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f9cc65cfda
M7350/base/services/audioflinger/AudioFlinger.cpp
T f9cc65cfda M7350v1_en_gpl
2024-09-09 08:52:07 +00:00

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/* //device/include/server/AudioFlinger/AudioFlinger.cpp
**
** Copyright 2007, The Android Open Source Project
** Copyright (c) 2010-2011, The Linux Foundation. All rights reserved.
**
** 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
#define LOG_NDDEBUG 0
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <cutils/properties.h>
#include <media/AudioTrack.h>
#include <media/AudioRecord.h>
#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>
#include <hardware_legacy/AudioHardwareInterface.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#ifdef WITH_A2DP
#include "A2dpAudioInterface.h"
#endif
#ifdef SRS_PROCESSING
#include "srs_processing.h"
#endif
#ifdef LVMX
#include "lifevibes.h"
#endif
#include <media/EffectsFactoryApi.h>
#include <media/EffectVisualizerApi.h>
// ----------------------------------------------------------------------------
// the sim build doesn't have gettid
#ifndef HAVE_GETTID
# define gettid getpid
#endif
// ----------------------------------------------------------------------------
extern const char * const gEffectLibPath;
namespace android {
static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
static const char* kHardwareLockedString = "Hardware lock is taken\n";
//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const float MAX_GAIN = 4096.0f;
static const float MAX_GAIN_INT = 0x1000;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
static const int8_t kMaxTrackRetries = 50;
static const int8_t kMaxTrackStartupRetries = 50;
// allow less retry attempts on direct output thread.
// direct outputs can be a scarce resource in audio hardware and should
// be released as quickly as possible.
static const int8_t kMaxTrackRetriesDirect = 2;
static const int kDumpLockRetries = 50;
static const int kDumpLockSleep = 20000;
static const nsecs_t kWarningThrottle = seconds(5);
#define AUDIOFLINGER_SECURITY_ENABLED 1
// ----------------------------------------------------------------------------
static bool recordingAllowed() {
#ifndef HAVE_ANDROID_OS
return true;
#endif
#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
return ok;
#else
if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
return true;
#endif
}
static bool settingsAllowed() {
#ifndef HAVE_ANDROID_OS
return true;
#endif
#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
return ok;
#else
if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
return true;
#endif
}
static uint32_t getInputChannelCount(uint32_t channels) {
// only mono or stereo is supported for input sources
return AudioSystem::popCount((channels) & (AudioSystem::CHANNEL_IN_STEREO | AudioSystem::CHANNEL_IN_MONO));
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), mFmOn(false)
{
mHardwareStatus = AUDIO_HW_IDLE;
mLPAOutput = NULL;
mLPAHandle = -1;
mLPAStreamIsActive = false;
mLPASessionId = -2; // -2 is invalid session ID
mIsEffectConfigChanged = false;
mLPAEffectChain = NULL;
mAudioHardware = AudioHardwareInterface::create();
mHardwareStatus = AUDIO_HW_INIT;
if (mAudioHardware->initCheck() == NO_ERROR) {
// open 16-bit output stream for s/w mixer
mMode = AudioSystem::MODE_NORMAL;
setMode(mMode);
setMasterVolume(1.0f);
setMasterMute(false);
} else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
#ifdef LVMX
LifeVibes::init();
mLifeVibesClientPid = -1;
#endif
}
AudioFlinger::~AudioFlinger()
{
while (!mRecordThreads.isEmpty()) {
// closeInput() will remove first entry from mRecordThreads
closeInput(mRecordThreads.keyAt(0));
}
while (!mPlaybackThreads.isEmpty()) {
// closeOutput() will remove first entry from mPlaybackThreads
closeOutput(mPlaybackThreads.keyAt(0));
}
if (mLPAOutput) {
// Close the Output
closeSession(mLPAHandle);
}
if (mAudioHardware) {
delete mAudioHardware;
}
}
status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
wp<Client> wClient = mClients.valueAt(i);
if (wClient != 0) {
sp<Client> client = wClient.promote();
if (client != 0) {
snprintf(buffer, SIZE, " pid: %d\n", client->pid());
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
int hardwareStatus = mHardwareStatus;
snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Permission Denial: "
"can't dump AudioFlinger from pid=%d, uid=%d\n",
IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
static bool tryLock(Mutex& mutex)
{
bool locked = false;
for (int i = 0; i < kDumpLockRetries; ++i) {
if (mutex.tryLock() == NO_ERROR) {
locked = true;
break;
}
usleep(kDumpLockSleep);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
bool hardwareLocked = tryLock(mHardwareLock);
if (!hardwareLocked) {
String8 result(kHardwareLockedString);
write(fd, result.string(), result.size());
} else {
mHardwareLock.unlock();
}
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
String8 result(kDeadlockedString);
write(fd, result.string(), result.size());
}
dumpClients(fd, args);
dumpInternals(fd, args);
// dump playback threads
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->dump(fd, args);
}
// dump record threads
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->dump(fd, args);
}
if (mAudioHardware) {
mAudioHardware->dumpState(fd, args);
}
if (locked) mLock.unlock();
}
return NO_ERROR;
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int output,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
int lSessionId;
if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
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 != AudioSystem::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();
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
LOGD("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelCount, 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;
}
void AudioFlinger::createSession(
pid_t pid,
uint32_t sampleRate,
int channelCount,
int *sessionId,
status_t *status)
{
status_t lStatus = NO_ERROR;
{
// createSession can be called from same PID (mediaserver process) only
if(pid != getpid()){
lStatus = BAD_VALUE;
goto Exit;
}
Mutex::Autolock _l(mLock);
LOGV("createSession() sessionId: %d sampleRate %d channelCount %d",
*sessionId, sampleRate, channelCount);
if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
// Check if the session ID is already associated with a track
uint32_t sessions = t->hasAudioSession(*sessionId);
if (sessions & PlaybackThread::TRACK_SESSION) {
LOGE("There is a track already associated with this session %d", *sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
// check if an effect with same session ID is waiting for a ssession to be created
if (sessions & PlaybackThread::EFFECT_SESSION) {
// Clear reference to previous effect chain if any
if(mLPAEffectChain.get()) {
mLPAEffectChain.clear();
}
mLPAEffectChain = t->getEffectChain_l(*sessionId);
}
}
mLPASessionId = *sessionId;
LOGV("createSession() lSessionId: %d", mLPASessionId);
if (mLPAEffectChain != NULL) {
mLPAEffectChain->setLPAFlag(true);
// For LPA, the volume will be applied in DSP. No need for volume
// control in the Effect chain, so setting it to unity.
uint32_t volume = 0x1000000; // Equals to 1.0 in 8.24 format
mLPAEffectChain->setVolume_l(&volume,&volume);
} else {
LOGW("There was no effectChain created for the sessionId(%d)", mLPASessionId);
}
} else {
if(sessionId != NULL) {
LOGE("Error: Invalid sessionID (%d) for LPA playback", *sessionId);
}
}
mLPASampleRate = sampleRate;
mLPANumChannels = channelCount;
}
#ifdef SRS_PROCESSING
LOGD("SRS_Processing - CreateSession - OutNotify_Init: %p TID %d\n", this, gettid());
SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, true);
#endif
Exit:
if(status) {
*status = lStatus;
}
}
void AudioFlinger::deleteSession()
{
Mutex::Autolock _l(mLock);
LOGV("deleteSession");
// -2 is invalid session ID
mLPASessionId = -2;
if (mLPAEffectChain != NULL) {
mLPAEffectChain->setLPAFlag(false);
size_t i, numEffects = mLPAEffectChain->getNumEffects();
for(i = 0; i < numEffects; i++) {
sp<EffectModule> effect = mLPAEffectChain->getEffectFromIndex_l(i);
effect->setInBuffer(mLPAEffectChain->inBuffer());
if (i == numEffects-1) {
effect->setOutBuffer(mLPAEffectChain->outBuffer());
} else {
effect->setOutBuffer(mLPAEffectChain->inBuffer());
}
effect->configure();
}
mLPAEffectChain.clear();
mLPAEffectChain = NULL;
}
#ifdef SRS_PROCESSING
LOGD("SRS_Processing - deleteSession - OutNotify_Init: %p TID %d\n", this, gettid());
SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, false);
#endif
}
// ToDo: Should we go ahead with this frameCount?
