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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 <hardware_legacy/AudioHardwareInterface.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#ifdef WITH_A2DP
#include "A2dpAudioInterface.h"
#endif
#ifdef LVMX
#include "lifevibes.h"
#endif
// ----------------------------------------------------------------------------
// 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;
// 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), mNextThreadId(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
setMode(AudioSystem::MODE_NORMAL);
setMasterVolume(1.0f);
setMasterMute(false);
} else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
#ifdef LVMX
LifeVibes::init();
#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,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
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);
}
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, &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)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
LOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
status_t ret = mAudioHardware->setMode(mode);
#ifdef LVMX
if (NO_ERROR == ret) {
LifeVibes::setMode(mode);
}
#endif
mHardwareStatus = AUDIO_HW_IDLE;
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) {
musicEnabled = 1;
} else if (value == LifevibesDisable) {
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 ((NO_ERROR == result) && (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)
{
LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
sp<IBinder> binder = client->asBinder();
if (mNotificationClients.indexOf(binder) < 0) {
LOGV("Adding notification client %p", binder.get());
binder->linkToDeath(this);
mNotificationClients.add(binder);
}
// 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::binderDied(const wp<IBinder>& who) {
LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
IBinder *binder = who.unsafe_get();
if (binder != NULL) {
int index = mNotificationClients.indexOf(binder);
if (index >= 0) {
LOGV("Removing notification client %p", binder);
mNotificationClients.removeAt(index);
}
}
}
// 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++) {
sp<IBinder> binder = mNotificationClients.itemAt(i);
LOGV("audioConfigChanged_l() Notifying change to client %p", binder.get());
sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
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),
mFormat(0), mFrameSize(1), 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 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 because audioConfigChanged() will lock AudioFlinger mLock
// before calling Audioflinger::audioConfigChanged_l() thus creating
// potential cross deadlock between AudioFlinger::mLock and mLock
mLock.unlock();
audioConfigChanged(configEvent->mEvent, configEvent->mParam);
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)
: ThreadBase(audioFlinger, id),
mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
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);
}
// notify client processes that a new input has been opened
sendConfigEvent(AudioSystem::OUTPUT_OPENED);
}
AudioFlinger::PlaybackThread::~PlaybackThread()
{
delete [] mMixBuffer;
}
status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(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 Buf S M F SRate LeftV RighV Serv User\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 Buf S M F SRate LeftV RighV Serv User\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::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);
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,
status_t *status)
{
sp<Track> track;
status_t lStatus;
if (mType == DIRECT) {
if (sampleRate != mSampleRate || format != mFormat || channelCount != 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);
if (track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
mTracks.add(track);
}
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);
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);
}
void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param);
switch (event) {
case AudioSystem::OUTPUT_OPENED:
case AudioSystem::OUTPUT_CONFIG_CHANGED:
desc.channels = mChannelCount;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = latency();
param2 = &desc;
break;
case AudioSystem::STREAM_CONFIG_CHANGED:
param2 = ¶m;
case AudioSystem::OUTPUT_CLOSED:
default:
break;
}
Mutex::Autolock _l(mAudioFlinger->mLock);
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
void AudioFlinger::PlaybackThread::readOutputParameters()
{
mSampleRate = mOutput->sampleRate();
mChannelCount = AudioSystem::popCount(mOutput->channels());
mFormat = mOutput->format();
mFrameSize = 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));
}
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);
}
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id)
: PlaybackThread(audioFlinger, output, id),
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()
{
int16_t* curBuf = mMixBuffer;
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;
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);
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
mAudioMixer->process(curBuf);
sleepTime = 0;
standbyTime = systemTime() + kStandbyTimeInNsecs;
} 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 (curBuf, 0, mixBufferSize);
sleepTime = 0;
LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
}
}
if (mSuspended) {
sleepTime = idleSleepTime;
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::process(audioOutputType, curBuf, mixBufferSize);
}
#endif
