audioflinger: move legacy audio hw/policy out to libhardware_legacy
Change-Id: I4adcec73d3c08bcbe15bb19e1ba2ff18b195af45
Signed-off-by: Dima Zavin <dima@android.com>
diff --git a/services/audioflinger/A2dpAudioInterface.cpp b/services/audioflinger/A2dpAudioInterface.cpp
deleted file mode 100644
index d926cb1..0000000
--- a/services/audioflinger/A2dpAudioInterface.cpp
+++ /dev/null
@@ -1,498 +0,0 @@
-/*
- * Copyright (C) 2008 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.
- */
-
-#include <math.h>
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "A2dpAudioInterface"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include "A2dpAudioInterface.h"
-#include "audio/liba2dp.h"
-#include <hardware_legacy/power.h>
-
-namespace android {
-
-static const char *sA2dpWakeLock = "A2dpOutputStream";
-#define MAX_WRITE_RETRIES 5
-
-// ----------------------------------------------------------------------------
-
-//AudioHardwareInterface* A2dpAudioInterface::createA2dpInterface()
-//{
-// AudioHardwareInterface* hw = 0;
-//
-// hw = AudioHardwareInterface::create();
-// LOGD("new A2dpAudioInterface(hw: %p)", hw);
-// hw = new A2dpAudioInterface(hw);
-// return hw;
-//}
-
-A2dpAudioInterface::A2dpAudioInterface(AudioHardwareInterface* hw) :
- mOutput(0), mHardwareInterface(hw), mBluetoothEnabled(true), mSuspended(false)
-{
-}
-
-A2dpAudioInterface::~A2dpAudioInterface()
-{
- closeOutputStream((AudioStreamOut *)mOutput);
- delete mHardwareInterface;
-}
-
-status_t A2dpAudioInterface::initCheck()
-{
- if (mHardwareInterface == 0) return NO_INIT;
- return mHardwareInterface->initCheck();
-}
-
-AudioStreamOut* A2dpAudioInterface::openOutputStream(
- uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
- if (!AudioSystem::isA2dpDevice((AudioSystem::audio_devices)devices)) {
- LOGV("A2dpAudioInterface::openOutputStream() open HW device: %x", devices);
- return mHardwareInterface->openOutputStream(devices, format, channels, sampleRate, status);
- }
-
- status_t err = 0;
-
- // only one output stream allowed
- if (mOutput) {
- if (status)
- *status = -1;
- return NULL;
- }
-
- // create new output stream
- A2dpAudioStreamOut* out = new A2dpAudioStreamOut();
- if ((err = out->set(devices, format, channels, sampleRate)) == NO_ERROR) {
- mOutput = out;
- mOutput->setBluetoothEnabled(mBluetoothEnabled);
- mOutput->setSuspended(mSuspended);
- } else {
- delete out;
- }
-
- if (status)
- *status = err;
- return mOutput;
-}
-
-void A2dpAudioInterface::closeOutputStream(AudioStreamOut* out) {
- if (mOutput == 0 || mOutput != out) {
- mHardwareInterface->closeOutputStream(out);
- }
- else {
- delete mOutput;
- mOutput = 0;
- }
-}
-
-
-AudioStreamIn* A2dpAudioInterface::openInputStream(
- uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status,
- AudioSystem::audio_in_acoustics acoustics)
-{
- return mHardwareInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
-}
-
-void A2dpAudioInterface::closeInputStream(AudioStreamIn* in)
-{
- return mHardwareInterface->closeInputStream(in);
-}
-
-status_t A2dpAudioInterface::setMode(int mode)
-{
- return mHardwareInterface->setMode(mode);
-}
-
-status_t A2dpAudioInterface::setMicMute(bool state)
-{
- return mHardwareInterface->setMicMute(state);
-}
-
-status_t A2dpAudioInterface::getMicMute(bool* state)
-{
- return mHardwareInterface->getMicMute(state);
-}
-
-status_t A2dpAudioInterface::setParameters(const String8& keyValuePairs)
-{
- AudioParameter param = AudioParameter(keyValuePairs);
- String8 value;
- String8 key;
- status_t status = NO_ERROR;
-
- LOGV("setParameters() %s", keyValuePairs.string());
-
- key = "bluetooth_enabled";
- if (param.get(key, value) == NO_ERROR) {
- mBluetoothEnabled = (value == "true");
- if (mOutput) {
- mOutput->setBluetoothEnabled(mBluetoothEnabled);
- }
- param.remove(key);
- }
- key = String8("A2dpSuspended");
- if (param.get(key, value) == NO_ERROR) {
- mSuspended = (value == "true");
- if (mOutput) {
- mOutput->setSuspended(mSuspended);
- }
- param.remove(key);
- }
-
- if (param.size()) {
- status_t hwStatus = mHardwareInterface->setParameters(param.toString());
- if (status == NO_ERROR) {
- status = hwStatus;
- }
- }
-
- return status;
-}
-
-String8 A2dpAudioInterface::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- AudioParameter a2dpParam = AudioParameter();
- String8 value;
- String8 key;
-
- key = "bluetooth_enabled";
- if (param.get(key, value) == NO_ERROR) {
- value = mBluetoothEnabled ? "true" : "false";
- a2dpParam.add(key, value);
- param.remove(key);
- }
- key = "A2dpSuspended";
- if (param.get(key, value) == NO_ERROR) {
- value = mSuspended ? "true" : "false";
- a2dpParam.add(key, value);
- param.remove(key);
- }
-
- String8 keyValuePairs = a2dpParam.toString();
-
- if (param.size()) {
- if (keyValuePairs != "") {
- keyValuePairs += ";";
- }
- keyValuePairs += mHardwareInterface->getParameters(param.toString());
- }
-
- LOGV("getParameters() %s", keyValuePairs.string());
- return keyValuePairs;
-}
-
-size_t A2dpAudioInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
-{
- return mHardwareInterface->getInputBufferSize(sampleRate, format, channelCount);
-}
-
-status_t A2dpAudioInterface::setVoiceVolume(float v)
-{
- return mHardwareInterface->setVoiceVolume(v);
-}
-
-status_t A2dpAudioInterface::setMasterVolume(float v)
-{
- return mHardwareInterface->setMasterVolume(v);
-}
-
-status_t A2dpAudioInterface::dump(int fd, const Vector<String16>& args)
-{
- return mHardwareInterface->dumpState(fd, args);
-}
-
-// ----------------------------------------------------------------------------
-
-A2dpAudioInterface::A2dpAudioStreamOut::A2dpAudioStreamOut() :
- mFd(-1), mStandby(true), mStartCount(0), mRetryCount(0), mData(NULL),
- // assume BT enabled to start, this is safe because its only the
- // enabled->disabled transition we are worried about
- mBluetoothEnabled(true), mDevice(0), mClosing(false), mSuspended(false)
-{
- // use any address by default
- strcpy(mA2dpAddress, "00:00:00:00:00:00");
- init();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::set(
- uint32_t device, int *pFormat, uint32_t *pChannels, uint32_t *pRate)
-{
- int lFormat = pFormat ? *pFormat : 0;
- uint32_t lChannels = pChannels ? *pChannels : 0;
- uint32_t lRate = pRate ? *pRate : 0;
-
- LOGD("A2dpAudioStreamOut::set %x, %d, %d, %d\n", device, lFormat, lChannels, lRate);
-
- // fix up defaults
- if (lFormat == 0) lFormat = format();
- if (lChannels == 0) lChannels = channels();
- if (lRate == 0) lRate = sampleRate();
-
- // check values
- if ((lFormat != format()) ||
- (lChannels != channels()) ||
- (lRate != sampleRate())){
- if (pFormat) *pFormat = format();
- if (pChannels) *pChannels = channels();
- if (pRate) *pRate = sampleRate();
- return BAD_VALUE;
- }
-
- if (pFormat) *pFormat = lFormat;
- if (pChannels) *pChannels = lChannels;
- if (pRate) *pRate = lRate;
-
- mDevice = device;
- mBufferDurationUs = ((bufferSize() * 1000 )/ frameSize() / sampleRate()) * 1000;
- return NO_ERROR;
-}
-
-A2dpAudioInterface::A2dpAudioStreamOut::~A2dpAudioStreamOut()
-{
- LOGV("A2dpAudioStreamOut destructor");
- close();
- LOGV("A2dpAudioStreamOut destructor returning from close()");
-}
-
-ssize_t A2dpAudioInterface::A2dpAudioStreamOut::write(const void* buffer, size_t bytes)
-{
- status_t status = -1;
- {
- Mutex::Autolock lock(mLock);
-
- size_t remaining = bytes;
-
- if (!mBluetoothEnabled || mClosing || mSuspended) {
- LOGV("A2dpAudioStreamOut::write(), but bluetooth disabled \
- mBluetoothEnabled %d, mClosing %d, mSuspended %d",
- mBluetoothEnabled, mClosing, mSuspended);
- goto Error;
- }
-
- if (mStandby) {
- acquire_wake_lock (PARTIAL_WAKE_LOCK, sA2dpWakeLock);
- mStandby = false;
- mLastWriteTime = systemTime();
- }
-
- status = init();
- if (status < 0)
- goto Error;
-
- int retries = MAX_WRITE_RETRIES;
- while (remaining > 0 && retries) {
- status = a2dp_write(mData, buffer, remaining);
- if (status < 0) {
- LOGE("a2dp_write failed err: %d\n", status);
- goto Error;
- }
- if (status == 0) {
- retries--;
- }
- remaining -= status;
- buffer = (char *)buffer + status;
- }
-
- // if A2DP sink runs abnormally fast, sleep a little so that audioflinger mixer thread
- // does no spin and starve other threads.
- // NOTE: It is likely that the A2DP headset is being disconnected
- nsecs_t now = systemTime();
- if ((uint32_t)ns2us(now - mLastWriteTime) < (mBufferDurationUs >> 2)) {
- LOGV("A2DP sink runs too fast");
- usleep(mBufferDurationUs - (uint32_t)ns2us(now - mLastWriteTime));
- }
- mLastWriteTime = now;
- return bytes;
-
- }
-Error:
-
- standby();
-
- // Simulate audio output timing in case of error
- usleep(mBufferDurationUs);
-
- return status;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::init()
-{
- if (!mData) {
- status_t status = a2dp_init(44100, 2, &mData);
- if (status < 0) {
- LOGE("a2dp_init failed err: %d\n", status);
- mData = NULL;
- return status;
- }
- a2dp_set_sink(mData, mA2dpAddress);
- }
-
- return 0;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::standby()
-{
- Mutex::Autolock lock(mLock);
- return standby_l();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::standby_l()
-{
- int result = NO_ERROR;
-
- if (!mStandby) {
- LOGV_IF(mClosing || !mBluetoothEnabled, "Standby skip stop: closing %d enabled %d",
- mClosing, mBluetoothEnabled);
- if (!mClosing && mBluetoothEnabled) {
- result = a2dp_stop(mData);
- }
- release_wake_lock(sA2dpWakeLock);
- mStandby = true;
- }
-
- return result;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setParameters(const String8& keyValuePairs)
-{
- AudioParameter param = AudioParameter(keyValuePairs);
- String8 value;
- String8 key = String8("a2dp_sink_address");
- status_t status = NO_ERROR;
- int device;
- LOGV("A2dpAudioStreamOut::setParameters() %s", keyValuePairs.string());
-
- if (param.get(key, value) == NO_ERROR) {
- if (value.length() != strlen("00:00:00:00:00:00")) {
- status = BAD_VALUE;
- } else {
- setAddress(value.string());
- }
- param.remove(key);
- }
- key = String8("closing");
- if (param.get(key, value) == NO_ERROR) {
- mClosing = (value == "true");
- if (mClosing) {
- standby();
- }
- param.remove(key);
- }
- key = AudioParameter::keyRouting;
- if (param.getInt(key, device) == NO_ERROR) {
- if (AudioSystem::isA2dpDevice((AudioSystem::audio_devices)device)) {
- mDevice = device;
- status = NO_ERROR;
- } else {
- status = BAD_VALUE;
- }
- param.remove(key);
- }
-
- if (param.size()) {
- status = BAD_VALUE;
- }
- return status;
-}
-
-String8 A2dpAudioInterface::A2dpAudioStreamOut::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- String8 value;
- String8 key = String8("a2dp_sink_address");
-
- if (param.get(key, value) == NO_ERROR) {
- value = mA2dpAddress;
- param.add(key, value);
- }
- key = AudioParameter::keyRouting;
- if (param.get(key, value) == NO_ERROR) {
- param.addInt(key, (int)mDevice);
- }
-
- LOGV("A2dpAudioStreamOut::getParameters() %s", param.toString().string());
- return param.toString();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setAddress(const char* address)
-{
- Mutex::Autolock lock(mLock);
-
- if (strlen(address) != strlen("00:00:00:00:00:00"))
- return -EINVAL;
-
- strcpy(mA2dpAddress, address);
- if (mData)
- a2dp_set_sink(mData, mA2dpAddress);
-
- return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setBluetoothEnabled(bool enabled)
-{
- LOGD("setBluetoothEnabled %d", enabled);
-
- Mutex::Autolock lock(mLock);
-
- mBluetoothEnabled = enabled;
- if (!enabled) {
- return close_l();
- }
- return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::setSuspended(bool onOff)
-{
- LOGV("setSuspended %d", onOff);
- mSuspended = onOff;
- standby();
- return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::close()
-{
- Mutex::Autolock lock(mLock);
- LOGV("A2dpAudioStreamOut::close() calling close_l()");
- return close_l();
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::close_l()
-{
- standby_l();
- if (mData) {
- LOGV("A2dpAudioStreamOut::close_l() calling a2dp_cleanup(mData)");
- a2dp_cleanup(mData);
- mData = NULL;
- }
- return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::dump(int fd, const Vector<String16>& args)
-{
- return NO_ERROR;
-}
-
-status_t A2dpAudioInterface::A2dpAudioStreamOut::getRenderPosition(uint32_t *driverFrames)
-{
- //TODO: enable when supported by driver
- return INVALID_OPERATION;
-}
-
-}; // namespace android
diff --git a/services/audioflinger/A2dpAudioInterface.h b/services/audioflinger/A2dpAudioInterface.h
deleted file mode 100644
index dbe2c6a..0000000
--- a/services/audioflinger/A2dpAudioInterface.h
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef A2DP_AUDIO_HARDWARE_H
-#define A2DP_AUDIO_HARDWARE_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/threads.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-
-namespace android {
-
-class A2dpAudioInterface : public AudioHardwareBase
-{
- class A2dpAudioStreamOut;
-
-public:
- A2dpAudioInterface(AudioHardwareInterface* hw);
- virtual ~A2dpAudioInterface();
- virtual status_t initCheck();
-
- virtual status_t setVoiceVolume(float volume);
- virtual status_t setMasterVolume(float volume);
-
- virtual status_t setMode(int mode);
-
- // mic mute
- virtual status_t setMicMute(bool state);
- virtual status_t getMicMute(bool* state);
-
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
-
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
-
- // create I/O streams
- virtual AudioStreamOut* openOutputStream(
- uint32_t devices,
- int *format=0,
- uint32_t *channels=0,
- uint32_t *sampleRate=0,
- status_t *status=0);
- virtual void closeOutputStream(AudioStreamOut* out);
-
- virtual AudioStreamIn* openInputStream(
- uint32_t devices,
- int *format,
- uint32_t *channels,
- uint32_t *sampleRate,
- status_t *status,
- AudioSystem::audio_in_acoustics acoustics);
- virtual void closeInputStream(AudioStreamIn* in);
-// static AudioHardwareInterface* createA2dpInterface();
-
-protected:
- virtual status_t dump(int fd, const Vector<String16>& args);
-
-private:
- class A2dpAudioStreamOut : public AudioStreamOut {
- public:
- A2dpAudioStreamOut();
- virtual ~A2dpAudioStreamOut();
- status_t set(uint32_t device,
- int *pFormat,
- uint32_t *pChannels,
- uint32_t *pRate);
- virtual uint32_t sampleRate() const { return 44100; }
- // SBC codec wants a multiple of 512
- virtual size_t bufferSize() const { return 512 * 20; }
- virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
- virtual int format() const { return AudioSystem::PCM_16_BIT; }
- virtual uint32_t latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
- virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; }
- virtual ssize_t write(const void* buffer, size_t bytes);
- status_t standby();
- virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
- virtual status_t getRenderPosition(uint32_t *dspFrames);
-
- private:
- friend class A2dpAudioInterface;
- status_t init();
- status_t close();
- status_t close_l();
- status_t setAddress(const char* address);
- status_t setBluetoothEnabled(bool enabled);
- status_t setSuspended(bool onOff);
- status_t standby_l();
-
- private:
- int mFd;
- bool mStandby;
- int mStartCount;
- int mRetryCount;
- char mA2dpAddress[20];
- void* mData;
- Mutex mLock;
- bool mBluetoothEnabled;
- uint32_t mDevice;
- bool mClosing;
- bool mSuspended;
- nsecs_t mLastWriteTime;
- uint32_t mBufferDurationUs;
- };
-
- friend class A2dpAudioStreamOut;
-
- A2dpAudioStreamOut* mOutput;
- AudioHardwareInterface *mHardwareInterface;
- char mA2dpAddress[20];
- bool mBluetoothEnabled;
- bool mSuspended;
-};
-
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // A2DP_AUDIO_HARDWARE_H
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 69a4adc..6d78614 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -1,77 +1,5 @@
LOCAL_PATH:= $(call my-dir)
-#AUDIO_POLICY_TEST := true
-#ENABLE_AUDIO_DUMP := true
-
-include $(CLEAR_VARS)
-
-
-ifeq ($(AUDIO_POLICY_TEST),true)
- ENABLE_AUDIO_DUMP := true
-endif
-
-
-LOCAL_SRC_FILES:= \
- AudioHardwareGeneric.cpp \
- AudioHardwareStub.cpp \
- AudioHardwareInterface.cpp
-
-ifeq ($(ENABLE_AUDIO_DUMP),true)
- LOCAL_SRC_FILES += AudioDumpInterface.cpp
- LOCAL_CFLAGS += -DENABLE_AUDIO_DUMP
-endif
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libutils \
- libbinder \
- libmedia \
- libhardware_legacy
-
-ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
- LOCAL_CFLAGS += -DGENERIC_AUDIO
-endif
-
-LOCAL_MODULE:= libaudiointerface
-
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
- LOCAL_SRC_FILES += A2dpAudioInterface.cpp
- LOCAL_SHARED_LIBRARIES += liba2dp
- LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
- LOCAL_C_INCLUDES += $(call include-path-for, bluez)
-endif
-
