Split ManagerDefault into Manager and ManagerDefinitions
This patch splits the managerdefault into a manager and a manager
defintion library that contains all pillar elements of a policy manager.
It renames the file with the name of the main class they contains.
It splits the AudioPort into AudioPort and AudioPatch.
Change-Id: I992cf0b8aed895805cc003ba0980d2c9e92c985b
Signed-off-by: François Gaffie <francois.gaffie@intel.com>
diff --git a/services/audiopolicy/managerdefault/ApmImplDefinitions.h b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
deleted file mode 100644
index 62927da..0000000
--- a/services/audiopolicy/managerdefault/ApmImplDefinitions.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-namespace android {
-
-enum routing_strategy {
- STRATEGY_MEDIA,
- STRATEGY_PHONE,
- STRATEGY_SONIFICATION,
- STRATEGY_SONIFICATION_RESPECTFUL,
- STRATEGY_DTMF,
- STRATEGY_ENFORCED_AUDIBLE,
- STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
- STRATEGY_ACCESSIBILITY,
- STRATEGY_REROUTING,
- NUM_STRATEGIES
-};
-
-}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
deleted file mode 100644
index ce6b1e7..0000000
--- a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::AudioInputDescriptor"
-//#define LOG_NDEBUG 0
-
-#include "AudioInputDescriptor.h"
-#include "IOProfile.h"
-#include "Gains.h"
-#include "HwModule.h"
-#include <media/AudioPolicy.h>
-
-namespace android {
-
-AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0),
- mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
- mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
-{
- if (profile != NULL) {
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-void AudioInputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(mProfile != 0,
- "toAudioPortConfig() called on input with null profile %d", mIoHandle);
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SINK;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.source = mInputSource;
-}
-
-void AudioInputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
-
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioInputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.h b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
deleted file mode 100644
index ce96228..0000000
--- a/services/audiopolicy/managerdefault/AudioInputDescriptor.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include "Ports.h"
-#include <utils/Errors.h>
-#include <system/audio.h>
-#include <utils/SortedVector.h>
-
-namespace android {
-
-class IOProfile;
-class AudioMix;
-
-// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
-// and keep track of the usage of this input.
-class AudioInputDescriptor: public AudioPortConfig
-{
-public:
- AudioInputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // input handle
- audio_devices_t mDevice; // current device this input is routed to
- AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount; // number of AudioRecord clients using
- // this input
- uint32_t mOpenRefCount;
- 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
- bool mIsSoundTrigger; // used by a soundtrigger capture
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- void toAudioPort(struct audio_port *port) const;
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
deleted file mode 100644
index 4dd9316..0000000
--- a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::AudioOutputDescriptor"
-//#define LOG_NDEBUG 0
-
-#include "AudioOutputDescriptor.h"
-#include "IOProfile.h"
-#include "Gains.h"
-#include "HwModule.h"
-#include <media/AudioPolicy.h>
-
-namespace android {
-
-AudioOutputDescriptor::AudioOutputDescriptor(
- const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mLatency(0),
- mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
- mPatchHandle(0),
- mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- mRefCount[i] = 0;
- mCurVolume[i] = -1.0;
- mMuteCount[i] = 0;
- mStopTime[i] = 0;
- }
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mStrategyMutedByDevice[i] = false;
- }
- if (profile != NULL) {
- mFlags = (audio_output_flags_t)profile->mFlags;
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-audio_devices_t AudioOutputDescriptor::device() const
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
- } else {
- return mDevice;
- }
-}
-
-uint32_t AudioOutputDescriptor::latency()
-{
- if (isDuplicated()) {
- return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
- } else {
- return mLatency;
- }
-}
-
-bool AudioOutputDescriptor::sharesHwModuleWith(
- const sp<AudioOutputDescriptor> outputDesc)
-{
- if (isDuplicated()) {
- return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
- } else if (outputDesc->isDuplicated()){
- return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
- } else {
- return (mProfile->mModule == outputDesc->mProfile->mModule);
- }
-}
-
-void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
- int delta)
-{
- // forward usage count change to attached outputs
- if (isDuplicated()) {
- mOutput1->changeRefCount(stream, delta);
- mOutput2->changeRefCount(stream, delta);
- }
- if ((delta + (int)mRefCount[stream]) < 0) {
- ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
- delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-audio_devices_t AudioOutputDescriptor::supportedDevices()
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
- } else {
- return mProfile->mSupportedDevices.types() ;
- }
-}
-
-bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
- nsecs_t sysTime = 0;
- if (inPastMs != 0) {
- sysTime = systemTime();
- }
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- if (i == AUDIO_STREAM_PATCH) {
- continue;
- }
- if (isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if (mRefCount[stream] != 0) {
- return true;
- }
- if (inPastMs == 0) {
- return false;
- }
- if (sysTime == 0) {
- sysTime = systemTime();
- }
- if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
- return true;
- }
- return false;
-}
-
-void AudioOutputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
-
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
-}
-
-void AudioOutputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class =
- mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioOutputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
- result.append(buffer);
- snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", device());
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
- result.append(buffer);
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
- i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-
-
-}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
deleted file mode 100644
index f7a06ee..0000000
--- a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include "Ports.h"
-#include "ApmImplDefinitions.h"
-#include <utils/Errors.h>
-#include <utils/Timers.h>
-#include <system/audio.h>
-
-namespace android {
-
-class IOProfile;
-class AudioMix;
-
-// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
-// and keep track of the usage of this output by each audio stream type.
