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android / platform / frameworks / base / e1dfcb7ab01fb991079ec1f70f75281a0ca9073e / . / services / audioflinger / AudioFlinger.cpp
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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/IPCThreadState.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 <utils/Atomic.h>
#include <cutils/bitops.h>
#include <cutils/properties.h>
#include <media/AudioTrack.h>
#include <media/AudioRecord.h>
#include <media/IMediaPlayerService.h>
#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
// ----------------------------------------------------------------------------
namespace android {
static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
static const char* kHardwareLockedString = "Hardware lock is taken\n";
//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const float MAX_GAIN = 4096.0f;
static const float MAX_GAIN_INT = 0x1000;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
static const int8_t kMaxTrackRetries = 50;
static const int8_t kMaxTrackStartupRetries = 50;
// allow less retry attempts on direct output thread.
// direct outputs can be a scarce resource in audio hardware and should
// be released as quickly as possible.
static const int8_t kMaxTrackRetriesDirect = 2;
static const int kDumpLockRetries = 50;
static const int kDumpLockSleep = 20000;
static const nsecs_t kWarningThrottle = seconds(5);
// ----------------------------------------------------------------------------
static bool recordingAllowed() {
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;
}
static bool settingsAllowed() {
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;
}
// To collect the amplifier usage
static void addBatteryData(uint32_t params) {
sp<IBinder> binder =
defaultServiceManager()->getService(String16("media.player"));
sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
if (service.get() == NULL) {
LOGW("Cannot connect to the MediaPlayerService for battery tracking");
return;
}
service->addBatteryData(params);
}
static int load_audio_interface(const char *if_name, const hw_module_t **mod,
audio_hw_device_t **dev)
{
int rc;
rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
if (rc)
goto out;
rc = audio_hw_device_open(*mod, dev);
LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc)
goto out;
return 0;
out:
*mod = NULL;
*dev = NULL;
return rc;
}
static const char *audio_interfaces[] = {
"primary",
"a2dp",
"usb",
};
#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1)
{
}
void AudioFlinger::onFirstRef()
{
int rc = 0;
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
mHardwareStatus = AUDIO_HW_IDLE;
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
const hw_module_t *mod;
audio_hw_device_t *dev;
rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
if (rc)
continue;
LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
mod->name, mod->id);
mAudioHwDevs.push(dev);
if (!mPrimaryHardwareDev) {
mPrimaryHardwareDev = dev;
LOGI("Using '%s' (%s.%s) as the primary audio interface",
mod->name, mod->id, audio_interfaces[i]);
}
}
mHardwareStatus = AUDIO_HW_INIT;
if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
LOGE("Primary audio interface not found");
return;
}
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->init_check(dev);
if (rc == 0) {
AutoMutex lock(mHardwareLock);
mMode = AUDIO_MODE_NORMAL;
mHardwareStatus = AUDIO_HW_SET_MODE;
dev->set_mode(dev, mMode);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
dev->set_master_volume(dev, 1.0f);
mHardwareStatus = AUDIO_HW_IDLE;
}
}
}
status_t AudioFlinger::initCheck() const
{
Mutex::Autolock _l(mLock);
if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
return NO_INIT;
return NO_ERROR;
}
AudioFlinger::~AudioFlinger()
{
int num_devs = mAudioHwDevs.size();
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));
}
for (int i = 0; i < num_devs; i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
audio_hw_device_close(dev);
}
mAudioHwDevs.clear();
}
audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
{
/* first matching HW device is returned */
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
if ((dev->get_supported_devices(dev) & devices) == devices)
return dev;
}
return NULL;
}
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);
}
// dump all hardware devs
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
dev->dump(dev, fd);
}
if (locked) mLock.unlock();
}
return NO_ERROR;
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
uint32_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
int lSessionId;
if (streamType >= AUDIO_STREAM_CNT) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
PlaybackThread *effectThread = NULL;
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);
}
LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output) {
// prevent same audio session on different output threads
uint32_t sessions = t->hasAudioSession(*sessionId);
if (sessions & PlaybackThread::TRACK_SESSION) {
lStatus = BAD_VALUE;
goto Exit;
}
// check if an effect with same session ID is waiting for a track to be created
