION) ||
(strategy == STRATEGY_SONIFICATION_RESPECTFUL) ||
@@ -1350,57 +1401,66 @@
"session %d, flags %#x",
attr->source, samplingRate, format, channelMask, session, flags);
- audio_devices_t device = getDeviceForInputSource(attr->source);
-
- if (device == AUDIO_DEVICE_NONE) {
- ALOGW("getInputForAttr() could not find device for source %d", attr->source);
- return BAD_VALUE;
- }
-
- // adapt channel selection to input source
- switch (attr->source) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- case AUDIO_SOURCE_VOICE_CALL:
- channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
- break;
- default:
- break;
- }
-
*input = AUDIO_IO_HANDLE_NONE;
+ audio_devices_t device;
+ // handle legacy remote submix case where the address was not always specified
+ String8 address = String8("");
bool isSoundTrigger = false;
audio_source_t halInputSource = attr->source;
- if (attr->source == AUDIO_SOURCE_HOTWORD) {
- ssize_t index = mSoundTriggerSessions.indexOfKey(session);
- if (index >= 0) {
- *input = mSoundTriggerSessions.valueFor(session);
- isSoundTrigger = true;
- flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
- ALOGV("SoundTrigger capture on session %d input %d", session, *input);
- } else {
- halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
+
+ if (attr->source == AUDIO_SOURCE_REMOTE_SUBMIX &&
+ strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
+ device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ address = String8(attr->tags + strlen("addr="));
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index < 0) {
+ ALOGW("getInputForAttr() no policy for address %s", address.string());
+ return BAD_VALUE;
+ }
+ } else {
+ device = getDeviceForInputSource(attr->source, &address);
+
+ if (device == AUDIO_DEVICE_NONE) {
+ ALOGW("getInputForAttr() could not find device for source %d", attr->source);
+ return BAD_VALUE;
+ }
+ // adapt channel selection to input source
+ switch (attr->source) {
+ case AUDIO_SOURCE_VOICE_UPLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_DOWNLINK:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ case AUDIO_SOURCE_VOICE_CALL:
+ channelMask = AUDIO_CHANNEL_IN_VOICE_UPLINK | AUDIO_CHANNEL_IN_VOICE_DNLINK;
+ break;
+ default:
+ break;
+ }
+ if (attr->source == AUDIO_SOURCE_HOTWORD) {
+ ssize_t index = mSoundTriggerSessions.indexOfKey(session);
+ if (index >= 0) {
+ *input = mSoundTriggerSessions.valueFor(session);
+ isSoundTrigger = true;
+ flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_HW_HOTWORD);
+ ALOGV("SoundTrigger capture on session %d input %d", session, *input);
+ } else {
+ halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
+ }
}
}
- sp<IOProfile> profile = getInputProfile(device,
- samplingRate,
- format,
- channelMask,
- flags);
+ sp<IOProfile> profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
if (profile == 0) {
//retry without flags
audio_input_flags_t log_flags = flags;
flags = AUDIO_INPUT_FLAG_NONE;
- profile = getInputProfile(device,
- samplingRate,
- format,
- channelMask,
- flags);
+ profile = getInputProfile(device, address,
+ samplingRate, format, channelMask,
+ flags);
if (profile == 0) {
ALOGW("getInputForAttr() could not find profile for device 0x%X, samplingRate %u,"
"format %#x, channelMask 0x%X, flags %#x",
@@ -1419,9 +1479,6 @@
config.channel_mask = channelMask;
config.format = format;
- // handle legacy remote submix case where the address was not always specified
- String8 address = deviceDistinguishesOnAddress(device) ? String8("0") : String8("");
-
status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
input,
&config,
@@ -1451,7 +1508,7 @@
inputDesc->mFormat = format;
inputDesc->mChannelMask = channelMask;
inputDesc->mDevice = device;
- inputDesc->mSessions.add(session);
+ inputDesc->mSessions.add(session, address);
inputDesc->mIsSoundTrigger = isSoundTrigger;
addInput(*input, inputDesc);
@@ -1470,7 +1527,7 @@
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
- index = inputDesc->mSessions.indexOf(session);
+ index = inputDesc->mSessions.indexOfKey(session);
if (index < 0) {
ALOGW("startInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
@@ -1488,8 +1545,8 @@
sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
- stopInput(activeInput, activeDesc->mSessions.itemAt(0));
- releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
+ stopInput(activeInput, activeDesc->mSessions.keyAt(0));
+ releaseInput(activeInput, activeDesc->mSessions.keyAt(0));
} else {
ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
return INVALID_OPERATION;
@@ -1507,7 +1564,8 @@
// For remote submix (a virtual device), we open only one input per capture request.