-include $(BUILD_STATIC_LIBRARY)
-
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
- AudioPolicyManagerBase.cpp
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libutils \
- libmedia
-
-ifeq ($(TARGET_SIMULATOR),true)
- LOCAL_LDLIBS += -ldl
-else
- LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-LOCAL_MODULE:= libaudiopolicybase
-
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
- LOCAL_CFLAGS += -DWITH_A2DP
-endif
-
-ifeq ($(AUDIO_POLICY_TEST),true)
- LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
-endif
-
-include $(BUILD_STATIC_LIBRARY)
-
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
@@ -90,12 +18,6 @@
libhardware_legacy \
libeffects
-ifeq ($(strip $(BOARD_USES_GENERIC_AUDIO)),true)
- LOCAL_STATIC_LIBRARIES += libaudiointerface libaudiopolicybase
- LOCAL_CFLAGS += -DGENERIC_AUDIO
-else
- LOCAL_SHARED_LIBRARIES += libaudio libaudiopolicy
-endif
ifeq ($(TARGET_SIMULATOR),true)
LOCAL_LDLIBS += -ldl
@@ -105,15 +27,6 @@
LOCAL_MODULE:= libaudioflinger
-ifeq ($(BOARD_HAVE_BLUETOOTH),true)
- LOCAL_CFLAGS += -DWITH_BLUETOOTH -DWITH_A2DP
- LOCAL_SHARED_LIBRARIES += liba2dp
-endif
-
-ifeq ($(AUDIO_POLICY_TEST),true)
- LOCAL_CFLAGS += -DAUDIO_POLICY_TEST
-endif
-
ifeq ($(TARGET_SIMULATOR),true)
ifeq ($(HOST_OS),linux)
LOCAL_LDLIBS += -lrt -lpthread
diff --git a/services/audioflinger/AudioDumpInterface.cpp b/services/audioflinger/AudioDumpInterface.cpp
deleted file mode 100644
index 6c11114..0000000
--- a/services/audioflinger/AudioDumpInterface.cpp
+++ /dev/null
@@ -1,573 +0,0 @@
-/* //device/servers/AudioFlinger/AudioDumpInterface.cpp
-**
-** Copyright 2008, 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 "AudioFlingerDump"
-//#define LOG_NDEBUG 0
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <utils/Log.h>
-
-#include <stdlib.h>
-#include <unistd.h>
-
-#include "AudioDumpInterface.h"
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-AudioDumpInterface::AudioDumpInterface(AudioHardwareInterface* hw)
- : mPolicyCommands(String8("")), mFileName(String8(""))
-{
- if(hw == 0) {
- LOGE("Dump construct hw = 0");
- }
- mFinalInterface = hw;
- LOGV("Constructor %p, mFinalInterface %p", this, mFinalInterface);
-}
-
-
-AudioDumpInterface::~AudioDumpInterface()
-{
- for (size_t i = 0; i < mOutputs.size(); i++) {
- closeOutputStream((AudioStreamOut *)mOutputs[i]);
- }
-
- for (size_t i = 0; i < mInputs.size(); i++) {
- closeInputStream((AudioStreamIn *)mInputs[i]);
- }
-
- if(mFinalInterface) delete mFinalInterface;
-}
-
-
-AudioStreamOut* AudioDumpInterface::openOutputStream(
- uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
- AudioStreamOut* outFinal = NULL;
- int lFormat = AudioSystem::PCM_16_BIT;
- uint32_t lChannels = AudioSystem::CHANNEL_OUT_STEREO;
- uint32_t lRate = 44100;
-
-
- outFinal = mFinalInterface->openOutputStream(devices, format, channels, sampleRate, status);
- if (outFinal != 0) {
- lFormat = outFinal->format();
- lChannels = outFinal->channels();
- lRate = outFinal->sampleRate();
- } else {
- if (format != 0) {
- if (*format != 0) {
- lFormat = *format;
- } else {
- *format = lFormat;
- }
- }
- if (channels != 0) {
- if (*channels != 0) {
- lChannels = *channels;
- } else {
- *channels = lChannels;
- }
- }
- if (sampleRate != 0) {
- if (*sampleRate != 0) {
- lRate = *sampleRate;
- } else {
- *sampleRate = lRate;
- }
- }
- if (status) *status = NO_ERROR;
- }
- LOGV("openOutputStream(), outFinal %p", outFinal);
-
- AudioStreamOutDump *dumOutput = new AudioStreamOutDump(this, mOutputs.size(), outFinal,
- devices, lFormat, lChannels, lRate);
- mOutputs.add(dumOutput);
-
- return dumOutput;
-}
-
-void AudioDumpInterface::closeOutputStream(AudioStreamOut* out)
-{
- AudioStreamOutDump *dumpOut = (AudioStreamOutDump *)out;
-
- if (mOutputs.indexOf(dumpOut) < 0) {
- LOGW("Attempt to close invalid output stream");
- return;
- }
-
- LOGV("closeOutputStream() output %p", out);
-
- dumpOut->standby();
- if (dumpOut->finalStream() != NULL) {
- mFinalInterface->closeOutputStream(dumpOut->finalStream());
- }
-
- mOutputs.remove(dumpOut);
- delete dumpOut;
-}
-
-AudioStreamIn* AudioDumpInterface::openInputStream(uint32_t devices, int *format, uint32_t *channels,
- uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics)
-{
- AudioStreamIn* inFinal = NULL;
- int lFormat = AudioSystem::PCM_16_BIT;
- uint32_t lChannels = AudioSystem::CHANNEL_IN_MONO;
- uint32_t lRate = 8000;
-
- inFinal = mFinalInterface->openInputStream(devices, format, channels, sampleRate, status, acoustics);
- if (inFinal != 0) {
- lFormat = inFinal->format();
- lChannels = inFinal->channels();
- lRate = inFinal->sampleRate();
- } else {
- if (format != 0) {
- if (*format != 0) {
- lFormat = *format;
- } else {
- *format = lFormat;
- }
- }
- if (channels != 0) {
- if (*channels != 0) {
- lChannels = *channels;
- } else {
- *channels = lChannels;
- }
- }
- if (sampleRate != 0) {
- if (*sampleRate != 0) {
- lRate = *sampleRate;
- } else {
- *sampleRate = lRate;
- }
- }
- if (status) *status = NO_ERROR;
- }
- LOGV("openInputStream(), inFinal %p", inFinal);
-
- AudioStreamInDump *dumInput = new AudioStreamInDump(this, mInputs.size(), inFinal,
- devices, lFormat, lChannels, lRate);
- mInputs.add(dumInput);
-
- return dumInput;
-}
-void AudioDumpInterface::closeInputStream(AudioStreamIn* in)
-{
- AudioStreamInDump *dumpIn = (AudioStreamInDump *)in;
-
- if (mInputs.indexOf(dumpIn) < 0) {
- LOGW("Attempt to close invalid input stream");
- return;
- }
- dumpIn->standby();
- if (dumpIn->finalStream() != NULL) {
- mFinalInterface->closeInputStream(dumpIn->finalStream());
- }
-
- mInputs.remove(dumpIn);
- delete dumpIn;
-}
-
-
-status_t AudioDumpInterface::setParameters(const String8& keyValuePairs)
-{
- AudioParameter param = AudioParameter(keyValuePairs);
- String8 value;
- int valueInt;
- LOGV("setParameters %s", keyValuePairs.string());
-
- if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
- mFileName = value;
- param.remove(String8("test_cmd_file_name"));
- }
- if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
- Mutex::Autolock _l(mLock);
- param.remove(String8("test_cmd_policy"));
- mPolicyCommands = param.toString();
- LOGV("test_cmd_policy command %s written", mPolicyCommands.string());
- return NO_ERROR;
- }
-
- if (mFinalInterface != 0 ) return mFinalInterface->setParameters(keyValuePairs);
- return NO_ERROR;
-}
-
-String8 AudioDumpInterface::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- AudioParameter response;
- String8 value;
-
-// LOGV("getParameters %s", keys.string());
- if (param.get(String8("test_cmd_policy"), value) == NO_ERROR) {
- Mutex::Autolock _l(mLock);
- if (mPolicyCommands.length() != 0) {
- response = AudioParameter(mPolicyCommands);
- response.addInt(String8("test_cmd_policy"), 1);
- } else {
- response.addInt(String8("test_cmd_policy"), 0);
- }
- param.remove(String8("test_cmd_policy"));
-// LOGV("test_cmd_policy command %s read", mPolicyCommands.string());
- }
-
- if (param.get(String8("test_cmd_file_name"), value) == NO_ERROR) {
- response.add(String8("test_cmd_file_name"), mFileName);
- param.remove(String8("test_cmd_file_name"));
- }
-
- String8 keyValuePairs = response.toString();
-
- if (param.size() && mFinalInterface != 0 ) {
- keyValuePairs += ";";
- keyValuePairs += mFinalInterface->getParameters(param.toString());
- }
-
- return keyValuePairs;
-}
-
-status_t AudioDumpInterface::setMode(int mode)
-{
- return mFinalInterface->setMode(mode);
-}
-
-size_t AudioDumpInterface::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
-{
- return mFinalInterface->getInputBufferSize(sampleRate, format, channelCount);
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamOutDump::AudioStreamOutDump(AudioDumpInterface *interface,
- int id,
- AudioStreamOut* finalStream,
- uint32_t devices,
- int format,
- uint32_t channels,
- uint32_t sampleRate)
- : mInterface(interface), mId(id),
- mSampleRate(sampleRate), mFormat(format), mChannels(channels), mLatency(0), mDevice(devices),
- mBufferSize(1024), mFinalStream(finalStream), mFile(0), mFileCount(0)
-{
- LOGV("AudioStreamOutDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
-}
-
-
-AudioStreamOutDump::~AudioStreamOutDump()
-{
- LOGV("AudioStreamOutDump destructor");
- Close();
-}
-
-ssize_t AudioStreamOutDump::write(const void* buffer, size_t bytes)
-{
- ssize_t ret;
-
- if (mFinalStream) {
- ret = mFinalStream->write(buffer, bytes);
- } else {
- usleep((((bytes * 1000) / frameSize()) / sampleRate()) * 1000);
- ret = bytes;
- }
- if(!mFile) {
- if (mInterface->fileName() != "") {
- char name[255];
- sprintf(name, "%s_out_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount);
- mFile = fopen(name, "wb");
- LOGV("Opening dump file %s, fh %p", name, mFile);
- }
- }
- if (mFile) {
- fwrite(buffer, bytes, 1, mFile);
- }
- return ret;
-}
-
-status_t AudioStreamOutDump::standby()
-{
- LOGV("AudioStreamOutDump standby(), mFile %p, mFinalStream %p", mFile, mFinalStream);
-
- Close();
- if (mFinalStream != 0 ) return mFinalStream->standby();
- return NO_ERROR;
-}
-
-uint32_t AudioStreamOutDump::sampleRate() const
-{
- if (mFinalStream != 0 ) return mFinalStream->sampleRate();
- return mSampleRate;
-}
-
-size_t AudioStreamOutDump::bufferSize() const
-{
- if (mFinalStream != 0 ) return mFinalStream->bufferSize();
- return mBufferSize;
-}
-
-uint32_t AudioStreamOutDump::channels() const
-{
- if (mFinalStream != 0 ) return mFinalStream->channels();
- return mChannels;
-}
-int AudioStreamOutDump::format() const
-{
- if (mFinalStream != 0 ) return mFinalStream->format();
- return mFormat;
-}
-uint32_t AudioStreamOutDump::latency() const
-{
- if (mFinalStream != 0 ) return mFinalStream->latency();
- return 0;
-}
-status_t AudioStreamOutDump::setVolume(float left, float right)
-{
- if (mFinalStream != 0 ) return mFinalStream->setVolume(left, right);
- return NO_ERROR;
-}
-status_t AudioStreamOutDump::setParameters(const String8& keyValuePairs)
-{
- LOGV("AudioStreamOutDump::setParameters %s", keyValuePairs.string());
-
- if (mFinalStream != 0 ) {
- return mFinalStream->setParameters(keyValuePairs);
- }
-
- AudioParameter param = AudioParameter(keyValuePairs);
- String8 value;
- int valueInt;
- status_t status = NO_ERROR;
-
- if (param.getInt(String8("set_id"), valueInt) == NO_ERROR) {
- mId = valueInt;
- }
-
- if (param.getInt(String8("format"), valueInt) == NO_ERROR) {
- if (mFile == 0) {
- mFormat = valueInt;
- } else {
- status = INVALID_OPERATION;
- }
- }
- if (param.getInt(String8("channels"), valueInt) == NO_ERROR) {
- if (valueInt == AudioSystem::CHANNEL_OUT_STEREO || valueInt == AudioSystem::CHANNEL_OUT_MONO) {
- mChannels = valueInt;
- } else {
- status = BAD_VALUE;
- }
- }
- if (param.getInt(String8("sampling_rate"), valueInt) == NO_ERROR) {
- if (valueInt > 0 && valueInt <= 48000) {
- if (mFile == 0) {
- mSampleRate = valueInt;
- } else {
- status = INVALID_OPERATION;
- }
- } else {
- status = BAD_VALUE;
- }
- }
- return status;
-}
-
-String8 AudioStreamOutDump::getParameters(const String8& keys)
-{
- if (mFinalStream != 0 ) return mFinalStream->getParameters(keys);
-
- AudioParameter param = AudioParameter(keys);
- return param.toString();
-}
-
-status_t AudioStreamOutDump::dump(int fd, const Vector<String16>& args)
-{
- if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
- return NO_ERROR;
-}
-
-void AudioStreamOutDump::Close()
-{
- if(mFile) {
- fclose(mFile);
- mFile = 0;
- }
-}
-
-status_t AudioStreamOutDump::getRenderPosition(uint32_t *dspFrames)
-{
- if (mFinalStream != 0 ) return mFinalStream->getRenderPosition(dspFrames);
- return INVALID_OPERATION;
-}
-
-// ----------------------------------------------------------------------------
-
-AudioStreamInDump::AudioStreamInDump(AudioDumpInterface *interface,
- int id,
- AudioStreamIn* finalStream,
- uint32_t devices,
- int format,
- uint32_t channels,
- uint32_t sampleRate)
- : mInterface(interface), mId(id),
- mSampleRate(sampleRate), mFormat(format), mChannels(channels), mDevice(devices),
- mBufferSize(1024), mFinalStream(finalStream), mFile(0), mFileCount(0)
-{
- LOGV("AudioStreamInDump Constructor %p, mInterface %p, mFinalStream %p", this, mInterface, mFinalStream);
-}
-
-
-AudioStreamInDump::~AudioStreamInDump()
-{
- Close();
-}
-
-ssize_t AudioStreamInDump::read(void* buffer, ssize_t bytes)
-{
- ssize_t ret;
-
- if (mFinalStream) {
- ret = mFinalStream->read(buffer, bytes);
- if(!mFile) {
- if (mInterface->fileName() != "") {
- char name[255];
- sprintf(name, "%s_in_%d_%d.pcm", mInterface->fileName().string(), mId, ++mFileCount);
- mFile = fopen(name, "wb");
- LOGV("Opening input dump file %s, fh %p", name, mFile);
- }
- }
- if (mFile) {
- fwrite(buffer, bytes, 1, mFile);
- }
- } else {
- usleep((((bytes * 1000) / frameSize()) / sampleRate()) * 1000);
- ret = bytes;
- if(!mFile) {
- char name[255];
- strcpy(name, "/sdcard/music/sine440");
- if (channels() == AudioSystem::CHANNEL_IN_MONO) {
- strcat(name, "_mo");
- } else {
- strcat(name, "_st");
- }
- if (format() == AudioSystem::PCM_16_BIT) {
- strcat(name, "_16b");
- } else {
- strcat(name, "_8b");
- }
- if (sampleRate() < 16000) {
- strcat(name, "_8k");
- } else if (sampleRate() < 32000) {
- strcat(name, "_22k");
- } else if (sampleRate() < 48000) {
- strcat(name, "_44k");
- } else {
- strcat(name, "_48k");
- }
- strcat(name, ".wav");
- mFile = fopen(name, "rb");
- LOGV("Opening input read file %s, fh %p", name, mFile);
- if (mFile) {
- fseek(mFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
- }
- }
- if (mFile) {
- ssize_t bytesRead = fread(buffer, bytes, 1, mFile);
- if (bytesRead >=0 && bytesRead < bytes) {
- fseek(mFile, AUDIO_DUMP_WAVE_HDR_SIZE, SEEK_SET);
- fread((uint8_t *)buffer+bytesRead, bytes-bytesRead, 1, mFile);
- }
- }
- }
-
- return ret;
-}
-
-status_t AudioStreamInDump::standby()
-{
- LOGV("AudioStreamInDump standby(), mFile %p, mFinalStream %p", mFile, mFinalStream);
-
- Close();
- if (mFinalStream != 0 ) return mFinalStream->standby();
- return NO_ERROR;
-}
-
-status_t AudioStreamInDump::setGain(float gain)
-{
- if (mFinalStream != 0 ) return mFinalStream->setGain(gain);
- return NO_ERROR;
-}
-
-uint32_t AudioStreamInDump::sampleRate() const
-{
- if (mFinalStream != 0 ) return mFinalStream->sampleRate();
- return mSampleRate;
-}
-
-size_t AudioStreamInDump::bufferSize() const
-{
- if (mFinalStream != 0 ) return mFinalStream->bufferSize();
- return mBufferSize;
-}
-
-uint32_t AudioStreamInDump::channels() const
-{
- if (mFinalStream != 0 ) return mFinalStream->channels();
- return mChannels;
-}
-
-int AudioStreamInDump::format() const
-{
- if (mFinalStream != 0 ) return mFinalStream->format();
- return mFormat;
-}
-
-status_t AudioStreamInDump::setParameters(const String8& keyValuePairs)
-{
- LOGV("AudioStreamInDump::setParameters()");
- if (mFinalStream != 0 ) return mFinalStream->setParameters(keyValuePairs);
- return NO_ERROR;
-}
-
-String8 AudioStreamInDump::getParameters(const String8& keys)
-{
- if (mFinalStream != 0 ) return mFinalStream->getParameters(keys);
-
- AudioParameter param = AudioParameter(keys);
- return param.toString();
-}
-
-unsigned int AudioStreamInDump::getInputFramesLost() const
-{
- if (mFinalStream != 0 ) return mFinalStream->getInputFramesLost();
- return 0;
-}
-
-status_t AudioStreamInDump::dump(int fd, const Vector<String16>& args)
-{
- if (mFinalStream != 0 ) return mFinalStream->dump(fd, args);
- return NO_ERROR;
-}
-
-void AudioStreamInDump::Close()
-{
- if(mFile) {
- fclose(mFile);
- mFile = 0;
- }
-}
-}; // namespace android
diff --git a/services/audioflinger/AudioDumpInterface.h b/services/audioflinger/AudioDumpInterface.h
deleted file mode 100644
index 814ce5f..0000000
--- a/services/audioflinger/AudioDumpInterface.h
+++ /dev/null
@@ -1,170 +0,0 @@
-/* //device/servers/AudioFlinger/AudioDumpInterface.h
-**
-** Copyright 2008, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_AUDIO_DUMP_INTERFACE_H
-#define ANDROID_AUDIO_DUMP_INTERFACE_H
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <utils/String8.h>
-#include <utils/SortedVector.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-namespace android {
-
-#define AUDIO_DUMP_WAVE_HDR_SIZE 44
-
-class AudioDumpInterface;
-
-class AudioStreamOutDump : public AudioStreamOut {
-public:
- AudioStreamOutDump(AudioDumpInterface *interface,
- int id,
- AudioStreamOut* finalStream,
- uint32_t devices,
- int format,
- uint32_t channels,
- uint32_t sampleRate);
- ~AudioStreamOutDump();
-
- virtual ssize_t write(const void* buffer, size_t bytes);
- virtual uint32_t sampleRate() const;
- virtual size_t bufferSize() const;
- virtual uint32_t channels() const;
- virtual int format() const;
- virtual uint32_t latency() const;
- virtual status_t setVolume(float left, float right);
