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873f21c820
replicant-frameworks_native/libs/audioflinger/AudioFlinger.cpp
Eric Laurent 873f21c820 Added Visualizer effect. 
The visualizer enables application to retrieve part of the currently playing audio for visualization purpose.
It is not an audio recording interface and only returns partial and low quality audio content as a waveform or
a frequency representation (FFT).

Removed temporary hack made in MediaPlayer for animated wall papers based on audio visualization (snoop() method.

This commit also includes a change in AudioEffect class:
 - the enable()/disable() methods have been replaced bya more standard setEnabled() method.
 - some fixes in javadoc

Change-Id: Id092a1340e9e38dae68646ade7be054e3a36980e
2010-07-07 11:00:28 -07:00

6055 lines
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/* //device/include/server/AudioFlinger/AudioFlinger.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.
*/
#define LOG_TAG "AudioFlinger"
//
#define LOG_NDEBUG 0
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <cutils/properties.h>
#include <media/AudioTrack.h>
#include <media/AudioRecord.h>
#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>
#include <hardware_legacy/AudioHardwareInterface.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#ifdef WITH_A2DP
#include "A2dpAudioInterface.h"
#endif
#ifdef LVMX
#include "lifevibes.h"
#endif
#include <media/EffectsFactoryApi.h>
#include <media/EffectVisualizerApi.h>
// ----------------------------------------------------------------------------
// the sim build doesn't have gettid
#ifndef HAVE_GETTID
# define gettid getpid
#endif
// ----------------------------------------------------------------------------
namespace android {
static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
static const char* kHardwareLockedString = "Hardware lock is taken\n";
//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const float MAX_GAIN = 4096.0f;
static const float MAX_GAIN_INT = 0x1000;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
static const int8_t kMaxTrackRetries = 50;
static const int8_t kMaxTrackStartupRetries = 50;
// allow less retry attempts on direct output thread.
// direct outputs can be a scarce resource in audio hardware and should
// be released as quickly as possible.
static const int8_t kMaxTrackRetriesDirect = 2;
static const int kDumpLockRetries = 50;
static const int kDumpLockSleep = 20000;
static const nsecs_t kWarningThrottle = seconds(5);
#define AUDIOFLINGER_SECURITY_ENABLED 1
// ----------------------------------------------------------------------------
static bool recordingAllowed() {
#ifndef HAVE_ANDROID_OS
return true;
#endif
#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
return ok;
#else
if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
return true;
#endif
}
static bool settingsAllowed() {
#ifndef HAVE_ANDROID_OS
return true;
#endif
#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
return ok;
#else
if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
return true;
#endif
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0)
{
mHardwareStatus = AUDIO_HW_IDLE;
mAudioHardware = AudioHardwareInterface::create();
mHardwareStatus = AUDIO_HW_INIT;
if (mAudioHardware->initCheck() == NO_ERROR) {
// open 16-bit output stream for s/w mixer
mMode = AudioSystem::MODE_NORMAL;
setMode(mMode);
setMasterVolume(1.0f);
setMasterMute(false);
} else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
#ifdef LVMX
LifeVibes::init();
mLifeVibesClientPid = -1;
#endif
}
AudioFlinger::~AudioFlinger()
{
while (!mRecordThreads.isEmpty()) {
// closeInput() will remove first entry from mRecordThreads
closeInput(mRecordThreads.keyAt(0));
}
while (!mPlaybackThreads.isEmpty()) {
// closeOutput() will remove first entry from mPlaybackThreads
closeOutput(mPlaybackThreads.keyAt(0));
}
if (mAudioHardware) {
delete mAudioHardware;
}
}
status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
wp<Client> wClient = mClients.valueAt(i);
if (wClient != 0) {
sp<Client> client = wClient.promote();
if (client != 0) {
snprintf(buffer, SIZE, " pid: %d\n", client->pid());
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
int hardwareStatus = mHardwareStatus;
snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Permission Denial: "
"can't dump AudioFlinger from pid=%d, uid=%d\n",
IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
static bool tryLock(Mutex& mutex)
{
bool locked = false;
for (int i = 0; i < kDumpLockRetries; ++i) {
if (mutex.tryLock() == NO_ERROR) {
locked = true;
break;
}
usleep(kDumpLockSleep);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
bool hardwareLocked = tryLock(mHardwareLock);
if (!hardwareLocked) {
String8 result(kHardwareLockedString);
write(fd, result.string(), result.size());
} else {
mHardwareLock.unlock();
}
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
String8 result(kDeadlockedString);
write(fd, result.string(), result.size());
}
dumpClients(fd, args);
dumpInternals(fd, args);
// dump playback threads
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->dump(fd, args);
}
// dump record threads
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->dump(fd, args);
}
if (mAudioHardware) {
mAudioHardware->dumpState(fd, args);
}
if (locked) mLock.unlock();
}
return NO_ERROR;
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
int lSessionId;
if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGE("unknown output thread");
lStatus = BAD_VALUE;
goto Exit;
}
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
// If no audio session id is provided, create one here
// TODO: enforce same stream type for all tracks in same audio session?
// TODO: prevent same audio session on different output threads
LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
if (sessionId != NULL && *sessionId != 0) {
lSessionId = *sessionId;
} else {
lSessionId = nextUniqueId();
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
LOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, lSessionId, &lStatus);
}
if (lStatus == NO_ERROR) {
trackHandle = new TrackHandle(track);
} else {
// remove local strong reference to Client before deleting the Track so that the Client
// destructor is called by the TrackBase destructor with mLock held
client.clear();
track.clear();
}
Exit:
if(status) {
*status = lStatus;
}
return trackHandle;
}
uint32_t AudioFlinger::sampleRate(int output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGW("sampleRate() unknown thread %d", output);
return 0;
}
return thread->sampleRate();
}
int AudioFlinger::channelCount(int output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGW("channelCount() unknown thread %d", output);
return 0;
}
return thread->channelCount();
}
int AudioFlinger::format(int output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGW("format() unknown thread %d", output);
return 0;
}
return thread->format();
}
size_t AudioFlinger::frameCount(int output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGW("frameCount() unknown thread %d", output);
return 0;
}
return thread->frameCount();
}
uint32_t AudioFlinger::latency(int output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGW("latency() unknown thread %d", output);
return 0;
}
return thread->latency();
}
status_t AudioFlinger::setMasterVolume(float value)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
// when hw supports master volume, don't scale in sw mixer
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
value = 1.0f;
}
mHardwareStatus = AUDIO_HW_IDLE;
mMasterVolume = value;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterVolume(value);
return NO_ERROR;
}
status_t AudioFlinger::setMode(int mode)
{
status_t ret;
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
LOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
{ // scope for the lock
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
ret = mAudioHardware->setMode(mode);
mHardwareStatus = AUDIO_HW_IDLE;
}
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMode(mode);
#ifdef LVMX
LifeVibes::setMode(mode);
#endif
}
return ret;
}
status_t AudioFlinger::setMicMute(bool state)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
status_t ret = mAudioHardware->setMicMute(state);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
bool state = AudioSystem::MODE_INVALID;
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mAudioHardware->getMicMute(&state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
mMasterMute = muted;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterMute(muted);
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
return mMasterVolume;
}
bool AudioFlinger::masterMute() const
{
return mMasterMute;
}
status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
PlaybackThread *thread = NULL;
if (output) {
thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
}
mStreamTypes[stream].volume = value;
if (thread == NULL) {
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
}
} else {
thread->setStreamVolume(stream, value);
}
return NO_ERROR;
}
status_t AudioFlinger::setStreamMute(int stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
return BAD_VALUE;
}
mStreamTypes[stream].mute = muted;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
return NO_ERROR;
}
float AudioFlinger::streamVolume(int stream, int output) const
{
if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return 0.0f;
}
AutoMutex lock(mLock);
float volume;
if (output) {
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return 0.0f;
}
volume = thread->streamVolume(stream);
} else {
volume = mStreamTypes[stream].volume;
}
return volume;
}
bool AudioFlinger::streamMute(int stream) const
{
if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) {
return true;
}
return mStreamTypes[stream].mute;
}
bool AudioFlinger::isStreamActive(int stream) const
{
Mutex::Autolock _l(mLock);
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->isStreamActive(stream)) {
return true;
}
}
return false;
}
status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
{
status_t result;
LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
#ifdef LVMX
AudioParameter param = AudioParameter(keyValuePairs);
LifeVibes::setParameters(ioHandle,keyValuePairs);
String8 key = String8(AudioParameter::keyRouting);
int device;
if (NO_ERROR != param.getInt(key, device)) {
device = -1;
}
key = String8(LifevibesTag);
String8 value;
int musicEnabled = -1;
if (NO_ERROR == param.get(key, value)) {
if (value == LifevibesEnable) {
mLifeVibesClientPid = IPCThreadState::self()->getCallingPid();
musicEnabled = 1;
} else if (value == LifevibesDisable) {
mLifeVibesClientPid = -1;
musicEnabled = 0;
}
}
#endif
// ioHandle == 0 means the parameters are global to the audio hardware interface
if (ioHandle == 0) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
result = mAudioHardware->setParameters(keyValuePairs);
#ifdef LVMX
if (musicEnabled != -1) {
LifeVibes::enableMusic((bool) musicEnabled);
}
#endif
mHardwareStatus = AUDIO_HW_IDLE;
return result;
}
// hold a strong ref on thread in case closeOutput() or closeInput() is called
// and the thread is exited once the lock is released
sp<ThreadBase> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(ioHandle);
if (thread == NULL) {
thread = checkRecordThread_l(ioHandle);
}
}
if (thread != NULL) {
result = thread->setParameters(keyValuePairs);
#ifdef LVMX
if ((NO_ERROR == result) && (device != -1)) {
LifeVibes::setDevice(LifeVibes::threadIdToAudioOutputType(thread->id()), device);
}
#endif
return result;
}
return BAD_VALUE;
}
String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
{
// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
if (ioHandle == 0) {
return mAudioHardware->getParameters(keys);
}
Mutex::Autolock _l(mLock);
PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
if (playbackThread != NULL) {
return playbackThread->getParameters(keys);
}
RecordThread *recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getParameters(keys);
}
return String8("");
}
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
{
return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
}
unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
{
if (ioHandle == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
RecordThread *recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getInputFramesLost();
}
return 0;
}
status_t AudioFlinger::setVoiceVolume(float value)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
status_t ret = mAudioHardware->setVoiceVolume(value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
{
status_t status;
Mutex::Autolock _l(mLock);
PlaybackThread *playbackThread = checkPlaybackThread_l(output);
if (playbackThread != NULL) {
return playbackThread->getRenderPosition(halFrames, dspFrames);
}
return BAD_VALUE;
}
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{
Mutex::Autolock _l(mLock);
int pid = IPCThreadState::self()->getCallingPid();
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
pid);
LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
mNotificationClients.add(pid, notificationClient);
sp<IBinder> binder = client->asBinder();
binder->linkToDeath(notificationClient);
// the config change is always sent from playback or record threads to avoid deadlock
// with AudioSystem::gLock
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
}
}
}
void AudioFlinger::removeNotificationClient(pid_t pid)
{
Mutex::Autolock _l(mLock);
int index = mNotificationClients.indexOfKey(pid);
if (index >= 0) {
sp <NotificationClient> client = mNotificationClients.valueFor(pid);
LOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
#ifdef LVMX
if (pid == mLifeVibesClientPid) {
LOGV("Disabling lifevibes");
LifeVibes::enableMusic(false);
mLifeVibesClientPid = -1;
}
#endif
mNotificationClients.removeItem(pid);
}
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
}
}
// removeClient_l() must be called with AudioFlinger::mLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// ----------------------------------------------------------------------------
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id)
: Thread(false),
mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false)
{
}
AudioFlinger::ThreadBase::~ThreadBase()
{
mParamCond.broadcast();
mNewParameters.clear();
}
void AudioFlinger::ThreadBase::exit()
{
// keep a strong ref on ourself so that we wont get
// destroyed in the middle of requestExitAndWait()
sp <ThreadBase> strongMe = this;
LOGV("ThreadBase::exit");
{
AutoMutex lock(&mLock);
mExiting = true;
requestExit();
mWaitWorkCV.signal();
}
requestExitAndWait();
}
uint32_t AudioFlinger::ThreadBase::sampleRate() const
{
return mSampleRate;
}
int AudioFlinger::ThreadBase::channelCount() const
{
return (int)mChannelCount;
}
int AudioFlinger::ThreadBase::format() const
{
return mFormat;
}
size_t AudioFlinger::ThreadBase::frameCount() const
{