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ inputDesc->mSessions.valueAt(index));
}
}
@@ -1528,7 +1586,7 @@
}
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
- index = inputDesc->mSessions.indexOf(session);
+ index = inputDesc->mSessions.indexOfKey(session);
if (index < 0) {
ALOGW("stopInput() unknown session %d on input %d", session, input);
return BAD_VALUE;
@@ -1545,7 +1603,8 @@
// automatically disable the remote submix output when input is stopped
if (audio_is_remote_submix_device(inputDesc->mDevice)) {
setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS);
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ inputDesc->mSessions.valueAt(index));
}
resetInputDevice(input);
@@ -1569,12 +1628,12 @@
sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
ALOG_ASSERT(inputDesc != 0);
- index = inputDesc->mSessions.indexOf(session);
+ index = inputDesc->mSessions.indexOfKey(session);
if (index < 0) {
ALOGW("releaseInput() unknown session %d on input %d", session, input);
return;
}
- inputDesc->mSessions.remove(session);
+ inputDesc->mSessions.removeItem(session);
if (inputDesc->mOpenRefCount == 0) {
ALOGW("releaseInput() invalid open ref count %d", inputDesc->mOpenRefCount);
return;
@@ -1923,6 +1982,117 @@
return false;
}
+// Register a list of custom mixes with their attributes and format.
+// When a mix is registered, corresponding input and output profiles are
+// added to the remote submix hw module. The profile contains only the
+// parameters (sampling rate, format...) specified by the mix.
+// The corresponding input remote submix device is also connected.
+//
+// When a remote submix device is connected, the address is checked to select the
+// appropriate profile and the corresponding input or output stream is opened.
+//
+// When capture starts, getInputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, getDeviceForInputSource() will:
+// - 2.1 look for a mix matching the attributes source
+// - 2.2 if none found, default to device selection by policy rules
+// At this time, the corresponding output remote submix device is also connected
+// and active playback use cases can be transferred to this mix if needed when reconnecting
+// after AudioTracks are invalidated
+//
+// When playback starts, getOutputForAttr() will:
+// - 1 look for a mix matching the address passed in attribtutes tags if any
+// - 2 if none found, look for a mix matching the attributes usage
+// - 3 if none found, default to device and output selection by policy rules.
+
+status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("registerPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index >= 0) {
+ ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string());
+ continue;
+ }
+ audio_config_t outputConfig = mixes[i].mFormat;
+ audio_config_t inputConfig = mixes[i].mFormat;
+ // NOTE: audio flinger mixer does not support mono output: configure remote submix HAL in
+ // stereo and let audio flinger do the channel conversion if needed.
+ outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ module->addOutputProfile(address, &outputConfig,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
+ module->addInputProfile(address, &inputConfig,
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
+ sp<AudioPolicyMix> policyMix = new AudioPolicyMix();
+ policyMix->mMix = mixes[i];
+ mPolicyMixes.add(address, policyMix);
+ setDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.string());
+ }
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes)
+{
+ sp<HwModule> module;
+ for (size_t i = 0; i < mHwModules.size(); i++) {
+ if (strcmp(AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX, mHwModules[i]->mName) == 0 &&
+ mHwModules[i]->mHandle != 0) {
+ module = mHwModules[i];
+ break;
+ }
+ }
+
+ if (module == 0) {
+ return INVALID_OPERATION;
+ }
+
+ ALOGV("unregisterPolicyMixes() num mixes %d", mixes.size());
+
+ for (size_t i = 0; i < mixes.size(); i++) {
+ String8 address = mixes[i].mRegistrationId;
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index < 0) {
+ ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string());
+ continue;
+ }
+
+ mPolicyMixes.removeItemsAt(index);
+
+ setDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string());
+
+ if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
+ AUDIO_POLICY_DEVICE_STATE_AVAILABLE)
+ {
+ setDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.string());
+ }
+ module->removeOutputProfile(address);
+ module->removeInputProfile(address);
+ }
+ return NO_ERROR;
+}
+
status_t AudioPolicyManager::dump(int fd)
{
@@ -2323,6 +2493,7 @@
}
if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
+ devDesc->mAddress,
patch->sources[0].sample_rate,
NULL, // updatedSamplingRate
patch->sources[0].format,
@@ -2377,13 +2548,14 @@
}
if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType,
- patch->sinks[0].sample_rate,
- NULL, /*updatedSampleRate*/
- patch->sinks[0].format,
- patch->sinks[0].channel_mask,
- // FIXME for the parameter type,
- // and the NONE
- (audio_output_flags_t)
+ devDesc->mAddress,
+ patch->sinks[0].sample_rate,
+ NULL, /*updatedSampleRate*/
+ patch->sinks[0].format,
+ patch->sinks[0].channel_mask,
+ // FIXME for the parameter type,
+ // and the NONE
+ (audio_output_flags_t)
AUDIO_INPUT_FLAG_NONE)) {
return INVALID_OPERATION;
}
@@ -3225,9 +3397,13 @@
}
for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++)
{
- if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) {
- ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
- profiles.add(mHwModules[i]->mOutputProfiles[j]);
+ sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j];
+ if (profile->mSupportedDevices.types() & device) {
+ if (!deviceDistinguishesOnAddress(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i);
+ }
}
}
}
@@ -3359,6 +3535,15 @@
if (output != AUDIO_IO_HANDLE_NONE) {
addOutput(output, desc);
+ if (deviceDistinguishesOnAddress(device) && address != "0") {
+ ssize_t index = mPolicyMixes.indexOfKey(address);
+ if (index >= 0) {
+ mPolicyMixes[index]->mOutput = desc;
+ } else {
+ ALOGE("checkOutputsForDevice() cannot find policy for address %s",
+ address.string());
+ }
+ }
if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) {
audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE;
@@ -3422,15 +3607,15 @@
for (size_t i = 0; i < mOutputs.size(); i++) {
desc = mOutputs.valueAt(i);
if (!desc->isDuplicated()) {
- if (!(desc->mProfile->mSupportedDevices.types()
+ // exact match on device
+ if (deviceDistinguishesOnAddress(device) &&
+ (desc->mProfile->mSupportedDevices.types() == device)) {
+ findIoHandlesByAddress(desc, address, outputs);
+ } else if (!(desc->mProfile->mSupportedDevices.types()
& mAvailableOutputDevices.types())) {
ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
mOutputs.keyAt(i));
outputs.add(mOutputs.keyAt(i));
- } else if (deviceDistinguishesOnAddress(device) &&
- // exact match on device
- (desc->mProfile->mSupportedDevices.types() == device)) {
- findIoHandlesByAddress(desc, address, outputs);
}
}
}
@@ -3492,11 +3677,15 @@
profile_index < mHwModules[module_idx]->mInputProfiles.size();
profile_index++)
{
- if (mHwModules[module_idx]->mInputProfiles[profile_index]->mSupportedDevices.types()
- & (device & ~AUDIO_DEVICE_BIT_IN)) {
- ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
- profile_index, module_idx);
- profiles.add(mHwModules[module_idx]->mInputProfiles[profile_index]);
+ sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index];
+
+ if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) {
+ if (!deviceDistinguishesOnAddress(device) ||
+ address == profile->mSupportedDevices[0]->mAddress) {
+ profiles.add(profile);
+ ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
+ profile_index, module_idx);
+ }
}
}
}
@@ -3664,6 +3853,12 @@
return;
}
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ if (mPolicyMixes[i]->mOutput == outputDesc) {
+ mPolicyMixes[i]->mOutput.clear();
+ }
+ }
+
// look for duplicated outputs connected to the output being removed.