- virtual status_t standby();
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
- virtual status_t dump(int fd, const Vector<String16>& args);
- void Close(void);
- AudioStreamOut* finalStream() { return mFinalStream; }
- uint32_t device() { return mDevice; }
- int getId() { return mId; }
- virtual status_t getRenderPosition(uint32_t *dspFrames);
-
-private:
- AudioDumpInterface *mInterface;
- int mId;
- uint32_t mSampleRate; //
- uint32_t mFormat; //
- uint32_t mChannels; // output configuration
- uint32_t mLatency; //
- uint32_t mDevice; // current device this output is routed to
- size_t mBufferSize;
- AudioStreamOut *mFinalStream;
- FILE *mFile; // output file
- int mFileCount;
-};
-
-class AudioStreamInDump : public AudioStreamIn {
-public:
- AudioStreamInDump(AudioDumpInterface *interface,
- int id,
- AudioStreamIn* finalStream,
- uint32_t devices,
- int format,
- uint32_t channels,
- uint32_t sampleRate);
- ~AudioStreamInDump();
-
- virtual uint32_t sampleRate() const;
- virtual size_t bufferSize() const;
- virtual uint32_t channels() const;
- virtual int format() const;
-
- virtual status_t setGain(float gain);
- virtual ssize_t read(void* buffer, ssize_t bytes);
- virtual status_t standby();
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
- virtual unsigned int getInputFramesLost() const;
- virtual status_t dump(int fd, const Vector<String16>& args);
- void Close(void);
- AudioStreamIn* finalStream() { return mFinalStream; }
- uint32_t device() { return mDevice; }
-
-private:
- AudioDumpInterface *mInterface;
- int mId;
- uint32_t mSampleRate; //
- uint32_t mFormat; //
- uint32_t mChannels; // output configuration
- uint32_t mDevice; // current device this output is routed to
- size_t mBufferSize;
- AudioStreamIn *mFinalStream;
- FILE *mFile; // output file
- int mFileCount;
-};
-
-class AudioDumpInterface : public AudioHardwareBase
-{
-
-public:
- AudioDumpInterface(AudioHardwareInterface* hw);
- virtual AudioStreamOut* openOutputStream(
- uint32_t devices,
- int *format=0,
- uint32_t *channels=0,
- uint32_t *sampleRate=0,
- status_t *status=0);
- virtual void closeOutputStream(AudioStreamOut* out);
-
- virtual ~AudioDumpInterface();
-
- virtual status_t initCheck()
- {return mFinalInterface->initCheck();}
- virtual status_t setVoiceVolume(float volume)
- {return mFinalInterface->setVoiceVolume(volume);}
- virtual status_t setMasterVolume(float volume)
- {return mFinalInterface->setMasterVolume(volume);}
-
- virtual status_t setMode(int mode);
-
- // mic mute
- virtual status_t setMicMute(bool state)
- {return mFinalInterface->setMicMute(state);}
- virtual status_t getMicMute(bool* state)
- {return mFinalInterface->getMicMute(state);}
-
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
-
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
-
- virtual AudioStreamIn* openInputStream(uint32_t devices, int *format, uint32_t *channels,
- uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics);
- virtual void closeInputStream(AudioStreamIn* in);
-
- virtual status_t dump(int fd, const Vector<String16>& args) { return mFinalInterface->dumpState(fd, args); }
-
- String8 fileName() const { return mFileName; }
-protected:
-
- AudioHardwareInterface *mFinalInterface;
- SortedVector<AudioStreamOutDump *> mOutputs;
- SortedVector<AudioStreamInDump *> mInputs;
- Mutex mLock;
- String8 mPolicyCommands;
- String8 mFileName;
-};
-
-}; // namespace android
-
-#endif // ANDROID_AUDIO_DUMP_INTERFACE_H
diff --git a/services/audioflinger/AudioHardwareGeneric.cpp b/services/audioflinger/AudioHardwareGeneric.cpp
deleted file mode 100644
index d63c031..0000000
--- a/services/audioflinger/AudioHardwareGeneric.cpp
+++ /dev/null
@@ -1,411 +0,0 @@
-/*
-**
-** Copyright 2007, 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.
-*/
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <unistd.h>
-#include <sched.h>
-#include <fcntl.h>
-#include <sys/ioctl.h>
-
-#define LOG_TAG "AudioHardware"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include "AudioHardwareGeneric.h"
-#include <media/AudioRecord.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-static char const * const kAudioDeviceName = "/dev/eac";
-
-// ----------------------------------------------------------------------------
-
-AudioHardwareGeneric::AudioHardwareGeneric()
- : mOutput(0), mInput(0), mFd(-1), mMicMute(false)
-{
- mFd = ::open(kAudioDeviceName, O_RDWR);
-}
-
-AudioHardwareGeneric::~AudioHardwareGeneric()
-{
- if (mFd >= 0) ::close(mFd);
- closeOutputStream((AudioStreamOut *)mOutput);
- closeInputStream((AudioStreamIn *)mInput);
-}
-
-status_t AudioHardwareGeneric::initCheck()
-{
- if (mFd >= 0) {
- if (::access(kAudioDeviceName, O_RDWR) == NO_ERROR)
- return NO_ERROR;
- }
- return NO_INIT;
-}
-
-AudioStreamOut* AudioHardwareGeneric::openOutputStream(
- uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
- AutoMutex lock(mLock);
-
- // only one output stream allowed
- if (mOutput) {
- if (status) {
- *status = INVALID_OPERATION;
- }
- return 0;
- }
-
- // create new output stream
- AudioStreamOutGeneric* out = new AudioStreamOutGeneric();
- status_t lStatus = out->set(this, mFd, devices, format, channels, sampleRate);
- if (status) {
- *status = lStatus;
- }
- if (lStatus == NO_ERROR) {
- mOutput = out;
- } else {
- delete out;
- }
- return mOutput;
-}
-
-void AudioHardwareGeneric::closeOutputStream(AudioStreamOut* out) {
- if (mOutput && out == mOutput) {
- delete mOutput;
- mOutput = 0;
- }
-}
-
-AudioStreamIn* AudioHardwareGeneric::openInputStream(
- uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
- status_t *status, AudioSystem::audio_in_acoustics acoustics)
-{
- // check for valid input source
- if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
- return 0;
- }
-
- AutoMutex lock(mLock);
-
- // only one input stream allowed
- if (mInput) {
- if (status) {
- *status = INVALID_OPERATION;
- }
- return 0;
- }
-
- // create new output stream
- AudioStreamInGeneric* in = new AudioStreamInGeneric();
- status_t lStatus = in->set(this, mFd, devices, format, channels, sampleRate, acoustics);
- if (status) {
- *status = lStatus;
- }
- if (lStatus == NO_ERROR) {
- mInput = in;
- } else {
- delete in;
- }
- return mInput;
-}
-
-void AudioHardwareGeneric::closeInputStream(AudioStreamIn* in) {
- if (mInput && in == mInput) {
- delete mInput;
- mInput = 0;
- }
-}
-
-status_t AudioHardwareGeneric::setVoiceVolume(float v)
-{
- // Implement: set voice volume
- return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::setMasterVolume(float v)
-{
- // Implement: set master volume
- // return error - software mixer will handle it
- return INVALID_OPERATION;
-}
-
-status_t AudioHardwareGeneric::setMicMute(bool state)
-{
- mMicMute = state;
- return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::getMicMute(bool* state)
-{
- *state = mMicMute;
- return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::dumpInternals(int fd, const Vector<String16>& args)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- result.append("AudioHardwareGeneric::dumpInternals\n");
- snprintf(buffer, SIZE, "\tmFd: %d mMicMute: %s\n", mFd, mMicMute? "true": "false");
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-status_t AudioHardwareGeneric::dump(int fd, const Vector<String16>& args)
-{
- dumpInternals(fd, args);
- if (mInput) {
- mInput->dump(fd, args);
- }
- if (mOutput) {
- mOutput->dump(fd, args);
- }
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-status_t AudioStreamOutGeneric::set(
- AudioHardwareGeneric *hw,
- int fd,
- uint32_t devices,
- int *pFormat,
- uint32_t *pChannels,
- uint32_t *pRate)
-{
- int lFormat = pFormat ? *pFormat : 0;
- uint32_t lChannels = pChannels ? *pChannels : 0;
- uint32_t lRate = pRate ? *pRate : 0;
-
- // fix up defaults
- if (lFormat == 0) lFormat = format();
- if (lChannels == 0) lChannels = channels();
- if (lRate == 0) lRate = sampleRate();
-
- // check values
- if ((lFormat != format()) ||
- (lChannels != channels()) ||
- (lRate != sampleRate())) {
- if (pFormat) *pFormat = format();
- if (pChannels) *pChannels = channels();
- if (pRate) *pRate = sampleRate();
- return BAD_VALUE;
- }
-
- if (pFormat) *pFormat = lFormat;
- if (pChannels) *pChannels = lChannels;
- if (pRate) *pRate = lRate;
-
- mAudioHardware = hw;
- mFd = fd;
- mDevice = devices;
- return NO_ERROR;
-}
-
-AudioStreamOutGeneric::~AudioStreamOutGeneric()
-{
-}
-
-ssize_t AudioStreamOutGeneric::write(const void* buffer, size_t bytes)
-{
- Mutex::Autolock _l(mLock);
- return ssize_t(::write(mFd, buffer, bytes));
-}
-
-status_t AudioStreamOutGeneric::standby()
-{
- // Implement: audio hardware to standby mode
- return NO_ERROR;
-}
-
-status_t AudioStreamOutGeneric::dump(int fd, const Vector<String16>& args)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- snprintf(buffer, SIZE, "AudioStreamOutGeneric::dump\n");
- result.append(buffer);
- snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tformat: %d\n", format());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
- result.append(buffer);
- snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-status_t AudioStreamOutGeneric::setParameters(const String8& keyValuePairs)
-{
- AudioParameter param = AudioParameter(keyValuePairs);
- String8 key = String8(AudioParameter::keyRouting);
- status_t status = NO_ERROR;
- int device;
- LOGV("setParameters() %s", keyValuePairs.string());
-
- if (param.getInt(key, device) == NO_ERROR) {
- mDevice = device;
- param.remove(key);
- }
-
- if (param.size()) {
- status = BAD_VALUE;
- }
- return status;
-}
-
-String8 AudioStreamOutGeneric::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- String8 value;
- String8 key = String8(AudioParameter::keyRouting);
-
- if (param.get(key, value) == NO_ERROR) {
- param.addInt(key, (int)mDevice);
- }
-
- LOGV("getParameters() %s", param.toString().string());
- return param.toString();
-}
-
-status_t AudioStreamOutGeneric::getRenderPosition(uint32_t *dspFrames)
-{
- return INVALID_OPERATION;
-}
-
-// ----------------------------------------------------------------------------
-
-// record functions
-status_t AudioStreamInGeneric::set(
- AudioHardwareGeneric *hw,
- int fd,
- uint32_t devices,
- int *pFormat,
- uint32_t *pChannels,
- uint32_t *pRate,
- AudioSystem::audio_in_acoustics acoustics)
-{
- if (pFormat == 0 || pChannels == 0 || pRate == 0) return BAD_VALUE;
- LOGV("AudioStreamInGeneric::set(%p, %d, %d, %d, %u)", hw, fd, *pFormat, *pChannels, *pRate);
- // check values
- if ((*pFormat != format()) ||
- (*pChannels != channels()) ||
- (*pRate != sampleRate())) {
- LOGE("Error opening input channel");
- *pFormat = format();
- *pChannels = channels();
- *pRate = sampleRate();
- return BAD_VALUE;
- }
-
- mAudioHardware = hw;
- mFd = fd;
- mDevice = devices;
- return NO_ERROR;
-}
-
-AudioStreamInGeneric::~AudioStreamInGeneric()
-{
-}
-
-ssize_t AudioStreamInGeneric::read(void* buffer, ssize_t bytes)
-{
- AutoMutex lock(mLock);
- if (mFd < 0) {
- LOGE("Attempt to read from unopened device");
- return NO_INIT;
- }
- return ::read(mFd, buffer, bytes);
-}
-
-status_t AudioStreamInGeneric::dump(int fd, const Vector<String16>& args)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- snprintf(buffer, SIZE, "AudioStreamInGeneric::dump\n");
- result.append(buffer);
- snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tformat: %d\n", format());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tdevice: %d\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, "\tmAudioHardware: %p\n", mAudioHardware);
- result.append(buffer);
- snprintf(buffer, SIZE, "\tmFd: %d\n", mFd);
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-status_t AudioStreamInGeneric::setParameters(const String8& keyValuePairs)
-{
- AudioParameter param = AudioParameter(keyValuePairs);
- String8 key = String8(AudioParameter::keyRouting);
- status_t status = NO_ERROR;
- int device;
- LOGV("setParameters() %s", keyValuePairs.string());
-
- if (param.getInt(key, device) == NO_ERROR) {
- mDevice = device;
- param.remove(key);
- }
-
- if (param.size()) {
- status = BAD_VALUE;
- }
- return status;
-}
-
-String8 AudioStreamInGeneric::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- String8 value;
- String8 key = String8(AudioParameter::keyRouting);
-
- if (param.get(key, value) == NO_ERROR) {
- param.addInt(key, (int)mDevice);
- }
-
- LOGV("getParameters() %s", param.toString().string());
- return param.toString();
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
diff --git a/services/audioflinger/AudioHardwareGeneric.h b/services/audioflinger/AudioHardwareGeneric.h
deleted file mode 100644
index aa4e78d..0000000
--- a/services/audioflinger/AudioHardwareGeneric.h
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_AUDIO_HARDWARE_GENERIC_H
-#define ANDROID_AUDIO_HARDWARE_GENERIC_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <utils/threads.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class AudioHardwareGeneric;
-
-class AudioStreamOutGeneric : public AudioStreamOut {
-public:
- AudioStreamOutGeneric() : mAudioHardware(0), mFd(-1) {}
- virtual ~AudioStreamOutGeneric();
-
- virtual status_t set(
- AudioHardwareGeneric *hw,
- int mFd,
- uint32_t devices,
- int *pFormat,
- uint32_t *pChannels,
- uint32_t *pRate);
-
- virtual uint32_t sampleRate() const { return 44100; }
- virtual size_t bufferSize() const { return 4096; }
- virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
- virtual int format() const { return AudioSystem::PCM_16_BIT; }
- virtual uint32_t latency() const { return 20; }
- virtual status_t setVolume(float left, float right) { return INVALID_OPERATION; }
- virtual ssize_t write(const void* buffer, size_t bytes);
- virtual status_t standby();
- virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
- virtual status_t getRenderPosition(uint32_t *dspFrames);
-
-private:
- AudioHardwareGeneric *mAudioHardware;
- Mutex mLock;
- int mFd;
- uint32_t mDevice;
-};
-
-class AudioStreamInGeneric : public AudioStreamIn {
-public:
- AudioStreamInGeneric() : mAudioHardware(0), mFd(-1) {}
- virtual ~AudioStreamInGeneric();
-
- virtual status_t set(
- AudioHardwareGeneric *hw,
- int mFd,
- uint32_t devices,
- int *pFormat,
- uint32_t *pChannels,
- uint32_t *pRate,
- AudioSystem::audio_in_acoustics acoustics);
-
- virtual uint32_t sampleRate() const { return 8000; }
- virtual size_t bufferSize() const { return 320; }
- virtual uint32_t channels() const { return AudioSystem::CHANNEL_IN_MONO; }
- virtual int format() const { return AudioSystem::PCM_16_BIT; }
- virtual status_t setGain(float gain) { return INVALID_OPERATION; }
- virtual ssize_t read(void* buffer, ssize_t bytes);
- virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t standby() { return NO_ERROR; }
- virtual status_t setParameters(const String8& keyValuePairs);
- virtual String8 getParameters(const String8& keys);
- virtual unsigned int getInputFramesLost() const { return 0; }
-
-private:
- AudioHardwareGeneric *mAudioHardware;
- Mutex mLock;
- int mFd;
- uint32_t mDevice;
-};
-
-
-class AudioHardwareGeneric : public AudioHardwareBase
-{
-public:
- AudioHardwareGeneric();
- virtual ~AudioHardwareGeneric();
- virtual status_t initCheck();
- virtual status_t setVoiceVolume(float volume);
- virtual status_t setMasterVolume(float volume);
-
- // mic mute
- virtual status_t setMicMute(bool state);
- virtual status_t getMicMute(bool* state);
-
- // create I/O streams
- virtual AudioStreamOut* openOutputStream(
- uint32_t devices,
- int *format=0,
- uint32_t *channels=0,
- uint32_t *sampleRate=0,
- status_t *status=0);
- virtual void closeOutputStream(AudioStreamOut* out);
-
- virtual AudioStreamIn* openInputStream(
- uint32_t devices,
- int *format,
- uint32_t *channels,
- uint32_t *sampleRate,
- status_t *status,
- AudioSystem::audio_in_acoustics acoustics);
- virtual void closeInputStream(AudioStreamIn* in);
-
- void closeOutputStream(AudioStreamOutGeneric* out);
- void closeInputStream(AudioStreamInGeneric* in);
-protected:
- virtual status_t dump(int fd, const Vector<String16>& args);
-
-private:
- status_t dumpInternals(int fd, const Vector<String16>& args);
-
- Mutex mLock;
- AudioStreamOutGeneric *mOutput;
- AudioStreamInGeneric *mInput;
- int mFd;
- bool mMicMute;
-};
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // ANDROID_AUDIO_HARDWARE_GENERIC_H
diff --git a/services/audioflinger/AudioHardwareInterface.cpp b/services/audioflinger/AudioHardwareInterface.cpp
deleted file mode 100644
index f58e4c0..0000000
--- a/services/audioflinger/AudioHardwareInterface.cpp
+++ /dev/null
@@ -1,183 +0,0 @@
-/*
-**
-** Copyright 2007, 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.