if (sessions & PlaybackThread::EFFECT_SESSION) {
effectThread = t.get();
}
}
}
lSessionId = *sessionId;
} else {
// if no audio session id is provided, create one here
lSessionId = nextUniqueId_l();
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
LOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
moveEffectChain_l(lSessionId, effectThread, thread, true);
}
}
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();
}
uint32_t 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
{ // scope for the lock
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
value = 1.0f;
}
mHardwareStatus = AUDIO_HW_IDLE;
}
Mutex::Autolock _l(mLock);
mMasterVolume = value;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterVolume(value);
return NO_ERROR;
}
status_t AudioFlinger::setMode(int mode)
{
status_t ret;
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
LOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
{ // scope for the lock
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
mHardwareStatus = AUDIO_HW_IDLE;
}
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMode(mode);
}
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 = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
bool state = AUDIO_MODE_INVALID;
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
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) >= AUDIO_STREAM_CNT) {
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) >= AUDIO_STREAM_CNT ||
uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
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) >= AUDIO_STREAM_CNT) {
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)AUDIO_STREAM_CNT) {
return true;
}
return mStreamTypes[stream].mute;
}
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;
}
// ioHandle == 0 means the parameters are global to the audio hardware interface
if (ioHandle == 0) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
status_t final_result = NO_ERROR;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
result = dev->set_parameters(dev, keyValuePairs.string());
final_result = result ?: final_result;
}
mHardwareStatus = AUDIO_HW_IDLE;
return final_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);
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) {
String8 out_s8;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
char *s = dev->get_parameters(dev, keys.string());
out_s8 += String8(s);
free(s);
}
return out_s8;
}
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 mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, 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 = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
{
status_t status;
Mutex::Autolock _l(mLock);
PlaybackThread *playbackThread = checkPlaybackThread_l(output);
if (playbackThread != NULL) {
return playbackThread->getRenderPosition(halFrames, dspFrames);
}
return BAD_VALUE;
}
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{
Mutex::Autolock _l(mLock);
int pid = IPCThreadState::self()->getCallingPid();
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
pid);
LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
mNotificationClients.add(pid, notificationClient);
sp<IBinder> binder = client->asBinder();
binder->linkToDeath(notificationClient);
// the config change is always sent from playback or record threads to avoid deadlock
// with AudioSystem::gLock
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
}
}
}
void AudioFlinger::removeNotificationClient(pid_t pid)
{
Mutex::Autolock _l(mLock);
int index = mNotificationClients.indexOfKey(pid);
if (index >= 0) {
sp <NotificationClient> client = mNotificationClients.valueFor(pid);
LOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
mNotificationClients.removeItem(pid);
}
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
}
}
// removeClient_l() must be called with AudioFlinger::mLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// ----------------------------------------------------------------------------
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id)
: Thread(false),
mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false)
{
}
AudioFlinger::ThreadBase::~ThreadBase()
{
mParamCond.broadcast();
mNewParameters.clear();
}
void AudioFlinger::ThreadBase::exit()
{
// keep a strong ref on ourself so that we wont get
// destroyed in the middle of requestExitAndWait()
sp <ThreadBase> strongMe = this;
LOGV("ThreadBase::exit");
{
AutoMutex lock(&mLock);
mExiting = true;
requestExit();
mWaitWorkCV.signal();
}
requestExitAndWait();
}
uint32_t AudioFlinger::ThreadBase::sampleRate() const
{
return mSampleRate;
}
int AudioFlinger::ThreadBase::channelCount() const
{
return (int)mChannelCount;
}
uint32_t AudioFlinger::ThreadBase::format() const
{
return mFormat;
}
size_t AudioFlinger::ThreadBase::frameCount() const
{
return mFrameCount;
}
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
status_t status;
LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
Mutex::Autolock _l(mLock);
mNewParameters.add(keyValuePairs);