#define DEAFULT_FRAME_COUNT 1200
void AudioFlinger::applyEffectsOn(int16_t *inBuffer, int16_t *outBuffer, int size)
{
LOGV("applyEffectsOn: inBuf %p outBuf %p size %d", inBuffer, outBuffer, size);
// This might be the first buffer to apply effects after effect config change
// should not skip effects processing
mIsEffectConfigChanged = false;
volatile size_t numEffects = 0;
if(mLPAEffectChain != NULL) {
numEffects = mLPAEffectChain->getNumEffects();
}
if( numEffects > 0) {
size_t i = 0;
int16_t *pIn = inBuffer;
int16_t *pOut = outBuffer;
int frameCount = size / (sizeof(int16_t) * mLPANumChannels);
while(frameCount > 0) {
if(mLPAEffectChain == NULL) {
LOGV("LPA Effect Chain is removed - No effects processing !!");
numEffects = 0;
break;
}
mLPAEffectChain->lock();
numEffects = mLPAEffectChain->getNumEffects();
if(!numEffects) {
LOGV("applyEffectsOn: All the effects are removed - nothing to process");
mLPAEffectChain->unlock();
break;
}
int outFrameCount = (frameCount > DEAFULT_FRAME_COUNT ? DEAFULT_FRAME_COUNT: frameCount);
bool isEffectEnabled = false;
for(i = 0; i < numEffects; i++) {
// If effect configuration is changed while applying effects do not process further
if(mIsEffectConfigChanged) {
mLPAEffectChain->unlock();
LOGV("applyEffectsOn: mIsEffectConfigChanged is set - no further processing");
return;
}
sp<EffectModule> effect = mLPAEffectChain->getEffectFromIndex_l(i);
if(effect == NULL) {
LOGE("getEffectFromIndex_l(%d) returned NULL ptr", i);
mLPAEffectChain->unlock();
return;
}
if(i == 0) {
// For the first set input and output buffers different
isEffectEnabled = effect->isProcessEnabled();
effect->setInBuffer(pIn);
effect->setOutBuffer(pOut);
} else {
// For the remaining use previous effect's output buffer as input buffer
effect->setInBuffer(pOut);
effect->setOutBuffer(pOut);
}
// true indicates that it is being applied on LPA output
effect->configure(true, mLPASampleRate, mLPANumChannels, outFrameCount);
}
if(isEffectEnabled) {
// Clear the output buffer
memset(pOut, 0, (outFrameCount * mLPANumChannels * sizeof(int16_t)));
} else {
// Copy input buffer content to the output buffer
memcpy(pOut, pIn, (outFrameCount * mLPANumChannels * sizeof(int16_t)));
}
mLPAEffectChain->process_l();
mLPAEffectChain->unlock();
// Update input and output buffer pointers
pIn += (outFrameCount * mLPANumChannels);
pOut += (outFrameCount * mLPANumChannels);
frameCount -= outFrameCount;
}
}
if (!numEffects) {
LOGV("applyEffectsOn: There are no effects to be applied");
if(inBuffer != outBuffer) {
// No effect applied so just copy input buffer to output buffer
memcpy(outBuffer, inBuffer, size);
}
}
#ifdef SRS_PROCESSING
SRS_Processing::ProcessOut(SRS_Processing::AUTO, this, outBuffer, size, mLPASampleRate, mLPANumChannels);
#endif
}
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;
mA2DPHandle = -1;
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;
LOGD("setMode(%d)", 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;
}
{ // scope for the lock
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
ret = mAudioHardware->setMode(mode);
mHardwareStatus = AUDIO_HW_IDLE;
}
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMode(mode);
#ifdef LVMX
LifeVibes::setMode(mode);
#endif
}
return ret;
}
status_t AudioFlinger::setMicMute(bool state)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
status_t ret = mAudioHardware->setMicMute(state);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
bool state = AudioSystem::MODE_INVALID;
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mAudioHardware->getMicMute(&state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
mMasterMute = muted;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterMute(muted);
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
return mMasterVolume;
}
bool AudioFlinger::masterMute() const
{
return mMasterMute;
}
status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
if( (mLPAOutput != NULL) &&
(mLPAStreamType == stream) ) {
mLPAOutput->setVolume(value, value);
}
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;
}
}
if (mLPAStreamIsActive && mLPAOutput && mLPAStreamType == stream) {
return true;
}
if (stream == AudioSystem::FM) {
String8 key ("Fm-radio");
AudioParameter result(mAudioHardware->getParameters(key));
int value;
if(result.getInt(String8("isFMON"),value) == NO_ERROR){
return true;
}
}
return false;
}
status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
{
status_t result;
LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
#ifdef LVMX
AudioParameter param = AudioParameter(keyValuePairs);
LifeVibes::setParameters(ioHandle,keyValuePairs);
String8 key = String8(AudioParameter::keyRouting);
int device;
if (NO_ERROR != param.getInt(key, device)) {
device = -1;
}
key = String8(LifevibesTag);
String8 value;
int musicEnabled = -1;
if (NO_ERROR == param.get(key, value)) {
if (value == LifevibesEnable) {
mLifeVibesClientPid = IPCThreadState::self()->getCallingPid();
musicEnabled = 1;
} else if (value == LifevibesDisable) {