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i);
@@ -4373,7 +4568,9 @@
uint32_t device2 = AUDIO_DEVICE_NONE;
if (strategy != STRATEGY_SONIFICATION) {
// no sonification on remote submix (e.g. WFD)
- device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) {
+ device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ }
}
if ((device2 == AUDIO_DEVICE_NONE) &&
(mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
@@ -4783,6 +4980,7 @@
}
sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
+ String8 address,
uint32_t& samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -4800,9 +4998,10 @@
{
sp<IOProfile> profile = mHwModules[i]->mInputProfiles[j];
// profile->log();
- if (profile->isCompatibleProfile(device, samplingRate,
+ if (profile->isCompatibleProfile(device, address, samplingRate,
&samplingRate /*updatedSamplingRate*/,
format, channelMask, (audio_output_flags_t) flags)) {
+
return profile;
}
}
@@ -4810,11 +5009,33 @@
return NULL;
}
-audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource,
+ String8 *address)
{
uint32_t device = AUDIO_DEVICE_NONE;
audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() &
~AUDIO_DEVICE_BIT_IN;
+
+ for (size_t i = 0; i < mPolicyMixes.size(); i++) {
+ if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) {
+ continue;
+ }
+ for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) {
+ if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) ||
+ (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule &&
+ mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) {
+ if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ if (address != NULL) {
+ *address = mPolicyMixes[i]->mMix.mRegistrationId;
+ }
+ return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ }
+ break;
+ }
+ }
+ }
+
switch (inputSource) {
case AUDIO_SOURCE_VOICE_UPLINK:
if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
@@ -4902,6 +5123,9 @@
case AUDIO_SOURCE_REMOTE_SUBMIX:
if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ if (address != NULL) {
+ *address = "0";
+ }
}
break;
case AUDIO_SOURCE_FM_TUNER:
@@ -4928,7 +5152,7 @@
}
bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) {
- return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL) != 0);
+ return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0);
}
audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs)
@@ -5338,6 +5562,19 @@
}
float volume = computeVolume(stream, index, output, device);
+ // unit gain if rerouting to external policy
+ if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
+ ssize_t index = mOutputs.indexOfKey(output);
+ if (index >= 0) {
+ sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+ if (outputDesc->mPolicyMixAddress != String8("")
+ && outputDesc->mPolicyMixAddress != String8("0")) {
+ ALOGV("max gain when rerouting for output=%d", output);
+ volume = 1.0f;
+ }
+ }
+
+ }
// We actually change the volume if:
// - the float value returned by computeVolume() changed
// - the force flag is set
@@ -5520,7 +5757,8 @@
AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
const sp<IOProfile>& profile)
: mId(0), mIoHandle(0), mLatency(0),
- mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPatchHandle(0),
+ mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMixAddress(String8("")),
+ mPatchHandle(0),
mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
{
// clear usage count for all stream types
@@ -6005,6 +6243,69 @@
return NO_ERROR;
}
+status_t AudioPolicyManager::HwModule::addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ mOutputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::HwModule::removeOutputProfile(String8 name)
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ if (mOutputProfiles[i]->mName == name) {
+ mOutputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::HwModule::addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+
+ mInputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::HwModule::removeInputProfile(String8 name)
+{
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ if (mInputProfiles[i]->mName == name) {
+ mInputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+
void AudioPolicyManager::HwModule::dump(int fd)
{
const size_t SIZE = 256;
@@ -6911,17 +7212,18 @@
// Sampling rate, format and channel mask must be specified in order to
// get a valid a match
bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
- uint32_t samplingRate,
- uint32_t *updatedSamplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- uint32_t flags) const
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const
{
const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
ALOG_ASSERT(isPlaybackThread != isRecordThread);
- if ((mSupportedDevices.types() & device) != device) {
+ if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
return false;
}
@@ -7116,7 +7418,8 @@
devName);
if (type != AUDIO_DEVICE_NONE) {
sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(""), type);
- if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+ type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
dev->mAddress = String8("0");
}
add(dev);
@@ -7128,7 +7431,7 @@
}
}
}
- devName = strtok(NULL, "|");
+ devName = strtok(NULL, "|");
}
}
@@ -7138,13 +7441,15 @@
sp<DeviceDescriptor> device;
for (size_t i = 0; i < size(); i++) {
if (itemAt(i)->mDeviceType == type) {
- device = itemAt(i);
- if (itemAt(i)->mAddress = address) {
- break;
+ if (address == "" || itemAt(i)->mAddress == address) {
+ device = itemAt(i);
+ if (itemAt(i)->mAddress == address) {
+ break;
+ }
}
}
}
- ALOGV("DeviceVector::getDevice() for type %d address %s found %p",
+ ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
type, address.string(), device.get());
return device;
}
@@ -7182,13 +7487,9 @@
audio_devices_t type, String8 address) const
{
DeviceVector devices;
- //ALOGV(" looking for device=%x, addr=%s", type, address.string());
for (size_t i = 0; i < size(); i++) {
- //ALOGV(" at i=%d: device=%x, addr=%s",
- // i, itemAt(i)->mDeviceType, itemAt(i)->mAddress.string());
if (itemAt(i)->mDeviceType == type) {
if (itemAt(i)->mAddress == address) {
- //ALOGV(" found matching address %s", address.string());
devices.add(itemAt(i));
}
}
@@ -7616,6 +7917,7 @@
case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
case AUDIO_USAGE_GAME:
+ case AUDIO_USAGE_VIRTUAL_SOURCE:
break;
default:
return false;
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
index 17348e9..4214131 100644
--- a/services/audiopolicy/AudioPolicyManager.h
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -23,6 +23,7 @@
#include <utils/Errors.h>
#include <utils/KeyedVector.h>
#include <utils/SortedVector.h>
+#include <media/AudioPolicy.h>
#include "AudioPolicyInterface.h"
@@ -184,6 +185,9 @@
virtual status_t releaseSoundTriggerSession(audio_session_t session);
+ virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
+ virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
+
protected:
enum routing_strategy {
@@ -392,6 +396,7 @@
// For input, flags is interpreted as audio_input_flags_t.
// TODO: merge audio_output_flags_t and audio_input_flags_t.
bool isCompatibleProfile(audio_devices_t device,
+ String8 address,
uint32_t samplingRate,
uint32_t *updatedSamplingRate,
audio_format_t format,
@@ -415,6 +420,13 @@
status_t loadInput(cnode *root);
status_t loadDevice(cnode *root);
+ status_t addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeOutputProfile(String8 name);
+ status_t addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeInputProfile(String8 name);
+
void dump(int fd);
const char *const mName; // base name of the audio HW module (primary, a2dp ...)
@@ -483,6 +495,7 @@
uint32_t mLatency; //
audio_output_flags_t mFlags; //
audio_devices_t mDevice; // current device this output is routed to
+ String8 mPolicyMixAddress; // non empty or "0" when used by a dynamic policy
audio_patch_handle_t mPatchHandle;
uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
nsecs_t mStopTime[AUDIO_STREAM_CNT];
@@ -515,7 +528,9 @@
audio_source_t mInputSource; // input source selected by application
//(mediarecorder.h)
const sp<IOProfile> mProfile; // I/O profile this output derives from
- SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
+ // audio sessions attached to this input and the
+ // corresponding device address
+ DefaultKeyedVector<audio_session_t, String8> mSessions;
bool mIsSoundTrigger; // used by a soundtrigger capture
virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
@@ -594,7 +609,8 @@
audio_patch_handle_t *patchHandle = NULL);
// select input device corresponding to requested audio source
- virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+ virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource,
+ String8 *address = NULL);
// return io handle of active input or 0 if no input is active
// Only considers inputs from physical devices (e.g. main mic, headset mic) when
@@ -731,10 +747,11 @@
audio_format_t format);
// samplingRate parameter is an in/out and so may be modified
sp<IOProfile> getInputProfile(audio_devices_t device,
- uint32_t& samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags);
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags);
sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
uint32_t samplingRate,
audio_format_t format,
@@ -832,6 +849,17 @@
uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
bool mBeaconMuted; // has STREAM_TTS been muted
+ // custom mix entry in mPolicyMixes
+ class AudioPolicyMix : public RefBase {
+ public:
+ AudioPolicyMix() {}
+
+ AudioMix mMix; // Audio policy mix descriptor
+ sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
+ };
+ DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
+
+
#ifdef AUDIO_POLICY_TEST
Mutex mLock;
Condition mWaitWorkCV;