-class AudioOutputDescriptor: public AudioPortConfig
-{
-public:
- AudioOutputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_devices_t device() const;
- void changeRefCount(audio_stream_type_t stream, int delta);
-
- bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
- audio_devices_t supportedDevices();
- uint32_t latency();
- bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
- bool isActive(uint32_t inPastMs = 0) const;
- bool isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- void toAudioPort(struct audio_port *port) const;
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // output handle
- uint32_t mLatency; //
- audio_output_flags_t mFlags; //
- audio_devices_t mDevice; // current device this output is routed to
- AudioMix *mPolicyMix; // non NULL 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];
- sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
- sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output
- float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
- int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
- const sp<IOProfile> mProfile; // I/O profile this output derives from
- bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
- // device selection. See checkDeviceMuteStrategies()
- uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 730d32f..a110ada 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -393,8 +393,7 @@
ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch",
status);
if (status == NO_ERROR) {
- mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
+ mCallRxPatch = new AudioPatch(&patch, mUidCached);
mCallRxPatch->mAfPatchHandle = afPatchHandle;
mCallRxPatch->mUid = mUidCached;
}
@@ -436,8 +435,7 @@
ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch",
status);
if (status == NO_ERROR) {
- mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
+ mCallTxPatch = new AudioPatch(&patch, mUidCached);
mCallTxPatch->mAfPatchHandle = afPatchHandle;
mCallTxPatch->mUid = mUidCached;
}
@@ -2634,8 +2632,7 @@
status, afPatchHandle);
if (status == NO_ERROR) {
if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &newPatch, uid);
+ patchDesc = new AudioPatch(&newPatch, uid);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = newPatch;
@@ -2880,19 +2877,11 @@
// ----------------------------------------------------------------------------
// AudioPolicyManager
// ----------------------------------------------------------------------------
-
-uint32_t AudioPolicyManager::nextUniqueId()
-{
- return android_atomic_inc(&mNextUniqueId);
-}
-
uint32_t AudioPolicyManager::nextAudioPortGeneration()
{
return android_atomic_inc(&mAudioPortGeneration);
}
-int32_t volatile AudioPolicyManager::mNextUniqueId = 1;
-
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
:
#ifdef AUDIO_POLICY_TEST
@@ -3314,16 +3303,14 @@
void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc)
{
- outputDesc->mIoHandle = output;
- outputDesc->mId = nextUniqueId();
+ outputDesc->setIoHandle(output);
mOutputs.add(output, outputDesc);
nextAudioPortGeneration();
}
void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc)
{
- inputDesc->mIoHandle = input;
- inputDesc->mId = nextUniqueId();
+ inputDesc->setIoHandle(input);
mInputs.add(input, inputDesc);
nextAudioPortGeneration();
}
@@ -4827,8 +4814,7 @@
status, afPatchHandle, patch.num_sources, patch.num_sinks);
if (status == NO_ERROR) {
if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
+ patchDesc = new AudioPatch(&patch, mUidCached);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = patch;
@@ -4934,8 +4920,7 @@
status, afPatchHandle);
if (status == NO_ERROR) {
if (index < 0) {
- patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
- &patch, mUidCached);
+ patchDesc = new AudioPatch(&patch, mUidCached);
addAudioPatch(patchDesc->mHandle, patchDesc);
} else {
patchDesc->mPatch = patch;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 8308d54..279dc93 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -27,14 +27,15 @@
#include <media/AudioPolicy.h>
#include "AudioPolicyInterface.h"
-#include "Gains.h"
-#include "Ports.h"
-#include "ConfigParsingUtils.h"
-#include "Devices.h"
-#include "IOProfile.h"
-#include "HwModule.h"
-#include "AudioInputDescriptor.h"
-#include "AudioOutputDescriptor.h"
+#include <AudioGain.h>
+#include <AudioPort.h>
+#include <AudioPatch.h>
+#include <ConfigParsingUtils.h>
+#include <DeviceDescriptor.h>
+#include <IOProfile.h>
+#include <HwModule.h>
+#include <AudioInputDescriptor.h>
+#include <AudioOutputDescriptor.h>
namespace android {
@@ -208,8 +209,6 @@
// return the strategy corresponding to a given stream type
static routing_strategy getStrategy(audio_stream_type_t stream);
-
- static uint32_t nextUniqueId();
protected:
class EffectDescriptor : public RefBase
@@ -453,7 +452,6 @@
// to boost soft sounds, used to adjust volume curves accordingly
Vector < sp<HwModule> > mHwModules;
- static volatile int32_t mNextUniqueId;
volatile int32_t mAudioPortGeneration;
DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
deleted file mode 100644
index 300f35a..0000000
--- a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::ConfigParsingUtils"
-//#define LOG_NDEBUG 0
-
-#include "ConfigParsingUtils.h"
-#include <utils/Log.h>
-
-namespace android {
-
-//static
-uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name)
-{
- for (size_t i = 0; i < size; i++) {
- if (strcmp(table[i].name, name) == 0) {
- ALOGV("stringToEnum() found %s", table[i].name);
- return table[i].value;
- }
- }
- return 0;
-}
-
-//static
-const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value)
-{
- for (size_t i = 0; i < size; i++) {
- if (table[i].value == value) {
- return table[i].name;
- }
- }
- return "";
-}
-
-//static
-bool ConfigParsingUtils::stringToBool(const char *value)
-{
- return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
-
-// --- audio_policy.conf file parsing
-//static
-uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable,
- ARRAY_SIZE(sOutputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flag |= AUDIO_OUTPUT_FLAG_DIRECT;
- }
-
- return flag;
-}
-
-//static
-uint32_t ConfigParsingUtils::parseInputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sInputFlagNameToEnumTable,
- ARRAY_SIZE(sInputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- return flag;
-}
-
-//static
-audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name)
-{
- uint32_t device = 0;
-
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- device |= stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- }
- devName = strtok(NULL, "|");
- }
- return device;
-}
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.h b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
deleted file mode 100644
index 45e96d9..0000000
--- a/services/audiopolicy/managerdefault/ConfigParsingUtils.h
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include <system/audio.h>
-#include <sys/types.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-// Definitions for audio_policy.conf file parsing
-// ----------------------------------------------------------------------------
-
-struct StringToEnum {
- const char *name;
- uint32_t value;
-};
-
-#define STRING_TO_ENUM(string) { #string, string }
-#ifndef ARRAY_SIZE
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-#endif
-
-const StringToEnum sDeviceNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
-};
-
-const StringToEnum sOutputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
-};
-
-const StringToEnum sInputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
-};
-
-const StringToEnum sFormatNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
- STRING_TO_ENUM(AUDIO_FORMAT_MP3),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
- STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
- STRING_TO_ENUM(AUDIO_FORMAT_AC3),
- STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
-};
-
-const StringToEnum sOutChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const StringToEnum sInChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-const StringToEnum sGainModeNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
-};
-
-class ConfigParsingUtils
-{
-public:
- static uint32_t stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name);
- static const char *enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value);
- static bool stringToBool(const char *value);
- static uint32_t parseOutputFlagNames(char *name);
- static uint32_t parseInputFlagNames(char *name);
- static audio_devices_t parseDeviceNames(char *name);
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.cpp b/services/audiopolicy/managerdefault/Devices.cpp
deleted file mode 100644
index 574cff5..0000000
--- a/services/audiopolicy/managerdefault/Devices.cpp
+++ /dev/null
@@ -1,289 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::Devices"
-//#define LOG_NDEBUG 0
-
-#include "Devices.h"
-#include "Gains.h"
-#include "HwModule.h"
-#include "ConfigParsingUtils.h"
-
-namespace android {
-
-String8 DeviceDescriptor::emptyNameStr = String8("");
-
-DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
- AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE,
- NULL),
- mDeviceType(type), mAddress("")
-{
-
-}
-
-bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
-{
- // Devices are considered equal if they:
- // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
- // - have the same address or one device does not specify the address
- // - have the same channel mask or one device does not specify the channel mask
- return (mDeviceType == other->mDeviceType) &&
- (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
- (mChannelMask == 0 || other->mChannelMask == 0 ||
- mChannelMask == other->mChannelMask);
-}
-
-void DeviceDescriptor::loadGains(cnode *root)
-{
- AudioPort::loadGains(root);
- if (mGains.size() > 0) {
- mGains[0]->getDefaultConfig(&mGain);
- }
-}
-