-*/
-
-#include <cutils/properties.h>
-#include <string.h>
-#include <unistd.h>
-//#define LOG_NDEBUG 0
-
-#define LOG_TAG "AudioHardwareInterface"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include "AudioHardwareStub.h"
-#include "AudioHardwareGeneric.h"
-#ifdef WITH_A2DP
-#include "A2dpAudioInterface.h"
-#endif
-
-#ifdef ENABLE_AUDIO_DUMP
-#include "AudioDumpInterface.h"
-#endif
-
-
-// change to 1 to log routing calls
-#define LOG_ROUTING_CALLS 1
-
-namespace android {
-
-#if LOG_ROUTING_CALLS
-static const char* routingModeStrings[] =
-{
- "OUT OF RANGE",
- "INVALID",
- "CURRENT",
- "NORMAL",
- "RINGTONE",
- "IN_CALL",
- "IN_COMMUNICATION"
-};
-
-static const char* routeNone = "NONE";
-
-static const char* displayMode(int mode)
-{
- if ((mode < AudioSystem::MODE_INVALID) || (mode >= AudioSystem::NUM_MODES))
- return routingModeStrings[0];
- return routingModeStrings[mode+3];
-}
-#endif
-
-// ----------------------------------------------------------------------------
-
-AudioHardwareInterface* AudioHardwareInterface::create()
-{
- /*
- * FIXME: This code needs to instantiate the correct audio device
- * interface. For now - we use compile-time switches.
- */
- AudioHardwareInterface* hw = 0;
- char value[PROPERTY_VALUE_MAX];
-
-#ifdef GENERIC_AUDIO
- hw = new AudioHardwareGeneric();
-#else
- // if running in emulation - use the emulator driver
- if (property_get("ro.kernel.qemu", value, 0)) {
- LOGD("Running in emulation - using generic audio driver");
- hw = new AudioHardwareGeneric();
- }
- else {
- LOGV("Creating Vendor Specific AudioHardware");
- hw = createAudioHardware();
- }
-#endif
- if (hw->initCheck() != NO_ERROR) {
- LOGW("Using stubbed audio hardware. No sound will be produced.");
- delete hw;
- hw = new AudioHardwareStub();
- }
-
-#ifdef WITH_A2DP
- hw = new A2dpAudioInterface(hw);
-#endif
-
-#ifdef ENABLE_AUDIO_DUMP
- // This code adds a record of buffers in a file to write calls made by AudioFlinger.
- // It replaces the current AudioHardwareInterface object by an intermediate one which
- // will record buffers in a file (after sending them to hardware) for testing purpose.
- // This feature is enabled by defining symbol ENABLE_AUDIO_DUMP.
- // The output file is set with setParameters("test_cmd_file_name=<name>"). Pause are not recorded in the file.
- LOGV("opening PCM dump interface");
- hw = new AudioDumpInterface(hw); // replace interface
-#endif
- return hw;
-}
-
-AudioStreamOut::~AudioStreamOut()
-{
-}
-
-AudioStreamIn::~AudioStreamIn() {}
-
-AudioHardwareBase::AudioHardwareBase()
-{
- mMode = 0;
-}
-
-status_t AudioHardwareBase::setMode(int mode)
-{
-#if LOG_ROUTING_CALLS
- LOGD("setMode(%s)", displayMode(mode));
-#endif
- if ((mode < 0) || (mode >= AudioSystem::NUM_MODES))
- return BAD_VALUE;
- if (mMode == mode)
- return ALREADY_EXISTS;
- mMode = mode;
- return NO_ERROR;
-}
-
-// default implementation
-status_t AudioHardwareBase::setParameters(const String8& keyValuePairs)
-{
- return NO_ERROR;
-}
-
-// default implementation
-String8 AudioHardwareBase::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- return param.toString();
-}
-
-// default implementation
-size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
-{
- if (sampleRate != 8000) {
- LOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
- return 0;
- }
- if (format != AudioSystem::PCM_16_BIT) {
- LOGW("getInputBufferSize bad format: %d", format);
- return 0;
- }
- if (channelCount != 1) {
- LOGW("getInputBufferSize bad channel count: %d", channelCount);
- return 0;
- }
-
- return 320;
-}
-
-status_t AudioHardwareBase::dumpState(int fd, const Vector<String16>& args)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- snprintf(buffer, SIZE, "AudioHardwareBase::dumpState\n");
- result.append(buffer);
- snprintf(buffer, SIZE, "\tmMode: %d\n", mMode);
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- dump(fd, args); // Dump the state of the concrete child.
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
diff --git a/services/audioflinger/AudioHardwareStub.cpp b/services/audioflinger/AudioHardwareStub.cpp
deleted file mode 100644
index d481150..0000000
--- a/services/audioflinger/AudioHardwareStub.cpp
+++ /dev/null
@@ -1,209 +0,0 @@
-/* //device/servers/AudioFlinger/AudioHardwareStub.cpp
-**
-** Copyright 2007, 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.
-*/
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <stdlib.h>
-#include <unistd.h>
-#include <utils/String8.h>
-
-#include "AudioHardwareStub.h"
-#include <media/AudioRecord.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-AudioHardwareStub::AudioHardwareStub() : mMicMute(false)
-{
-}
-
-AudioHardwareStub::~AudioHardwareStub()
-{
-}
-
-status_t AudioHardwareStub::initCheck()
-{
- return NO_ERROR;
-}
-
-AudioStreamOut* AudioHardwareStub::openOutputStream(
- uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status)
-{
- AudioStreamOutStub* out = new AudioStreamOutStub();
- status_t lStatus = out->set(format, channels, sampleRate);
- if (status) {
- *status = lStatus;
- }
- if (lStatus == NO_ERROR)
- return out;
- delete out;
- return 0;
-}
-
-void AudioHardwareStub::closeOutputStream(AudioStreamOut* out)
-{
- delete out;
-}
-
-AudioStreamIn* AudioHardwareStub::openInputStream(
- uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate,
- status_t *status, AudioSystem::audio_in_acoustics acoustics)
-{
- // check for valid input source
- if (!AudioSystem::isInputDevice((AudioSystem::audio_devices)devices)) {
- return 0;
- }
-
- AudioStreamInStub* in = new AudioStreamInStub();
- status_t lStatus = in->set(format, channels, sampleRate, acoustics);
- if (status) {
- *status = lStatus;
- }
- if (lStatus == NO_ERROR)
- return in;
- delete in;
- return 0;
-}
-
-void AudioHardwareStub::closeInputStream(AudioStreamIn* in)
-{
- delete in;
-}
-
-status_t AudioHardwareStub::setVoiceVolume(float volume)
-{
- return NO_ERROR;
-}
-
-status_t AudioHardwareStub::setMasterVolume(float volume)
-{
- return NO_ERROR;
-}
-
-status_t AudioHardwareStub::dumpInternals(int fd, const Vector<String16>& args)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- result.append("AudioHardwareStub::dumpInternals\n");
- snprintf(buffer, SIZE, "\tmMicMute: %s\n", mMicMute? "true": "false");
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-status_t AudioHardwareStub::dump(int fd, const Vector<String16>& args)
-{
- dumpInternals(fd, args);
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-
-status_t AudioStreamOutStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate)
-{
- if (pFormat) *pFormat = format();
- if (pChannels) *pChannels = channels();
- if (pRate) *pRate = sampleRate();
-
- return NO_ERROR;
-}
-
-ssize_t AudioStreamOutStub::write(const void* buffer, size_t bytes)
-{
- // fake timing for audio output
- usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
- return bytes;
-}
-
-status_t AudioStreamOutStub::standby()
-{
- return NO_ERROR;
-}
-
-status_t AudioStreamOutStub::dump(int fd, const Vector<String16>& args)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- snprintf(buffer, SIZE, "AudioStreamOutStub::dump\n");
- snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
- snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
- snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
- snprintf(buffer, SIZE, "\tformat: %d\n", format());
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-String8 AudioStreamOutStub::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- return param.toString();
-}
-
-status_t AudioStreamOutStub::getRenderPosition(uint32_t *dspFrames)
-{
- return INVALID_OPERATION;
-}
-
-// ----------------------------------------------------------------------------
-
-status_t AudioStreamInStub::set(int *pFormat, uint32_t *pChannels, uint32_t *pRate,
- AudioSystem::audio_in_acoustics acoustics)
-{
- return NO_ERROR;
-}
-
-ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
-{
- // fake timing for audio input
- usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
- memset(buffer, 0, bytes);
- return bytes;
-}
-
-status_t AudioStreamInStub::dump(int fd, const Vector<String16>& args)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- snprintf(buffer, SIZE, "AudioStreamInStub::dump\n");
- result.append(buffer);
- snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
- result.append(buffer);
- snprintf(buffer, SIZE, "\tformat: %d\n", format());
- result.append(buffer);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-String8 AudioStreamInStub::getParameters(const String8& keys)
-{
- AudioParameter param = AudioParameter(keys);
- return param.toString();
-}
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
diff --git a/services/audioflinger/AudioHardwareStub.h b/services/audioflinger/AudioHardwareStub.h
deleted file mode 100644
index 06a29de..0000000
--- a/services/audioflinger/AudioHardwareStub.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/* //device/servers/AudioFlinger/AudioHardwareStub.h
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_AUDIO_HARDWARE_STUB_H
-#define ANDROID_AUDIO_HARDWARE_STUB_H
-
-#include <stdint.h>
-#include <sys/types.h>
-
-#include <hardware_legacy/AudioHardwareBase.h>
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class AudioStreamOutStub : public AudioStreamOut {
-public:
- virtual status_t set(int *pFormat, uint32_t *pChannels, uint32_t *pRate);
- virtual uint32_t sampleRate() const { return 44100; }
- virtual size_t bufferSize() const { return 4096; }
- virtual uint32_t channels() const { return AudioSystem::CHANNEL_OUT_STEREO; }
- virtual int format() const { return AudioSystem::PCM_16_BIT; }
- virtual uint32_t latency() const { return 0; }
- virtual status_t setVolume(float left, float right) { return NO_ERROR; }
- virtual ssize_t write(const void* buffer, size_t bytes);
- virtual status_t standby();
- virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR;}
- virtual String8 getParameters(const String8& keys);
- virtual status_t getRenderPosition(uint32_t *dspFrames);
-};
-
-class AudioStreamInStub : public AudioStreamIn {
-public:
- virtual status_t set(int *pFormat, uint32_t *pChannels, uint32_t *pRate, AudioSystem::audio_in_acoustics acoustics);
- virtual uint32_t sampleRate() const { return 8000; }
- virtual size_t bufferSize() const { return 320; }
- virtual uint32_t channels() const { return AudioSystem::CHANNEL_IN_MONO; }
- virtual int format() const { return AudioSystem::PCM_16_BIT; }
- virtual status_t setGain(float gain) { return NO_ERROR; }
- virtual ssize_t read(void* buffer, ssize_t bytes);
- virtual status_t dump(int fd, const Vector<String16>& args);
- virtual status_t standby() { return NO_ERROR; }
- virtual status_t setParameters(const String8& keyValuePairs) { return NO_ERROR;}
- virtual String8 getParameters(const String8& keys);
- virtual unsigned int getInputFramesLost() const { return 0; }
-};
-
-class AudioHardwareStub : public AudioHardwareBase
-{
-public:
- AudioHardwareStub();
- virtual ~AudioHardwareStub();
- virtual status_t initCheck();
- virtual status_t setVoiceVolume(float volume);
- virtual status_t setMasterVolume(float volume);
-
- // mic mute
- virtual status_t setMicMute(bool state) { mMicMute = state; return NO_ERROR; }
- virtual status_t getMicMute(bool* state) { *state = mMicMute ; return NO_ERROR; }
-
- // create I/O streams
- virtual AudioStreamOut* openOutputStream(
- uint32_t devices,
- int *format=0,
- uint32_t *channels=0,
- uint32_t *sampleRate=0,
- status_t *status=0);
- virtual void closeOutputStream(AudioStreamOut* out);
-
- virtual AudioStreamIn* openInputStream(
- uint32_t devices,
- int *format,
- uint32_t *channels,
- uint32_t *sampleRate,
- status_t *status,
- AudioSystem::audio_in_acoustics acoustics);
- virtual void closeInputStream(AudioStreamIn* in);
-
-protected:
- virtual status_t dump(int fd, const Vector<String16>& args);
-
- bool mMicMute;
-private:
- status_t dumpInternals(int fd, const Vector<String16>& args);
-};
-
-// ----------------------------------------------------------------------------
-
-}; // namespace android
-
-#endif // ANDROID_AUDIO_HARDWARE_STUB_H
diff --git a/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp
deleted file mode 100644
index 32d92dc..0000000
--- a/services/audioflinger/AudioPolicyManagerBase.cpp
+++ /dev/null
@@ -1,2287 +0,0 @@
-/*
- * Copyright (C) 2009 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 "AudioPolicyManagerBase"
-//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-#include <hardware_legacy/AudioPolicyManagerBase.h>
-#include <media/mediarecorder.h>
-#include <math.h>
-
-namespace android {
-
-
-// ----------------------------------------------------------------------------
-// AudioPolicyInterface implementation
-// ----------------------------------------------------------------------------
-
-
-status_t AudioPolicyManagerBase::setDeviceConnectionState(AudioSystem::audio_devices device,
- AudioSystem::device_connection_state state,
- const char *device_address)
-{
-
- LOGV("setDeviceConnectionState() device: %x, state %d, address %s", device, state, device_address);
-
- // connect/disconnect only 1 device at a time
- if (AudioSystem::popCount(device) != 1) return BAD_VALUE;
-
- if (strlen(device_address) >= MAX_DEVICE_ADDRESS_LEN) {
- LOGE("setDeviceConnectionState() invalid address: %s", device_address);
- return BAD_VALUE;
- }
-
- // handle output devices
- if (AudioSystem::isOutputDevice(device)) {
-
-#ifndef WITH_A2DP
- if (AudioSystem::isA2dpDevice(device)) {
- LOGE("setDeviceConnectionState() invalid device: %x", device);
- return BAD_VALUE;
- }
-#endif
-
- switch (state)
- {
- // handle output device connection
- case AudioSystem::DEVICE_STATE_AVAILABLE:
- if (mAvailableOutputDevices & device) {
- LOGW("setDeviceConnectionState() device already connected: %x", device);
- return INVALID_OPERATION;
- }
- LOGV("setDeviceConnectionState() connecting device %x", device);
-
- // register new device as available
- mAvailableOutputDevices |= device;
-
-#ifdef WITH_A2DP
- // handle A2DP device connection
- if (AudioSystem::isA2dpDevice(device)) {
- status_t status = handleA2dpConnection(device, device_address);
- if (status != NO_ERROR) {
- mAvailableOutputDevices &= ~device;
- return status;
- }
- } else
-#endif
- {
- if (AudioSystem::isBluetoothScoDevice(device)) {
- LOGV("setDeviceConnectionState() BT SCO device, address %s", device_address);
- // keep track of SCO device address
- mScoDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
- }
- }
- break;
- // handle output device disconnection
- case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
- if (!(mAvailableOutputDevices & device)) {
- LOGW("setDeviceConnectionState() device not connected: %x", device);
- return INVALID_OPERATION;
- }
-
-
- LOGV("setDeviceConnectionState() disconnecting device %x", device);
- // remove device from available output devices
- mAvailableOutputDevices &= ~device;
-
-#ifdef WITH_A2DP
- // handle A2DP device disconnection
- if (AudioSystem::isA2dpDevice(device)) {
- status_t status = handleA2dpDisconnection(device, device_address);
- if (status != NO_ERROR) {
- mAvailableOutputDevices |= device;
- return status;
- }
- } else
-#endif
- {
- if (AudioSystem::isBluetoothScoDevice(device)) {
- mScoDeviceAddress = "";
- }
- }
- } break;
-
- default:
- LOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
-
- // request routing change if necessary
- uint32_t newDevice = getNewDevice(mHardwareOutput, false);
-#ifdef WITH_A2DP
- checkA2dpSuspend();
- checkOutputForAllStrategies();
- // A2DP outputs must be closed after checkOutputForAllStrategies() is executed
- if (state == AudioSystem::DEVICE_STATE_UNAVAILABLE && AudioSystem::isA2dpDevice(device)) {
- closeA2dpOutputs();
- }
-#endif
- updateDeviceForStrategy();
- setOutputDevice(mHardwareOutput, newDevice);
-
- if (device == AudioSystem::DEVICE_OUT_WIRED_HEADSET) {
- device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
- } else if (device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO ||
- device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET ||
- device == AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT) {
- device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else {
- return NO_ERROR;
- }
- }
- // handle input devices
- if (AudioSystem::isInputDevice(device)) {
-
- switch (state)
- {
- // handle input device connection
- case AudioSystem::DEVICE_STATE_AVAILABLE: {
- if (mAvailableInputDevices & device) {
- LOGW("setDeviceConnectionState() device already connected: %d", device);
- return INVALID_OPERATION;
- }
- mAvailableInputDevices |= device;
- }
- break;
-
- // handle input device disconnection