mWaitWorkCV.signal();
// wait condition with timeout in case the thread loop has exited
// before the request could be processed
if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) {
status = mParamStatus;
mWaitWorkCV.signal();
} else {
status = TIMED_OUT;
}
return status;
}
void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
{
Mutex::Autolock _l(mLock);
sendConfigEvent_l(event, param);
}
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
ConfigEvent *configEvent = new ConfigEvent();
configEvent->mEvent = event;
configEvent->mParam = param;
mConfigEvents.add(configEvent);
LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
}
void AudioFlinger::ThreadBase::processConfigEvents()
{
mLock.lock();
while(!mConfigEvents.isEmpty()) {
LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
ConfigEvent *configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
mAudioFlinger->mLock.lock();
audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
mAudioFlinger->mLock.unlock();
delete configEvent;
mLock.lock();
}
mLock.unlock();
}
status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
bool locked = tryLock(mLock);
if (!locked) {
snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
write(fd, buffer, strlen(buffer));
}
snprintf(buffer, SIZE, "standby: %d\n", mStandby);
result.append(buffer);
snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
result.append(buffer);
snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
result.append(buffer);
snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
result.append(buffer);
result.append(" Index Command");
for (size_t i = 0; i < mNewParameters.size(); ++i) {
snprintf(buffer, SIZE, "\n %02d ", i);
result.append(buffer);
result.append(mNewParameters[i]);
}
snprintf(buffer, SIZE, "\n\nPending config events: \n");
result.append(buffer);
snprintf(buffer, SIZE, " Index event param\n");
result.append(buffer);
for (size_t i = 0; i < mConfigEvents.size(); i++) {
snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
result.append(buffer);
}
result.append("\n");
write(fd, result.string(), result.size());
if (locked) {
mLock.unlock();
}
return NO_ERROR;
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: ThreadBase(audioFlinger, id),
mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
mDevice(device)
{
readOutputParameters();
mMasterVolume = mAudioFlinger->masterVolume();
mMasterMute = mAudioFlinger->masterMute();
for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
}
}
AudioFlinger::PlaybackThread::~PlaybackThread()
{
delete [] mMixBuffer;
}
status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
dumpEffectChains(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
wp<Track> wTrack = mActiveTracks[i];
if (wTrack != 0) {
sp<Track> track = wTrack.promote();
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mEffectChains.size(); ++i) {
sp<EffectChain> chain = mEffectChains[i];
if (chain != 0) {
chain->dump(fd, args);
}
}
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
result.append(buffer);
snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
result.append(buffer);
snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
result.append(buffer);
snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
result.append(buffer);
snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
result.append(buffer);
write(fd, result.string(), result.size());
dumpBase(fd, args);
return NO_ERROR;
}
// Thread virtuals
status_t AudioFlinger::PlaybackThread::readyToRun()
{
if (mSampleRate == 0) {
LOGE("No working audio driver found.");
return NO_INIT;
}
LOGI("AudioFlinger's thread %p ready to run", this);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::onFirstRef()
{
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "Playback Thread %p", this);
run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
}
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
uint32_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
status_t *status)
{
sp<Track> track;
status_t lStatus;
if (mType == DIRECT) {
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
"for output %p with format %d",
sampleRate, format, channelMask, mOutput, mFormat);
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);
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
uint32_t strategy =
AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0) {
if (sessionId == t->sessionId() &&
strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
lStatus = BAD_VALUE;
goto Exit;
}
}
}
track = new Track(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId);
if (track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
mTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
chain->incTrackCnt();
}
}
lStatus = NO_ERROR;
Exit:
if(status) {
*status = lStatus;
}
return track;
}
uint32_t AudioFlinger::PlaybackThread::latency() const
{
if (mOutput) {
return mOutput->stream->get_latency(mOutput->stream);
}
else {
return 0;
}
}
status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
mMasterVolume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