-void DeviceVector::refreshTypes()
-{
- mDeviceTypes = AUDIO_DEVICE_NONE;
- for(size_t i = 0; i < size(); i++) {
- mDeviceTypes |= itemAt(i)->mDeviceType;
- }
- ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
-}
-
-ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
-{
- for(size_t i = 0; i < size(); i++) {
- if (item->equals(itemAt(i))) {
- return i;
- }
- }
- return -1;
-}
-
-ssize_t DeviceVector::add(const sp<DeviceDescriptor>& item)
-{
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ret = SortedVector::add(item);
- if (ret >= 0) {
- refreshTypes();
- }
- } else {
- ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
- ret = -1;
- }
- return ret;
-}
-
-ssize_t DeviceVector::remove(const sp<DeviceDescriptor>& item)
-{
- size_t i;
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
- } else {
- ret = SortedVector::removeAt(ret);
- if (ret >= 0) {
- refreshTypes();
- }
- }
- return ret;
-}
-
-void DeviceVector::loadDevicesFromType(audio_devices_t types)
-{
- DeviceVector deviceList;
-
- uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
- types &= ~role_bit;
-
- while (types) {
- uint32_t i = 31 - __builtin_clz(types);
- uint32_t type = 1 << i;
- types &= ~type;
- add(new DeviceDescriptor(String8("device_type"), type | role_bit));
- }
-}
-
-void DeviceVector::loadDevicesFromName(char *name,
- const DeviceVector& declaredDevices)
-{
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- if (type != AUDIO_DEVICE_NONE) {
- sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
- if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
- type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
- dev->mAddress = String8("0");
- }
- add(dev);
- } else {
- sp<DeviceDescriptor> deviceDesc =
- declaredDevices.getDeviceFromName(String8(devName));
- if (deviceDesc != 0) {
- add(deviceDesc);
- }
- }
- }
- devName = strtok(NULL, "|");
- }
-}
-
-sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, String8 address) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (address == "" || itemAt(i)->mAddress == address) {
- device = itemAt(i);
- if (itemAt(i)->mAddress == address) {
- break;
- }
- }
- }
- }
- ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
- type, address.string(), device.get());
- return device;
-}
-
-sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->getHandle() == id) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
-{
- DeviceVector devices;
- bool isOutput = audio_is_output_devices(type);
- type &= ~AUDIO_DEVICE_BIT_IN;
- for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
- bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType);
- audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN;
- if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
- devices.add(itemAt(i));
- type &= ~curType;
- ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
- itemAt(i)->mDeviceType, itemAt(i).get());
- }
- }
- return devices;
-}
-
-DeviceVector DeviceVector::getDevicesFromTypeAddr(
- audio_devices_t type, String8 address) const
-{
- DeviceVector devices;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (itemAt(i)->mAddress == address) {
- devices.add(itemAt(i));
- }
- }
- }
- return devices;
-}
-
-sp<DeviceDescriptor> DeviceVector::getDeviceFromName(const String8& name) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mName == name) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = audio_is_output_device(mDeviceType) ?
- AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
- dstConfig->ext.device.type = mDeviceType;
-
- //TODO Understand why this test is necessary. i.e. why at boot time does it crash
- // without the test?
- // This has been demonstrated to NOT be true (at start up)
- // ALOG_ASSERT(mModule != NULL);
- dstConfig->ext.device.hw_module = mModule != 0 ? mModule->mHandle : AUDIO_IO_HANDLE_NONE;
- strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-void DeviceDescriptor::toAudioPort(struct audio_port *port) const
-{
- ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
- AudioPort::toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.device.type = mDeviceType;
- port->ext.device.hw_module = mModule->mHandle;
- strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
- result.append(buffer);
- if (mId != 0) {
- snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
- ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mDeviceType));
- result.append(buffer);
- if (mAddress.size() != 0) {
- snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
- AudioPort::dump(fd, spaces);
-
- return NO_ERROR;
-}
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.h b/services/audiopolicy/managerdefault/Devices.h
deleted file mode 100644
index 7ccf559..0000000
--- a/services/audiopolicy/managerdefault/Devices.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include "Ports.h"
-#include <utils/Errors.h>
-#include <utils/String8.h>
-#include <utils/SortedVector.h>
-#include <cutils/config_utils.h>
-#include <system/audio.h>
-
-namespace android {
-
-class DeviceDescriptor: public AudioPort, public AudioPortConfig
-{
-public:
- DeviceDescriptor(const String8& name, audio_devices_t type);
-
- virtual ~DeviceDescriptor() {}
-
- bool equals(const sp<DeviceDescriptor>& other) const;
-
- // AudioPortConfig
- virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
-
- // AudioPort
- virtual void loadGains(cnode *root);
- virtual void toAudioPort(struct audio_port *port) const;
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_devices_t mDeviceType;
- String8 mAddress;
-
- static String8 emptyNameStr;
-};
-
-class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
-{
-public:
- DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
-
- ssize_t add(const sp<DeviceDescriptor>& item);
- ssize_t remove(const sp<DeviceDescriptor>& item);
- ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
-
- audio_devices_t types() const { return mDeviceTypes; }
-
- void loadDevicesFromType(audio_devices_t types);
- void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
-
- sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
- DeviceVector getDevicesFromType(audio_devices_t types) const;
- sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
- sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
- DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
- const;
-
-private:
- void refreshTypes();
- audio_devices_t mDeviceTypes;
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.cpp b/services/audiopolicy/managerdefault/Gains.cpp
deleted file mode 100644
index 98a8d1c..0000000
--- a/services/audiopolicy/managerdefault/Gains.cpp
+++ /dev/null
@@ -1,448 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::Gains"
-//#define LOG_NDEBUG 0
-
-//#define VERY_VERBOSE_LOGGING
-#ifdef VERY_VERBOSE_LOGGING
-#define ALOGVV ALOGV
-#else
-#define ALOGVV(a...) do { } while(0)
-#endif
-
-#include "Gains.h"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include <math.h>
-
-namespace android {
-
-const VolumeCurvePoint
-ApmGains::sDefaultVolumeCurve[ApmGains::VOLCNT] = {
- {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-
-const VolumeCurvePoint
-ApmGains::sDefaultMediaVolumeCurve[ApmGains::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sExtMediaSystemVolumeCurve[ApmGains::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sSpeakerMediaVolumeCurve[ApmGains::VOLCNT] = {
- {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT] = {
- {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT] = {
- {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT] = {
- {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
-};
-
-// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
-// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
-// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
-// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
-
-const VolumeCurvePoint
-ApmGains::sDefaultSystemVolumeCurve[ApmGains::VOLCNT] = {
- {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT] = {
- {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sHeadsetSystemVolumeCurve[ApmGains::VOLCNT] = {
- {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sDefaultVoiceVolumeCurve[ApmGains::VOLCNT] = {
- {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT] = {
- {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sLinearVolumeCurve[ApmGains::VOLCNT] = {
- {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sSilentVolumeCurve[ApmGains::VOLCNT] = {
- {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
-};
-
-const VolumeCurvePoint
-ApmGains::sFullScaleVolumeCurve[ApmGains::VOLCNT] = {
- {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
-};
-
-const VolumeCurvePoint *ApmGains::sVolumeProfiles[AUDIO_STREAM_CNT]
- [ApmGains::DEVICE_CATEGORY_CNT] = {
- { // AUDIO_STREAM_VOICE_CALL
- ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_SYSTEM
- ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_RING
- ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_MUSIC
- ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ALARM
- ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_NOTIFICATION
- ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_BLUETOOTH_SCO
- ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ENFORCED_AUDIBLE
- ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_DTMF
- ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_TTS
- // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
- ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ACCESSIBILITY
- ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_REROUTING
- ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_PATCH
- ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
-};
-
-//static
-audio_devices_t ApmGains::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 (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.
- // - HDMI-CEC system audio mode only output: give priority to available item in order.