return mFrameCount;
}
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
status_t status;
LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
Mutex::Autolock _l(mLock);
mNewParameters.add(keyValuePairs);
mWaitWorkCV.signal();
// wait condition with timeout in case the thread loop has exited
// before the request could be processed
if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) {
status = mParamStatus;
mWaitWorkCV.signal();
} else {
status = TIMED_OUT;
}
return status;
}
void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
{
Mutex::Autolock _l(mLock);
sendConfigEvent_l(event, param);
}
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
ConfigEvent *configEvent = new ConfigEvent();
configEvent->mEvent = event;
configEvent->mParam = param;
mConfigEvents.add(configEvent);
LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
}
void AudioFlinger::ThreadBase::processConfigEvents()
{
mLock.lock();
while(!mConfigEvents.isEmpty()) {
LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
ConfigEvent *configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
mAudioFlinger->mLock.lock();
audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
mAudioFlinger->mLock.unlock();
delete configEvent;
mLock.lock();
}
mLock.unlock();
}
status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
bool locked = tryLock(mLock);
if (!locked) {
snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
write(fd, buffer, strlen(buffer));
}
snprintf(buffer, SIZE, "standby: %d\n", mStandby);
result.append(buffer);
snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
result.append(buffer);
snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
result.append(buffer);
snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
result.append(buffer);
result.append(" Index Command");
for (size_t i = 0; i < mNewParameters.size(); ++i) {
snprintf(buffer, SIZE, "\n %02d ", i);
result.append(buffer);
result.append(mNewParameters[i]);
}
snprintf(buffer, SIZE, "\n\nPending config events: \n");
result.append(buffer);
snprintf(buffer, SIZE, " Index event param\n");
result.append(buffer);
for (size_t i = 0; i < mConfigEvents.size(); i++) {
snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
result.append(buffer);
}
result.append("\n");
write(fd, result.string(), result.size());
if (locked) {
mLock.unlock();
}
return NO_ERROR;
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: ThreadBase(audioFlinger, id),
mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
mDevice(device)
{
readOutputParameters();
mMasterVolume = mAudioFlinger->masterVolume();
mMasterMute = mAudioFlinger->masterMute();
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
}
}
AudioFlinger::PlaybackThread::~PlaybackThread()
{
delete [] mMixBuffer;
}
status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
dumpEffectChains(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
wp<Track> wTrack = mActiveTracks[i];
if (wTrack != 0) {
sp<Track> track = wTrack.promote();
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mEffectChains.size(); ++i) {
sp<EffectChain> chain = mEffectChains[i];
if (chain != 0) {
chain->dump(fd, args);
}
}
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
result.append(buffer);
snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
result.append(buffer);
snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
result.append(buffer);
snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
result.append(buffer);
snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
result.append(buffer);
write(fd, result.string(), result.size());
dumpBase(fd, args);
return NO_ERROR;
}
// Thread virtuals
status_t AudioFlinger::PlaybackThread::readyToRun()
{
if (mSampleRate == 0) {
LOGE("No working audio driver found.");
return NO_INIT;
}
LOGI("AudioFlinger's thread %p ready to run", this);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::onFirstRef()
{
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "Playback Thread %p", this);
run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
}
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
status_t *status)
{
sp<Track> track;
status_t lStatus;
if (mType == DIRECT) {
if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) {
LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
sampleRate, format, channelCount, mOutput);
lStatus = BAD_VALUE;
goto Exit;
}
} else {
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (sampleRate > mSampleRate*2) {
LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
}
if (mOutput == 0) {
LOGE("Audio driver not initialized.");
lStatus = NO_INIT;
goto Exit;
}
{ // scope for mLock
Mutex::Autolock _l(mLock);
track = new Track(this, client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, sessionId);
if (track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
mTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
}
}
lStatus = NO_ERROR;
Exit:
if(status) {
*status = lStatus;
}
return track;
}
uint32_t AudioFlinger::PlaybackThread::latency() const
{
if (mOutput) {
return mOutput->latency();
}
else {
return 0;
}
}
status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setMasterVolume(audioOutputType, value);
}
#endif
mMasterVolume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setMasterMute(audioOutputType, muted);
}
#endif
mMasterMute = muted;
return NO_ERROR;
}
float AudioFlinger::PlaybackThread::masterVolume() const
{
return mMasterVolume;
}
bool AudioFlinger::PlaybackThread::masterMute() const
{
return mMasterMute;
}
status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setStreamVolume(audioOutputType, stream, value);
}
#endif
mStreamTypes[stream].volume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setStreamMute(audioOutputType, stream, muted);
}
#endif
mStreamTypes[stream].mute = muted;
return NO_ERROR;
}
float AudioFlinger::PlaybackThread::streamVolume(int stream) const
{
return mStreamTypes[stream].volume;
}
bool AudioFlinger::PlaybackThread::streamMute(int stream) const
{
return mStreamTypes[stream].mute;
}
bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const
{
Mutex::Autolock _l(mLock);
size_t count = mActiveTracks.size();
for (size_t i = 0 ; i < count ; ++i) {
sp<Track> t = mActiveTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
if (t->type() == stream)
return true;
}
return false;
}
// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
status_t status = ALREADY_EXISTS;
// set retry count for buffer fill
track->mRetryCount = kMaxTrackStartupRetries;
if (mActiveTracks.indexOf(track) < 0) {
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
track->mFillingUpStatus = Track::FS_FILLING;
track->mResetDone = false;
mActiveTracks.add(track);
if (track->mainBuffer() != mMixBuffer) {
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
chain->startTrack();
}
}
status = NO_ERROR;
}
LOGV("mWaitWorkCV.broadcast");
mWaitWorkCV.broadcast();
return status;
}
// destroyTrack_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
{
track->mState = TrackBase::TERMINATED;
if (mActiveTracks.indexOf(track) < 0) {
mTracks.remove(track);
deleteTrackName_l(track->name());
}
}
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
return mOutput->getParameters(keys);
}
// destroyTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
switch (event) {
case AudioSystem::OUTPUT_OPENED:
case AudioSystem::OUTPUT_CONFIG_CHANGED:
desc.channels = mChannels;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = latency();
param2 = &desc;
break;
case AudioSystem::STREAM_CONFIG_CHANGED:
param2 = &param;
case AudioSystem::OUTPUT_CLOSED:
default:
break;
}
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
void AudioFlinger::PlaybackThread::readOutputParameters()
{
mSampleRate = mOutput->sampleRate();
mChannels = mOutput->channels();
mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
mFormat = mOutput->format();
mFrameSize = (uint16_t)mOutput->frameSize();
mFrameCount = mOutput->bufferSize() / mFrameSize;
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
if (mMixBuffer != NULL) delete[] mMixBuffer;
mMixBuffer = new int16_t[mFrameCount * 2];
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
//TODO handle effects reconfig
}
status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
{
if (halFrames == 0 || dspFrames == 0) {
return BAD_VALUE;
}
if (mOutput == 0) {
return INVALID_OPERATION;
}
*halFrames = mBytesWritten/mOutput->frameSize();
return mOutput->getRenderPosition(dspFrames);
}
bool AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
{
Mutex::Autolock _l(mLock);
if (getEffectChain_l(sessionId) != 0) {
return true;
}
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId()) {
return true;
}
}
return false;
}
sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId)
{
Mutex::Autolock _l(mLock);
return getEffectChain_l(sessionId);
}
sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId)
{
sp<EffectChain> chain;
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() == sessionId) {
chain = mEffectChains[i];
break;
}
}
return chain;
}
void AudioFlinger::PlaybackThread::setMode(uint32_t mode)
{
Mutex::Autolock _l(mLock);
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
mEffectChains[i]->setMode(mode);
}
}
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device),
mAudioMixer(0)
{
mType = PlaybackThread::MIXER;
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
LOGE("Invalid audio hardware channel count");
}
}
AudioFlinger::MixerThread::~MixerThread()
{
delete mAudioMixer;
}
bool AudioFlinger::MixerThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
uint32_t mixerStatus = MIXER_IDLE;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount * mFrameSize;
// FIXME: Relaxed timing because of a certain device that can't meet latency
// Should be reduced to 2x after the vendor fixes the driver issue
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
nsecs_t lastWarning = 0;
bool longStandbyExit = false;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
Vector< sp<EffectChain> > effectChains;
while (!exitPending())
{
processConfigEvents();
mixerStatus = MIXER_IDLE;
{ // scope for mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
mixBufferSize = mFrameCount * mFrameSize;
// FIXME: Relaxed timing because of a certain device that can't meet latency
// Should be reduced to 2x after the vendor fixes the driver issue
maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
}
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
mSuspended) {
if (!mStandby) {
LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
mOutput->standby();
mStandby = true;
mBytesWritten = 0;
}
if (!activeTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) break;
// wait until we have something to do...
LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
mWaitWorkCV.wait(mLock);
LOGV("MixerThread %p TID %d waking up\n", this, gettid());
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + kStandbyTimeInNsecs;
sleepTime = idleSleepTime;
continue;
}
}
mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
effectChains = mEffectChains;
lockEffectChains_l();
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
mAudioMixer->process();
sleepTime = 0;
standbyTime = systemTime() + kStandbyTimeInNsecs;
//TODO: delay standby when effects have a tail
} else {
// If no tracks are ready, sleep once for the duration of an output
// buffer size, then write 0s to the output
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_ENABLED) {
sleepTime = activeSleepTime;
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 ||
(mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
memset (mMixBuffer, 0, mixBufferSize);
sleepTime = 0;
LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
}
// TODO add standby time extension fct of effect tail
}
if (mSuspended) {
sleepTime = idleSleepTime;
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
// enable changes in effect chain
unlockEffectChains();
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize);
}
#endif
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
nsecs_t now = systemTime();
nsecs_t delta = now - mLastWriteTime;
if (delta > maxPeriod) {
mNumDelayedWrites++;
if ((now - lastWarning) > kWarningThrottle) {
LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
}
if (mStandby) {
longStandbyExit = true;
}
}
mStandby = false;
} else {
// enable changes in effect chain
unlockEffectChains();
usleep(sleepTime);
}
// finally let go of all our tracks, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
if (!mStandby) {
mOutput->standby();
}
LOGV("MixerThread %p exiting", this);
return false;
}
// prepareTracks_l() must be called with ThreadBase::mLock held
uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
{
uint32_t mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
size_t count = activeTracks.size();
size_t mixedTracks = 0;
size_t tracksWithEffect = 0;
float masterVolume = mMasterVolume;
bool masterMute = mMasterMute;
#ifdef LVMX
bool tracksConnectedChanged = false;
bool stateChanged = false;
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
{
int activeTypes = 0;
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
int iTracktype=track->type();
activeTypes |= 1<<track->type();
}
LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute);
}
#endif
// Delegate master volume control to effect in output mix effect chain if needed
sp<EffectChain> chain = getEffectChain_l(0);
if (chain != 0) {
uint32_t v = (uint32_t)(masterVolume * (1 << 24));
chain->setVolume(&v, &v);
masterVolume = (float)((v + (1 << 23)) >> 24);
chain.clear();
}
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
mAudioMixer->setActiveTrack(track->name());
if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
!track->isPaused() && !track->isTerminated())
{
//LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
mixedTracks++;
// track->mainBuffer() != mMixBuffer means there is an effect chain
// connected to the track
chain.clear();
if (track->mainBuffer() != mMixBuffer) {
chain = getEffectChain_l(track->sessionId());
// Delegate volume control to effect in track effect chain if needed
if (chain != 0) {
tracksWithEffect++;
} else {
LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
track->name(), track->sessionId());
}
}
int param = AudioMixer::VOLUME;
if (track->mFillingUpStatus == Track::FS_FILLED) {
// no ramp for the first volume setting
track->mFillingUpStatus = Track::FS_ACTIVE;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
}
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
param = AudioMixer::RAMP_VOLUME;
}
// compute volume for this track
int16_t left, right, aux;
if (track->isMuted() || masterMute || track->isPausing() ||
mStreamTypes[track->type()].mute) {
left = right = aux = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
// read original volumes with volume control
float typeVolume = mStreamTypes[track->type()].volume;
#ifdef LVMX
bool streamMute=false;
// read the volume from the LivesVibes audio engine.