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)
{
mStreamTypes[stream].volume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
{
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;
}
// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
status_t status = ALREADY_EXISTS;
// set retry count for buffer fill
track->mRetryCount = kMaxTrackStartupRetries;
if (mActiveTracks.indexOf(track) < 0) {
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
track->mFillingUpStatus = Track::FS_FILLING;
track->mResetDone = false;
mActiveTracks.add(track);
if (track->mainBuffer() != mMixBuffer) {
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
chain->incActiveTrackCnt();
}
}
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) {
removeTrack_l(track);
}
}
void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
{
mTracks.remove(track);
deleteTrackName_l(track->name());
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
chain->decTrackCnt();
}
}
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
String8 out_s8;
char *s;
s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
out_s8 = String8(s);
free(s);
return out_s8;
}
// destroyTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
switch (event) {
case AudioSystem::OUTPUT_OPENED:
case AudioSystem::OUTPUT_CONFIG_CHANGED:
desc.channels = mChannelMask;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = latency();
param2 = &desc;
break;
case AudioSystem::STREAM_CONFIG_CHANGED:
param2 = &param;
case AudioSystem::OUTPUT_CLOSED:
default:
break;
}
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
void AudioFlinger::PlaybackThread::readOutputParameters()
{
mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / 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));
// force reconfiguration of effect chains and engines to take new buffer size and audio
// parameters into account
// Note that mLock is not held when readOutputParameters() is called from the constructor
// but in this case nothing is done below as no audio sessions have effect yet so it doesn't
// matter.
// create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains
Vector< sp<EffectChain> > effectChains = mEffectChains;
for (size_t i = 0; i < effectChains.size(); i ++) {
mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false);
}
}
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 / audio_stream_frame_size(&mOutput->stream->common);
return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
}
uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
{
Mutex::Autolock _l(mLock);
uint32_t result = 0;
if (getEffectChain_l(sessionId) != 0) {
result = EFFECT_SESSION;
}
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
result |= TRACK_SESSION;
break;
}
}
return result;
}
uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
{
// session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
// it is moved to correct output by audio policy manager when A2DP is connected or disconnected
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
for (size_t i = 0; i < mTracks.size(); i++) {
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
}
}
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId)
{
Mutex::Autolock _l(mLock);
return getEffectChain_l(sessionId);
}
sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId)
{
sp<EffectChain> chain;
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() == sessionId) {
chain = mEffectChains[i];
break;
}
}
return chain;
}
void AudioFlinger::PlaybackThread::setMode(uint32_t mode)
{
Mutex::Autolock _l(mLock);
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
mEffectChains[i]->setMode_l(mode);
}
}
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device),
mAudioMixer(0)
{
mType = PlaybackThread::MIXER;
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
LOGE("Invalid audio hardware channel count");
}
}
AudioFlinger::MixerThread::~MixerThread()
{
delete mAudioMixer;
}
bool AudioFlinger::MixerThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
uint32_t mixerStatus = MIXER_IDLE;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount * mFrameSize;
// FIXME: Relaxed timing because of a certain device that can't meet latency
// Should be reduced to 2x after the vendor fixes the driver issue
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
nsecs_t lastWarning = 0;
bool longStandbyExit = false;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
Vector< sp<EffectChain> > effectChains;
while (!exitPending())
{
processConfigEvents();
mixerStatus = MIXER_IDLE;
{ // scope for mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
mixBufferSize = mFrameCount * mFrameSize;
// FIXME: Relaxed timing because of a certain device that can't meet latency
// Should be reduced to 2x after the vendor fixes the driver issue
maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
}
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
mSuspended) {
if (!mStandby) {
LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
}
if (!activeTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) break;
// wait until we have something to do...
LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
mWaitWorkCV.wait(mLock);
LOGV("MixerThread %p TID %d waking up\n", this, gettid());
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + kStandbyTimeInNsecs;
sleepTime = idleSleepTime;
continue;
}
}
mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
mAudioMixer->process();
sleepTime = 0;
standbyTime = systemTime() + kStandbyTimeInNsecs;
//TODO: delay standby when effects have a tail
} else {
// If no tracks are ready, sleep once for the duration of an output
// buffer size, then write 0s to the output
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_ENABLED) {
sleepTime = activeSleepTime;
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 ||
(mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
memset (mMixBuffer, 0, mixBufferSize);
sleepTime = 0;
LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
}
// TODO add standby time extension fct of effect tail
}
if (mSuspended) {
sleepTime = suspendSleepTimeUs();
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
// enable changes in effect chain
unlockEffectChains(effectChains);
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
nsecs_t now = systemTime();
nsecs_t delta = now - mLastWriteTime;
if (delta > maxPeriod) {
mNumDelayedWrites++;
if ((now - lastWarning) > kWarningThrottle) {
LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
}
if (mStandby) {
longStandbyExit = true;
}
}
mStandby = false;
} else {
// enable changes in effect chain
unlockEffectChains(effectChains);
usleep(sleepTime);
}
// finally let go of all our tracks, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
if (!mStandby) {
mOutput->stream->common.standby(&mOutput->stream->common);
}
LOGV("MixerThread %p exiting", this);
return false;
}
// prepareTracks_l() must be called with ThreadBase::mLock held
uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
{
uint32_t mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
size_t count = activeTracks.size();
size_t mixedTracks = 0;
size_t tracksWithEffect = 0;
float masterVolume = mMasterVolume;
bool masterMute = mMasterMute;
if (masterMute) {
masterVolume = 0;
}
// Delegate master volume control to effect in output mix effect chain if needed
sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (chain != 0) {
uint32_t v = (uint32_t)(masterVolume * (1 << 24));
chain->setVolume_l(&v, &v);
masterVolume = (float)((v + (1 << 23)) >> 24);
chain.clear();
}
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
mAudioMixer->setActiveTrack(track->name());
if (cblk->framesReady() && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
//LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
mixedTracks++;
// track->mainBuffer() != mMixBuffer means there is an effect chain
// connected to the track
chain.clear();
if (track->mainBuffer() != mMixBuffer) {
chain = getEffectChain_l(track->sessionId());
// Delegate volume control to effect in track effect chain if needed
if (chain != 0) {
tracksWithEffect++;
} else {
LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
track->name(), track->sessionId());
}
}
int param = AudioMixer::VOLUME;
if (track->mFillingUpStatus == Track::FS_FILLED) {
// no ramp for the first volume setting
track->mFillingUpStatus = Track::FS_ACTIVE;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
}
mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
param = AudioMixer::RAMP_VOLUME;
}
// compute volume for this track
uint32_t vl, vr, va;
if (track->isMuted() || track->isPausing() ||
mStreamTypes[track->type()].mute) {
vl = vr = va = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
// read original volumes with volume control
float typeVolume = mStreamTypes[track->type()].volume;
float v = masterVolume * typeVolume;
vl = (uint32_t)(v * cblk->volume[0]) << 12;
vr = (uint32_t)(v * cblk->volume[1]) << 12;
va = (uint32_t)(v * cblk->sendLevel);
}
// Delegate volume control to effect in track effect chain if needed
if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
// Do not ramp volume if volume is controlled by effect
param = AudioMixer::VOLUME;
track->mHasVolumeController = true;
} else {
// force no volume ramp when volume controller was just disabled or removed
// from effect chain to avoid volume spike
if (track->mHasVolumeController) {
param = AudioMixer::VOLUME;
}
track->mHasVolumeController = false;
}
// Convert volumes from 8.24 to 4.12 format
int16_t left, right, aux;
uint32_t v_clamped = (vl + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
left = int16_t(v_clamped);
v_clamped = (vr + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
right = int16_t(v_clamped);
if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
aux = int16_t(va);
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(track);
mAudioMixer->enable(AudioMixer::MIXING);
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(cblk->sampleRate));
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
// reset retry count
track->mRetryCount = kMaxTrackRetries;
mixerStatus = MIXER_TRACKS_READY;
} else {
//LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
if (track->isStopped()) {
track->reset();
}
if (track->isTerminated() || track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
tracksToRemove->add(track);
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun
android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
} else if (mixerStatus != MIXER_TRACKS_READY) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
mAudioMixer->disable(AudioMixer::MIXING);
}
}
// remove all the tracks that need to be...