- if (device & AUDIO_DEVICE_OUT_SPEAKER) {
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
- device = AUDIO_DEVICE_OUT_HDMI_ARC;
- } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
- device = AUDIO_DEVICE_OUT_AUX_LINE;
- } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
- device = AUDIO_DEVICE_OUT_SPDIF;
- } else {
- device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
- }
- }
-
- /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
- if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
- device = AUDIO_DEVICE_OUT_SPEAKER;
-
- ALOGW_IF(popcount(device) != 1,
- "getDeviceForVolume() invalid device combination: %08x",
- device);
-
- return device;
-}
-
-//static
-ApmGains::device_category ApmGains::getDeviceCategory(audio_devices_t device)
-{
- switch(getDeviceForVolume(device)) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- return ApmGains::DEVICE_CATEGORY_EARPIECE;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
- return ApmGains::DEVICE_CATEGORY_HEADSET;
- case AUDIO_DEVICE_OUT_LINE:
- case AUDIO_DEVICE_OUT_AUX_DIGITAL:
- /*USB? Remote submix?*/
- return ApmGains::DEVICE_CATEGORY_EXT_MEDIA;
- case AUDIO_DEVICE_OUT_SPEAKER:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
- case AUDIO_DEVICE_OUT_USB_ACCESSORY:
- case AUDIO_DEVICE_OUT_USB_DEVICE:
- case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
- default:
- return ApmGains::DEVICE_CATEGORY_SPEAKER;
- }
-}
-
-//static
-float ApmGains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi)
-{
- ApmGains::device_category deviceCategory = ApmGains::getDeviceCategory(device);
- const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
-
- // the volume index in the UI is relative to the min and max volume indices for this stream type
- int nbSteps = 1 + curve[ApmGains::VOLMAX].mIndex -
- curve[ApmGains::VOLMIN].mIndex;
- int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
- (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
- // find what part of the curve this index volume belongs to, or if it's out of bounds
- int segment = 0;
- if (volIdx < curve[ApmGains::VOLMIN].mIndex) { // out of bounds
- return 0.0f;
- } else if (volIdx < curve[ApmGains::VOLKNEE1].mIndex) {
- segment = 0;
- } else if (volIdx < curve[ApmGains::VOLKNEE2].mIndex) {
- segment = 1;
- } else if (volIdx <= curve[ApmGains::VOLMAX].mIndex) {
- segment = 2;
- } else { // out of bounds
- return 1.0f;
- }
-
- // linear interpolation in the attenuation table in dB
- float decibels = curve[segment].mDBAttenuation +
- ((float)(volIdx - curve[segment].mIndex)) *
- ( (curve[segment+1].mDBAttenuation -
- curve[segment].mDBAttenuation) /
- ((float)(curve[segment+1].mIndex -
- curve[segment].mIndex)) );
-
- float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
- ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
- curve[segment].mIndex, volIdx,
- curve[segment+1].mIndex,
- curve[segment].mDBAttenuation,
- decibels,
- curve[segment+1].mDBAttenuation,
- amplification);
-
- return amplification;
-}
-
-
-
-AudioGain::AudioGain(int index, bool useInChannelMask)
-{
- mIndex = index;
- mUseInChannelMask = useInChannelMask;
- memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
- config->index = mIndex;
- config->mode = mGain.mode;
- config->channel_mask = mGain.channel_mask;
- if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- config->values[0] = mGain.default_value;
- } else {
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- config->values[i] = mGain.default_value;
- }
- }
- if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- config->ramp_duration_ms = mGain.min_ramp_ms;
- }
-}
-
-status_t AudioGain::checkConfig(const struct audio_gain_config *config)
-{
- if ((config->mode & ~mGain.mode) != 0) {
- return BAD_VALUE;
- }
- if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- if ((config->values[0] < mGain.min_value) ||
- (config->values[0] > mGain.max_value)) {
- return BAD_VALUE;
- }
- } else {
- if ((config->channel_mask & ~mGain.channel_mask) != 0) {
- return BAD_VALUE;
- }
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(config->channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(config->channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- if ((config->values[i] < mGain.min_value) ||
- (config->values[i] > mGain.max_value)) {
- return BAD_VALUE;
- }
- }
- }
- if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
- (config->ramp_duration_ms > mGain.max_ramp_ms)) {
- return BAD_VALUE;
- }
- }
- return NO_ERROR;
-}
-
-void AudioGain::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-}
-
-
-// --- StreamDescriptor class implementation
-
-StreamDescriptor::StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
- mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
- device = ApmGains::getDeviceForVolume(device);
- // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
- if (mIndexCur.indexOfKey(device) < 0) {
- device = AUDIO_DEVICE_OUT_DEFAULT;
- }
- return mIndexCur.valueFor(device);
-}
-
-void StreamDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%s %02d %02d ",
- mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
- result.append(buffer);
- for (size_t i = 0; i < mIndexCur.size(); i++) {
- snprintf(buffer, SIZE, "%04x : %02d, ",
- mIndexCur.keyAt(i),
- mIndexCur.valueAt(i));
- result.append(buffer);
- }
- result.append("\n");
-
- write(fd, result.string(), result.size());
-}
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.h b/services/audiopolicy/managerdefault/Gains.h
deleted file mode 100644
index f638c8e..0000000
--- a/services/audiopolicy/managerdefault/Gains.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include <utils/Errors.h>
-#include <utils/RefBase.h>
-#include <system/audio.h>
-#include <utils/KeyedVector.h>
-
-namespace android {
-
-class VolumeCurvePoint
-{
-public:
- int mIndex;
- float mDBAttenuation;
-};
-
-class StreamDescriptor;
-
-class ApmGains
-{
-public :
- // 4 points to define the volume attenuation curve, each characterized by the volume
- // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
- // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
- enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
-
- // device categories used for volume curve management.