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
{
LifeVibes::getStreamVolumes(audioOutputType, track->type(), &typeVolume, &streamMute);
if (streamMute) {
typeVolume = 0;
}
}
#endif
float v = masterVolume * typeVolume;
uint32_t vl = (uint32_t)(v * cblk->volume[0]) << 12;
uint32_t vr = (uint32_t)(v * cblk->volume[1]) << 12;
// Delegate volume control to effect in track effect chain if needed
if (chain != 0 && chain->setVolume(&vl, &vr)) {
// Do not ramp volume is volume is controlled by effect
param = AudioMixer::VOLUME;
}
// Convert volumes from 8.24 to 4.12 format
uint32_t v_clamped = (vl + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
left = int16_t(v_clamped);
v_clamped = (vr + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
right = int16_t(v_clamped);
v_clamped = (uint32_t)(v * cblk->sendLevel);
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
aux = int16_t(v_clamped);
}
#ifdef LVMX
if ( tracksConnectedChanged || stateChanged )
{
// only do the ramp when the volume is changed by the user / application
param = AudioMixer::VOLUME;
}
#endif
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(track);
mAudioMixer->enable(AudioMixer::MIXING);
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::CHANNEL_COUNT, (void *)track->channelCount());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(cblk->sampleRate));
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
// reset retry count
track->mRetryCount = kMaxTrackRetries;
mixerStatus = MIXER_TRACKS_READY;
} else {
//LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
if (track->isStopped()) {
track->reset();
}
if (track->isTerminated() || track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
tracksToRemove->add(track);
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
tracksToRemove->add(track);
} else if (mixerStatus != MIXER_TRACKS_READY) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
mAudioMixer->disable(AudioMixer::MIXING);
}
}
// remove all the tracks that need to be...
count = tracksToRemove->size();
if (UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove->itemAt(i);
mActiveTracks.remove(track);
if (track->mainBuffer() != mMixBuffer) {
chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
chain->stopTrack();
}
}
if (track->isTerminated()) {
mTracks.remove(track);
deleteTrackName_l(track->mName);
}
}
}
// mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to
// mix buffer and track effects will accumulate into it
if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
}
return mixerStatus;
}
void AudioFlinger::MixerThread::invalidateTracks(int streamType)
{
LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this, streamType, mTracks.size());
Mutex::Autolock _l(mLock);
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
if (t->type() == streamType) {
t->mCblk->lock.lock();
t->mCblk->flags |= CBLK_INVALID_ON;
t->mCblk->cv.signal();
t->mCblk->lock.unlock();
}
}
}
// getTrackName_l() must be called with ThreadBase::mLock held
int AudioFlinger::MixerThread::getTrackName_l()
{
return mAudioMixer->getTrackName();
}
// deleteTrackName_l() must be called with ThreadBase::mLock held
void AudioFlinger::MixerThread::deleteTrackName_l(int name)
{
LOGV("remove track (%d) and delete from mixer", name);
mAudioMixer->deleteTrackName(name);
}
// checkForNewParameters_l() must be called with ThreadBase::mLock held
bool AudioFlinger::MixerThread::checkForNewParameters_l()
{
bool reconfig = false;
while (!mNewParameters.isEmpty()) {
status_t status = NO_ERROR;
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
if (value != AudioSystem::PCM_16_BIT) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
if (value != AudioSystem::CHANNEL_OUT_STEREO) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
// forward device change to effects that have requested to be
// aware of attached audio device.
mDevice = (uint32_t)value;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevice(mDevice);
}
}
if (status == NO_ERROR) {
status = mOutput->setParameters(keyValuePair);
if (!mStandby && status == INVALID_OPERATION) {
mOutput->standby();
mStandby = true;
mBytesWritten = 0;
status = mOutput->setParameters(keyValuePair);
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
readOutputParameters();
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
int name = getTrackName_l();
if (name < 0) break;
mTracks[i]->mName = name;
// limit track sample rate to 2 x new output sample rate
if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
}
}
sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
}
}
mNewParameters.removeAt(0);
mParamStatus = status;
mParamCond.signal();
mWaitWorkCV.wait(mLock);
}
return reconfig;
}
status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
PlaybackThread::dumpInternals(fd, args);
snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
{
return (uint32_t)(mOutput->latency() * 1000) / 2;
}
uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
{
return (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000;
}
// ----------------------------------------------------------------------------
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device)
{
mType = PlaybackThread::DIRECT;
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
static inline int16_t clamp16(int32_t sample)
{
if ((sample>>15) ^ (sample>>31))
sample = 0x7FFF ^ (sample>>31);
return sample;
}
static inline
int32_t mul(int16_t in, int16_t v)
{
#if defined(__arm__) && !defined(__thumb__)
int32_t out;
asm( "smulbb %[out], %[in], %[v] \n"
: [out]"=r"(out)
: [in]"%r"(in), [v]"r"(v)
: );
return out;
#else
return in * int32_t(v);
#endif
}
void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
{
// Do not apply volume on compressed audio
if (!AudioSystem::isLinearPCM(mFormat)) {
return;
}
// convert to signed 16 bit before volume calculation
if (mFormat == AudioSystem::PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
uint8_t *src = (uint8_t *)mMixBuffer + count-1;
int16_t *dst = mMixBuffer + count-1;
while(count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
}
size_t frameCount = mFrameCount;
int16_t *out = mMixBuffer;
if (ramp) {
if (mChannelCount == 1) {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out++;
vl += vlInc;
} while (--frameCount);
} else {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
int32_t vrInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
int32_t vr = ((int32_t)mRightVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
out += 2;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
}
} else {
if (mChannelCount == 1) {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out++;
} while (--frameCount);
} else {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out[1] = clamp16(mul(out[1], rightVol) >> 12);
out += 2;
} while (--frameCount);
}
}
// convert back to unsigned 8 bit after volume calculation
if (mFormat == AudioSystem::PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
int16_t *src = mMixBuffer;
uint8_t *dst = (uint8_t *)mMixBuffer;
while(count--) {
*dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
}
}
mLeftVolShort = leftVol;
mRightVolShort = rightVol;
}
bool AudioFlinger::DirectOutputThread::threadLoop()
{
uint32_t mixerStatus = MIXER_IDLE;
sp<Track> trackToRemove;
sp<Track> activeTrack;
nsecs_t standbyTime = systemTime();
int8_t *curBuf;
size_t mixBufferSize = mFrameCount*mFrameSize;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
// use shorter standby delay as on normal output to release
// hardware resources as soon as possible
nsecs_t standbyDelay = microseconds(activeSleepTime*2);
while (!exitPending())
{
bool rampVolume;
uint16_t leftVol;
uint16_t rightVol;
Vector< sp<EffectChain> > effectChains;
processConfigEvents();
mixerStatus = MIXER_IDLE;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
mixBufferSize = mFrameCount*mFrameSize;
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
standbyDelay = microseconds(activeSleepTime*2);
}
// put audio hardware into standby after short delay
if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
mSuspended) {
// wait until we have something to do...
if (!mStandby) {
LOGV("Audio hardware entering standby, mixer %p\n", this);
mOutput->standby();
mStandby = true;
mBytesWritten = 0;
}
if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) break;
LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
mWaitWorkCV.wait(mLock);
LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + standbyDelay;
sleepTime = idleSleepTime;
continue;
}
}
effectChains = mEffectChains;
// find out which tracks need to be processed
if (mActiveTracks.size() != 0) {
sp<Track> t = mActiveTracks[0].promote();
if (t == 0) continue;
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
!track->isPaused() && !track->isTerminated())
{
//LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
mLeftVolFloat = mRightVolFloat = 0;
mLeftVolShort = mRightVolShort = 0;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
rampVolume = true;
}
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
rampVolume = true;
}
// compute volume for this track
float left, right;
if (track->isMuted() || mMasterMute || track->isPausing() ||
mStreamTypes[track->type()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
float typeVolume = mStreamTypes[track->type()].volume;
float v = mMasterVolume * typeVolume;
float v_clamped = v * cblk->volume[0];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = v_clamped/MAX_GAIN;
v_clamped = v * cblk->volume[1];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = v_clamped/MAX_GAIN;
}
if (left != mLeftVolFloat || right != mRightVolFloat) {
mLeftVolFloat = left;
mRightVolFloat = right;
// If audio HAL implements volume control,
// force software volume to nominal value
if (mOutput->setVolume(left, right) == NO_ERROR) {
left = 1.0f;
right = 1.0f;
}
// Convert volumes from float to 8.24
uint32_t vl = (uint32_t)(left * (1 << 24));
uint32_t vr = (uint32_t)(right * (1 << 24));
// Delegate volume control to effect in track effect chain if needed
// only one effect chain can be present on DirectOutputThread, so if
// there is one, the track is connected to it
if (!effectChains.isEmpty()) {
// Do not ramp volume is volume is controlled by effect
if(effectChains[0]->setVolume(&vl, &vr)) {
rampVolume = false;
}
}
// Convert volumes from 8.24 to 4.12 format
uint32_t v_clamped = (vl + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
leftVol = (uint16_t)v_clamped;
v_clamped = (vr + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
rightVol = (uint16_t)v_clamped;
} else {
leftVol = mLeftVolShort;
rightVol = mRightVolShort;
rampVolume = false;
}
// reset retry count
track->mRetryCount = kMaxTrackRetriesDirect;
activeTrack = t;
mixerStatus = MIXER_TRACKS_READY;
} else {
//LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
if (track->isStopped()) {
track->reset();
}
if (track->isTerminated() || track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
trackToRemove = track;
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
trackToRemove = track;
} else {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
}
}
// remove all the tracks that need to be...
if (UNLIKELY(trackToRemove != 0)) {
mActiveTracks.remove(trackToRemove);
if (!effectChains.isEmpty()) {
LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), trackToRemove->sessionId());
effectChains[0]->stopTrack();
}
if (trackToRemove->isTerminated()) {
mTracks.remove(trackToRemove);
deleteTrackName_l(trackToRemove->mName);
}
}
lockEffectChains_l();
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
AudioBufferProvider::Buffer buffer;
size_t frameCount = mFrameCount;
curBuf = (int8_t *)mMixBuffer;
// output audio to hardware
while (frameCount) {
buffer.frameCount = frameCount;
activeTrack->getNextBuffer(&buffer);
if (UNLIKELY(buffer.raw == 0)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
frameCount -= buffer.frameCount;
curBuf += buffer.frameCount * mFrameSize;
activeTrack->releaseBuffer(&buffer);
}
sleepTime = 0;
standbyTime = systemTime() + standbyDelay;
} else {
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_ENABLED) {
sleepTime = activeSleepTime;
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) {
memset (mMixBuffer, 0, mFrameCount * mFrameSize);
sleepTime = 0;
}
}
if (mSuspended) {
sleepTime = idleSleepTime;
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_READY) {
applyVolume(leftVol, rightVol, rampVolume);
}
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
unlockEffectChains();
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
mStandby = false;
} else {
unlockEffectChains();
usleep(sleepTime);
}
// finally let go of removed track, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
trackToRemove.clear();
activeTrack.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
if (!mStandby) {
mOutput->standby();
}
LOGV("DirectOutputThread %p exiting", this);
return false;
}
// getTrackName_l() must be called with ThreadBase::mLock held
int AudioFlinger::DirectOutputThread::getTrackName_l()
{
return 0;
}
// deleteTrackName_l() must be called with ThreadBase::mLock held
void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
{
}
// checkForNewParameters_l() must be called with ThreadBase::mLock held
bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
{
bool reconfig = false;
while (!mNewParameters.isEmpty()) {
status_t status = NO_ERROR;
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (status == NO_ERROR) {
status = mOutput->setParameters(keyValuePair);
if (!mStandby && status == INVALID_OPERATION) {
mOutput->standby();
mStandby = true;
mBytesWritten = 0;
status = mOutput->setParameters(keyValuePair);
}
if (status == NO_ERROR && reconfig) {
readOutputParameters();
sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
}
}
mNewParameters.removeAt(0);
mParamStatus = status;
mParamCond.signal();
mWaitWorkCV.wait(mLock);
}
return reconfig;
}
uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
{
uint32_t time;
if (AudioSystem::isLinearPCM(mFormat)) {
time = (uint32_t)(mOutput->latency() * 1000) / 2;
} else {
time = 10000;
}
return time;
}
uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
{
uint32_t time;
if (AudioSystem::isLinearPCM(mFormat)) {
time = (uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000;
} else {
time = 10000;
}
return time;
}
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
: MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
{
mType = PlaybackThread::DUPLICATING;
addOutputTrack(mainThread);
}
AudioFlinger::DuplicatingThread::~DuplicatingThread()
{
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
mOutputTracks.clear();
}
bool AudioFlinger::DuplicatingThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
uint32_t mixerStatus = MIXER_IDLE;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount*mFrameSize;
SortedVector< sp<OutputTrack> > outputTracks;
uint32_t writeFrames = 0;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
Vector< sp<EffectChain> > effectChains;
while (!exitPending())
{
processConfigEvents();
mixerStatus = MIXER_IDLE;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
mixBufferSize = mFrameCount*mFrameSize;
updateWaitTime();
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
}
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
outputTracks.add(mOutputTracks[i]);
}
// put audio hardware into standby after short delay
if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
mSuspended) {
if (!mStandby) {
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->stop();
}
mStandby = true;
mBytesWritten = 0;
}
if (!activeTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
outputTracks.clear();
if (exitPending()) break;
LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
mWaitWorkCV.wait(mLock);
LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + kStandbyTimeInNsecs;
sleepTime = idleSleepTime;
continue;
}
}
mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
effectChains = mEffectChains;
lockEffectChains_l();
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
if (outputsReady(outputTracks)) {
mAudioMixer->process();
} else {
memset(mMixBuffer, 0, mixBufferSize);
}
sleepTime = 0;
writeFrames = mFrameCount;