- case AudioSystem::DEVICE_STATE_UNAVAILABLE: {
- if (!(mAvailableInputDevices & device)) {
- LOGW("setDeviceConnectionState() device not connected: %d", device);
- return INVALID_OPERATION;
- }
- mAvailableInputDevices &= ~device;
- } break;
-
- default:
- LOGE("setDeviceConnectionState() invalid state: %x", state);
- return BAD_VALUE;
- }
-
- audio_io_handle_t activeInput = getActiveInput();
- if (activeInput != 0) {
- AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
- uint32_t newDevice = getDeviceForInputSource(inputDesc->mInputSource);
- if (newDevice != inputDesc->mDevice) {
- LOGV("setDeviceConnectionState() changing device from %x to %x for input %d",
- inputDesc->mDevice, newDevice, activeInput);
- inputDesc->mDevice = newDevice;
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
- mpClientInterface->setParameters(activeInput, param.toString());
- }
- }
-
- return NO_ERROR;
- }
-
- LOGW("setDeviceConnectionState() invalid device: %x", device);
- return BAD_VALUE;
-}
-
-AudioSystem::device_connection_state AudioPolicyManagerBase::getDeviceConnectionState(AudioSystem::audio_devices device,
- const char *device_address)
-{
- AudioSystem::device_connection_state state = AudioSystem::DEVICE_STATE_UNAVAILABLE;
- String8 address = String8(device_address);
- if (AudioSystem::isOutputDevice(device)) {
- if (device & mAvailableOutputDevices) {
-#ifdef WITH_A2DP
- if (AudioSystem::isA2dpDevice(device) &&
- address != "" && mA2dpDeviceAddress != address) {
- return state;
- }
-#endif
- if (AudioSystem::isBluetoothScoDevice(device) &&
- address != "" && mScoDeviceAddress != address) {
- return state;
- }
- state = AudioSystem::DEVICE_STATE_AVAILABLE;
- }
- } else if (AudioSystem::isInputDevice(device)) {
- if (device & mAvailableInputDevices) {
- state = AudioSystem::DEVICE_STATE_AVAILABLE;
- }
- }
-
- return state;
-}
-
-void AudioPolicyManagerBase::setPhoneState(int state)
-{
- LOGV("setPhoneState() state %d", state);
- uint32_t newDevice = 0;
- if (state < 0 || state >= AudioSystem::NUM_MODES) {
- LOGW("setPhoneState() invalid state %d", state);
- return;
- }
-
- if (state == mPhoneState ) {
- LOGW("setPhoneState() setting same state %d", state);
- return;
- }
-
- // if leaving call state, handle special case of active streams
- // pertaining to sonification strategy see handleIncallSonification()
- if (isInCall()) {
- LOGV("setPhoneState() in call state management: new state is %d", state);
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- handleIncallSonification(stream, false, true);
- }
- }
-
- // store previous phone state for management of sonification strategy below
- int oldState = mPhoneState;
- mPhoneState = state;
- bool force = false;
-
- // are we entering or starting a call
- if (!isStateInCall(oldState) && isStateInCall(state)) {
- LOGV(" Entering call in setPhoneState()");
- // force routing command to audio hardware when starting a call
- // even if no device change is needed
- force = true;
- } else if (isStateInCall(oldState) && !isStateInCall(state)) {
- LOGV(" Exiting call in setPhoneState()");
- // force routing command to audio hardware when exiting a call
- // even if no device change is needed
- force = true;
- } else if (isStateInCall(state) && (state != oldState)) {
- LOGV(" Switching between telephony and VoIP in setPhoneState()");
- // force routing command to audio hardware when switching between telephony and VoIP
- // even if no device change is needed
- force = true;
- }
-
- // check for device and output changes triggered by new phone state
- newDevice = getNewDevice(mHardwareOutput, false);
-#ifdef WITH_A2DP
- checkA2dpSuspend();
- checkOutputForAllStrategies();
-#endif
- updateDeviceForStrategy();
-
- AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
-
- // force routing command to audio hardware when ending call
- // even if no device change is needed
- if (isStateInCall(oldState) && newDevice == 0) {
- newDevice = hwOutputDesc->device();
- }
-
- // when changing from ring tone to in call mode, mute the ringing tone
- // immediately and delay the route change to avoid sending the ring tone
- // tail into the earpiece or headset.
- int delayMs = 0;
- if (isStateInCall(state) && oldState == AudioSystem::MODE_RINGTONE) {
- // delay the device change command by twice the output latency to have some margin
- // and be sure that audio buffers not yet affected by the mute are out when
- // we actually apply the route change
- delayMs = hwOutputDesc->mLatency*2;
- setStreamMute(AudioSystem::RING, true, mHardwareOutput);
- }
-
- // change routing is necessary
- setOutputDevice(mHardwareOutput, newDevice, force, delayMs);
-
- // if entering in call state, handle special case of active streams
- // pertaining to sonification strategy see handleIncallSonification()
- if (isStateInCall(state)) {
- LOGV("setPhoneState() in call state management: new state is %d", state);
- // unmute the ringing tone after a sufficient delay if it was muted before
- // setting output device above
- if (oldState == AudioSystem::MODE_RINGTONE) {
- setStreamMute(AudioSystem::RING, false, mHardwareOutput, MUTE_TIME_MS);
- }
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- handleIncallSonification(stream, true, true);
- }
- }
-
- // Flag that ringtone volume must be limited to music volume until we exit MODE_RINGTONE
- if (state == AudioSystem::MODE_RINGTONE &&
- isStreamActive(AudioSystem::MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY)) {
- mLimitRingtoneVolume = true;
- } else {
- mLimitRingtoneVolume = false;
- }
-}
-
-void AudioPolicyManagerBase::setRingerMode(uint32_t mode, uint32_t mask)
-{
- LOGV("setRingerMode() mode %x, mask %x", mode, mask);
-
- mRingerMode = mode;
-}
-
-void AudioPolicyManagerBase::setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config)
-{
- LOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState);
-
- bool forceVolumeReeval = false;
- switch(usage) {
- case AudioSystem::FOR_COMMUNICATION:
- if (config != AudioSystem::FORCE_SPEAKER && config != AudioSystem::FORCE_BT_SCO &&
- config != AudioSystem::FORCE_NONE) {
- LOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config);
- return;
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- case AudioSystem::FOR_MEDIA:
- if (config != AudioSystem::FORCE_HEADPHONES && config != AudioSystem::FORCE_BT_A2DP &&
- config != AudioSystem::FORCE_WIRED_ACCESSORY &&
- config != AudioSystem::FORCE_ANALOG_DOCK &&
- config != AudioSystem::FORCE_DIGITAL_DOCK && config != AudioSystem::FORCE_NONE) {
- LOGW("setForceUse() invalid config %d for FOR_MEDIA", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AudioSystem::FOR_RECORD:
- if (config != AudioSystem::FORCE_BT_SCO && config != AudioSystem::FORCE_WIRED_ACCESSORY &&
- config != AudioSystem::FORCE_NONE) {
- LOGW("setForceUse() invalid config %d for FOR_RECORD", config);
- return;
- }
- mForceUse[usage] = config;
- break;
- case AudioSystem::FOR_DOCK:
- if (config != AudioSystem::FORCE_NONE && config != AudioSystem::FORCE_BT_CAR_DOCK &&
- config != AudioSystem::FORCE_BT_DESK_DOCK &&
- config != AudioSystem::FORCE_WIRED_ACCESSORY &&
- config != AudioSystem::FORCE_ANALOG_DOCK &&
- config != AudioSystem::FORCE_DIGITAL_DOCK) {
- LOGW("setForceUse() invalid config %d for FOR_DOCK", config);
- }
- forceVolumeReeval = true;
- mForceUse[usage] = config;
- break;
- default:
- LOGW("setForceUse() invalid usage %d", usage);
- break;
- }
-
- // check for device and output changes triggered by new phone state
- uint32_t newDevice = getNewDevice(mHardwareOutput, false);
-#ifdef WITH_A2DP
- checkA2dpSuspend();
- checkOutputForAllStrategies();
-#endif
- updateDeviceForStrategy();
- setOutputDevice(mHardwareOutput, newDevice);
- if (forceVolumeReeval) {
- applyStreamVolumes(mHardwareOutput, newDevice, 0, true);
- }
-
- audio_io_handle_t activeInput = getActiveInput();
- if (activeInput != 0) {
- AudioInputDescriptor *inputDesc = mInputs.valueFor(activeInput);
- newDevice = getDeviceForInputSource(inputDesc->mInputSource);
- if (newDevice != inputDesc->mDevice) {
- LOGV("setForceUse() changing device from %x to %x for input %d",
- inputDesc->mDevice, newDevice, activeInput);
- inputDesc->mDevice = newDevice;
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)newDevice);
- mpClientInterface->setParameters(activeInput, param.toString());
- }
- }
-
-}
-
-AudioSystem::forced_config AudioPolicyManagerBase::getForceUse(AudioSystem::force_use usage)
-{
- return mForceUse[usage];
-}
-
-void AudioPolicyManagerBase::setSystemProperty(const char* property, const char* value)
-{
- LOGV("setSystemProperty() property %s, value %s", property, value);
- if (strcmp(property, "ro.camera.sound.forced") == 0) {
- if (atoi(value)) {
- LOGV("ENFORCED_AUDIBLE cannot be muted");
- mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = false;
- } else {
- LOGV("ENFORCED_AUDIBLE can be muted");
- mStreams[AudioSystem::ENFORCED_AUDIBLE].mCanBeMuted = true;
- }
- }
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutput(AudioSystem::stream_type stream,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::output_flags flags)
-{
- audio_io_handle_t output = 0;
- uint32_t latency = 0;
- routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
- uint32_t device = getDeviceForStrategy(strategy);
- LOGV("getOutput() stream %d, samplingRate %d, format %d, channels %x, flags %x", stream, samplingRate, format, channels, flags);
-
-#ifdef AUDIO_POLICY_TEST
- if (mCurOutput != 0) {
- LOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channels %x, mDirectOutput %d",
- mCurOutput, mTestSamplingRate, mTestFormat, mTestChannels, mDirectOutput);
-
- if (mTestOutputs[mCurOutput] == 0) {
- LOGV("getOutput() opening test output");
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = mTestDevice;
- outputDesc->mSamplingRate = mTestSamplingRate;
- outputDesc->mFormat = mTestFormat;
- outputDesc->mChannels = mTestChannels;
- outputDesc->mLatency = mTestLatencyMs;
- outputDesc->mFlags = (AudioSystem::output_flags)(mDirectOutput ? AudioSystem::OUTPUT_FLAG_DIRECT : 0);
- outputDesc->mRefCount[stream] = 0;
- mTestOutputs[mCurOutput] = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (mTestOutputs[mCurOutput]) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"),mCurOutput);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput],outputCmd.toString());
- addOutput(mTestOutputs[mCurOutput], outputDesc);
- }
- }
- return mTestOutputs[mCurOutput];
- }
-#endif //AUDIO_POLICY_TEST
-
- // open a direct output if required by specified parameters
- if (needsDirectOuput(stream, samplingRate, format, channels, flags, device)) {
-
- LOGV("getOutput() opening direct output device %x", device);
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = device;
- outputDesc->mSamplingRate = samplingRate;
- outputDesc->mFormat = format;
- outputDesc->mChannels = channels;
- outputDesc->mLatency = 0;
- outputDesc->mFlags = (AudioSystem::output_flags)(flags | AudioSystem::OUTPUT_FLAG_DIRECT);
- outputDesc->mRefCount[stream] = 0;
- outputDesc->mStopTime[stream] = 0;
- output = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
-
- // only accept an output with the requeted parameters
- if (output == 0 ||
- (samplingRate != 0 && samplingRate != outputDesc->mSamplingRate) ||
- (format != 0 && format != outputDesc->mFormat) ||
- (channels != 0 && channels != outputDesc->mChannels)) {
- LOGV("getOutput() failed opening direct output: samplingRate %d, format %d, channels %d",
- samplingRate, format, channels);
- if (output != 0) {
- mpClientInterface->closeOutput(output);
- }
- delete outputDesc;
- return 0;
- }
- addOutput(output, outputDesc);
- return output;
- }
-
- if (channels != 0 && channels != AudioSystem::CHANNEL_OUT_MONO &&
- channels != AudioSystem::CHANNEL_OUT_STEREO) {
- return 0;
- }
- // open a non direct output
-
- // get which output is suitable for the specified stream. The actual routing change will happen
- // when startOutput() will be called
- uint32_t a2dpDevice = device & AudioSystem::DEVICE_OUT_ALL_A2DP;
- if (AudioSystem::popCount((AudioSystem::audio_devices)device) == 2) {
-#ifdef WITH_A2DP
- if (a2dpUsedForSonification() && a2dpDevice != 0) {
- // if playing on 2 devices among which one is A2DP, use duplicated output
- LOGV("getOutput() using duplicated output");
- LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device in multiple %x selected but A2DP output not opened", device);
- output = mDuplicatedOutput;
- } else
-#endif
- {
- // if playing on 2 devices among which none is A2DP, use hardware output
- output = mHardwareOutput;
- }
- LOGV("getOutput() using output %d for 2 devices %x", output, device);
- } else {
-#ifdef WITH_A2DP
- if (a2dpDevice != 0) {
- // if playing on A2DP device, use a2dp output
- LOGW_IF((mA2dpOutput == 0), "getOutput() A2DP device %x selected but A2DP output not opened", device);
- output = mA2dpOutput;
- } else
-#endif
- {
- // if playing on not A2DP device, use hardware output
- output = mHardwareOutput;
- }
- }
-
-
- LOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
- stream, samplingRate, format, channels, flags);
-
- return output;
-}
-
-status_t AudioPolicyManagerBase::startOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
- int session)
-{
- LOGV("startOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- LOGW("startOutput() unknow output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
- routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
-
-#ifdef WITH_A2DP
- if (mA2dpOutput != 0 && !a2dpUsedForSonification() && strategy == STRATEGY_SONIFICATION) {
- setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
- }
-#endif
-
- // incremenent usage count for this stream on the requested output:
- // NOTE that the usage count is the same for duplicated output and hardware output which is
- // necassary for a correct control of hardware output routing by startOutput() and stopOutput()
- outputDesc->changeRefCount(stream, 1);
-
- setOutputDevice(output, getNewDevice(output));
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, true, false);
- }
-
- // apply volume rules for current stream and device if necessary
- checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, outputDesc->device());
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::stopOutput(audio_io_handle_t output,
- AudioSystem::stream_type stream,
- int session)
-{
- LOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- LOGW("stopOutput() unknow output %d", output);
- return BAD_VALUE;
- }
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
- routing_strategy strategy = getStrategy((AudioSystem::stream_type)stream);
-
- // handle special case for sonification while in call
- if (isInCall()) {
- handleIncallSonification(stream, false, false);
- }
-
- if (outputDesc->mRefCount[stream] > 0) {
- // decrement usage count of this stream on the output
- outputDesc->changeRefCount(stream, -1);
- // store time at which the stream was stopped - see isStreamActive()
- outputDesc->mStopTime[stream] = systemTime();
-
- setOutputDevice(output, getNewDevice(output), false, outputDesc->mLatency*2);
-
-#ifdef WITH_A2DP
- if (mA2dpOutput != 0 && !a2dpUsedForSonification() &&
- strategy == STRATEGY_SONIFICATION) {
- setStrategyMute(STRATEGY_MEDIA,
- false,
- mA2dpOutput,
- mOutputs.valueFor(mHardwareOutput)->mLatency*2);
- }
-#endif
- if (output != mHardwareOutput) {
- setOutputDevice(mHardwareOutput, getNewDevice(mHardwareOutput), true);
- }
- return NO_ERROR;
- } else {
- LOGW("stopOutput() refcount is already 0 for output %d", output);
- return INVALID_OPERATION;
- }
-}
-
-void AudioPolicyManagerBase::releaseOutput(audio_io_handle_t output)
-{
- LOGV("releaseOutput() %d", output);
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- LOGW("releaseOutput() releasing unknown output %d", output);
- return;
- }
-
-#ifdef AUDIO_POLICY_TEST
- int testIndex = testOutputIndex(output);
- if (testIndex != 0) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueAt(index);
- if (outputDesc->refCount() == 0) {
- mpClientInterface->closeOutput(output);
- delete mOutputs.valueAt(index);
- mOutputs.removeItem(output);
- mTestOutputs[testIndex] = 0;
- }
- return;
- }
-#endif //AUDIO_POLICY_TEST
-
- if (mOutputs.valueAt(index)->mFlags & AudioSystem::OUTPUT_FLAG_DIRECT) {
- mpClientInterface->closeOutput(output);
- delete mOutputs.valueAt(index);
- mOutputs.removeItem(output);
- }
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getInput(int inputSource,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::audio_in_acoustics acoustics)
-{
- audio_io_handle_t input = 0;
- uint32_t device = getDeviceForInputSource(inputSource);
-