- enum device_category {
- DEVICE_CATEGORY_HEADSET,
- DEVICE_CATEGORY_SPEAKER,
- DEVICE_CATEGORY_EARPIECE,
- DEVICE_CATEGORY_EXT_MEDIA,
- DEVICE_CATEGORY_CNT
- };
-
- // returns the category the device belongs to with regard to volume curve management
- static ApmGains::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);
-
- static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi);
-
- // default volume curve
- static const VolumeCurvePoint sDefaultVolumeCurve[ApmGains::VOLCNT];
- // default volume curve for media strategy
- static const VolumeCurvePoint sDefaultMediaVolumeCurve[ApmGains::VOLCNT];
- // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
- static const VolumeCurvePoint sExtMediaSystemVolumeCurve[ApmGains::VOLCNT];
- // volume curve for media strategy on speakers
- static const VolumeCurvePoint sSpeakerMediaVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT];
- // volume curve for sonification strategy on speakers
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT];
- static const VolumeCurvePoint sHeadsetSystemVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sDefaultVoiceVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sLinearVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sSilentVolumeCurve[ApmGains::VOLCNT];
- static const VolumeCurvePoint sFullScaleVolumeCurve[ApmGains::VOLCNT];
- // default volume curves per stream and device category. See initializeVolumeCurves()
- static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][ApmGains::DEVICE_CATEGORY_CNT];
-};
-
-
-class AudioGain: public RefBase
-{
-public:
- AudioGain(int index, bool useInChannelMask);
- virtual ~AudioGain() {}
-
- void dump(int fd, int spaces, int index) const;
-
- void getDefaultConfig(struct audio_gain_config *config);
- status_t checkConfig(const struct audio_gain_config *config);
- int mIndex;
- struct audio_gain mGain;
- bool mUseInChannelMask;
-};
-
-
-// stream descriptor used for volume control
-class StreamDescriptor
-{
-public:
- StreamDescriptor();
-
- int getVolumeIndex(audio_devices_t device);
- void dump(int fd);
-
- int mIndexMin; // min volume index
- int mIndexMax; // max volume index
- KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
- bool mCanBeMuted; // true is the stream can be muted
-
- const VolumeCurvePoint *mVolumeCurve[ApmGains::DEVICE_CATEGORY_CNT];
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.cpp b/services/audiopolicy/managerdefault/HwModule.cpp
deleted file mode 100644
index 39dc889..0000000
--- a/services/audiopolicy/managerdefault/HwModule.cpp
+++ /dev/null
@@ -1,282 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::HwModule"
-//#define LOG_NDEBUG 0
-
-#include "HwModule.h"
-#include "IOProfile.h"
-#include "Gains.h"
-#include "ConfigParsingUtils.h"
-#include "audio_policy_conf.h"
-#include <hardware/audio.h>
-
-namespace android {
-
-HwModule::HwModule(const char *name)
- : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
- mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
-{
-}
-
-HwModule::~HwModule()
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- mOutputProfiles[i]->mSupportedDevices.clear();
- }
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- mInputProfiles[i]->mSupportedDevices.clear();
- }
- free((void *)mName);
-}
-
-status_t HwModule::loadInput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadInChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadInput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadInput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadInput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadInput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadInput() adding input Supported Devices %04x",
- profile->mSupportedDevices.types());
-
- mInputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t HwModule::loadOutput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadOutChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadOutput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadOutput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadOutput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadOutput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
- profile->mSupportedDevices.types(), profile->mFlags);
-
- mOutputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t HwModule::loadDevice(cnode *root)
-{
- cnode *node = root->first_child;
-
- audio_devices_t type = AUDIO_DEVICE_NONE;
- while (node) {
- if (strcmp(node->name, DEVICE_TYPE) == 0) {
- type = ConfigParsingUtils::parseDeviceNames((char *)node->value);
- break;
- }
- node = node->next;
- }
- if (type == AUDIO_DEVICE_NONE ||
- (!audio_is_input_device(type) && !audio_is_output_device(type))) {
- ALOGW("loadDevice() bad type %08x", type);
- return BAD_VALUE;
- }
- sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
- deviceDesc->mModule = this;
-
- node = root->first_child;
- while (node) {
- if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
- deviceDesc->mAddress = String8((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- if (audio_is_input_device(type)) {
- deviceDesc->loadInChannels((char *)node->value);
- } else {
- deviceDesc->loadOutChannels((char *)node->value);
- }
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- deviceDesc->loadGains(node);
- }
- node = node->next;
- }
-
- ALOGV("loadDevice() adding device name %s type %08x address %s",
- deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
-
- mDeclaredDevices.add(deviceDesc);
-
- return NO_ERROR;
-}
-
-status_t 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(name, device);
- devDesc->mAddress = address;
- profile->mSupportedDevices.add(devDesc);
-
- mOutputProfiles.add(profile);
-
- return NO_ERROR;
-}
-
-status_t 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 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(name, 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 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 HwModule::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " - name: %s\n", mName);
- result.append(buffer);
- snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
- result.append(buffer);
- write(fd, result.string(), result.size());
- if (mOutputProfiles.size()) {
- write(fd, " - outputs:\n", strlen(" - outputs:\n"));
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " output %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mOutputProfiles[i]->dump(fd);
- }
- }
- if (mInputProfiles.size()) {
- write(fd, " - inputs:\n", strlen(" - inputs:\n"));
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " input %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mInputProfiles[i]->dump(fd);
- }
- }
- if (mDeclaredDevices.size()) {
- write(fd, " - devices:\n", strlen(" - devices:\n"));
- for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
- mDeclaredDevices[i]->dump(fd, 4, i);
- }
- }
-}
-
-} //namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.h b/services/audiopolicy/managerdefault/HwModule.h
deleted file mode 100644
index d9e6cdf..0000000
--- a/services/audiopolicy/managerdefault/HwModule.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include "Devices.h"
-#include <utils/RefBase.h>
-#include <utils/String8.h>
-#include <utils/Errors.h>
-#include <utils/Vector.h>
-#include <system/audio.h>
-#include <cutils/config_utils.h>
-
-namespace android {
-
-class IOProfile;
-
-class HwModule : public RefBase
-{
-public:
- HwModule(const char *name);
- ~HwModule();
-
- status_t loadOutput(cnode *root);
- 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 ...)
- uint32_t mHalVersion; // audio HAL API version
- audio_module_handle_t mHandle;
- Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
- Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
- DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.cpp b/services/audiopolicy/managerdefault/IOProfile.cpp
deleted file mode 100644
index f82ff92..0000000
--- a/services/audiopolicy/managerdefault/IOProfile.cpp
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::IOProfile"
-//#define LOG_NDEBUG 0
-
-#include "IOProfile.h"
-#include "HwModule.h"
-#include "Gains.h"
-
-namespace android {
-
-IOProfile::IOProfile(const String8& name, audio_port_role_t role,
- const sp<HwModule>& module)
- : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
-{
-}
-
-IOProfile::~IOProfile()
-{
-}
-
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool IOProfile::isCompatibleProfile(audio_devices_t device,
- 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 (device != AUDIO_DEVICE_NONE) {
- // just check types if multiple devices are selected
- if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) {
- if ((mSupportedDevices.types() & device) != device) {
- return false;
- }
- } else if (mSupportedDevices.getDevice(device, address) == 0) {
- return false;
- }
- }
-
- if (samplingRate == 0) {
- return false;
- }
- uint32_t myUpdatedSamplingRate = samplingRate;
- if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
- return false;
- }
- if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
- NO_ERROR) {
- return false;
- }
-
- if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
- return false;
- }
-
- if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
- checkExactChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
- if (isRecordThread && (!audio_is_input_channel(channelMask) ||
- checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
-
- if (isPlaybackThread && (mFlags & flags) != flags) {
- return false;
- }
- // The only input flag that is allowed to be different is the fast flag.
- // An existing fast stream is compatible with a normal track request.
- // An existing normal stream is compatible with a fast track request,
- // but the fast request will be denied by AudioFlinger and converted to normal track.