} else {
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_ENABLED) {
sleepTime = activeSleepTime;
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0) {
// flush remaining overflow buffers in output tracks
for (size_t i = 0; i < outputTracks.size(); i++) {
if (outputTracks[i]->isActive()) {
sleepTime = 0;
writeFrames = 0;
memset(mMixBuffer, 0, mixBufferSize);
break;
}
}
}
}
if (mSuspended) {
sleepTime = idleSleepTime;
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
// enable changes in effect chain
unlockEffectChains();
standbyTime = systemTime() + kStandbyTimeInNsecs;
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->write(mMixBuffer, writeFrames);
}
mStandby = false;
mBytesWritten += mixBufferSize;
} else {
// enable changes in effect chain
unlockEffectChains();
usleep(sleepTime);
}
// finally let go of all our tracks, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
outputTracks.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
return false;
}
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
this,
mSampleRate,
mFormat,
mChannelCount,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
mOutputTracks.add(outputTrack);
LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
updateWaitTime();
}
}
void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mOutputTracks.size(); i++) {
if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
mOutputTracks[i]->destroy();
mOutputTracks.removeAt(i);
updateWaitTime();
return;
}
}
LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
}
void AudioFlinger::DuplicatingThread::updateWaitTime()
{
mWaitTimeMs = UINT_MAX;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
if (strong != NULL) {
uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
if (waitTimeMs < mWaitTimeMs) {
mWaitTimeMs = waitTimeMs;
}
}
}
}
bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
{
for (size_t i = 0; i < outputTracks.size(); i++) {
sp <ThreadBase> thread = outputTracks[i]->thread().promote();
if (thread == 0) {
LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
return false;
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->standby() && !playbackThread->isSuspended()) {
LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
return false;
}
}
return true;
}
uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
{
return (mWaitTimeMs * 1000) / 2;
}
// ----------------------------------------------------------------------------
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(0),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK),
mSessionId(sessionId)
{
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
if (sharedBuffer == 0) {
size += bufferSize;
}
if (client != NULL) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
if (mCblk) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mCblk->channelCount = (uint8_t)channelCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flags = CBLK_UNDERRUN_ON;
} else {
mBuffer = sharedBuffer->pointer();
}
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
} else {
LOGE("not enough memory for AudioTrack size=%u", size);
client->heap()->dump("AudioTrack");
return;
}
} else {
mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
if (mCblk) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mCblk->channelCount = (uint8_t)channelCount;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flags = CBLK_UNDERRUN_ON;
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
}
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
if (mCblk) {
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
if (mClient == NULL) {
delete mCblk;
}
}
mCblkMemory.clear(); // and free the shared memory
if (mClient != NULL) {
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
}
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
}
bool AudioFlinger::ThreadBase::TrackBase::step() {
bool result;
audio_track_cblk_t* cblk = this->cblk();
result = cblk->stepServer(mFrameCount);
if (!result) {
LOGV("stepServer failed acquiring cblk mutex");
mFlags |= STEPSERVER_FAILED;
}
return result;
}
void AudioFlinger::ThreadBase::TrackBase::reset() {
audio_track_cblk_t* cblk = this->cblk();
cblk->user = 0;
cblk->server = 0;
cblk->userBase = 0;
cblk->serverBase = 0;
mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
LOGV("TrackBase::reset");
}
sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
{
return mCblkMemory;
}
int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
return (int)mCblk->channelCount;
}
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d, channelCount %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);
return 0;
}
return bufferStart;
}
// ----------------------------------------------------------------------------
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
const wp<ThreadBase>& thread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
: TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), mAuxEffectId(0)
{
if (mCblk != NULL) {
sp<ThreadBase> baseThread = thread.promote();
if (baseThread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
mName = playbackThread->getTrackName_l();
mMainBuffer = playbackThread->mixBuffer();
}
LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
if (mName < 0) {
LOGE("no more track names available");
}
mVolume[0] = 1.0f;
mVolume[1] = 1.0f;
mStreamType = streamType;
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
}
}
AudioFlinger::PlaybackThread::Track::~Track()
{
LOGV("PlaybackThread::Track destructor");
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
mState = TERMINATED;
}
}
void AudioFlinger::PlaybackThread::Track::destroy()
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
// desctructor is called. As the destructor needs to lock mLock,
// we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
// On the other hand, as long as Track::destroy() is only called by
// TrackHandle destructor, the TrackHandle still holds a strong ref on
// this Track with its member mTrack.
sp<Track> keep(this);
{ // scope for mLock
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
}
AudioSystem::releaseOutput(thread->id());
}
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->destroyTrack_l(this);
}
}
}
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
mFormat,
mCblk->channelCount,
mSessionId,
mFrameCount,
mState,
mMute,
mFillingUpStatus,
mCblk->sampleRate,
mCblk->volume[0],
mCblk->volume[1],
mCblk->server,
mCblk->user,
(int)mMainBuffer,
(int)mAuxBuffer);
}
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
LOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesReady = cblk->framesReady();
if (LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
if (framesReq > framesReady) {
framesReq = framesReady;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
return NOT_ENOUGH_DATA;
}
bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING) return true;
if (mCblk->framesReady() >= mCblk->frameCount ||
(mCblk->flags & CBLK_FORCEREADY_MSK)) {
mFillingUpStatus = FS_FILLED;
mCblk->flags &= ~CBLK_FORCEREADY_MSK;
return true;
}
return false;
}
status_t AudioFlinger::PlaybackThread::Track::start()
{
status_t status = NO_ERROR;
LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
int state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (mState == PAUSED) {
mState = TrackBase::RESUMING;
LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
} else {
mState = TrackBase::ACTIVE;
LOGV("? => ACTIVE (%d) on thread %p", mName, this);
}
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
status = AudioSystem::startOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
thread->mLock.lock();
}
if (status == NO_ERROR) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->addTrack_l(this);
} else {
mState = state;
}
} else {
status = BAD_VALUE;
}
return status;
}
void AudioFlinger::PlaybackThread::Track::stop()
{
LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
int state = mState;
if (mState > STOPPED) {
mState = STOPPED;
// If the track is not active (PAUSED and buffers full), flush buffers
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
thread->mLock.lock();
}
}
}
void AudioFlinger::PlaybackThread::Track::pause()
{
LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
if (mState == ACTIVE || mState == RESUMING) {
mState = PAUSING;
LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(), (AudioSystem::stream_type)mStreamType);
thread->mLock.lock();
}
}
}
}
void AudioFlinger::PlaybackThread::Track::flush()
{
LOGV("flush(%d)", mName);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
return;
}
// No point remaining in PAUSED state after a flush => go to
// STOPPED state
mState = STOPPED;
mCblk->lock.lock();
// NOTE: reset() will reset cblk->user and cblk->server with
// the risk that at the same time, the AudioMixer is trying to read
// data. In this case, getNextBuffer() would return a NULL pointer
// as audio buffer => the AudioMixer code MUST always test that pointer
// returned by getNextBuffer() is not NULL!
reset();
mCblk->lock.unlock();
}
}
void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
if (!mResetDone) {
TrackBase::reset();
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flags |= CBLK_UNDERRUN_ON;
mCblk->flags &= ~CBLK_FORCEREADY_MSK;
mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
}
void AudioFlinger::PlaybackThread::Track::mute(bool muted)
{
mMute = muted;
}
void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
{
mVolume[0] = left;
mVolume[1] = right;
}
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
{
status_t status = DEAD_OBJECT;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
status = playbackThread->attachAuxEffect(this, EffectId);
}
return status;
}
void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
{
mAuxEffectId = EffectId;
mAuxBuffer = buffer;
}
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
channelCount, frameCount, flags, 0, sessionId),
mOverflow(false)
{
if (mCblk != NULL) {
LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
if (format == AudioSystem::PCM_16_BIT) {
mCblk->frameSize = channelCount * sizeof(int16_t);
} else if (format == AudioSystem::PCM_8_BIT) {
mCblk->frameSize = channelCount * sizeof(int8_t);
} else {
mCblk->frameSize = sizeof(int8_t);
}
}
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
AudioSystem::releaseInput(thread->id());
}
}
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
LOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesAvail = cblk->framesAvailable_l();
if (LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
status_t AudioFlinger::RecordThread::RecordTrack::start()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->start(this);
} else {
return BAD_VALUE;
}
}
void AudioFlinger::RecordThread::RecordTrack::stop()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
recordThread->stop(this);
TrackBase::reset();
// Force overerrun condition to avoid false overrun callback until first data is
// read from buffer
mCblk->flags |= CBLK_UNDERRUN_ON;
}
}
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n",
(mClient == NULL) ? getpid() : mClient->pid(),
mFormat,
mCblk->channelCount,
mSessionId,
mFrameCount,
mState,
mCblk->sampleRate,
mCblk->server,
mCblk->user);
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount)
: Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);
} else {
LOGW("Error creating output track on thread %p", playbackThread);
}
}
AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
{
clearBufferQueue();
}
status_t AudioFlinger::PlaybackThread::OutputTrack::start()
{
status_t status = Track::start();
if (status != NO_ERROR) {
return status;
}
mActive = true;
mRetryCount = 127;
return status;
}
void AudioFlinger::PlaybackThread::OutputTrack::stop()
{
Track::stop();
clearBufferQueue();
mOutBuffer.frameCount = 0;
mActive = false;
}
bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
uint32_t channelCount = mCblk->channelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
start();
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
MixerThread *mixerThread = (MixerThread *)thread.get();
if (mCblk->frameCount > frames){
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
uint32_t startFrames = (mCblk->frameCount - frames);
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
pInBuffer->frameCount = startFrames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else {
LOGW ("OutputTrack::write() %p no more buffers in queue", this);
}
}
}
}
while (waitTimeLeftMs) {
// First write pending buffers, then new data
if (mBufferQueue.size()) {
pInBuffer = mBufferQueue.itemAt(0);
} else {
pInBuffer = &inBuffer;
}
if (pInBuffer->frameCount == 0) {
break;
}
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
outputBufferFull = true;
break;
}
uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
if (waitTimeLeftMs >= waitTimeMs) {
waitTimeLeftMs -= waitTimeMs;
} else {
waitTimeLeftMs = 0;
}
}
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
mCblk->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channelCount;
mOutBuffer.frameCount -= outFrames;
mOutBuffer.i16 += outFrames * channelCount;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
delete [] pInBuffer->mBuffer;
delete pInBuffer;
LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
} else {
break;
}
}
}
// If we could not write all frames, allocate a buffer and queue it for next time.
if (inBuffer.frameCount) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->i16 = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
} else {
LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
}
}
}
// Calling write() with a 0 length buffer, means that no more data will be written:
// If no more buffers are pending, fill output track buffer to make sure it is started
// by output mixer.
if (frames == 0 && mBufferQueue.size() == 0) {
if (mCblk->user < mCblk->frameCount) {
frames = mCblk->frameCount - mCblk->user;
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[frames * channelCount];
pInBuffer->frameCount = frames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else if (mActive) {
stop();
}
}
return outputBufferFull;
}
status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
int active;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = buffer->frameCount;
// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
buffer->frameCount = 0;
uint32_t framesAvail = cblk->framesAvailable();
if (framesAvail == 0) {
Mutex::Autolock _l(cblk->lock);
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
if (UNLIKELY(!active)) {
LOGV("Not active and NO_MORE_BUFFERS");
return AudioTrack::NO_MORE_BUFFERS;
}
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (result != NO_ERROR) {
return AudioTrack::NO_MORE_BUFFERS;
}
// read the server count again
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
}
// if (framesAvail < framesReq) {
// return AudioTrack::NO_MORE_BUFFERS;
// }
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (u + framesReq > bufferEnd) {
framesReq = bufferEnd - u;
}
buffer->frameCount = framesReq;
buffer->raw = (void *)cblk->buffer(u);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
Buffer *pBuffer;
for (size_t i = 0; i < size; i++) {
pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
delete pBuffer;
}
mBufferQueue.clear();
}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
mPid(pid)
{
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}
// Client destructor must be called with AudioFlinger::mLock held
AudioFlinger::Client::~Client()
{
mAudioFlinger->removeClient_l(mPid);
}
const sp<MemoryDealer>& AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid)
: mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
mClient.clear();
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
{
sp<NotificationClient> keep(this);
{
mAudioFlinger->removeNotificationClient(mPid);
}
}
// ----------------------------------------------------------------------------
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
}
AudioFlinger::TrackHandle::~TrackHandle() {
// just stop the track on deletion, associated resources
// will be freed from the main thread once all pending buffers have
// been played. Unless it's not in the active track list, in which
// case we free everything now...