- LOGV("getInput() inputSource %d, samplingRate %d, format %d, channels %x, acoustics %x", inputSource, samplingRate, format, channels, acoustics);
-
- if (device == 0) {
- return 0;
- }
-
- // adapt channel selection to input source
- switch(inputSource) {
- case AUDIO_SOURCE_VOICE_UPLINK:
- channels = AudioSystem::CHANNEL_IN_VOICE_UPLINK;
- break;
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- channels = AudioSystem::CHANNEL_IN_VOICE_DNLINK;
- break;
- case AUDIO_SOURCE_VOICE_CALL:
- channels = (AudioSystem::CHANNEL_IN_VOICE_UPLINK | AudioSystem::CHANNEL_IN_VOICE_DNLINK);
- break;
- default:
- break;
- }
-
- AudioInputDescriptor *inputDesc = new AudioInputDescriptor();
-
- inputDesc->mInputSource = inputSource;
- inputDesc->mDevice = device;
- inputDesc->mSamplingRate = samplingRate;
- inputDesc->mFormat = format;
- inputDesc->mChannels = channels;
- inputDesc->mAcoustics = acoustics;
- inputDesc->mRefCount = 0;
- input = mpClientInterface->openInput(&inputDesc->mDevice,
- &inputDesc->mSamplingRate,
- &inputDesc->mFormat,
- &inputDesc->mChannels,
- inputDesc->mAcoustics);
-
- // only accept input with the exact requested set of parameters
- if (input == 0 ||
- (samplingRate != inputDesc->mSamplingRate) ||
- (format != inputDesc->mFormat) ||
- (channels != inputDesc->mChannels)) {
- LOGV("getInput() failed opening input: samplingRate %d, format %d, channels %d",
- samplingRate, format, channels);
- if (input != 0) {
- mpClientInterface->closeInput(input);
- }
- delete inputDesc;
- return 0;
- }
- mInputs.add(input, inputDesc);
- return input;
-}
-
-status_t AudioPolicyManagerBase::startInput(audio_io_handle_t input)
-{
- LOGV("startInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- LOGW("startInput() unknow input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
-#ifdef AUDIO_POLICY_TEST
- if (mTestInput == 0)
-#endif //AUDIO_POLICY_TEST
- {
- // refuse 2 active AudioRecord clients at the same time
- if (getActiveInput() != 0) {
- LOGW("startInput() input %d failed: other input already started", input);
- return INVALID_OPERATION;
- }
- }
-
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)inputDesc->mDevice);
-
- param.addInt(String8(AudioParameter::keyInputSource), (int)inputDesc->mInputSource);
- LOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);
-
- mpClientInterface->setParameters(input, param.toString());
-
- inputDesc->mRefCount = 1;
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::stopInput(audio_io_handle_t input)
-{
- LOGV("stopInput() input %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- LOGW("stopInput() unknow input %d", input);
- return BAD_VALUE;
- }
- AudioInputDescriptor *inputDesc = mInputs.valueAt(index);
-
- if (inputDesc->mRefCount == 0) {
- LOGW("stopInput() input %d already stopped", input);
- return INVALID_OPERATION;
- } else {
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), 0);
- mpClientInterface->setParameters(input, param.toString());
- inputDesc->mRefCount = 0;
- return NO_ERROR;
- }
-}
-
-void AudioPolicyManagerBase::releaseInput(audio_io_handle_t input)
-{
- LOGV("releaseInput() %d", input);
- ssize_t index = mInputs.indexOfKey(input);
- if (index < 0) {
- LOGW("releaseInput() releasing unknown input %d", input);
- return;
- }
- mpClientInterface->closeInput(input);
- delete mInputs.valueAt(index);
- mInputs.removeItem(input);
- LOGV("releaseInput() exit");
-}
-
-void AudioPolicyManagerBase::initStreamVolume(AudioSystem::stream_type stream,
- int indexMin,
- int indexMax)
-{
- LOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
- if (indexMin < 0 || indexMin >= indexMax) {
- LOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax);
- return;
- }
- mStreams[stream].mIndexMin = indexMin;
- mStreams[stream].mIndexMax = indexMax;
-}
-
-status_t AudioPolicyManagerBase::setStreamVolumeIndex(AudioSystem::stream_type stream, int index)
-{
-
- if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) {
- return BAD_VALUE;
- }
-
- // Force max volume if stream cannot be muted
- if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax;
-
- LOGV("setStreamVolumeIndex() stream %d, index %d", stream, index);
- mStreams[stream].mIndexCur = index;
-
- // compute and apply stream volume on all outputs according to connected device
- status_t status = NO_ERROR;
- for (size_t i = 0; i < mOutputs.size(); i++) {
- status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), mOutputs.valueAt(i)->device());
- if (volStatus != NO_ERROR) {
- status = volStatus;
- }
- }
- return status;
-}
-
-status_t AudioPolicyManagerBase::getStreamVolumeIndex(AudioSystem::stream_type stream, int *index)
-{
- if (index == 0) {
- return BAD_VALUE;
- }
- LOGV("getStreamVolumeIndex() stream %d", stream);
- *index = mStreams[stream].mIndexCur;
- return NO_ERROR;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getOutputForEffect(effect_descriptor_t *desc)
-{
- LOGV("getOutputForEffect()");
- // apply simple rule where global effects are attached to the same output as MUSIC streams
- return getOutput(AudioSystem::MUSIC);
-}
-
-status_t AudioPolicyManagerBase::registerEffect(effect_descriptor_t *desc,
- audio_io_handle_t output,
- uint32_t strategy,
- int session,
- int id)
-{
- ssize_t index = mOutputs.indexOfKey(output);
- if (index < 0) {
- LOGW("registerEffect() unknown output %d", output);
- return INVALID_OPERATION;
- }
-
- if (mTotalEffectsCpuLoad + desc->cpuLoad > getMaxEffectsCpuLoad()) {
- LOGW("registerEffect() CPU Load limit exceeded for Fx %s, CPU %f MIPS",
- desc->name, (float)desc->cpuLoad/10);
- return INVALID_OPERATION;
- }
- if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) {
- LOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB",
- desc->name, desc->memoryUsage);
- return INVALID_OPERATION;
- }
- mTotalEffectsCpuLoad += desc->cpuLoad;
- mTotalEffectsMemory += desc->memoryUsage;
- LOGV("registerEffect() effect %s, output %d, strategy %d session %d id %d",
- desc->name, output, strategy, session, id);
-
- LOGV("registerEffect() CPU %d, memory %d", desc->cpuLoad, desc->memoryUsage);
- LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
-
- EffectDescriptor *pDesc = new EffectDescriptor();
- memcpy (&pDesc->mDesc, desc, sizeof(effect_descriptor_t));
- pDesc->mOutput = output;
- pDesc->mStrategy = (routing_strategy)strategy;
- pDesc->mSession = session;
- mEffects.add(id, pDesc);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::unregisterEffect(int id)
-{
- ssize_t index = mEffects.indexOfKey(id);
- if (index < 0) {
- LOGW("unregisterEffect() unknown effect ID %d", id);
- return INVALID_OPERATION;
- }
-
- EffectDescriptor *pDesc = mEffects.valueAt(index);
-
- if (mTotalEffectsCpuLoad < pDesc->mDesc.cpuLoad) {
- LOGW("unregisterEffect() CPU load %d too high for total %d",
- pDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad);
- pDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad;
- }
- mTotalEffectsCpuLoad -= pDesc->mDesc.cpuLoad;
- if (mTotalEffectsMemory < pDesc->mDesc.memoryUsage) {
- LOGW("unregisterEffect() memory %d too big for total %d",
- pDesc->mDesc.memoryUsage, mTotalEffectsMemory);
- pDesc->mDesc.memoryUsage = mTotalEffectsMemory;
- }
- mTotalEffectsMemory -= pDesc->mDesc.memoryUsage;
- LOGV("unregisterEffect() effect %s, ID %d, CPU %d, memory %d",
- pDesc->mDesc.name, id, pDesc->mDesc.cpuLoad, pDesc->mDesc.memoryUsage);
- LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
-
- mEffects.removeItem(id);
- delete pDesc;
-
- return NO_ERROR;
-}
-
-bool AudioPolicyManagerBase::isStreamActive(int stream, uint32_t inPastMs) const
-{
- nsecs_t sysTime = systemTime();
- for (size_t i = 0; i < mOutputs.size(); i++) {
- if (mOutputs.valueAt(i)->mRefCount[stream] != 0 ||
- ns2ms(sysTime - mOutputs.valueAt(i)->mStopTime[stream]) < inPastMs) {
- return true;
- }
- }
- return false;
-}
-
-
-status_t AudioPolicyManagerBase::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
- result.append(buffer);
- snprintf(buffer, SIZE, " Hardware Output: %d\n", mHardwareOutput);
- result.append(buffer);
-#ifdef WITH_A2DP
- snprintf(buffer, SIZE, " A2DP Output: %d\n", mA2dpOutput);
- result.append(buffer);
- snprintf(buffer, SIZE, " Duplicated Output: %d\n", mDuplicatedOutput);
- result.append(buffer);
- snprintf(buffer, SIZE, " A2DP device address: %s\n", mA2dpDeviceAddress.string());
- result.append(buffer);
-#endif
- snprintf(buffer, SIZE, " SCO device address: %s\n", mScoDeviceAddress.string());
- result.append(buffer);
- snprintf(buffer, SIZE, " Output devices: %08x\n", mAvailableOutputDevices);
- result.append(buffer);
- snprintf(buffer, SIZE, " Input devices: %08x\n", mAvailableInputDevices);
- result.append(buffer);
- snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ringer mode: %d\n", mRingerMode);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for communications %d\n", mForceUse[AudioSystem::FOR_COMMUNICATION]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AudioSystem::FOR_MEDIA]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AudioSystem::FOR_RECORD]);
- result.append(buffer);
- snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AudioSystem::FOR_DOCK]);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- snprintf(buffer, SIZE, "\nOutputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mOutputs.size(); i++) {
- snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mOutputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nInputs dump:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mInputs.size(); i++) {
- snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mInputs.valueAt(i)->dump(fd);
- }
-
- snprintf(buffer, SIZE, "\nStreams dump:\n");
- write(fd, buffer, strlen(buffer));
- snprintf(buffer, SIZE, " Stream Index Min Index Max Index Cur Can be muted\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- snprintf(buffer, SIZE, " %02d", i);
- mStreams[i].dump(buffer + 3, SIZE);
- write(fd, buffer, strlen(buffer));
- }
-
- snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n",
- (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory);
- write(fd, buffer, strlen(buffer));
-
- snprintf(buffer, SIZE, "Registered effects:\n");
- write(fd, buffer, strlen(buffer));
- for (size_t i = 0; i < mEffects.size(); i++) {
- snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i));
- write(fd, buffer, strlen(buffer));
- mEffects.valueAt(i)->dump(fd);
- }
-
-
- return NO_ERROR;
-}
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManagerBase
-// ----------------------------------------------------------------------------
-
-AudioPolicyManagerBase::AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface)
- :
-#ifdef AUDIO_POLICY_TEST
- Thread(false),
-#endif //AUDIO_POLICY_TEST
- mPhoneState(AudioSystem::MODE_NORMAL), mRingerMode(0),
- mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
- mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
- mA2dpSuspended(false)
-{
- mpClientInterface = clientInterface;
-
- for (int i = 0; i < AudioSystem::NUM_FORCE_USE; i++) {
- mForceUse[i] = AudioSystem::FORCE_NONE;
- }
-
- initializeVolumeCurves();
-
- // devices available by default are speaker, ear piece and microphone
- mAvailableOutputDevices = AudioSystem::DEVICE_OUT_EARPIECE |
- AudioSystem::DEVICE_OUT_SPEAKER;
- mAvailableInputDevices = AudioSystem::DEVICE_IN_BUILTIN_MIC;
-
-#ifdef WITH_A2DP
- mA2dpOutput = 0;
- mDuplicatedOutput = 0;
- mA2dpDeviceAddress = String8("");
-#endif
- mScoDeviceAddress = String8("");
-
- // open hardware output
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
- mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
-
- if (mHardwareOutput == 0) {
- LOGE("Failed to initialize hardware output stream, samplingRate: %d, format %d, channels %d",
- outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
- } else {
- addOutput(mHardwareOutput, outputDesc);
- setOutputDevice(mHardwareOutput, (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER, true);
- //TODO: configure audio effect output stage here
- }
-
- updateDeviceForStrategy();
-#ifdef AUDIO_POLICY_TEST
- if (mHardwareOutput != 0) {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
-
- mTestDevice = AudioSystem::DEVICE_OUT_SPEAKER;
- mTestSamplingRate = 44100;
- mTestFormat = AudioSystem::PCM_16_BIT;
- mTestChannels = AudioSystem::CHANNEL_OUT_STEREO;
- mTestLatencyMs = 0;
- mCurOutput = 0;
- mDirectOutput = false;
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- mTestOutputs[i] = 0;
- }
-
- const size_t SIZE = 256;
- char buffer[SIZE];
- snprintf(buffer, SIZE, "AudioPolicyManagerTest");
- run(buffer, ANDROID_PRIORITY_AUDIO);
- }
-#endif //AUDIO_POLICY_TEST
-}
-
-AudioPolicyManagerBase::~AudioPolicyManagerBase()
-{
-#ifdef AUDIO_POLICY_TEST
- exit();
-#endif //AUDIO_POLICY_TEST
- for (size_t i = 0; i < mOutputs.size(); i++) {
- mpClientInterface->closeOutput(mOutputs.keyAt(i));
- delete mOutputs.valueAt(i);
- }
- mOutputs.clear();
- for (size_t i = 0; i < mInputs.size(); i++) {
- mpClientInterface->closeInput(mInputs.keyAt(i));
- delete mInputs.valueAt(i);
- }
- mInputs.clear();
-}
-
-status_t AudioPolicyManagerBase::initCheck()
-{
- return (mHardwareOutput == 0) ? NO_INIT : NO_ERROR;
-}
-
-#ifdef AUDIO_POLICY_TEST
-bool AudioPolicyManagerBase::threadLoop()
-{
- LOGV("entering threadLoop()");
- while (!exitPending())
- {
- String8 command;
- int valueInt;
- String8 value;
-
- Mutex::Autolock _l(mLock);
- mWaitWorkCV.waitRelative(mLock, milliseconds(50));
-
- command = mpClientInterface->getParameters(0, String8("test_cmd_policy"));
- AudioParameter param = AudioParameter(command);
-
- if (param.getInt(String8("test_cmd_policy"), valueInt) == NO_ERROR &&
- valueInt != 0) {
- LOGV("Test command %s received", command.string());
- String8 target;
- if (param.get(String8("target"), target) != NO_ERROR) {
- target = "Manager";
- }
- if (param.getInt(String8("test_cmd_policy_output"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_output"));
- mCurOutput = valueInt;
- }
- if (param.get(String8("test_cmd_policy_direct"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_direct"));
- if (value == "false") {
- mDirectOutput = false;
- } else if (value == "true") {
- mDirectOutput = true;
- }
- }
- if (param.getInt(String8("test_cmd_policy_input"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_input"));
- mTestInput = valueInt;
- }
-
- if (param.get(String8("test_cmd_policy_format"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_format"));
- int format = AudioSystem::INVALID_FORMAT;
- if (value == "PCM 16 bits") {
- format = AudioSystem::PCM_16_BIT;
- } else if (value == "PCM 8 bits") {
- format = AudioSystem::PCM_8_BIT;
- } else if (value == "Compressed MP3") {
- format = AudioSystem::MP3;
- }
- if (format != AudioSystem::INVALID_FORMAT) {
- if (target == "Manager") {
- mTestFormat = format;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("format"), format);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.get(String8("test_cmd_policy_channels"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_channels"));
- int channels = 0;
-
- if (value == "Channels Stereo") {
- channels = AudioSystem::CHANNEL_OUT_STEREO;
- } else if (value == "Channels Mono") {
- channels = AudioSystem::CHANNEL_OUT_MONO;
- }
- if (channels != 0) {
- if (target == "Manager") {
- mTestChannels = channels;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("channels"), channels);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
- if (param.getInt(String8("test_cmd_policy_sampleRate"), valueInt) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_sampleRate"));
- if (valueInt >= 0 && valueInt <= 96000) {
- int samplingRate = valueInt;
- if (target == "Manager") {
- mTestSamplingRate = samplingRate;
- } else if (mTestOutputs[mCurOutput] != 0) {
- AudioParameter outputParam = AudioParameter();
- outputParam.addInt(String8("sampling_rate"), samplingRate);
- mpClientInterface->setParameters(mTestOutputs[mCurOutput], outputParam.toString());
- }
- }
- }
-
- if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) {
- param.remove(String8("test_cmd_policy_reopen"));
-
- mpClientInterface->closeOutput(mHardwareOutput);
- delete mOutputs.valueFor(mHardwareOutput);
- mOutputs.removeItem(mHardwareOutput);
-
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = (uint32_t)AudioSystem::DEVICE_OUT_SPEAKER;
- mHardwareOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (mHardwareOutput == 0) {
- LOGE("Failed to reopen hardware output stream, samplingRate: %d, format %d, channels %d",
- outputDesc->mSamplingRate, outputDesc->mFormat, outputDesc->mChannels);
- } else {
- AudioParameter outputCmd = AudioParameter();
- outputCmd.addInt(String8("set_id"), 0);
- mpClientInterface->setParameters(mHardwareOutput, outputCmd.toString());