- if (isRecordThread && ((mFlags ^ flags) &
- ~AUDIO_INPUT_FLAG_FAST)) {
- return false;
- }
-
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = myUpdatedSamplingRate;
- }
- return true;
-}
-
-void IOProfile::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- AudioPort::dump(fd, 4);
-
- snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " - devices:\n");
- result.append(buffer);
- write(fd, result.string(), result.size());
- for (size_t i = 0; i < mSupportedDevices.size(); i++) {
- mSupportedDevices[i]->dump(fd, 6, i);
- }
-}
-
-void IOProfile::log()
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- ALOGV(" - sampling rates: ");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- ALOGV(" %d", mSamplingRates[i]);
- }
-
- ALOGV(" - channel masks: ");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV(" 0x%04x", mChannelMasks[i]);
- }
-
- ALOGV(" - formats: ");
- for (size_t i = 0; i < mFormats.size(); i++) {
- ALOGV(" 0x%08x", mFormats[i]);
- }
-
- ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
- ALOGV(" - flags: 0x%04x\n", mFlags);
-}
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.h b/services/audiopolicy/managerdefault/IOProfile.h
deleted file mode 100644
index a9d93d1..0000000
--- a/services/audiopolicy/managerdefault/IOProfile.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include "Ports.h"
-#include "Devices.h"
-#include <utils/String8.h>
-#include <system/audio.h>
-
-namespace android {
-
-class HwModule;
-
-// the IOProfile class describes the capabilities of an output or input stream.
-// It is currently assumed that all combination of listed parameters are supported.
-// It is used by the policy manager to determine if an output or input is suitable for
-// a given use case, open/close it accordingly and connect/disconnect audio tracks
-// to/from it.
-class IOProfile : public AudioPort
-{
-public:
- IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
- virtual ~IOProfile();
-
- // This method is used for both output and input.
- // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
- // 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,
- audio_channel_mask_t channelMask,
- uint32_t flags) const;
-
- void dump(int fd);
- void log();
-
- DeviceVector mSupportedDevices; // supported devices
- // (devices this output can be routed to)
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.cpp b/services/audiopolicy/managerdefault/Ports.cpp
deleted file mode 100644
index 326f9c5..0000000
--- a/services/audiopolicy/managerdefault/Ports.cpp
+++ /dev/null
@@ -1,848 +0,0 @@
-/*
- * Copyright (C) 2015 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 "APM::Ports"
-//#define LOG_NDEBUG 0
-
-#include "Ports.h"
-#include "HwModule.h"
-#include "Gains.h"
-#include "ConfigParsingUtils.h"
-#include "audio_policy_conf.h"
-
-namespace android {
-
-int32_t volatile AudioPort::mNextUniqueId = 1;
-
-// --- AudioPort class implementation
-
-AudioPort::AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module) :
- mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
-{
- mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
- ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
-}
-
-void AudioPort::attach(const sp<HwModule>& module) {
- mId = android_atomic_inc(&mNextUniqueId);
- mModule = module;
-}
-
-void AudioPort::toAudioPort(struct audio_port *port) const
-{
- port->role = mRole;
- port->type = mType;
- strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
- unsigned int i;
- for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
- if (mSamplingRates[i] != 0) {
- port->sample_rates[i] = mSamplingRates[i];
- }
- }
- port->num_sample_rates = i;
- for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
- if (mChannelMasks[i] != 0) {
- port->channel_masks[i] = mChannelMasks[i];
- }
- }
- port->num_channel_masks = i;
- for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
- if (mFormats[i] != 0) {
- port->formats[i] = mFormats[i];
- }
- }
- port->num_formats = i;
-
- ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
-
- for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
- port->gains[i] = mGains[i]->mGain;
- }
- port->num_gains = i;
-}
-
-void AudioPort::importAudioPort(const sp<AudioPort> port) {
- for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
- const uint32_t rate = port->mSamplingRates.itemAt(k);
- if (rate != 0) { // skip "dynamic" rates
- bool hasRate = false;
- for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
- if (rate == mSamplingRates.itemAt(l)) {
- hasRate = true;
- break;
- }
- }
- if (!hasRate) { // never import a sampling rate twice
- mSamplingRates.add(rate);
- }
- }
- }
- for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
- const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
- if (mask != 0) { // skip "dynamic" masks
- bool hasMask = false;
- for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
- if (mask == mChannelMasks.itemAt(l)) {
- hasMask = true;
- break;
- }
- }
- if (!hasMask) { // never import a channel mask twice
- mChannelMasks.add(mask);
- }
- }
- }
- for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
- const audio_format_t format = port->mFormats.itemAt(k);
- if (format != 0) { // skip "dynamic" formats
- bool hasFormat = false;
- for (size_t l = 0 ; l < mFormats.size() ; l++) {
- if (format == mFormats.itemAt(l)) {
- hasFormat = true;
- break;
- }
- }
- if (!hasFormat) { // never import a channel mask twice
- mFormats.add(format);
- }
- }
- }
- for (size_t k = 0 ; k < port->mGains.size() ; k++) {
- sp<AudioGain> gain = port->mGains.itemAt(k);
- if (gain != 0) {
- bool hasGain = false;
- for (size_t l = 0 ; l < mGains.size() ; l++) {
- if (gain == mGains.itemAt(l)) {
- hasGain = true;
- break;
- }
- }
- if (!hasGain) { // never import a gain twice
- mGains.add(gain);
- }
- }
- }
-}
-
-void AudioPort::clearCapabilities() {
- mChannelMasks.clear();
- mFormats.clear();
- mSamplingRates.clear();
- mGains.clear();
-}
-
-void AudioPort::loadSamplingRates(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
- // rates should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mSamplingRates.add(0);
- return;
- }
-
- while (str != NULL) {
- uint32_t rate = atoi(str);
- if (rate != 0) {
- ALOGV("loadSamplingRates() adding rate %d", rate);
- mSamplingRates.add(rate);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPort::loadFormats(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mFormats indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mFormats.add(AUDIO_FORMAT_DEFAULT);
- return;
- }
-
- while (str != NULL) {
- audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- str);
- if (format != AUDIO_FORMAT_DEFAULT) {
- mFormats.add(format);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPort::loadInChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadInChannels() %s", name);
-
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- ALOGV("loadInChannels() adding channelMask %04x", channelMask);
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPort::loadOutChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadOutChannels() %s", name);
-
- // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
- // masks should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-audio_gain_mode_t AudioPort::loadGainMode(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadGainMode() %s", name);
- audio_gain_mode_t mode = 0;
- while (str != NULL) {
- mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable,
- ARRAY_SIZE(sGainModeNameToEnumTable),
- str);
- str = strtok(NULL, "|");
- }
- return mode;
-}
-
-void AudioPort::loadGain(cnode *root, int index)
-{
- cnode *node = root->first_child;
-
- sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
-
- while (node) {
- if (strcmp(node->name, GAIN_MODE) == 0) {
- gain->mGain.mode = loadGainMode((char *)node->value);
- } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
- if (mUseInChannelMask) {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- (char *)node->value);
- } else {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- (char *)node->value);
- }
- } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
- gain->mGain.min_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
- gain->mGain.max_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
- gain->mGain.default_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
- gain->mGain.step_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
- gain->mGain.min_ramp_ms = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
- gain->mGain.max_ramp_ms = atoi((char *)node->value);
- }
- node = node->next;
- }
-
- ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
- gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
-
- if (gain->mGain.mode == 0) {
- return;
- }
- mGains.add(gain);
-}
-
-void AudioPort::loadGains(cnode *root)
-{
- cnode *node = root->first_child;
- int index = 0;
- while (node) {
- ALOGV("loadGains() loading gain %s", node->name);
- loadGain(node, index++);
- node = node->next;
- }
-}
-
-status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if (mSamplingRates[i] == samplingRate) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- // Search for the closest supported sampling rate that is above (preferred)
- // or below (acceptable) the desired sampling rate, within a permitted ratio.