mTrack->destroy();
}
status_t AudioFlinger::TrackHandle::start() {
return mTrack->start();
}
void AudioFlinger::TrackHandle::stop() {
mTrack->stop();
}
void AudioFlinger::TrackHandle::flush() {
mTrack->flush();
}
void AudioFlinger::TrackHandle::mute(bool e) {
mTrack->mute(e);
}
void AudioFlinger::TrackHandle::pause() {
mTrack->pause();
}
void AudioFlinger::TrackHandle::setVolume(float left, float right) {
mTrack->setVolume(left, right);
}
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
return mTrack->getCblk();
}
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
}
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioTrack::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
int input,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
int *sessionId,
status_t *status)
{
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
RecordThread *thread;
size_t inFrameCount;
int lSessionId;
// check calling permissions
if (!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// add client to list
{ // scope for mLock
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
if (thread == NULL) {
lStatus = BAD_VALUE;
goto Exit;
}
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
// If no audio session id is provided, create one here
if (sessionId != NULL && *sessionId != 0) {
lSessionId = *sessionId;
} else {
lSessionId = nextUniqueId();
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
format, channelCount, frameCount, flags, lSessionId);
}
if (recordTrack->getCblk() == NULL) {
// remove local strong reference to Client before deleting the RecordTrack so that the Client
// destructor is called by the TrackBase destructor with mLock held
client.clear();
recordTrack.clear();
lStatus = NO_MEMORY;
goto Exit;
}
// return to handle to client
recordHandle = new RecordHandle(recordTrack);
lStatus = NO_ERROR;
Exit:
if (status) {
*status = lStatus;
}
return recordHandle;
}
// ----------------------------------------------------------------------------
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
}
AudioFlinger::RecordHandle::~RecordHandle() {
stop();
}
status_t AudioFlinger::RecordHandle::start() {
LOGV("RecordHandle::start()");
return mRecordTrack->start();
}
void AudioFlinger::RecordHandle::stop() {
LOGV("RecordHandle::stop()");
mRecordTrack->stop();
}
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
return mRecordTrack->getCblk();
}
status_t AudioFlinger::RecordHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioRecord::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) :
ThreadBase(audioFlinger, id),
mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
{
mReqChannelCount = AudioSystem::popCount(channels);
mReqSampleRate = sampleRate;
readInputParameters();
}
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
if (mResampler != 0) {
delete mResampler;
delete[] mRsmpOutBuffer;
}
}
void AudioFlinger::RecordThread::onFirstRef()
{
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "Record Thread %p", this);
run(buffer, PRIORITY_URGENT_AUDIO);
}
bool AudioFlinger::RecordThread::threadLoop()
{
AudioBufferProvider::Buffer buffer;
sp<RecordTrack> activeTrack;
// start recording
while (!exitPending()) {
processConfigEvents();
{ // scope for mLock
Mutex::Autolock _l(mLock);
checkForNewParameters_l();
if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
if (!mStandby) {
mInput->standby();
mStandby = true;
}
if (exitPending()) break;
LOGV("RecordThread: loop stopping");
// go to sleep
mWaitWorkCV.wait(mLock);
LOGV("RecordThread: loop starting");
continue;
}
if (mActiveTrack != 0) {
if (mActiveTrack->mState == TrackBase::PAUSING) {
if (!mStandby) {
mInput->standby();
mStandby = true;
}
mActiveTrack.clear();
mStartStopCond.broadcast();
} else if (mActiveTrack->mState == TrackBase::RESUMING) {
if (mReqChannelCount != mActiveTrack->channelCount()) {
mActiveTrack.clear();
mStartStopCond.broadcast();
} else if (mBytesRead != 0) {
// record start succeeds only if first read from audio input
// succeeds
if (mBytesRead > 0) {
mActiveTrack->mState = TrackBase::ACTIVE;
} else {
mActiveTrack.clear();
}
mStartStopCond.broadcast();
}
mStandby = false;
}
}
}
if (mActiveTrack != 0) {
if (mActiveTrack->mState != TrackBase::ACTIVE &&
mActiveTrack->mState != TrackBase::RESUMING) {
usleep(5000);
continue;
}
buffer.frameCount = mFrameCount;
if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
if (mResampler == 0) {
// no resampling
while (framesOut) {
size_t framesIn = mFrameCount - mRsmpInIndex;
if (framesIn) {
int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
if (framesIn > framesOut)
framesIn = framesOut;
mRsmpInIndex += framesIn;
framesOut -= framesIn;
if ((int)mChannelCount == mReqChannelCount ||
mFormat != AudioSystem::PCM_16_BIT) {
memcpy(dst, src, framesIn * mFrameSize);
} else {
int16_t *src16 = (int16_t *)src;
int16_t *dst16 = (int16_t *)dst;
if (mChannelCount == 1) {
while (framesIn--) {
*dst16++ = *src16;
*dst16++ = *src16++;
}
} else {
while (framesIn--) {
*dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
src16 += 2;
}
}
}
}
if (framesOut && mFrameCount == mRsmpInIndex) {
if (framesOut == mFrameCount &&
((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
mBytesRead = mInput->read(buffer.raw, mInputBytes);
framesOut = 0;
} else {
mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
mRsmpInIndex = 0;
}
if (mBytesRead < 0) {
LOGE("Error reading audio input");
if (mActiveTrack->mState == TrackBase::ACTIVE) {
// Force input into standby so that it tries to
// recover at next read attempt
mInput->standby();
usleep(5000);
}
mRsmpInIndex = mFrameCount;
framesOut = 0;
buffer.frameCount = 0;
}
}
}
} else {
// resampling
memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
// alter output frame count as if we were expecting stereo samples
if (mChannelCount == 1 && mReqChannelCount == 1) {
framesOut >>= 1;
}
mResampler->resample(mRsmpOutBuffer, framesOut, this);
// ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
// are 32 bit aligned which should be always true.
if (mChannelCount == 2 && mReqChannelCount == 1) {
AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
// the resampler always outputs stereo samples: do post stereo to mono conversion
int16_t *src = (int16_t *)mRsmpOutBuffer;
int16_t *dst = buffer.i16;
while (framesOut--) {
*dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
src += 2;
}
} else {
AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
}
}
mActiveTrack->releaseBuffer(&buffer);
mActiveTrack->overflow();
}
// client isn't retrieving buffers fast enough
else {
if (!mActiveTrack->setOverflow())
LOGW("RecordThread: buffer overflow");
// Release the processor for a while before asking for a new buffer.
// This will give the application more chance to read from the buffer and
// clear the overflow.
usleep(5000);
}
}
}
if (!mStandby) {
mInput->standby();
}
mActiveTrack.clear();
mStartStopCond.broadcast();
LOGV("RecordThread %p exiting", this);
return false;
}
status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
{
LOGV("RecordThread::start");
sp <ThreadBase> strongMe = this;
status_t status = NO_ERROR;
{
AutoMutex lock(&mLock);
if (mActiveTrack != 0) {
if (recordTrack != mActiveTrack.get()) {
status = -EBUSY;
} else if (mActiveTrack->mState == TrackBase::PAUSING) {
mActiveTrack->mState = TrackBase::ACTIVE;
}
return status;
}
recordTrack->mState = TrackBase::IDLE;
mActiveTrack = recordTrack;
mLock.unlock();
status_t status = AudioSystem::startInput(mId);
mLock.lock();
if (status != NO_ERROR) {
mActiveTrack.clear();
return status;
}
mActiveTrack->mState = TrackBase::RESUMING;
mRsmpInIndex = mFrameCount;
mBytesRead = 0;
// signal thread to start
LOGV("Signal record thread");
mWaitWorkCV.signal();
// do not wait for mStartStopCond if exiting
if (mExiting) {
mActiveTrack.clear();
status = INVALID_OPERATION;
goto startError;
}
mStartStopCond.wait(mLock);
if (mActiveTrack == 0) {
LOGV("Record failed to start");
status = BAD_VALUE;
goto startError;
}
LOGV("Record started OK");
return status;
}
startError:
AudioSystem::stopInput(mId);
return status;
}
void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
LOGV("RecordThread::stop");
sp <ThreadBase> strongMe = this;
{
AutoMutex lock(&mLock);
if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
mActiveTrack->mState = TrackBase::PAUSING;
// do not wait for mStartStopCond if exiting
if (mExiting) {
return;
}
mStartStopCond.wait(mLock);
// if we have been restarted, recordTrack == mActiveTrack.get() here
if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
mLock.unlock();
AudioSystem::stopInput(mId);
mLock.lock();
LOGV("Record stopped OK");
}
}
}
}
status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
pid_t pid = 0;
snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
result.append(buffer);
if (mActiveTrack != 0) {
result.append("Active Track:\n");
result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n");
mActiveTrack->dump(buffer, SIZE);
result.append(buffer);
snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
result.append(buffer);
snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
result.append(buffer);
snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
result.append(buffer);
} else {
result.append("No record client\n");
}
write(fd, result.string(), result.size());
dumpBase(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
size_t framesReq = buffer->frameCount;
size_t framesReady = mFrameCount - mRsmpInIndex;
int channelCount;
if (framesReady == 0) {
mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
if (mBytesRead < 0) {
LOGE("RecordThread::getNextBuffer() Error reading audio input");
if (mActiveTrack->mState == TrackBase::ACTIVE) {
// Force input into standby so that it tries to
// recover at next read attempt
mInput->standby();
usleep(5000);
}
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
mRsmpInIndex = 0;
framesReady = mFrameCount;
}
if (framesReq > framesReady) {
framesReq = framesReady;
}
if (mChannelCount == 1 && mReqChannelCount == 2) {
channelCount = 1;
} else {
channelCount = 2;
}
buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
buffer->frameCount = framesReq;
return NO_ERROR;
}
void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
mRsmpInIndex += buffer->frameCount;
buffer->frameCount = 0;
}
bool AudioFlinger::RecordThread::checkForNewParameters_l()
{
bool reconfig = false;
while (!mNewParameters.isEmpty()) {
status_t status = NO_ERROR;
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
int reqFormat = mFormat;
int reqSamplingRate = mReqSampleRate;
int reqChannelCount = mReqChannelCount;
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
reqSamplingRate = value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
reqFormat = value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
reqChannelCount = AudioSystem::popCount(value);
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// if frame count is changed after track creation
if (mActiveTrack != 0) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (status == NO_ERROR) {
status = mInput->setParameters(keyValuePair);
if (status == INVALID_OPERATION) {
mInput->standby();
status = mInput->setParameters(keyValuePair);
}
if (reconfig) {
if (status == BAD_VALUE &&
reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
(AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
readInputParameters();
sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
}
}
}
mNewParameters.removeAt(0);
mParamStatus = status;
mParamCond.signal();
mWaitWorkCV.wait(mLock);
}
return reconfig;
}
String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
{
return mInput->getParameters(keys);
}
void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
switch (event) {
case AudioSystem::INPUT_OPENED:
case AudioSystem::INPUT_CONFIG_CHANGED:
desc.channels = mChannels;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = 0;
param2 = &desc;
break;
case AudioSystem::INPUT_CLOSED:
default:
break;
}
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
void AudioFlinger::RecordThread::readInputParameters()
{
if (mRsmpInBuffer) delete mRsmpInBuffer;
if (mRsmpOutBuffer) delete mRsmpOutBuffer;
if (mResampler) delete mResampler;
mResampler = 0;
mSampleRate = mInput->sampleRate();
mChannels = mInput->channels();
mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
mFormat = mInput->format();
mFrameSize = (uint16_t)mInput->frameSize();
mInputBytes = mInput->bufferSize();
mFrameCount = mInputBytes / mFrameSize;
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
{
int channelCount;
// optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
// stereo to mono post process as the resampler always outputs stereo.