count = tracksToRemove->size();
if (UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove->itemAt(i);
mActiveTracks.remove(track);
if (track->mainBuffer() != mMixBuffer) {
chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
chain->decActiveTrackCnt();
}
}
if (track->isTerminated()) {
removeTrack_l(track);
}
}
}
// mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to
// mix buffer and track effects will accumulate into it
if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
}
return mixerStatus;
}
void AudioFlinger::MixerThread::invalidateTracks(int streamType)
{
LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
this, streamType, mTracks.size());
Mutex::Autolock _l(mLock);
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
if (t->type() == streamType) {
android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
t->mCblk->cv.signal();
}
}
}
// 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 != AUDIO_FORMAT_PCM_16_BIT) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
if (value != AUDIO_CHANNEL_OUT_STEREO) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
// when changing the audio output device, call addBatteryData to notify
// the change
if ((int)mDevice != value) {
uint32_t params = 0;
// check whether speaker is on
if (value & AUDIO_DEVICE_OUT_SPEAKER) {
params |= IMediaPlayerService::kBatteryDataSpeakerOn;
}
int deviceWithoutSpeaker
= AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
// check if any other device (except speaker) is on
if (value & deviceWithoutSpeaker ) {
params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
}
if (params != 0) {
addBatteryData(params);
}
}
// forward device change to effects that have requested to be
// aware of attached audio device.
mDevice = (uint32_t)value;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevice_l(mDevice);
}
}
if (status == NO_ERROR) {
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
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->stream->get_latency(mOutput->stream) * 1000) / 2;
}
uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
{
return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
}
uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs()
{
return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
}
// ----------------------------------------------------------------------------
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device)
{
mType = PlaybackThread::DIRECT;
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
static inline int16_t clamp16(int32_t sample)
{
if ((sample>>15) ^ (sample>>31))
sample = 0x7FFF ^ (sample>>31);
return sample;
}
static inline
int32_t mul(int16_t in, int16_t v)
{
#if defined(__arm__) && !defined(__thumb__)
int32_t out;
asm( "smulbb %[out], %[in], %[v] \n"
: [out]"=r"(out)
: [in]"%r"(in), [v]"r"(v)
: );
return out;
#else
return in * int32_t(v);
#endif
}
void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
{
// Do not apply volume on compressed audio
if (!audio_is_linear_pcm(mFormat)) {
return;
}
// convert to signed 16 bit before volume calculation
if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
uint8_t *src = (uint8_t *)mMixBuffer + count-1;
int16_t *dst = mMixBuffer + count-1;
while(count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
}
size_t frameCount = mFrameCount;
int16_t *out = mMixBuffer;
if (ramp) {
if (mChannelCount == 1) {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out++;
vl += vlInc;
} while (--frameCount);
} else {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
int32_t vrInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
int32_t vr = ((int32_t)mRightVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
out += 2;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
}
} else {
if (mChannelCount == 1) {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out++;
} while (--frameCount);
} else {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out[1] = clamp16(mul(out[1], rightVol) >> 12);
out += 2;
} while (--frameCount);
}
}
// convert back to unsigned 8 bit after volume calculation
if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
int16_t *src = mMixBuffer;
uint8_t *dst = (uint8_t *)mMixBuffer;
while(count--) {
*dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
}
}
mLeftVolShort = leftVol;
mRightVolShort = rightVol;
}
bool AudioFlinger::DirectOutputThread::threadLoop()
{
uint32_t mixerStatus = MIXER_IDLE;
sp<Track> trackToRemove;
sp<Track> activeTrack;
nsecs_t standbyTime = systemTime();
int8_t *curBuf;
size_t mixBufferSize = mFrameCount*mFrameSize;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
// use shorter standby delay as on normal output to release
// hardware resources as soon as possible
nsecs_t standbyDelay = microseconds(activeSleepTime*2);
while (!exitPending())
{
bool rampVolume;
uint16_t leftVol;
uint16_t rightVol;
Vector< sp<EffectChain> > effectChains;
processConfigEvents();
mixerStatus = MIXER_IDLE;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
mixBufferSize = mFrameCount*mFrameSize;
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
standbyDelay = microseconds(activeSleepTime*2);
}
// put audio hardware into standby after short delay
if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
mSuspended) {
// wait until we have something to do...