- addOutput(mHardwareOutput, outputDesc);
- }
- }
-
-
- mpClientInterface->setParameters(0, String8("test_cmd_policy="));
- }
- }
- return false;
-}
-
-void AudioPolicyManagerBase::exit()
-{
- {
- AutoMutex _l(mLock);
- requestExit();
- mWaitWorkCV.signal();
- }
- requestExitAndWait();
-}
-
-int AudioPolicyManagerBase::testOutputIndex(audio_io_handle_t output)
-{
- for (int i = 0; i < NUM_TEST_OUTPUTS; i++) {
- if (output == mTestOutputs[i]) return i;
- }
- return 0;
-}
-#endif //AUDIO_POLICY_TEST
-
-// ---
-
-void AudioPolicyManagerBase::addOutput(audio_io_handle_t id, AudioOutputDescriptor *outputDesc)
-{
- outputDesc->mId = id;
- mOutputs.add(id, outputDesc);
-}
-
-
-#ifdef WITH_A2DP
-status_t AudioPolicyManagerBase::handleA2dpConnection(AudioSystem::audio_devices device,
- const char *device_address)
-{
- // when an A2DP device is connected, open an A2DP and a duplicated output
- LOGV("opening A2DP output for device %s", device_address);
- AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor();
- outputDesc->mDevice = device;
- mA2dpOutput = mpClientInterface->openOutput(&outputDesc->mDevice,
- &outputDesc->mSamplingRate,
- &outputDesc->mFormat,
- &outputDesc->mChannels,
- &outputDesc->mLatency,
- outputDesc->mFlags);
- if (mA2dpOutput) {
- // add A2DP output descriptor
- addOutput(mA2dpOutput, outputDesc);
-
- //TODO: configure audio effect output stage here
-
- // set initial stream volume for A2DP device
- applyStreamVolumes(mA2dpOutput, device);
- if (a2dpUsedForSonification()) {
- mDuplicatedOutput = mpClientInterface->openDuplicateOutput(mA2dpOutput, mHardwareOutput);
- }
- if (mDuplicatedOutput != 0 ||
- !a2dpUsedForSonification()) {
- // If both A2DP and duplicated outputs are open, send device address to A2DP hardware
- // interface
- AudioParameter param;
- param.add(String8("a2dp_sink_address"), String8(device_address));
- mpClientInterface->setParameters(mA2dpOutput, param.toString());
- mA2dpDeviceAddress = String8(device_address, MAX_DEVICE_ADDRESS_LEN);
-
- if (a2dpUsedForSonification()) {
- // add duplicated output descriptor
- AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor();
- dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput);
- dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput);
- dupOutputDesc->mSamplingRate = outputDesc->mSamplingRate;
- dupOutputDesc->mFormat = outputDesc->mFormat;
- dupOutputDesc->mChannels = outputDesc->mChannels;
- dupOutputDesc->mLatency = outputDesc->mLatency;
- addOutput(mDuplicatedOutput, dupOutputDesc);
- applyStreamVolumes(mDuplicatedOutput, device);
- }
- } else {
- LOGW("getOutput() could not open duplicated output for %d and %d",
- mHardwareOutput, mA2dpOutput);
- mpClientInterface->closeOutput(mA2dpOutput);
- mOutputs.removeItem(mA2dpOutput);
- mA2dpOutput = 0;
- delete outputDesc;
- return NO_INIT;
- }
- } else {
- LOGW("setDeviceConnectionState() could not open A2DP output for device %x", device);
- delete outputDesc;
- return NO_INIT;
- }
- AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
-
- if (!a2dpUsedForSonification()) {
- // mute music on A2DP output if a notification or ringtone is playing
- uint32_t refCount = hwOutputDesc->strategyRefCount(STRATEGY_SONIFICATION);
- for (uint32_t i = 0; i < refCount; i++) {
- setStrategyMute(STRATEGY_MEDIA, true, mA2dpOutput);
- }
- }
-
- mA2dpSuspended = false;
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManagerBase::handleA2dpDisconnection(AudioSystem::audio_devices device,
- const char *device_address)
-{
- if (mA2dpOutput == 0) {
- LOGW("setDeviceConnectionState() disconnecting A2DP and no A2DP output!");
- return INVALID_OPERATION;
- }
-
- if (mA2dpDeviceAddress != device_address) {
- LOGW("setDeviceConnectionState() disconnecting unknow A2DP sink address %s", device_address);
- return INVALID_OPERATION;
- }
-
- // mute media strategy to avoid outputting sound on hardware output while music stream
- // is switched from A2DP output and before music is paused by music application
- setStrategyMute(STRATEGY_MEDIA, true, mHardwareOutput);
- setStrategyMute(STRATEGY_MEDIA, false, mHardwareOutput, MUTE_TIME_MS);
-
- if (!a2dpUsedForSonification()) {
- // unmute music on A2DP output if a notification or ringtone is playing
- uint32_t refCount = mOutputs.valueFor(mHardwareOutput)->strategyRefCount(STRATEGY_SONIFICATION);
- for (uint32_t i = 0; i < refCount; i++) {
- setStrategyMute(STRATEGY_MEDIA, false, mA2dpOutput);
- }
- }
- mA2dpDeviceAddress = "";
- mA2dpSuspended = false;
- return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::closeA2dpOutputs()
-{
-
- LOGV("setDeviceConnectionState() closing A2DP and duplicated output!");
-
- if (mDuplicatedOutput != 0) {
- AudioOutputDescriptor *dupOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
- AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
- // As all active tracks on duplicated output will be deleted,
- // and as they were also referenced on hardware output, the reference
- // count for their stream type must be adjusted accordingly on
- // hardware output.
- for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
- int refCount = dupOutputDesc->mRefCount[i];
- hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
- }
-
- mpClientInterface->closeOutput(mDuplicatedOutput);
- delete mOutputs.valueFor(mDuplicatedOutput);
- mOutputs.removeItem(mDuplicatedOutput);
- mDuplicatedOutput = 0;
- }
- if (mA2dpOutput != 0) {
- AudioParameter param;
- param.add(String8("closing"), String8("true"));
- mpClientInterface->setParameters(mA2dpOutput, param.toString());
-
- mpClientInterface->closeOutput(mA2dpOutput);
- delete mOutputs.valueFor(mA2dpOutput);
- mOutputs.removeItem(mA2dpOutput);
- mA2dpOutput = 0;
- }
-}
-
-void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy)
-{
- uint32_t prevDevice = getDeviceForStrategy(strategy);
- uint32_t curDevice = getDeviceForStrategy(strategy, false);
- bool a2dpWasUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(prevDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
- bool a2dpIsUsed = AudioSystem::isA2dpDevice((AudioSystem::audio_devices)(curDevice & ~AudioSystem::DEVICE_OUT_SPEAKER));
- audio_io_handle_t srcOutput = 0;
- audio_io_handle_t dstOutput = 0;
-
- if (a2dpWasUsed && !a2dpIsUsed) {
- bool dupUsed = a2dpUsedForSonification() && a2dpWasUsed && (AudioSystem::popCount(prevDevice) == 2);
- dstOutput = mHardwareOutput;
- if (dupUsed) {
- LOGV("checkOutputForStrategy() moving strategy %d from duplicated", strategy);
- srcOutput = mDuplicatedOutput;
- } else {
- LOGV("checkOutputForStrategy() moving strategy %d from a2dp", strategy);
- srcOutput = mA2dpOutput;
- }
- }
- if (a2dpIsUsed && !a2dpWasUsed) {
- bool dupUsed = a2dpUsedForSonification() && a2dpIsUsed && (AudioSystem::popCount(curDevice) == 2);
- srcOutput = mHardwareOutput;
- if (dupUsed) {
- LOGV("checkOutputForStrategy() moving strategy %d to duplicated", strategy);
- dstOutput = mDuplicatedOutput;
- } else {
- LOGV("checkOutputForStrategy() moving strategy %d to a2dp", strategy);
- dstOutput = mA2dpOutput;
- }
- }
-
- if (srcOutput != 0 && dstOutput != 0) {
- // Move effects associated to this strategy from previous output to new output
- for (size_t i = 0; i < mEffects.size(); i++) {
- EffectDescriptor *desc = mEffects.valueAt(i);
- if (desc->mSession != AudioSystem::SESSION_OUTPUT_STAGE &&
- desc->mStrategy == strategy &&
- desc->mOutput == srcOutput) {
- LOGV("checkOutputForStrategy() moving effect %d to output %d", mEffects.keyAt(i), dstOutput);
- mpClientInterface->moveEffects(desc->mSession, srcOutput, dstOutput);
- desc->mOutput = dstOutput;
- }
- }
- // Move tracks associated to this strategy from previous output to new output
- for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
- if (getStrategy((AudioSystem::stream_type)i) == strategy) {
- mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, dstOutput);
- }
- }
- }
-}
-
-void AudioPolicyManagerBase::checkOutputForAllStrategies()
-{
- checkOutputForStrategy(STRATEGY_PHONE);
- checkOutputForStrategy(STRATEGY_SONIFICATION);
- checkOutputForStrategy(STRATEGY_MEDIA);
- checkOutputForStrategy(STRATEGY_DTMF);
-}
-
-void AudioPolicyManagerBase::checkA2dpSuspend()
-{
- // suspend A2DP output if:
- // (NOT already suspended) &&
- // ((SCO device is connected &&
- // (forced usage for communication || for record is SCO))) ||
- // (phone state is ringing || in call)
- //
- // restore A2DP output if:
- // (Already suspended) &&
- // ((SCO device is NOT connected ||
- // (forced usage NOT for communication && NOT for record is SCO))) &&
- // (phone state is NOT ringing && NOT in call)
- //
- if (mA2dpOutput == 0) {
- return;
- }
-
- if (mA2dpSuspended) {
- if (((mScoDeviceAddress == "") ||
- ((mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO) &&
- (mForceUse[AudioSystem::FOR_RECORD] != AudioSystem::FORCE_BT_SCO))) &&
- ((mPhoneState != AudioSystem::MODE_IN_CALL) &&
- (mPhoneState != AudioSystem::MODE_RINGTONE))) {
-
- mpClientInterface->restoreOutput(mA2dpOutput);
- mA2dpSuspended = false;
- }
- } else {
- if (((mScoDeviceAddress != "") &&
- ((mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
- (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO))) ||
- ((mPhoneState == AudioSystem::MODE_IN_CALL) ||
- (mPhoneState == AudioSystem::MODE_RINGTONE))) {
-
- mpClientInterface->suspendOutput(mA2dpOutput);
- mA2dpSuspended = true;
- }
- }
-}
-
-
-#endif
-
-uint32_t AudioPolicyManagerBase::getNewDevice(audio_io_handle_t output, bool fromCache)
-{
- uint32_t device = 0;
-
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- // check the following by order of priority to request a routing change if necessary:
- // 1: we are in call or the strategy phone is active on the hardware output:
- // use device for strategy phone
- // 2: the strategy sonification is active on the hardware output:
- // use device for strategy sonification
- // 3: the strategy media is active on the hardware output:
- // use device for strategy media
- // 4: the strategy DTMF is active on the hardware output:
- // use device for strategy DTMF
- if (isInCall() ||
- outputDesc->isUsedByStrategy(STRATEGY_PHONE)) {
- device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
- } else if (outputDesc->isUsedByStrategy(STRATEGY_SONIFICATION)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
- } else if (outputDesc->isUsedByStrategy(STRATEGY_MEDIA)) {
- device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
- } else if (outputDesc->isUsedByStrategy(STRATEGY_DTMF)) {
- device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
- }
-
- LOGV("getNewDevice() selected device %x", device);
- return device;
-}
-
-uint32_t AudioPolicyManagerBase::getStrategyForStream(AudioSystem::stream_type stream) {
- return (uint32_t)getStrategy(stream);
-}
-
-uint32_t AudioPolicyManagerBase::getDevicesForStream(AudioSystem::stream_type stream) {
- uint32_t devices;
- // By checking the range of stream before calling getStrategy, we avoid
- // getStrategy's behavior for invalid streams. getStrategy would do a LOGE
- // and then return STRATEGY_MEDIA, but we want to return the empty set.
- if (stream < (AudioSystem::stream_type) 0 || stream >= AudioSystem::NUM_STREAM_TYPES) {
- devices = 0;
- } else {
- AudioPolicyManagerBase::routing_strategy strategy = getStrategy(stream);
- devices = getDeviceForStrategy(strategy, true);
- }
- return devices;
-}
-
-AudioPolicyManagerBase::routing_strategy AudioPolicyManagerBase::getStrategy(
- AudioSystem::stream_type stream) {
- // stream to strategy mapping
- switch (stream) {
- case AudioSystem::VOICE_CALL:
- case AudioSystem::BLUETOOTH_SCO:
- return STRATEGY_PHONE;
- case AudioSystem::RING:
- case AudioSystem::NOTIFICATION:
- case AudioSystem::ALARM:
- case AudioSystem::ENFORCED_AUDIBLE:
- return STRATEGY_SONIFICATION;
- case AudioSystem::DTMF:
- return STRATEGY_DTMF;
- default:
- LOGE("unknown stream type");
- case AudioSystem::SYSTEM:
- // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs
- // while key clicks are played produces a poor result
- case AudioSystem::TTS:
- case AudioSystem::MUSIC:
- return STRATEGY_MEDIA;
- }
-}
-
-uint32_t AudioPolicyManagerBase::getDeviceForStrategy(routing_strategy strategy, bool fromCache)
-{
- uint32_t device = 0;
-
- if (fromCache) {
- LOGV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]);
- return mDeviceForStrategy[strategy];
- }
-
- switch (strategy) {
- case STRATEGY_DTMF:
- if (!isInCall()) {
- // when off call, DTMF strategy follows the same rules as MEDIA strategy
- device = getDeviceForStrategy(STRATEGY_MEDIA, false);
- break;
- }
- // when in call, DTMF and PHONE strategies follow the same rules
- // FALL THROUGH
-
- case STRATEGY_PHONE:
- // for phone strategy, we first consider the forced use and then the available devices by order
- // of priority
- switch (mForceUse[AudioSystem::FOR_COMMUNICATION]) {
- case AudioSystem::FORCE_BT_SCO:
- if (!isInCall() || strategy != STRATEGY_DTMF) {
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
- if (device) break;
- }
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_SCO;
- if (device) break;
- // if SCO device is requested but no SCO device is available, fall back to default case
- // FALL THROUGH
-
- default: // FORCE_NONE
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
- if (device) break;
-#ifdef WITH_A2DP
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
- if (!isInCall() && !mA2dpSuspended) {
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- if (device) break;
- }
-#endif
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_EARPIECE;
- if (device == 0) {
- LOGE("getDeviceForStrategy() earpiece device not found");
- }
- break;
-
- case AudioSystem::FORCE_SPEAKER:
-#ifdef WITH_A2DP
- // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to
- // A2DP speaker when forcing to speaker output
- if (!isInCall() && !mA2dpSuspended) {
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- if (device) break;
- }
-#endif
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
- if (device == 0) {
- LOGE("getDeviceForStrategy() speaker device not found");
- }
- break;
- }
- break;
-
- case STRATEGY_SONIFICATION:
-
- // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
- // handleIncallSonification().
- if (isInCall()) {
- device = getDeviceForStrategy(STRATEGY_PHONE, false);
- break;
- }
- device = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
- if (device == 0) {
- LOGE("getDeviceForStrategy() speaker device not found");
- }
- // The second device used for sonification is the same as the device used by media strategy
- // FALL THROUGH
-
- case STRATEGY_MEDIA: {
- uint32_t device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADPHONE;
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_WIRED_HEADSET;
- }
-#ifdef WITH_A2DP
- if ((mA2dpOutput != 0) && !mA2dpSuspended &&
- (strategy != STRATEGY_SONIFICATION || a2dpUsedForSonification())) {
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- }
- }
-#endif
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_AUX_DIGITAL;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_DGTL_DOCK_HEADSET;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_ANLG_DOCK_HEADSET;
- }
- if (device2 == 0) {
- device2 = mAvailableOutputDevices & AudioSystem::DEVICE_OUT_SPEAKER;
- }
-
- // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION, 0 otherwise
- device |= device2;
- if (device == 0) {
- LOGE("getDeviceForStrategy() speaker device not found");
- }
- } break;
-
- default:
- LOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
- break;
- }
-
- LOGV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
- return device;
-}
-
-void AudioPolicyManagerBase::updateDeviceForStrategy()
-{
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false);
- }
-}
-
-void AudioPolicyManagerBase::setOutputDevice(audio_io_handle_t output, uint32_t device, bool force, int delayMs)
-{
- LOGV("setOutputDevice() output %d device %x delayMs %d", output, device, delayMs);
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-
-
- if (outputDesc->isDuplicated()) {
- setOutputDevice(outputDesc->mOutput1->mId, device, force, delayMs);
- setOutputDevice(outputDesc->mOutput2->mId, device, force, delayMs);
- return;
- }
-#ifdef WITH_A2DP
- // filter devices according to output selected
- if (output == mA2dpOutput) {
- device &= AudioSystem::DEVICE_OUT_ALL_A2DP;
- } else {
- device &= ~AudioSystem::DEVICE_OUT_ALL_A2DP;
- }
-#endif
-
- uint32_t prevDevice = (uint32_t)outputDesc->device();
- // Do not change the routing if:
- // - the requestede device is 0
- // - the requested device is the same as current device and force is not specified.