int bytesWritten = (int)mOutput->write(curBuf, 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 {
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();
}
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();
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
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);
// compute volume for this track
int16_t left, right;
if (track->isMuted() || masterMute || track->isPausing() ||
mStreamTypes[track->type()].mute) {
left = right = 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;
float v_clamped = v * cblk->volume[0];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = int16_t(v_clamped);
v_clamped = v * cblk->volume[1];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = int16_t(v_clamped);
}
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(track);
mAudioMixer->enable(AudioMixer::MIXING);
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;
}
#ifdef LVMX
if ( tracksConnectedChanged || stateChanged )
{
// only do the ramp when the volume is changed by the user / application
param = AudioMixer::VOLUME;
}
#endif
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::FORMAT, track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::CHANNEL_COUNT, track->channelCount());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
int(cblk->sampleRate));
// 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);
mAudioMixer->disable(AudioMixer::MIXING);
} 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->isTerminated()) {
mTracks.remove(track);
deleteTrackName_l(track->mName);
}
}
}
return mixerStatus;
}
void AudioFlinger::MixerThread::getTracks(
SortedVector < sp<Track> >& tracks,
SortedVector < wp<Track> >& activeTracks,
int streamType)
{
LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this, mTracks.size(), mActiveTracks.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) {
tracks.add(t);
int j = mActiveTracks.indexOf(t);
if (j >= 0) {
t = mActiveTracks[j].promote();
if (t != NULL) {
activeTracks.add(t);
}
}
}
}
size = activeTracks.size();
for (size_t i = 0; i < size; i++) {
mActiveTracks.remove(activeTracks[i]);
}
size = tracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = tracks[i];
mTracks.remove(t);
deleteTrackName_l(t->name());
}
}
void AudioFlinger::MixerThread::putTracks(
SortedVector < sp<Track> >& tracks,
SortedVector < wp<Track> >& activeTracks)
{
LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this, tracks.size(), activeTracks.size());
Mutex::Autolock _l(mLock);
size_t size = tracks.size();
for (size_t i = 0; i < size ; i++) {
sp<Track> t = tracks[i];
int name = getTrackName_l();
if (name < 0) return;
t->mName = name;
t->mThread = this;
mTracks.add(t);
int j = activeTracks.indexOf(t);
if (j >= 0) {
mActiveTracks.add(t);
// force buffer refilling and no ramp volume when the track is mixed for the first time
t->mFillingUpStatus = Track::FS_FILLING;
}
}
}
// 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 (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)
: PlaybackThread(audioFlinger, output, id),
mLeftVolume (1.0), mRightVolume(1.0)
{
mType = PlaybackThread::DIRECT;
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
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())
{
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;
}
}
// 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);
// 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 != mLeftVolume || right != mRightVolume) {
mOutput->setVolume(left, right);
left = mLeftVolume;
right = mRightVolume;
}
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
}
}
// 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 (trackToRemove->isTerminated()) {
mTracks.remove(trackToRemove);
deleteTrackName_l(trackToRemove->mName);
}
}
}
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) {
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
mStandby = false;
} else {
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();
}
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), 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()
{
int16_t* curBuf = mMixBuffer;
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;
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);
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
if (outputsReady(outputTracks)) {
mAudioMixer->process(curBuf);
} else {
memset(curBuf, 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;
break;
}
}
}
}
if (mSuspended) {
sleepTime = idleSleepTime;
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
standbyTime = systemTime() + kStandbyTimeInNsecs;
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->write(curBuf, writeFrames);
}
mStandby = false;
mBytesWritten += mixBufferSize;
} else {
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();
}
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)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(0),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK)
{
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->channels = (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->flowControlFlag = 1;
} 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->channels = (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->flowControlFlag = 1;
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->channels;
}
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, channels %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels);
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)
: TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1)
{
if (mCblk != NULL) {
sp<ThreadBase> baseThread = thread.promote();