if (!mStandby) {
LOGV("Audio hardware entering standby, mixer %p\n", this);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
}
if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) break;
LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
mWaitWorkCV.wait(mLock);
LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + standbyDelay;
sleepTime = idleSleepTime;
continue;
}
}
effectChains = mEffectChains;
// find out which tracks need to be processed
if (mActiveTracks.size() != 0) {
sp<Track> t = mActiveTracks[0].promote();
if (t == 0) continue;
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
if (cblk->framesReady() && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
//LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
mLeftVolFloat = mRightVolFloat = 0;
mLeftVolShort = mRightVolShort = 0;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
rampVolume = true;
}
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
rampVolume = true;
}
// compute volume for this track
float left, right;
if (track->isMuted() || mMasterMute || track->isPausing() ||
mStreamTypes[track->type()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
float typeVolume = mStreamTypes[track->type()].volume;
float v = mMasterVolume * typeVolume;
float v_clamped = v * cblk->volume[0];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = v_clamped/MAX_GAIN;
v_clamped = v * cblk->volume[1];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = v_clamped/MAX_GAIN;
}
if (left != mLeftVolFloat || right != mRightVolFloat) {
mLeftVolFloat = left;
mRightVolFloat = right;
// If audio HAL implements volume control,
// force software volume to nominal value
if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
left = 1.0f;
right = 1.0f;
}
// Convert volumes from float to 8.24
uint32_t vl = (uint32_t)(left * (1 << 24));
uint32_t vr = (uint32_t)(right * (1 << 24));
// Delegate volume control to effect in track effect chain if needed
// only one effect chain can be present on DirectOutputThread, so if
// there is one, the track is connected to it
if (!effectChains.isEmpty()) {
// Do not ramp volume if volume is controlled by effect
if(effectChains[0]->setVolume_l(&vl, &vr)) {
rampVolume = false;
}
}
// Convert volumes from 8.24 to 4.12 format
uint32_t v_clamped = (vl + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
leftVol = (uint16_t)v_clamped;
v_clamped = (vr + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
rightVol = (uint16_t)v_clamped;
} else {
leftVol = mLeftVolShort;
rightVol = mRightVolShort;
rampVolume = false;
}
// reset retry count
track->mRetryCount = kMaxTrackRetriesDirect;
activeTrack = t;
mixerStatus = MIXER_TRACKS_READY;
} else {
//LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
if (track->isStopped()) {
track->reset();
}
if (track->isTerminated() || track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
trackToRemove = track;
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
trackToRemove = track;
} else {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
}
}
// remove all the tracks that need to be...