if (mChannelCount == 1 && mReqChannelCount == 2) {
channelCount = 1;
} else {
channelCount = 2;
}
mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
mResampler->setSampleRate(mSampleRate);
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
mRsmpOutBuffer = new int32_t[mFrameCount * 2];
// optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
if (mChannelCount == 1 && mReqChannelCount == 1) {
mFrameCount >>= 1;
}
}
mRsmpInIndex = mFrameCount;
}
unsigned int AudioFlinger::RecordThread::getInputFramesLost()
{
return mInput->getInputFramesLost();
}
// ----------------------------------------------------------------------------
int AudioFlinger::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags)
{
status_t status;
PlaybackThread *thread = NULL;
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
uint32_t format = pFormat ? *pFormat : 0;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
pDevices ? *pDevices : 0,
samplingRate,
format,
channels,
flags);
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
(int *)&format,
&channels,
&samplingRate,
&status);
LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
output,
samplingRate,
format,
channels,
status);
mHardwareStatus = AUDIO_HW_IDLE;
if (output != 0) {
int id = nextUniqueId();
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
(format != AudioSystem::PCM_16_BIT) ||
(channels != AudioSystem::CHANNEL_OUT_STEREO)) {
thread = new DirectOutputThread(this, output, id, *pDevices);
LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
} else {
thread = new MixerThread(this, output, id, *pDevices);
LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
#ifdef LVMX
unsigned bitsPerSample =
(format == AudioSystem::PCM_16_BIT) ? 16 :
((format == AudioSystem::PCM_8_BIT) ? 8 : 0);
unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1;
int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id());
LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount);
LifeVibes::setDevice(audioOutputType, *pDevices);
#endif
}
mPlaybackThreads.add(id, thread);
if (pSamplingRate) *pSamplingRate = samplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = channels;
if (pLatencyMs) *pLatencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return id;
}
return 0;
}
int AudioFlinger::openDuplicateOutput(int output1, int output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
MixerThread *thread2 = checkMixerThread_l(output2);
if (thread1 == NULL || thread2 == NULL) {
LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
return 0;
}
int id = nextUniqueId();
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return id;
}
status_t AudioFlinger::closeOutput(int output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
sp <PlaybackThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
LOGV("closeOutput() %d", output);
if (thread->type() == PlaybackThread::MIXER) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) {
DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
dupThread->removeOutputTrack((MixerThread *)thread.get());
}
}
}
void *param2 = 0;
audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
mPlaybackThreads.removeItem(output);
}
thread->exit();
if (thread->type() != PlaybackThread::DUPLICATING) {
mAudioHardware->closeOutputStream(thread->getOutput());
}
return NO_ERROR;
}
status_t AudioFlinger::suspendOutput(int output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
LOGV("suspendOutput() %d", output);
thread->suspend();
return NO_ERROR;
}
status_t AudioFlinger::restoreOutput(int output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
LOGV("restoreOutput() %d", output);
thread->restore();
return NO_ERROR;
}
int AudioFlinger::openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t acoustics)
{
status_t status;
RecordThread *thread = NULL;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
uint32_t format = pFormat ? *pFormat : 0;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t reqSamplingRate = samplingRate;
uint32_t reqFormat = format;
uint32_t reqChannels = channels;
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
(int *)&format,
&channels,
&samplingRate,
&status,
(AudioSystem::audio_in_acoustics)acoustics);
LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
input,
samplingRate,
format,
channels,
acoustics,
status);
// If the input could not be opened with the requested parameters and we can handle the conversion internally,
// try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
// or stereo to mono conversions on 16 bit PCM inputs.
if (input == 0 && status == BAD_VALUE &&
reqFormat == format && format == AudioSystem::PCM_16_BIT &&
(samplingRate <= 2 * reqSamplingRate) &&
(AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
LOGV("openInput() reopening with proposed sampling rate and channels");
input = mAudioHardware->openInputStream(*pDevices,
(int *)&format,
&channels,
&samplingRate,
&status,
(AudioSystem::audio_in_acoustics)acoustics);
}
if (input != 0) {
int id = nextUniqueId();
// Start record thread
thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id);
mRecordThreads.add(id, thread);
LOGV("openInput() created record thread: ID %d thread %p", id, thread);
if (pSamplingRate) *pSamplingRate = reqSamplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = reqChannels;
input->standby();
// notify client processes of the new input creation
thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
return id;
}
return 0;
}
status_t AudioFlinger::closeInput(int input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
sp <RecordThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
if (thread == NULL) {
return BAD_VALUE;
}
LOGV("closeInput() %d", input);
void *param2 = 0;
audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
mRecordThreads.removeItem(input);
}
thread->exit();
mAudioHardware->closeInputStream(thread->getInput());
return NO_ERROR;
}
status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
{
Mutex::Autolock _l(mLock);
MixerThread *dstThread = checkMixerThread_l(output);
if (dstThread == NULL) {
LOGW("setStreamOutput() bad output id %d", output);
return BAD_VALUE;
}
LOGV("setStreamOutput() stream %d to output %d", stream, output);
audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if (thread != dstThread &&
thread->type() != PlaybackThread::DIRECT) {
MixerThread *srcThread = (MixerThread *)thread;
srcThread->invalidateTracks(stream);
}
}
return NO_ERROR;
}
int AudioFlinger::newAudioSessionId()
{
return nextUniqueId();
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
{
PlaybackThread *thread = NULL;
if (mPlaybackThreads.indexOfKey(output) >= 0) {
thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
}
return thread;
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread != NULL) {
if (thread->type() == PlaybackThread::DIRECT) {
thread = NULL;
}
}
return (MixerThread *)thread;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
{
RecordThread *thread = NULL;
if (mRecordThreads.indexOfKey(input) >= 0) {
thread = (RecordThread *)mRecordThreads.valueFor(input).get();
}
return thread;
}
int AudioFlinger::nextUniqueId()
{
return android_atomic_inc(&mNextUniqueId);
}
// ----------------------------------------------------------------------------
// Effect management
// ----------------------------------------------------------------------------
status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle)
{
Mutex::Autolock _l(mLock);
return EffectLoadLibrary(libPath, handle);
}
status_t AudioFlinger::unloadEffectLibrary(int handle)
{
Mutex::Autolock _l(mLock);
return EffectUnloadLibrary(handle);
}
status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
{
Mutex::Autolock _l(mLock);
return EffectQueryNumberEffects(numEffects);
}
status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
{
Mutex::Autolock _l(mLock);
return EffectQueryEffect(index, descriptor);
}
status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
{
Mutex::Autolock _l(mLock);
return EffectGetDescriptor(pUuid, descriptor);
}
// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp
static const effect_uuid_t VISUALIZATION_UUID_ =
{0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
sp<IEffect> AudioFlinger::createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
int output,
int sessionId,
status_t *status,
int *id,
int *enabled)
{
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
effect_interface_t itfe;
effect_descriptor_t desc;
sp<Client> client;
wp<Client> wclient;
LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", pid, effectClient.get(), priority, sessionId, output);
if (pDesc == NULL) {
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
// check recording permission for visualizer
if (memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 ||
memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) {
if (!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
}
if (!EffectIsNullUuid(&pDesc->uuid)) {
// if uuid is specified, request effect descriptor
lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
if (lStatus < 0) {
LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
goto Exit;
}
} else {
// if uuid is not specified, look for an available implementation
// of the required type in effect factory
if (EffectIsNullUuid(&pDesc->type)) {
LOGW("createEffect() no effect type");
lStatus = BAD_VALUE;
goto Exit;
}
uint32_t numEffects = 0;
effect_descriptor_t d;
bool found = false;
lStatus = EffectQueryNumberEffects(&numEffects);
if (lStatus < 0) {
LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
goto Exit;
}
for (uint32_t i = 0; i < numEffects; i++) {
lStatus = EffectQueryEffect(i, &desc);
if (lStatus < 0) {
LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
continue;
}
if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
// If matching type found save effect descriptor. If the session is
// 0 and the effect is not auxiliary, continue enumeration in case
// an auxiliary version of this effect type is available
found = true;
memcpy(&d, &desc, sizeof(effect_descriptor_t));
if (sessionId != 0 ||
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
break;
}
}
}
if (!found) {
lStatus = BAD_VALUE;
LOGW("createEffect() effect not found");
goto Exit;
}
// For same effect type, chose auxiliary version over insert version if
// connect to output mix (Compliance to OpenSL ES)
if (sessionId == 0 &&
(d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
memcpy(&desc, &d, sizeof(effect_descriptor_t));
}
}
// Do not allow auxiliary effects on a session different from 0 (output mix)
if (sessionId != 0 &&
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
lStatus = INVALID_OPERATION;
goto Exit;
}
// Session -1 is reserved for output stage effects that can only be created
// by audio policy manager (running in same process)
if (sessionId == -1 && getpid() != IPCThreadState::self()->getCallingPid()) {
lStatus = INVALID_OPERATION;
goto Exit;
}
// return effect descriptor
memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
// If output is not specified try to find a matching audio session ID in one of the
// output threads.
// TODO: allow attachment of effect to inputs
if (output == 0) {
if (sessionId <= 0) {
// default to first output
// TODO: define criteria to choose output when not specified. Or
// receive output from audio policy manager
if (mPlaybackThreads.size() != 0) {
output = mPlaybackThreads.keyAt(0);
}
} else {
// look for the thread where the specified audio session is present
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId)) {
output = mPlaybackThreads.keyAt(i);
break;
}
}
}
}
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGE("unknown output thread");
lStatus = BAD_VALUE;
goto Exit;
}
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
// create effect on selected output trhead
handle = thread->createEffect_l(client, effectClient, priority, sessionId, &desc, enabled, &lStatus);
if (handle != 0 && id != NULL) {
*id = handle->id();
}
}
Exit:
if(status) {
*status = lStatus;
}
return handle;
}
status_t AudioFlinger::registerEffectResource_l(effect_descriptor_t *desc) {
if (mTotalEffectsCpuLoad + desc->cpuLoad > MAX_EFFECTS_CPU_LOAD) {
LOGW("registerEffectResource() CPU Load limit exceeded for Fx %s, CPU %f MIPS",
desc->name, (float)desc->cpuLoad/10);
return INVALID_OPERATION;
}
if (mTotalEffectsMemory + desc->memoryUsage > MAX_EFFECTS_MEMORY) {
LOGW("registerEffectResource() memory limit exceeded for Fx %s, Memory %d KB",
desc->name, desc->memoryUsage);
return INVALID_OPERATION;
}
mTotalEffectsCpuLoad += desc->cpuLoad;
mTotalEffectsMemory += desc->memoryUsage;
LOGV("registerEffectResource_l() effect %s, CPU %d, memory %d",
desc->name, desc->cpuLoad, desc->memoryUsage);
LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
return NO_ERROR;
}
void AudioFlinger::unregisterEffectResource_l(effect_descriptor_t *desc) {
mTotalEffectsCpuLoad -= desc->cpuLoad;
mTotalEffectsMemory -= desc->memoryUsage;
LOGV("unregisterEffectResource_l() effect %s, CPU %d, memory %d",
desc->name, desc->cpuLoad, desc->memoryUsage);
LOGV(" total CPU %d, total memory %d", mTotalEffectsCpuLoad, mTotalEffectsMemory);
}
// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l(
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority,
int sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status
)
{
sp<EffectModule> effect;
sp<EffectHandle> handle;
status_t lStatus;
sp<Track> track;
sp<EffectChain> chain;
bool effectCreated = false;
bool effectRegistered = false;
if (mOutput == 0) {
LOGW("createEffect_l() Audio driver not initialized.");
lStatus = NO_INIT;
goto Exit;
}
// Do not allow auxiliary effect on session other than 0
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY &&
sessionId != 0) {
LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
// Do not allow effects with session ID 0 on direct output or duplicating threads
// TODO: add rule for hw accelerated effects on direct outputs with non PCM format
if (sessionId == 0 && mType != MIXER) {
LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", desc->name, sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
{ // scope for mLock
Mutex::Autolock _l(mLock);
// check for existing effect chain with the requested audio session
chain = getEffectChain_l(sessionId);
if (chain == 0) {
// create a new chain for this session
LOGV("createEffect_l() new effect chain for session %d", sessionId);
chain = new EffectChain(this, sessionId);
addEffectChain_l(chain);
} else {
effect = chain->getEffectFromDesc(desc);
}
LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
if (effect == 0) {
// Check CPU and memory usage
lStatus = mAudioFlinger->registerEffectResource_l(desc);
if (lStatus != NO_ERROR) {
goto Exit;
}
effectRegistered = true;
// create a new effect module if none present in the chain
effect = new EffectModule(this, chain, desc, mAudioFlinger->nextUniqueId(), sessionId);
lStatus = effect->status();
if (lStatus != NO_ERROR) {
goto Exit;
}
lStatus = chain->addEffect(effect);
if (lStatus != NO_ERROR) {
goto Exit;
}
effectCreated = true;
effect->setDevice(mDevice);
effect->setMode(mAudioFlinger->getMode());
}
// create effect handle and connect it to effect module
handle = new EffectHandle(effect, client, effectClient, priority);
lStatus = effect->addHandle(handle);
if (enabled) {
*enabled = (int)effect->isEnabled();
}
}
Exit:
if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
if (effectCreated) {
if (chain->removeEffect(effect) == 0) {
removeEffectChain_l(chain);
}
}
if (effectRegistered) {
mAudioFlinger->unregisterEffectResource_l(desc);
}
handle.clear();
}
if(status) {
*status = lStatus;
}
return handle;
}
void AudioFlinger::PlaybackThread::disconnectEffect(const sp< EffectModule>& effect,
const wp<EffectHandle>& handle) {
effect_descriptor_t desc = effect->desc();
Mutex::Autolock _l(mLock);
// delete the effect module if removing last handle on it
if (effect->removeHandle(handle) == 0) {
if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
detachAuxEffect_l(effect->id());
}
sp<EffectChain> chain = effect->chain().promote();
if (chain != 0) {
// remove effect chain if remove last effect
if (chain->removeEffect(effect) == 0) {
removeEffectChain_l(chain);
}
}
mLock.unlock();
mAudioFlinger->mLock.lock();
mAudioFlinger->unregisterEffectResource_l(&desc);
mAudioFlinger->mLock.unlock();
}
}
status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
int16_t *buffer = mMixBuffer;
bool ownsBuffer = false;
LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
if (session > 0) {
// Only one effect chain can be present in direct output thread and it uses
// the mix buffer as input
if (mType != DIRECT) {
size_t numSamples = mFrameCount * mChannelCount;
buffer = new int16_t[numSamples];
memset(buffer, 0, numSamples * sizeof(int16_t));
LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
ownsBuffer = true;
}
// Attach all tracks with same session ID to this chain.