if (baseThread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
mName = playbackThread->getTrackName_l();
}
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, " %5d %5d %3u %3u %3u %04u %1d %1d %1d %5u %5u %5u %08x %08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
mFormat,
mCblk->channels,
mFrameCount,
mState,
mMute,
mFillingUpStatus,
mCblk->sampleRate,
mCblk->volume[0],
mCblk->volume[1],
mCblk->server,
mCblk->user);
}
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->forceReady) {
mFillingUpStatus = FS_FILLED;
mCblk->forceReady = 0;
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->flowControlFlag = 1;
mCblk->forceReady = 0;
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;
}
// ----------------------------------------------------------------------------
// 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)
: TrackBase(thread, client, sampleRate, format,
channelCount, frameCount, flags, 0),
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->flowControlFlag = 1;
}
}
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u %03u %04u %01d %05u %08x %08x\n",
(mClient == NULL) ? getpid() : mClient->pid(),
mFormat,
mCblk->channels,
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),
mActive(false), mSourceThread(sourceThread)
{
PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
if (mCblk != NULL) {
mCblk->out = 1;
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->channels %d mBufferEnd %p",
mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, 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 channels = mCblk->channels;
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 * channels];
pInBuffer->frameCount = startFrames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, startFrames * channels * 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 * channels * sizeof(int16_t));
mCblk->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channels;
mOutBuffer.frameCount -= outFrames;
mOutBuffer.i16 += outFrames * channels;
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 * channels];
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->i16 = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * 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 * channels];
pInBuffer->frameCount = frames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, frames * channels * 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::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::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,
status_t *status)
{
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
RecordThread *thread;
size_t inFrameCount;
// 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);
}
// 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);
}
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();
sendConfigEvent(AudioSystem::INPUT_OPENED);
}
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 (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 &&
(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 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(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
switch (event) {
case AudioSystem::INPUT_OPENED:
case AudioSystem::INPUT_CONFIG_CHANGED:
desc.channels = mChannelCount;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = 0;
param2 = &desc;
break;
case AudioSystem::INPUT_CLOSED:
default:
break;
}
Mutex::Autolock _l(mAudioFlinger->mLock);
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();
mChannelCount = AudioSystem::popCount(mInput->channels());
mFormat = mInput->format();
mFrameSize = 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) {
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
(format != AudioSystem::PCM_16_BIT) ||
(channels != AudioSystem::CHANNEL_OUT_STEREO)) {
thread = new DirectOutputThread(this, output, ++mNextThreadId);
LOGV("openOutput() created direct output: ID %d thread %p", mNextThreadId, thread);
} else {
thread = new MixerThread(this, output, ++mNextThreadId);
LOGV("openOutput() created mixer output: ID %d thread %p", mNextThreadId, 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(mNextThreadId, thread);
if (pSamplingRate) *pSamplingRate = samplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = channels;
if (pLatencyMs) *pLatencyMs = thread->latency();
return mNextThreadId;
}
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;
}
DuplicatingThread *thread = new DuplicatingThread(this, thread1, ++mNextThreadId);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(mNextThreadId, thread);
return mNextThreadId;
}
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) {
// Start record thread
thread = new RecordThread(this, input, reqSamplingRate, reqChannels, ++mNextThreadId);
mRecordThreads.add(mNextThreadId, thread);
LOGV("openInput() created record thread: ID %d thread %p", mNextThreadId, thread);
if (pSamplingRate) *pSamplingRate = reqSamplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = reqChannels;
input->standby();
return mNextThreadId;
}
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);
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;
SortedVector < sp<MixerThread::Track> > tracks;
SortedVector < wp<MixerThread::Track> > activeTracks;
srcThread->getTracks(tracks, activeTracks, stream);
if (tracks.size()) {
dstThread->putTracks(tracks, activeTracks);
}
}
}
dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream);
return NO_ERROR;
}
// 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;
}
// ----------------------------------------------------------------------------
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