if (UNLIKELY(trackToRemove != 0)) {
mActiveTracks.remove(trackToRemove);
if (!effectChains.isEmpty()) {
LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
trackToRemove->sessionId());
effectChains[0]->decActiveTrackCnt();
}
if (trackToRemove->isTerminated()) {
removeTrack_l(trackToRemove);
}
}
lockEffectChains_l(effectChains);
}
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 && audio_is_linear_pcm(mFormat)) {
memset (mMixBuffer, 0, mFrameCount * mFrameSize);
sleepTime = 0;
}
}
if (mSuspended) {
sleepTime = suspendSleepTimeUs();
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_READY) {
applyVolume(leftVol, rightVol, rampVolume);
}
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
unlockEffectChains(effectChains);
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
mStandby = false;
} else {
unlockEffectChains(effectChains);
usleep(sleepTime);
}
// finally let go of removed track, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
trackToRemove.clear();
activeTrack.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
if (!mStandby) {
mOutput->stream->common.standby(&mOutput->stream->common);
}
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->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
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 (audio_is_linear_pcm(mFormat)) {
time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
} else {
time = 10000;
}
return time;
}
uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
} else {
time = 10000;
}
return time;
}
uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
} else {
time = 10000;
}
return time;
}
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
: MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
{
mType = PlaybackThread::DUPLICATING;
addOutputTrack(mainThread);
}
AudioFlinger::DuplicatingThread::~DuplicatingThread()
{
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
mOutputTracks.clear();
}
bool AudioFlinger::DuplicatingThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
uint32_t mixerStatus = MIXER_IDLE;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount*mFrameSize;
SortedVector< sp<OutputTrack> > outputTracks;
uint32_t writeFrames = 0;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
Vector< sp<EffectChain> > effectChains;
while (!exitPending())
{
processConfigEvents();
mixerStatus = MIXER_IDLE;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
mixBufferSize = mFrameCount*mFrameSize;
updateWaitTime();
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
}
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
outputTracks.add(mOutputTracks[i]);
}
// put audio hardware into standby after short delay
if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
mSuspended) {
if (!mStandby) {
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->stop();
}
mStandby = true;
mBytesWritten = 0;
}
if (!activeTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
outputTracks.clear();
if (exitPending()) break;
LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
mWaitWorkCV.wait(mLock);
LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + kStandbyTimeInNsecs;
sleepTime = idleSleepTime;
continue;
}
}
mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove);
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
}
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
if (outputsReady(outputTracks)) {
mAudioMixer->process();
} else {
memset(mMixBuffer, 0, mixBufferSize);
}
sleepTime = 0;
writeFrames = mFrameCount;
} else {
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_ENABLED) {
sleepTime = activeSleepTime;
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0) {
// flush remaining overflow buffers in output tracks
for (size_t i = 0; i < outputTracks.size(); i++) {
if (outputTracks[i]->isActive()) {
sleepTime = 0;
writeFrames = 0;
memset(mMixBuffer, 0, mixBufferSize);
break;
}
}
}
}
if (mSuspended) {
sleepTime = suspendSleepTimeUs();
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
// enable changes in effect chain
unlockEffectChains(effectChains);
standbyTime = systemTime() + kStandbyTimeInNsecs;
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->write(mMixBuffer, writeFrames);
}
mStandby = false;
mBytesWritten += mixBufferSize;
} else {
// enable changes in effect chain
unlockEffectChains(effectChains);
usleep(sleepTime);
}
// finally let go of all our tracks, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
outputTracks.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
return false;
}
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
this,
mSampleRate,
mFormat,
mChannelMask,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_CNT, 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,
uint32_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(0),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK),
mSessionId(sessionId)
{
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
uint8_t channelCount = popcount(channelMask);
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;
mChannelCount = channelCount;
mChannelMask = channelMask;
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 (other flags are cleared)
mCblk->flags = CBLK_UNDERRUN_ON;
} else {
mBuffer = sharedBuffer->pointer();
}
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
} else {
LOGE("not enough memory for AudioTrack size=%u", size);
client->heap()->dump("AudioTrack");
return;
}
} else {
mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
if (mCblk) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mChannelCount = channelCount;
mChannelMask = channelMask;
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 (other flags are cleared)
mCblk->flags = CBLK_UNDERRUN_ON;
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
}
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
if (mCblk) {
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
if (mClient == NULL) {
delete mCblk;
}
}
mCblkMemory.clear(); // and free the shared memory
if (mClient != NULL) {
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
}
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
}
bool AudioFlinger::ThreadBase::TrackBase::step() {
bool result;
audio_track_cblk_t* cblk = this->cblk();
result = cblk->stepServer(mFrameCount);
if (!result) {
LOGV("stepServer failed acquiring cblk mutex");
mFlags |= STEPSERVER_FAILED;
}
return result;
}
void AudioFlinger::ThreadBase::TrackBase::reset() {
audio_track_cblk_t* cblk = this->cblk();
cblk->user = 0;
cblk->server = 0;
cblk->userBase = 0;
cblk->serverBase = 0;
mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
LOGV("TrackBase::reset");
}
sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
{
return mCblkMemory;
}
int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
return (const int)mChannelCount;
}
uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
return mChannelMask;
}<