- // Doing this check here allows the caller to call setOutputDevice() without conditions
- if ((device == 0 || device == prevDevice) && !force) {
- LOGV("setOutputDevice() setting same device %x or null device for output %d", device, output);
- return;
- }
-
- outputDesc->mDevice = device;
- // mute media streams if both speaker and headset are selected
- if (output == mHardwareOutput && AudioSystem::popCount(device) == 2) {
- setStrategyMute(STRATEGY_MEDIA, true, output);
- // wait for the PCM output buffers to empty before proceeding with the rest of the command
- usleep(outputDesc->mLatency*2*1000);
- }
-
- // do the routing
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)device);
- mpClientInterface->setParameters(mHardwareOutput, param.toString(), delayMs);
- // update stream volumes according to new device
- applyStreamVolumes(output, device, delayMs);
-
- // if changing from a combined headset + speaker route, unmute media streams
- if (output == mHardwareOutput && AudioSystem::popCount(prevDevice) == 2) {
- setStrategyMute(STRATEGY_MEDIA, false, output, delayMs);
- }
-}
-
-uint32_t AudioPolicyManagerBase::getDeviceForInputSource(int inputSource)
-{
- uint32_t device;
-
- switch(inputSource) {
- case AUDIO_SOURCE_DEFAULT:
- case AUDIO_SOURCE_MIC:
- case AUDIO_SOURCE_VOICE_RECOGNITION:
- case AUDIO_SOURCE_VOICE_COMMUNICATION:
- if (mForceUse[AudioSystem::FOR_RECORD] == AudioSystem::FORCE_BT_SCO &&
- mAvailableInputDevices & AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AudioSystem::DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (mAvailableInputDevices & AudioSystem::DEVICE_IN_WIRED_HEADSET) {
- device = AudioSystem::DEVICE_IN_WIRED_HEADSET;
- } else {
- device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_CAMCORDER:
- if (hasBackMicrophone()) {
- device = AudioSystem::DEVICE_IN_BACK_MIC;
- } else {
- device = AudioSystem::DEVICE_IN_BUILTIN_MIC;
- }
- break;
- case AUDIO_SOURCE_VOICE_UPLINK:
- case AUDIO_SOURCE_VOICE_DOWNLINK:
- case AUDIO_SOURCE_VOICE_CALL:
- device = AudioSystem::DEVICE_IN_VOICE_CALL;
- break;
- default:
- LOGW("getInput() invalid input source %d", inputSource);
- device = 0;
- break;
- }
- LOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
- return device;
-}
-
-audio_io_handle_t AudioPolicyManagerBase::getActiveInput()
-{
- for (size_t i = 0; i < mInputs.size(); i++) {
- if (mInputs.valueAt(i)->mRefCount > 0) {
- return mInputs.keyAt(i);
- }
- }
- return 0;
-}
-
-float AudioPolicyManagerBase::volIndexToAmpl(uint32_t device, const StreamDescriptor& streamDesc,
- int indexInUi) {
- // the volume index in the UI is relative to the min and max volume indices for this stream type
- int nbSteps = 1 + streamDesc.mVolIndex[StreamDescriptor::VOLMAX] -
- streamDesc.mVolIndex[StreamDescriptor::VOLMIN];
- 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 < streamDesc.mVolIndex[StreamDescriptor::VOLMIN]) { // out of bounds
- return 0.0f;
- } else if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLKNEE1]) {
- segment = 0;
- } else if (volIdx < streamDesc.mVolIndex[StreamDescriptor::VOLKNEE2]) {
- segment = 1;
- } else if (volIdx <= streamDesc.mVolIndex[StreamDescriptor::VOLMAX]) {
- segment = 2;
- } else { // out of bounds
- return 1.0f;
- }
-
- // linear interpolation in the attenuation table in dB
- float decibels = streamDesc.mVolDbAtt[segment] +
- ((float)(volIdx - streamDesc.mVolIndex[segment])) *
- ( (streamDesc.mVolDbAtt[segment+1] - streamDesc.mVolDbAtt[segment]) /
- ((float)(streamDesc.mVolIndex[segment+1] - streamDesc.mVolIndex[segment])) );
-
- float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
- LOGV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
- streamDesc.mVolIndex[segment], volIdx, streamDesc.mVolIndex[segment+1],
- streamDesc.mVolDbAtt[segment], decibels, streamDesc.mVolDbAtt[segment+1],
- amplification);
-
- return amplification;
-}
-
-void AudioPolicyManagerBase::initializeVolumeCurves() {
- // initialize the volume curves to a (-49.5 - 0 dB) attenuation in 0.5dB steps
- for (int i=0 ; i< AudioSystem::NUM_STREAM_TYPES ; i++) {
- mStreams[i].mVolIndex[StreamDescriptor::VOLMIN] = 1;
- mStreams[i].mVolDbAtt[StreamDescriptor::VOLMIN] = -49.5f;
- mStreams[i].mVolIndex[StreamDescriptor::VOLKNEE1] = 33;
- mStreams[i].mVolDbAtt[StreamDescriptor::VOLKNEE1] = -33.5f;
- mStreams[i].mVolIndex[StreamDescriptor::VOLKNEE2] = 66;
- mStreams[i].mVolDbAtt[StreamDescriptor::VOLKNEE2] = -17.0f;
- // here we use 100 steps to avoid rounding errors
- // when computing the volume in volIndexToAmpl()
- mStreams[i].mVolIndex[StreamDescriptor::VOLMAX] = 100;
- mStreams[i].mVolDbAtt[StreamDescriptor::VOLMAX] = 0.0f;
- }
-
- // Modification for music: more attenuation for lower volumes, finer steps at high volumes
- mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLMIN] = 1;
- mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLMIN] = -58.0f;
- mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLKNEE1] = 20;
- mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLKNEE1] = -40.0f;
- mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLKNEE2] = 60;
- mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLKNEE2] = -17.0f;
- mStreams[AudioSystem::MUSIC].mVolIndex[StreamDescriptor::VOLMAX] = 100;
- mStreams[AudioSystem::MUSIC].mVolDbAtt[StreamDescriptor::VOLMAX] = 0.0f;
-}
-
-float AudioPolicyManagerBase::computeVolume(int stream, int index, audio_io_handle_t output, uint32_t device)
-{
- float volume = 1.0;
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
- StreamDescriptor &streamDesc = mStreams[stream];
-
- if (device == 0) {
- device = outputDesc->device();
- }
-
- // if volume is not 0 (not muted), force media volume to max on digital output
- if (stream == AudioSystem::MUSIC &&
- index != mStreams[stream].mIndexMin &&
- device == AudioSystem::DEVICE_OUT_AUX_DIGITAL) {
- return 1.0;
- }
-
- volume = volIndexToAmpl(device, streamDesc, index);
-
- // if a headset is connected, apply the following rules to ring tones and notifications
- // to avoid sound level bursts in user's ears:
- // - always attenuate ring tones and notifications volume by 6dB
- // - if music is playing, always limit the volume to current music volume,
- // with a minimum threshold at -36dB so that notification is always perceived.
- if ((device &
- (AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP |
- AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
- AudioSystem::DEVICE_OUT_WIRED_HEADSET |
- AudioSystem::DEVICE_OUT_WIRED_HEADPHONE)) &&
- ((getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) ||
- (stream == AudioSystem::SYSTEM)) &&
- streamDesc.mCanBeMuted) {
- volume *= SONIFICATION_HEADSET_VOLUME_FACTOR;
- // when the phone is ringing we must consider that music could have been paused just before
- // by the music application and behave as if music was active if the last music track was
- // just stopped
- if (outputDesc->mRefCount[AudioSystem::MUSIC] || mLimitRingtoneVolume) {
- float musicVol = computeVolume(AudioSystem::MUSIC, mStreams[AudioSystem::MUSIC].mIndexCur, output, device);
- float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? musicVol : SONIFICATION_HEADSET_VOLUME_MIN;
- if (volume > minVol) {
- volume = minVol;
- LOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol);
- }
- }
- }
-
- return volume;
-}
-
-status_t AudioPolicyManagerBase::checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs, bool force)
-{
-
- // do not change actual stream volume if the stream is muted
- if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) {
- LOGV("checkAndSetVolume() stream %d muted count %d", stream, mOutputs.valueFor(output)->mMuteCount[stream]);
- return NO_ERROR;
- }
-
- // do not change in call volume if bluetooth is connected and vice versa
- if ((stream == AudioSystem::VOICE_CALL && mForceUse[AudioSystem::FOR_COMMUNICATION] == AudioSystem::FORCE_BT_SCO) ||
- (stream == AudioSystem::BLUETOOTH_SCO && mForceUse[AudioSystem::FOR_COMMUNICATION] != AudioSystem::FORCE_BT_SCO)) {
- LOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm",
- stream, mForceUse[AudioSystem::FOR_COMMUNICATION]);
- return INVALID_OPERATION;
- }
-
- float volume = computeVolume(stream, index, output, device);
- // We actually change the volume if:
- // - the float value returned by computeVolume() changed
- // - the force flag is set
- if (volume != mOutputs.valueFor(output)->mCurVolume[stream] ||
- force) {
- mOutputs.valueFor(output)->mCurVolume[stream] = volume;
- LOGV("setStreamVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs);
- if (stream == AudioSystem::VOICE_CALL ||
- stream == AudioSystem::DTMF ||
- stream == AudioSystem::BLUETOOTH_SCO) {
- // offset value to reflect actual hardware volume that never reaches 0
- // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
- volume = 0.01 + 0.99 * volume;
- // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is
- // enabled
- if (stream == AudioSystem::BLUETOOTH_SCO) {
- mpClientInterface->setStreamVolume(AudioSystem::VOICE_CALL, volume, output, delayMs);
- }
- }
-
- mpClientInterface->setStreamVolume((AudioSystem::stream_type)stream, volume, output, delayMs);
- }
-
- if (stream == AudioSystem::VOICE_CALL ||
- stream == AudioSystem::BLUETOOTH_SCO) {
- float voiceVolume;
- // Force voice volume to max for bluetooth SCO as volume is managed by the headset
- if (stream == AudioSystem::VOICE_CALL) {
- voiceVolume = (float)index/(float)mStreams[stream].mIndexMax;
- } else {
- voiceVolume = 1.0;
- }
-
- if (voiceVolume != mLastVoiceVolume && output == mHardwareOutput) {
- mpClientInterface->setVoiceVolume(voiceVolume, delayMs);
- mLastVoiceVolume = voiceVolume;
- }
- }
-
- return NO_ERROR;
-}
-
-void AudioPolicyManagerBase::applyStreamVolumes(audio_io_handle_t output, uint32_t device, int delayMs, bool force)
-{
- LOGV("applyStreamVolumes() for output %d and device %x", output, device);
-
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- checkAndSetVolume(stream, mStreams[stream].mIndexCur, output, device, delayMs, force);
- }
-}
-
-void AudioPolicyManagerBase::setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs)
-{
- LOGV("setStrategyMute() strategy %d, mute %d, output %d", strategy, on, output);
- for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
- if (getStrategy((AudioSystem::stream_type)stream) == strategy) {
- setStreamMute(stream, on, output, delayMs);
- }
- }
-}
-
-void AudioPolicyManagerBase::setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs)
-{
- StreamDescriptor &streamDesc = mStreams[stream];
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(output);
-
- LOGV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d", stream, on, output, outputDesc->mMuteCount[stream]);
-
- if (on) {
- if (outputDesc->mMuteCount[stream] == 0) {
- if (streamDesc.mCanBeMuted) {
- checkAndSetVolume(stream, 0, output, outputDesc->device(), delayMs);
- }
- }
- // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored
- outputDesc->mMuteCount[stream]++;
- } else {
- if (outputDesc->mMuteCount[stream] == 0) {
- LOGW("setStreamMute() unmuting non muted stream!");
- return;
- }
- if (--outputDesc->mMuteCount[stream] == 0) {
- checkAndSetVolume(stream, streamDesc.mIndexCur, output, outputDesc->device(), delayMs);
- }
- }
-}
-
-void AudioPolicyManagerBase::handleIncallSonification(int stream, bool starting, bool stateChange)
-{
- // if the stream pertains to sonification strategy and we are in call we must
- // mute the stream if it is low visibility. If it is high visibility, we must play a tone
- // in the device used for phone strategy and play the tone if the selected device does not
- // interfere with the device used for phone strategy
- // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
- // many times as there are active tracks on the output
-
- if (getStrategy((AudioSystem::stream_type)stream) == STRATEGY_SONIFICATION) {
- AudioOutputDescriptor *outputDesc = mOutputs.valueFor(mHardwareOutput);
- LOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
- stream, starting, outputDesc->mDevice, stateChange);
- if (outputDesc->mRefCount[stream]) {
- int muteCount = 1;
- if (stateChange) {
- muteCount = outputDesc->mRefCount[stream];
- }
- if (AudioSystem::isLowVisibility((AudioSystem::stream_type)stream)) {
- LOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mHardwareOutput);
- }
- } else {
- LOGV("handleIncallSonification() high visibility");
- if (outputDesc->device() & getDeviceForStrategy(STRATEGY_PHONE)) {
- LOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
- for (int i = 0; i < muteCount; i++) {
- setStreamMute(stream, starting, mHardwareOutput);
- }
- }
- if (starting) {
- mpClientInterface->startTone(ToneGenerator::TONE_SUP_CALL_WAITING, AudioSystem::VOICE_CALL);
- } else {
- mpClientInterface->stopTone();
- }
- }
- }
- }
-}
-
-bool AudioPolicyManagerBase::isInCall()
-{
- return isStateInCall(mPhoneState);
-}
-
-bool AudioPolicyManagerBase::isStateInCall(int state) {
- return ((state == AudioSystem::MODE_IN_CALL) ||
- (state == AudioSystem::MODE_IN_COMMUNICATION));
-}
-
-bool AudioPolicyManagerBase::needsDirectOuput(AudioSystem::stream_type stream,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::output_flags flags,
- uint32_t device)
-{
- return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
- (format != 0 && !AudioSystem::isLinearPCM(format)));
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
-{
- return MAX_EFFECTS_CPU_LOAD;
-}
-
-uint32_t AudioPolicyManagerBase::getMaxEffectsMemory()
-{
- return MAX_EFFECTS_MEMORY;
-}
-
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManagerBase::AudioOutputDescriptor::AudioOutputDescriptor()
- : mId(0), mSamplingRate(0), mFormat(0), mChannels(0), mLatency(0),
- mFlags((AudioSystem::output_flags)0), mDevice(0), mOutput1(0), mOutput2(0)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- mRefCount[i] = 0;
- mCurVolume[i] = -1.0;
- mMuteCount[i] = 0;
- mStopTime[i] = 0;
- }
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::device()
-{
- uint32_t device = 0;
- if (isDuplicated()) {
- device = mOutput1->mDevice | mOutput2->mDevice;
- } else {
- device = mDevice;
- }
- return device;
-}
-
-void AudioPolicyManagerBase::AudioOutputDescriptor::changeRefCount(AudioSystem::stream_type stream, int delta)
-{
- // forward usage count change to attached outputs
- if (isDuplicated()) {
- mOutput1->changeRefCount(stream, delta);
- mOutput2->changeRefCount(stream, delta);
- }
- if ((delta + (int)mRefCount[stream]) < 0) {
- LOGW("changeRefCount() invalid delta %d for stream %d, refCount %d", delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- LOGV("changeRefCount() delta %d, stream %d, refCount %d", delta, stream, mRefCount[stream]);
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
-{
- uint32_t refcount = 0;
- for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
- refcount += mRefCount[i];
- }
- return refcount;
-}
-
-uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::strategyRefCount(routing_strategy strategy)
-{
- uint32_t refCount = 0;
- for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
- if (getStrategy((AudioSystem::stream_type)i) == strategy) {
- refCount += mRefCount[i];
- }
- }
- return refCount;
-}
-
-status_t AudioPolicyManagerBase::AudioOutputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
- result.append(buffer);
- snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
- result.append(buffer);
- snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", device());
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
- result.append(buffer);
- for (int i = 0; i < AudioSystem::NUM_STREAM_TYPES; i++) {
- snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n", i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
- : mSamplingRate(0), mFormat(0), mChannels(0),
- mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0),
- mInputSource(0)
-{
-}
-
-status_t AudioPolicyManagerBase::AudioInputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannels);
- result.append(buffer);
- snprintf(buffer, SIZE, " Acoustics %08x\n", mAcoustics);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-void AudioPolicyManagerBase::StreamDescriptor::dump(char* buffer, size_t size)
-{
- snprintf(buffer, size, " %02d %02d %02d %d\n",
- mIndexMin,
- mIndexMax,
- mIndexCur,
- mCanBeMuted);
-}
-
-// --- EffectDescriptor class implementation
-
-status_t AudioPolicyManagerBase::EffectDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " Output: %d\n", mOutput);
- result.append(buffer);
- snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy);
- result.append(buffer);
- snprintf(buffer, SIZE, " Session: %d\n", mSession);
- result.append(buffer);
- snprintf(buffer, SIZE, " Name: %s\n", mDesc.name);
- result.append(buffer);
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-
-
-}; // namespace android