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
track->setMainBuffer(buffer);
}
}
// indicate all active tracks in the chain
for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
sp<Track> track = mActiveTracks[i].promote();
if (track == 0) continue;
if (session == track->sessionId()) {
LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
chain->startTrack();
}
}
}
chain->setInBuffer(buffer, ownsBuffer);
chain->setOutBuffer(mMixBuffer);
// Effect chain for session -1 is inserted at end of effect chains list
// in order to be processed last as it contains output stage effects
// Effect chain for session 0 is inserted before session -1 to be processed
// after track specific effects and before output stage
// Effect chain for session other than 0 is inserted at beginning of effect
// chains list to be processed before output mix effects. Relative order between
// sessions other than 0 is not important
size_t size = mEffectChains.size();
size_t i = 0;
for (i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() < session) break;
}
mEffectChains.insertAt(chain, i);
return NO_ERROR;
}
size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
for (size_t i = 0; i < mEffectChains.size(); i++) {
if (chain == mEffectChains[i]) {
mEffectChains.removeAt(i);
// detach all tracks with same session ID from this chain
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
track->setMainBuffer(mMixBuffer);
}
}
}
}
return mEffectChains.size();
}
void AudioFlinger::PlaybackThread::lockEffectChains_l()
{
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->lock();
}
}
void AudioFlinger::PlaybackThread::unlockEffectChains()
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->unlock();
}
}
sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId)
{
sp<EffectModule> effect;
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
effect = chain->getEffectFromId(effectId);
}
return effect;
}
status_t AudioFlinger::PlaybackThread::attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
{
Mutex::Autolock _l(mLock);
return attachAuxEffect_l(track, EffectId);
}
status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
{
status_t status = NO_ERROR;
if (EffectId == 0) {
track->setAuxBuffer(0, NULL);
} else {
// Auxiliary effects are always in audio session 0
sp<EffectModule> effect = getEffect_l(0, EffectId);
if (effect != 0) {
if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
} else {
status = INVALID_OPERATION;
}
} else {
status = BAD_VALUE;
}
}
return status;
}
void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
{
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track->auxEffectId() == effectId) {
attachAuxEffect_l(track, 0);
}
}
}
// ----------------------------------------------------------------------------
// EffectModule implementation
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectModule"
AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId)
: mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
mStatus(NO_INIT), mState(IDLE)
{
LOGV("Constructor %p", this);
int lStatus;
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return;
}
PlaybackThread *p = (PlaybackThread *)thread.get();
memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
// create effect engine from effect factory
mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface);
if (mStatus != NO_ERROR) {
return;
}
lStatus = init();
if (lStatus < 0) {
mStatus = lStatus;
goto Error;
}
LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
return;
Error:
EffectRelease(mEffectInterface);
mEffectInterface = NULL;
LOGV("Constructor Error %d", mStatus);
}
AudioFlinger::EffectModule::~EffectModule()
{
LOGV("Destructor %p", this);
if (mEffectInterface != NULL) {
// release effect engine
EffectRelease(mEffectInterface);
}
}
status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
{
status_t status;
Mutex::Autolock _l(mLock);
// First handle in mHandles has highest priority and controls the effect module
int priority = handle->priority();
size_t size = mHandles.size();
sp<EffectHandle> h;
size_t i;
for (i = 0; i < size; i++) {
h = mHandles[i].promote();
if (h == 0) continue;
if (h->priority() <= priority) break;
}
// if inserted in first place, move effect control from previous owner to this handle
if (i == 0) {
if (h != 0) {
h->setControl(false, true);
}
handle->setControl(true, false);
status = NO_ERROR;
} else {
status = ALREADY_EXISTS;
}
mHandles.insertAt(handle, i);
return status;
}
size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
{
Mutex::Autolock _l(mLock);
size_t size = mHandles.size();
size_t i;
for (i = 0; i < size; i++) {
if (mHandles[i] == handle) break;
}
if (i == size) {
return size;
}
mHandles.removeAt(i);
size = mHandles.size();
// if removed from first place, move effect control from this handle to next in line
if (i == 0 && size != 0) {
sp<EffectHandle> h = mHandles[0].promote();
if (h != 0) {
h->setControl(true, true);
}
}
return size;
}
void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle)
{
// keep a strong reference on this EffectModule to avoid calling the
// destructor before we exit
sp<EffectModule> keep(this);
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->disconnectEffect(keep, handle);
}
}
}
void AudioFlinger::EffectModule::process()
{
Mutex::Autolock _l(mLock);
if (mEffectInterface == NULL || mConfig.inputCfg.buffer.raw == NULL || mConfig.outputCfg.buffer.raw == NULL) {
return;
}
if (mState != IDLE) {
// do 32 bit to 16 bit conversion for auxiliary effect input buffer
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount);
}
// TODO: handle effects with buffer provider
if (mState != ACTIVE) {
switch (mState) {
case RESET:
reset_l();
mState = STARTING;
// clear auxiliary effect input buffer for next accumulation
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
}
return;
case STARTING:
start_l();
mState = ACTIVE;
break;
case STOPPING:
mState = STOPPED;
break;
case STOPPED:
stop_l();
mState = IDLE;
return;
}
}
// do the actual processing in the effect engine
(*mEffectInterface)->process(mEffectInterface, &mConfig.inputCfg.buffer, &mConfig.outputCfg.buffer);
// clear auxiliary effect input buffer for next accumulation
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
memset(mConfig.inputCfg.buffer.raw, 0, mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
}
} else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw){
// If an insert effect is idle and input buffer is different from output buffer, copy input to
// output
sp<EffectChain> chain = mChain.promote();
if (chain != 0 && chain->activeTracks() != 0) {
size_t size = mConfig.inputCfg.buffer.frameCount * sizeof(int16_t);
if (mConfig.inputCfg.channels == CHANNEL_STEREO) {
size *= 2;
}
memcpy(mConfig.outputCfg.buffer.raw, mConfig.inputCfg.buffer.raw, size);
}
}
}
void AudioFlinger::EffectModule::reset_l()
{
if (mEffectInterface == NULL) {
return;
}
(*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
}
status_t AudioFlinger::EffectModule::configure()
{
uint32_t channels;
if (mEffectInterface == NULL) {
return NO_INIT;
}
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return DEAD_OBJECT;
}
// TODO: handle configuration of effects replacing track process
if (thread->channelCount() == 1) {
channels = CHANNEL_MONO;
} else {
channels = CHANNEL_STEREO;
}
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
mConfig.inputCfg.channels = CHANNEL_MONO;
} else {
mConfig.inputCfg.channels = channels;
}
mConfig.outputCfg.channels = channels;
mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15;
mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15;
mConfig.inputCfg.samplingRate = thread->sampleRate();
mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
mConfig.inputCfg.bufferProvider.cookie = NULL;
mConfig.inputCfg.bufferProvider.getBuffer = NULL;
mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
mConfig.outputCfg.bufferProvider.cookie = NULL;
mConfig.outputCfg.bufferProvider.getBuffer = NULL;
mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
// Insert effect:
// - in session 0 or -1, always overwrites output buffer: input buffer == output buffer
// - in other sessions:
// last effect in the chain accumulates in output buffer: input buffer != output buffer
// other effect: overwrites output buffer: input buffer == output buffer
// Auxiliary effect:
// accumulates in output buffer: input buffer != output buffer
// Therefore: accumulate <=> input buffer != output buffer
if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
} else {
mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
}
mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
mConfig.inputCfg.buffer.frameCount = thread->frameCount();
mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
status_t cmdStatus;
int size = sizeof(int);
status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_CONFIGURE, sizeof(effect_config_t), &mConfig, &size, &cmdStatus);
if (status == 0) {
status = cmdStatus;
}
return status;
}
status_t AudioFlinger::EffectModule::init()
{
Mutex::Autolock _l(mLock);
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t cmdStatus;
int size = sizeof(status_t);
status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_INIT, 0, NULL, &size, &cmdStatus);
if (status == 0) {
status = cmdStatus;
}
return status;
}
status_t AudioFlinger::EffectModule::start_l()
{
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t cmdStatus;
int size = sizeof(status_t);
status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_ENABLE, 0, NULL, &size, &cmdStatus);
if (status == 0) {
status = cmdStatus;
}
return status;
}
status_t AudioFlinger::EffectModule::stop_l()
{
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t cmdStatus;
int size = sizeof(status_t);
status_t status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_DISABLE, 0, NULL, &size, &cmdStatus);
if (status == 0) {
status = cmdStatus;
}
return status;
}
status_t AudioFlinger::EffectModule::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData)
{
Mutex::Autolock _l(mLock);
// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t status = (*mEffectInterface)->command(mEffectInterface, cmdCode, cmdSize, pCmdData, replySize, pReplyData);
if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
int size = (replySize == NULL) ? 0 : *replySize;
for (size_t i = 1; i < mHandles.size(); i++) {
sp<EffectHandle> h = mHandles[i].promote();
if (h != 0) {
h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
}
}
}
return status;
}
status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
{
Mutex::Autolock _l(mLock);
LOGV("setEnabled %p enabled %d", this, enabled);
if (enabled != isEnabled()) {
switch (mState) {
// going from disabled to enabled
case IDLE:
mState = RESET;
break;
case STOPPING:
mState = ACTIVE;
break;
case STOPPED:
mState = STARTING;
break;
// going from enabled to disabled
case RESET:
mState = IDLE;
break;
case STARTING:
mState = STOPPED;
break;
case ACTIVE:
mState = STOPPING;
break;
}
for (size_t i = 1; i < mHandles.size(); i++) {
sp<EffectHandle> h = mHandles[i].promote();
if (h != 0) {
h->setEnabled(enabled);
}
}
}
return NO_ERROR;
}
bool AudioFlinger::EffectModule::isEnabled()
{
switch (mState) {
case RESET:
case STARTING:
case ACTIVE:
return true;
case IDLE:
case STOPPING:
case STOPPED:
default:
return false;
}
}
status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
// Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
// if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
if ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) & (EFFECT_FLAG_VOLUME_CTRL|EFFECT_FLAG_VOLUME_IND)) {
status_t cmdStatus;
uint32_t volume[2];
uint32_t *pVolume = NULL;
int size = sizeof(volume);
volume[0] = *left;
volume[1] = *right;
if (controller) {
pVolume = volume;
}
status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_VOLUME, size, volume, &size, pVolume);
if (controller && status == NO_ERROR && size == sizeof(volume)) {
*left = volume[0];
*right = volume[1];
}
}
return status;
}
status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
// convert device bit field from AudioSystem to EffectApi format.
device = deviceAudioSystemToEffectApi(device);
if (device == 0) {
return BAD_VALUE;
}
status_t cmdStatus;
int size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_DEVICE, sizeof(uint32_t), &device, &size, &cmdStatus);
if (status == NO_ERROR) {
status = cmdStatus;
}
}
return status;
}
status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
// convert audio mode from AudioSystem to EffectApi format.