mLifeVibesClientPid = -1;
musicEnabled = 0;
}
}
#endif
AudioParameter param = AudioParameter(keyValuePairs);
String8 key = String8(AudioParameter::keyRouting);
int device;
if (param.getInt(key, device) == NO_ERROR) {
#ifdef SRS_PROCESSING
if (mLPAOutput && mLPAStreamIsActive) {
LOGV("setParameters:: routing change to device %d", device);
SRS_Processing::ProcessOutRoute(SRS_Processing::AUTO, this, device);
if(ioHandle > 0) {
audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
}
}
#endif
if((device & AudioSystem::DEVICE_OUT_FM) && mFmOn == false){
mFmOn=true;
} else if (mFmOn == true && !(device & AudioSystem::DEVICE_OUT_FM)){
mFmOn=false;
}
}
// ioHandle == 0 means the parameters are global to the audio hardware interface
if (ioHandle == 0) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
#ifdef SRS_PROCESSING
bool status = SRS_Processing::ParamsSet(SRS_Processing::AUTO, keyValuePairs);
if(status && mLPAOutput && mLPAStreamIsActive) {
LOGV("setParameters:: Notifying EFFECT_CONFIG_CHANGED");
audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
}
#endif
result = mAudioHardware->setParameters(keyValuePairs);
#ifdef LVMX
if (musicEnabled != -1) {
LifeVibes::enableMusic((bool) musicEnabled);
}
#endif
mHardwareStatus = AUDIO_HW_IDLE;
return result;
}
// Ensure that the routing to LPA is invoked only when the LPA stream is
// active. Otherwise if there is a input routing request and if there is a
// Valid LPA handle, routing gets applied for the output descriptor rather
// than to the input descriptor.
if ( mLPAOutput && mLPAStreamIsActive && mLPAHandle == ioHandle ) {
if(mLPAEffectChain != NULL) {
AudioParameter param = AudioParameter(keyValuePairs);
String8 key = String8(AudioParameter::keyRouting);
int device;
if (param.getInt(key, device) == NO_ERROR) {
// forward device change to effects that have requested to be
// aware of attached audio device.
LOGD("mLPAEffectChain->setDevice_l(device)");
mLPAEffectChain->setDevice_l(device);
}
}
result = mLPAOutput->setParameters(keyValuePairs);
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) {
#ifdef SRS_PROCESSING
String8 srs_params = SRS_Processing::ParamsGet(SRS_Processing::AUTO, keys);
if (srs_params != "") srs_params += ";";
srs_params += mAudioHardware->getParameters(keys);
return srs_params;
#else
return mAudioHardware->getParameters(keys);
#endif
}
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;
}
status_t AudioFlinger::setFmVolume(float value)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_FM_VOLUME;
status_t ret = mAudioHardware->setFmVolume(value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{
Mutex::Autolock _l(mLock);
sp<IBinder> binder = client->asBinder();
if (mNotificationClients.indexOfKey(binder) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
binder);
LOGV("registerClient() client %p, binder %p", notificationClient.get(), binder.get());
mNotificationClients.add(binder, notificationClient);
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);
}
}
// Send the notification to the client only once.
if (mA2DPHandle != -1) {
LOGV("A2DP active. Notifying the registered client");
client->ioConfigChanged(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle, NULL);
}
}
status_t AudioFlinger::deregisterClient(const sp<IAudioFlingerClient>& client)
{
LOGV("deregisterClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
sp<IBinder> binder = client->asBinder();
int index = mNotificationClients.indexOfKey(binder);
if (index >= 0) {
mNotificationClients.removeItemsAt(index);
return true;
}
return false;
}
void AudioFlinger::removeNotificationClient(sp<IBinder> binder)
{
Mutex::Autolock _l(mLock);
int index = mNotificationClients.indexOfKey(binder);
if (index >= 0) {
sp <NotificationClient> client = mNotificationClients.valueFor(binder);
LOGV("removeNotificationClient() %p, binder %p", client.get(), binder.get());
#ifdef LVMX
if (pid == mLifeVibesClientPid) {
LOGV("Disabling lifevibes");
LifeVibes::enableMusic(false);
mLifeVibesClientPid = -1;
}
#endif
mNotificationClients.removeItem(binder);
}
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
{
LOGV("AudioFlinger::audioConfigChanged_l: event %d", event);
if (event == AudioSystem::EFFECT_CONFIG_CHANGED) {
mIsEffectConfigChanged = true;
}
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
}
}
// removeClient_l() must be called with AudioFlinger::mLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// ----------------------------------------------------------------------------
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id)
: Thread(false),
mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false)
{
}
AudioFlinger::ThreadBase::~ThreadBase()
{
mParamCond.broadcast();
mNewParameters.clear();
}
void AudioFlinger::ThreadBase::exit()
{
// keep a strong ref on ourself so that we wont get
// destroyed in the middle of requestExitAndWait()
sp <ThreadBase> strongMe = this;
LOGV("ThreadBase::exit");
{
AutoMutex lock(&mLock);
mExiting = true;
requestExit();
mWaitWorkCV.signal();