- // The sampling rates do not need to be sorted in ascending order.
- ssize_t maxBelow = -1;
- ssize_t minAbove = -1;
- uint32_t candidate;
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- candidate = mSamplingRates[i];
- if (candidate == samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- // candidate < desired
- if (candidate < samplingRate) {
- if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
- maxBelow = i;
- }
- // candidate > desired
- } else {
- if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
- minAbove = i;
- }
- }
- }
- // This uses hard-coded knowledge about AudioFlinger resampling ratios.
- // TODO Move these assumptions out.
- static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
- static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
- // due to approximation by an int32_t of the
- // phase increments
- // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
- if (minAbove >= 0) {
- candidate = mSamplingRates[minAbove];
- if (candidate / kMaxDownSampleRatio <= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // But if we have to up-sample from a lower sampling rate, that's OK.
- if (maxBelow >= 0) {
- candidate = mSamplingRates[maxBelow];
- if (candidate * kMaxUpSampleRatio >= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // leave updatedSamplingRate unmodified
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- if (mChannelMasks[i] == channelMask) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
- const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- // FIXME Does not handle multi-channel automatic conversions yet
- audio_channel_mask_t supported = mChannelMasks[i];
- if (supported == channelMask) {
- return NO_ERROR;
- }
- if (isRecordThread) {
- // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
- // FIXME Abstract this out to a table.
- if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
- && channelMask == AUDIO_CHANNEL_IN_MONO) ||
- (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
- || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
- return NO_ERROR;
- }
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPort::checkFormat(audio_format_t format) const
-{
- if (mFormats.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if (mFormats[i] == format) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-
-uint32_t AudioPort::pickSamplingRate() const
-{
- // special case for uninitialized dynamic profile
- if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
- return 0;
- }
-
- // For direct outputs, pick minimum sampling rate: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t samplingRate = UINT_MAX;
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return (samplingRate == UINT_MAX) ? 0 : samplingRate;
- }
-
- uint32_t samplingRate = 0;
- uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
-
- // For mixed output and inputs, use max mixer sampling rates. Do not
- // limit sampling rate otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxRate = UINT_MAX;
- }
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return samplingRate;
-}
-
-audio_channel_mask_t AudioPort::pickChannelMask() const
-{
- // special case for uninitialized dynamic profile
- if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
- return AUDIO_CHANNEL_NONE;
- }
- audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
-
- // For direct outputs, pick minimum channel count: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t channelCount = UINT_MAX;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount < channelCount) && (cnlCount > 0)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
- }
-
- uint32_t channelCount = 0;
- uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
-
- // For mixed output and inputs, use max mixer channel count. Do not
- // limit channel count otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxCount = UINT_MAX;
- }
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
-}
-
-/* format in order of increasing preference */
-const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
- AUDIO_FORMAT_DEFAULT,
- AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_FORMAT_PCM_8_24_BIT,
- AUDIO_FORMAT_PCM_24_BIT_PACKED,
- AUDIO_FORMAT_PCM_32_BIT,
- AUDIO_FORMAT_PCM_FLOAT,
-};
-
-int AudioPort::compareFormats(audio_format_t format1,
- audio_format_t format2)
-{
- // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
- // compressed format and better than any PCM format. This is by design of pickFormat()
- if (!audio_is_linear_pcm(format1)) {
- if (!audio_is_linear_pcm(format2)) {
- return 0;
- }
- return 1;
- }
- if (!audio_is_linear_pcm(format2)) {
- return -1;
- }
-
- int index1 = -1, index2 = -1;
- for (size_t i = 0;
- (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
- i ++) {
- if (sPcmFormatCompareTable[i] == format1) {
- index1 = i;
- }
- if (sPcmFormatCompareTable[i] == format2) {
- index2 = i;
- }
- }
- // format1 not found => index1 < 0 => format2 > format1
- // format2 not found => index2 < 0 => format2 < format1
- return index1 - index2;
-}
-
-audio_format_t AudioPort::pickFormat() const
-{
- // special case for uninitialized dynamic profile
- if (mFormats.size() == 1 && mFormats[0] == 0) {
- return AUDIO_FORMAT_DEFAULT;
- }
-
- audio_format_t format = AUDIO_FORMAT_DEFAULT;
- audio_format_t bestFormat =
- AudioPort::sPcmFormatCompareTable[
- ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
- // For mixed output and inputs, use best mixer output format. Do not
- // limit format otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) ||
- ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
- bestFormat = AUDIO_FORMAT_INVALID;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if ((compareFormats(mFormats[i], format) > 0) &&
- (compareFormats(mFormats[i], bestFormat) <= 0)) {
- format = mFormats[i];
- }
- }
- return format;
-}
-
-status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
- int index) const
-{
- if (index < 0 || (size_t)index >= mGains.size()) {
- return BAD_VALUE;
- }
- return mGains[index]->checkConfig(gainConfig);
-}
-
-void AudioPort::dump(int fd, int spaces) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- if (mName.size() != 0) {
- snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
- result.append(buffer);
- }
-
- if (mSamplingRates.size() != 0) {
- snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- if (i == 0 && mSamplingRates[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
- }
- result.append(buffer);
- result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mChannelMasks.size() != 0) {
- snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
-
- if (i == 0 && mChannelMasks[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
- }
- result.append(buffer);
- result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mFormats.size() != 0) {
- snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mFormats.size(); i++) {
- const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- mFormats[i]);
- if (i == 0 && strcmp(formatStr, "") == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%s", formatStr);
- }
- result.append(buffer);
- result.append(i == (mFormats.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
- write(fd, result.string(), result.size());
- if (mGains.size() != 0) {
- snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
- write(fd, buffer, strlen(buffer) + 1);
- result.append(buffer);
- for (size_t i = 0; i < mGains.size(); i++) {
- mGains[i]->dump(fd, spaces + 2, i);
- }
- }
-}
-
-
-// --- AudioPortConfig class implementation
-
-AudioPortConfig::AudioPortConfig()
-{
- mSamplingRate = 0;
- mChannelMask = AUDIO_CHANNEL_NONE;
- mFormat = AUDIO_FORMAT_INVALID;
- mGain.index = -1;
-}
-
-status_t AudioPortConfig::applyAudioPortConfig(
- const struct audio_port_config *config,
- struct audio_port_config *backupConfig)
-{
- struct audio_port_config localBackupConfig;
- status_t status = NO_ERROR;
-
- localBackupConfig.config_mask = config->config_mask;
- toAudioPortConfig(&localBackupConfig);
-
- sp<AudioPort> audioport = getAudioPort();
- if (audioport == 0) {
- status = NO_INIT;
- goto exit;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- status = audioport->checkExactSamplingRate(config->sample_rate);
- if (status != NO_ERROR) {
- goto exit;
- }
- mSamplingRate = config->sample_rate;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- status = audioport->checkExactChannelMask(config->channel_mask);
- if (status != NO_ERROR) {
- goto exit;
- }
- mChannelMask = config->channel_mask;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- status = audioport->checkFormat(config->format);
- if (status != NO_ERROR) {
- goto exit;
- }
- mFormat = config->format;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- status = audioport->checkGain(&config->gain, config->gain.index);
- if (status != NO_ERROR) {
- goto exit;
- }
- mGain = config->gain;
- }
-
-exit:
- if (status != NO_ERROR) {
- applyAudioPortConfig(&localBackupConfig);
- }
- if (backupConfig != NULL) {
- *backupConfig = localBackupConfig;
- }
- return status;
-}
-
-void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = mSamplingRate;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- } else {
- dstConfig->sample_rate = 0;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = mChannelMask;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- } else {
- dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = mFormat;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
- dstConfig->format = srcConfig->format;
- }
- } else {
- dstConfig->format = AUDIO_FORMAT_INVALID;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = mGain;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
- dstConfig->gain = srcConfig->gain;
- }
- } else {
- dstConfig->gain.index = -1;
- }
- if (dstConfig->gain.index != -1) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- } else {
- dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
- }
-}
-
-
-// --- AudioPatch class implementation
-
-AudioPatch::AudioPatch(audio_patch_handle_t handle,
- const struct audio_patch *patch, uid_t uid) :
- mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0)
-{}
-
-status_t AudioPatch::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sources; i++) {
- if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sources[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
- }
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sinks; i++) {
- if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sinks[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
- }
- result.append(buffer);
- }
-
- write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.h b/services/audiopolicy/managerdefault/Ports.h
deleted file mode 100644
index 6e0e2fe..0000000
--- a/services/audiopolicy/managerdefault/Ports.h
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (C) 2015 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.