int effectMode = modeAudioSystemToEffectApi(mode);
if (effectMode < 0) {
return BAD_VALUE;
}
status_t cmdStatus;
int size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_SET_AUDIO_MODE, sizeof(int), &effectMode, &size, &cmdStatus);
if (status == NO_ERROR) {
status = cmdStatus;
}
}
return status;
}
// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified
const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = {
DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE
DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER
DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET
DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE
DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO
DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET
DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT
DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP
DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
DEVICE_AUX_DIGITAL // AudioSystem::DEVICE_OUT_AUX_DIGITAL
};
uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device)
{
uint32_t deviceOut = 0;
while (device) {
const uint32_t i = 31 - __builtin_clz(device);
device &= ~(1 << i);
if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) {
LOGE("device convertion error for AudioSystem device 0x%08x", device);
return 0;
}
deviceOut |= (uint32_t)sDeviceConvTable[i];
}
return deviceOut;
}
// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified
const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL
AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE
AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_CALL
};
int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)
{
int modeOut = -1;
if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) {
modeOut = (int)sModeConvTable[mode];
}
return modeOut;
}
status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
result.append(buffer);
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
result.append("\t\tCould not lock Fx mutex:\n");
}
result.append("\t\tSession Status State Engine:\n");
snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n",
mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
result.append(buffer);
result.append("\t\tDescriptor:\n");
snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n",
mDescriptor.apiVersion,
mDescriptor.flags);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- name: %s\n",
mDescriptor.name);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
mDescriptor.implementor);
result.append(buffer);
result.append("\t\t- Input configuration:\n");
result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n",
(uint32_t)mConfig.inputCfg.buffer.raw,
mConfig.inputCfg.buffer.frameCount,
mConfig.inputCfg.samplingRate,
mConfig.inputCfg.channels,
mConfig.inputCfg.format);
result.append(buffer);
result.append("\t\t- Output configuration:\n");
result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n",
(uint32_t)mConfig.outputCfg.buffer.raw,
mConfig.outputCfg.buffer.frameCount,
mConfig.outputCfg.samplingRate,
mConfig.outputCfg.channels,
mConfig.outputCfg.format);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
result.append(buffer);
result.append("\t\t\tPid Priority Ctrl Locked client server\n");
for (size_t i = 0; i < mHandles.size(); ++i) {
sp<EffectHandle> handle = mHandles[i].promote();
if (handle != 0) {
handle->dump(buffer, SIZE);
result.append(buffer);
}
}
result.append("\n");
write(fd, result.string(), result.length());
if (locked) {
mLock.unlock();
}
return NO_ERROR;
}
// ----------------------------------------------------------------------------
// EffectHandle implementation
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectHandle"
AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority)
: BnEffect(),
mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false)
{
LOGV("constructor %p", this);
int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
if (mCblkMemory != 0) {
mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
if (mCblk) {
new(mCblk) effect_param_cblk_t();
mBuffer = (uint8_t *)mCblk + bufOffset;
}
} else {
LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
return;
}
}
AudioFlinger::EffectHandle::~EffectHandle()
{
LOGV("Destructor %p", this);
disconnect();
}
status_t AudioFlinger::EffectHandle::enable()
{
if (!mHasControl) return INVALID_OPERATION;
if (mEffect == 0) return DEAD_OBJECT;
return mEffect->setEnabled(true);
}
status_t AudioFlinger::EffectHandle::disable()
{
if (!mHasControl) return INVALID_OPERATION;
if (mEffect == NULL) return DEAD_OBJECT;
return mEffect->setEnabled(false);
}
void AudioFlinger::EffectHandle::disconnect()
{
if (mEffect == 0) {
return;
}
mEffect->disconnect(this);
// release sp on module => module destructor can be called now
mEffect.clear();
if (mCblk) {
mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
}
mCblkMemory.clear(); // and free the shared memory
if (mClient != 0) {
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
}
status_t AudioFlinger::EffectHandle::command(int cmdCode, int cmdSize, void *pCmdData, int *replySize, void *pReplyData)
{
// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
// only get parameter command is permitted for applications not controlling the effect
if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
return INVALID_OPERATION;
}
if (mEffect == 0) return DEAD_OBJECT;
// handle commands that are not forwarded transparently to effect engine
if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
// No need to trylock() here as this function is executed in the binder thread serving a particular client process:
// no risk to block the whole media server process or mixer threads is we are stuck here
Mutex::Autolock _l(mCblk->lock);
if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
mCblk->serverIndex = 0;
mCblk->clientIndex = 0;
return BAD_VALUE;
}
status_t status = NO_ERROR;
while (mCblk->serverIndex < mCblk->clientIndex) {
int reply;
int rsize = sizeof(int);
int *p = (int *)(mBuffer + mCblk->serverIndex);
int size = *p++;
if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
LOGW("command(): invalid parameter block size");
break;
}
effect_param_t *param = (effect_param_t *)p;
if (param->psize == 0 || param->vsize == 0) {
LOGW("command(): null parameter or value size");
mCblk->serverIndex += size;
continue;
}
int psize = sizeof(effect_param_t) + ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + param->vsize;
status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, psize, p, &rsize, &reply);
if (ret == NO_ERROR) {
if (reply != NO_ERROR) {
status = reply;
}
} else {
status = ret;
}
mCblk->serverIndex += size;
}
mCblk->serverIndex = 0;
mCblk->clientIndex = 0;
return status;
} else if (cmdCode == EFFECT_CMD_ENABLE) {
return enable();
} else if (cmdCode == EFFECT_CMD_DISABLE) {
return disable();
}
return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
return mCblkMemory;
}
void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal)
{
LOGV("setControl %p control %d", this, hasControl);
mHasControl = hasControl;
if (signal && mEffectClient != 0) {
mEffectClient->controlStatusChanged(hasControl);
}
}
void AudioFlinger::EffectHandle::commandExecuted(int cmdCode, int cmdSize, void *pCmdData, int replySize, void *pReplyData)
{
if (mEffectClient != 0) {
mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
}
void AudioFlinger::EffectHandle::setEnabled(bool enabled)
{
if (mEffectClient != 0) {
mEffectClient->enableStatusChanged(enabled);
}
}
status_t AudioFlinger::EffectHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnEffect::onTransact(code, data, reply, flags);
}
void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
{
bool locked = tryLock(mCblk->lock);
snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
(mClient == NULL) ? getpid() : mClient->pid(),
mPriority,
mHasControl,
!locked,
mCblk->clientIndex,
mCblk->serverIndex
);
if (locked) {
mCblk->lock.unlock();
}
}
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectChain"
AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
int sessionId)
: mThread(wThread), mSessionId(sessionId), mVolumeCtrlIdx(-1), mActiveTrackCnt(0), mOwnInBuffer(false)
{
}
AudioFlinger::EffectChain::~EffectChain()
{
if (mOwnInBuffer) {
delete mInBuffer;
}
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc(effect_descriptor_t *descriptor)
{
sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
effect = mEffects[i];
break;
}
}
return effect;
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId(int id)
{
sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (mEffects[i]->id() == id) {
effect = mEffects[i];
break;
}
}
return effect;
}
// Must be called with EffectChain::mLock locked
void AudioFlinger::EffectChain::process_l()
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
mEffects[i]->process();
}
// if no track is active, input buffer must be cleared here as the mixer process
// will not do it
if (mSessionId > 0 && activeTracks() == 0) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
size_t numSamples = thread->frameCount() * thread->channelCount();
memset(mInBuffer, 0, numSamples * sizeof(int16_t));
}
}
}
status_t AudioFlinger::EffectChain::addEffect(sp<EffectModule>& effect)
{
effect_descriptor_t desc = effect->desc();
uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
Mutex::Autolock _l(mLock);
if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
// Auxiliary effects are inserted at the beginning of mEffects vector as
// they are processed first and accumulated in chain input buffer
mEffects.insertAt(effect, 0);
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return NO_INIT;
}
// the input buffer for auxiliary effect contains mono samples in
// 32 bit format. This is to avoid saturation in AudoMixer
// accumulation stage. Saturation is done in EffectModule::process() before
// calling the process in effect engine
size_t numSamples = thread->frameCount();
int32_t *buffer = new int32_t[numSamples];
memset(buffer, 0, numSamples * sizeof(int32_t));
effect->setInBuffer((int16_t *)buffer);
// auxiliary effects output samples to chain input buffer for further processing
// by insert effects
effect->setOutBuffer(mInBuffer);
} else {
// Insert effects are inserted at the end of mEffects vector as they are processed
// after track and auxiliary effects.
// Insert effect order as a function of indicated preference:
// if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
// another effect is present
// else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
// last effect claiming first position
// else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
// first effect claiming last position
// else if EFFECT_FLAG_INSERT_ANY insert after first or before last
// Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
// already present
int size = (int)mEffects.size();
int idx_insert = size;
int idx_insert_first = -1;
int idx_insert_last = -1;
for (int i = 0; i < size; i++) {
effect_descriptor_t d = mEffects[i]->desc();
uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
if (iMode == EFFECT_FLAG_TYPE_INSERT) {
// check invalid effect chaining combinations
if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
LOGW("addEffect() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
return INVALID_OPERATION;
}
// remember position of first insert effect and by default
// select this as insert position for new effect
if (idx_insert == size) {
idx_insert = i;
}
// remember position of last insert effect claiming
// first position
if (iPref == EFFECT_FLAG_INSERT_FIRST) {
idx_insert_first = i;
}
// remember position of first insert effect claiming
// last position
if (iPref == EFFECT_FLAG_INSERT_LAST &&
idx_insert_last == -1) {
idx_insert_last = i;
}
}
}
// modify idx_insert from first position if needed
if (insertPref == EFFECT_FLAG_INSERT_LAST) {
if (idx_insert_last != -1) {
idx_insert = idx_insert_last;
} else {
idx_insert = size;
}
} else {
if (idx_insert_first != -1) {
idx_insert = idx_insert_first + 1;
}
}
// always read samples from chain input buffer
effect->setInBuffer(mInBuffer);
// if last effect in the chain, output samples to chain
// output buffer, otherwise to chain input buffer
if (idx_insert == size) {
if (idx_insert != 0) {
mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
mEffects[idx_insert-1]->configure();
}
effect->setOutBuffer(mOutBuffer);
} else {
effect->setOutBuffer(mInBuffer);
}
mEffects.insertAt(effect, idx_insert);
// Always give volume control to last effect in chain with volume control capability
if (((desc.flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) &&
mVolumeCtrlIdx < idx_insert) {
mVolumeCtrlIdx = idx_insert;
}
LOGV("addEffect() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
}
effect->configure();
return NO_ERROR;
}
size_t AudioFlinger::EffectChain::removeEffect(const sp<EffectModule>& effect)
{
Mutex::Autolock _l(mLock);
int size = (int)mEffects.size();
int i;
uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
for (i = 0; i < size; i++) {
if (effect == mEffects[i]) {
if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
delete[] effect->inBuffer();
} else {
if (i == size - 1 && i != 0) {
mEffects[i - 1]->setOutBuffer(mOutBuffer);
mEffects[i - 1]->configure();
}
}
mEffects.removeAt(i);
LOGV("removeEffect() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
break;
}
}
// Return volume control to last effect in chain with volume control capability
if (mVolumeCtrlIdx == i) {
size = (int)mEffects.size();
for (i = size; i > 0; i--) {
if ((mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) & EFFECT_FLAG_VOLUME_CTRL) {
break;
}
}
// mVolumeCtrlIdx reset to -1 if no effect found with volume control flag set
mVolumeCtrlIdx = i - 1;
}
return mEffects.size();
}
void AudioFlinger::EffectChain::setDevice(uint32_t device)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
mEffects[i]->setDevice(device);
}
}
void AudioFlinger::EffectChain::setMode(uint32_t mode)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
mEffects[i]->setMode(mode);
}
}
bool AudioFlinger::EffectChain::setVolume(uint32_t *left, uint32_t *right)
{
uint32_t newLeft = *left;
uint32_t newRight = *right;
bool hasControl = false;
// first get volume update from volume controller
if (mVolumeCtrlIdx >= 0) {
mEffects[mVolumeCtrlIdx]->setVolume(&newLeft, &newRight, true);
hasControl = true;
}
// then indicate volume to all other effects in chain.
// Pass altered volume to effects before volume controller
// and requested volume to effects after controller
uint32_t lVol = newLeft;
uint32_t rVol = newRight;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if ((int)i == mVolumeCtrlIdx) continue;
// this also works for mVolumeCtrlIdx == -1 when there is no volume controller
if ((int)i > mVolumeCtrlIdx) {
lVol = *left;
rVol = *right;
}
mEffects[i]->setVolume(&lVol, &rVol, false);
}
*left = newLeft;
*right = newRight;
return hasControl;
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getVolumeController()
{
sp<EffectModule> effect;
if (mVolumeCtrlIdx >= 0) {
effect = mEffects[mVolumeCtrlIdx];
}
return effect;
}
status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
result.append(buffer);
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
result.append("\tCould not lock mutex:\n");
}
result.append("\tNum fx In buffer Out buffer Vol ctrl Active tracks:\n");
snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %02d %d\n",
mEffects.size(),
(uint32_t)mInBuffer,
(uint32_t)mOutBuffer,
(mVolumeCtrlIdx == -1) ? 0 : mEffects[mVolumeCtrlIdx]->id(),
mActiveTrackCnt);
result.append(buffer);
write(fd, result.string(), result.size());
for (size_t i = 0; i < mEffects.size(); ++i) {
sp<EffectModule> effect = mEffects[i];
if (effect != 0) {
effect->dump(fd, args);
}
}
if (locked) {
mLock.unlock();
}
return NO_ERROR;
}
#undef LOG_TAG
#define LOG_TAG "AudioFlinger"
// ----------------------------------------------------------------------------
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioFlinger::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
void AudioFlinger::instantiate() {
defaultServiceManager()->addService(
String16("media.audio_flinger"), new AudioFlinger());
}
}; // namespace android
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