- */
-
-#pragma once
-
-#include <utils/String8.h>
-#include <utils/Vector.h>
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
-#include <system/audio.h>
-#include <cutils/config_utils.h>
-
-#define MAX_MIXER_SAMPLING_RATE 48000
-#define MAX_MIXER_CHANNEL_COUNT 8
-
-namespace android {
-
-class HwModule;
-class AudioGain;
-
-class AudioPort: public virtual RefBase
-{
-public:
- AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module);
- virtual ~AudioPort() {}
-
- audio_port_handle_t getHandle() { return mId; }
-
- void attach(const sp<HwModule>& module);
- bool isAttached() { return mId != 0; }
-
- virtual void toAudioPort(struct audio_port *port) const;
-
- void importAudioPort(const sp<AudioPort> port);
- void clearCapabilities();
-
- void loadSamplingRates(char *name);
- void loadFormats(char *name);
- void loadOutChannels(char *name);
- void loadInChannels(char *name);
-
- audio_gain_mode_t loadGainMode(char *name);
- void loadGain(cnode *root, int index);
- virtual void loadGains(cnode *root);
-
- // searches for an exact match
- status_t checkExactSamplingRate(uint32_t samplingRate) const;
- // searches for a compatible match, and returns the best match via updatedSamplingRate
- status_t checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const;
- // searches for an exact match
- status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
- // searches for a compatible match, currently implemented for input channel masks only
- status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
- status_t checkFormat(audio_format_t format) const;
- status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
-
- uint32_t pickSamplingRate() const;
- audio_channel_mask_t pickChannelMask() const;
- audio_format_t pickFormat() const;
-
- static const audio_format_t sPcmFormatCompareTable[];
- static int compareFormats(audio_format_t format1, audio_format_t format2);
-
- void dump(int fd, int spaces) const;
-
- String8 mName;
- audio_port_type_t mType;
- audio_port_role_t mRole;
- bool mUseInChannelMask;
- // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
- // indicates the supported parameters should be read from the output stream
- // after it is opened for the first time
- Vector <uint32_t> mSamplingRates; // supported sampling rates
- Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
- Vector <audio_format_t> mFormats; // supported audio formats
- Vector < sp<AudioGain> > mGains; // gain controllers
- sp<HwModule> mModule; // audio HW module exposing this I/O stream
- uint32_t mFlags; // attribute flags (e.g primary output,
- // direct output...).
-
-
-protected:
- //TODO - clarify the role of mId in this case, both an "attached" indicator
- // and a unique ID for identifying a port to the (upcoming) selection API,
- // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
- audio_port_handle_t mId;
-
-private:
- static volatile int32_t mNextUniqueId;
-};
-
-class AudioPortConfig: public virtual RefBase
-{
-public:
- AudioPortConfig();
- virtual ~AudioPortConfig() {}
-
- status_t applyAudioPortConfig(const struct audio_port_config *config,
- struct audio_port_config *backupConfig = NULL);
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const = 0;
- virtual sp<AudioPort> getAudioPort() const = 0;
- uint32_t mSamplingRate;
- audio_format_t mFormat;
- audio_channel_mask_t mChannelMask;
- struct audio_gain_config mGain;
-};
-
-
-class AudioPatch: public RefBase
-{
-public:
- AudioPatch(audio_patch_handle_t handle, const struct audio_patch *patch, uid_t uid);
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_patch_handle_t mHandle;
- struct audio_patch mPatch;
- uid_t mUid;
- audio_patch_handle_t mAfPatchHandle;
-};
-
-}; // namespace android
diff --git a/services/audiopolicy/managerdefault/audio_policy_conf.h b/services/audiopolicy/managerdefault/audio_policy_conf.h
deleted file mode 100644
index 441bf7b..0000000
--- a/services/audiopolicy/managerdefault/audio_policy_conf.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (C) 2012 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.
- */
-
-#pragma once
-
-/////////////////////////////////////////////////
-// Definitions for audio policy configuration file (audio_policy.conf)
-/////////////////////////////////////////////////
-
-#define AUDIO_HARDWARE_MODULE_ID_MAX_LEN 32
-
-#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf"
-#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf"
-
-// global configuration
-#define GLOBAL_CONFIG_TAG "global_configuration"
-
-#define ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices"
-#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device"
-#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices"
-#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled"
-#define AUDIO_HAL_VERSION_TAG "audio_hal_version"
-
-// hw modules descriptions
-#define AUDIO_HW_MODULE_TAG "audio_hw_modules"
-
-#define OUTPUTS_TAG "outputs"
-#define INPUTS_TAG "inputs"
-
-#define SAMPLING_RATES_TAG "sampling_rates"
-#define FORMATS_TAG "formats"
-#define CHANNELS_TAG "channel_masks"
-#define DEVICES_TAG "devices"
-#define FLAGS_TAG "flags"
-
-#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and
- // "formats" in outputs descriptors indicating that supported
- // values should be queried after opening the output.
-
-#define DEVICES_TAG "devices"
-#define DEVICE_TYPE "type"
-#define DEVICE_ADDRESS "address"
-
-#define MIXERS_TAG "mixers"
-#define MIXER_TYPE "type"
-#define MIXER_TYPE_MUX "mux"
-#define MIXER_TYPE_MIX "mix"
-
-#define GAINS_TAG "gains"
-#define GAIN_MODE "mode"
-#define GAIN_CHANNELS "channel_mask"
-#define GAIN_MIN_VALUE "min_value_mB"
-#define GAIN_MAX_VALUE "max_value_mB"
-#define GAIN_DEFAULT_VALUE "default_value_mB"
-#define GAIN_STEP_VALUE "step_value_mB"
-#define GAIN_MIN_RAMP_MS "min_ramp_ms"
-#define GAIN_MAX_RAMP_MS "max_ramp_ms"