}
requestExitAndWait();
}
uint32_t AudioFlinger::ThreadBase::sampleRate() const
{
return mSampleRate;
}
int AudioFlinger::ThreadBase::channelCount() const
{
return (int)mChannelCount;
}
int AudioFlinger::ThreadBase::format() const
{
return mFormat;
}
size_t AudioFlinger::ThreadBase::frameCount() const
{
return mFrameCount;
}
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
status_t status;
LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
Mutex::Autolock _l(mLock);
mNewParameters.add(keyValuePairs);
mWaitWorkCV.signal();
// wait condition with timeout in case the thread loop has exited
// before the request could be processed
if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) {
status = mParamStatus;
mWaitWorkCV.signal();
} else {
status = TIMED_OUT;
}
return status;
}
void AudioFlinger::ThreadBase::effectConfigChanged() {
mAudioFlinger->mLock.lock();
LOGV("New effect is being added to LPA chain, Notifying LPA Player");
mAudioFlinger->audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
mAudioFlinger->mLock.unlock();
}
void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
{
Mutex::Autolock _l(mLock);
sendConfigEvent_l(event, param);
}
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
ConfigEvent *configEvent = new ConfigEvent();
configEvent->mEvent = event;
configEvent->mParam = param;
mConfigEvents.add(configEvent);
LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
}
void AudioFlinger::ThreadBase::processConfigEvents()
{
mLock.lock();
while(!mConfigEvents.isEmpty()) {
LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
ConfigEvent *configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
mAudioFlinger->mLock.lock();
audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
mAudioFlinger->mLock.unlock();
delete configEvent;
mLock.lock();
}
mLock.unlock();
}
status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
bool locked = tryLock(mLock);
if (!locked) {
snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
write(fd, buffer, strlen(buffer));
}
snprintf(buffer, SIZE, "standby: %d\n", mStandby);
result.append(buffer);
snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
result.append(buffer);
snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
result.append(buffer);
snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
result.append(buffer);
result.append(" Index Command");
for (size_t i = 0; i < mNewParameters.size(); ++i) {
snprintf(buffer, SIZE, "\n %02d ", i);
result.append(buffer);
result.append(mNewParameters[i]);
}
snprintf(buffer, SIZE, "\n\nPending config events: \n");
result.append(buffer);
snprintf(buffer, SIZE, " Index event param\n");
result.append(buffer);
for (size_t i = 0; i < mConfigEvents.size(); i++) {
snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
result.append(buffer);
}
result.append("\n");
write(fd, result.string(), result.size());
if (locked) {
mLock.unlock();
}
return NO_ERROR;
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: ThreadBase(audioFlinger, id),
mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
mDevice(device)
{
readOutputParameters();
mMasterVolume = mAudioFlinger->masterVolume();
mMasterMute = mAudioFlinger->masterMute();
for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
}
}
AudioFlinger::PlaybackThread::~PlaybackThread()
{
delete [] mMixBuffer;
}
status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
dumpEffectChains(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
wp<Track> wTrack = mActiveTracks[i];
if (wTrack != 0) {
sp<Track> track = wTrack.promote();
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mEffectChains.size(); ++i) {
sp<EffectChain> chain = mEffectChains[i];
if (chain != 0) {
chain->dump(fd, args);
}
}
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
result.append(buffer);
snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
result.append(buffer);
snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
result.append(buffer);
snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
result.append(buffer);
snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer);
result.append(buffer);
write(fd, result.string(), result.size());
dumpBase(fd, args);
return NO_ERROR;
}
// Thread virtuals
status_t AudioFlinger::PlaybackThread::readyToRun()
{
if (mSampleRate == 0) {
LOGE("No working audio driver found.");
return NO_INIT;
}
LOGI("AudioFlinger's thread %p ready to run", this);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::onFirstRef()
{
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "Playback Thread %p", this);
run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
}
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
status_t *status)
{
sp<Track> track;
status_t lStatus;
LOGV("PlaybackThread::createTrack_l() sessionId %d mType %d", sessionId, mType);
if (mType == DIRECT) {
if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) {
LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
sampleRate, format, channelCount, mOutput);
lStatus = BAD_VALUE;
goto Exit;
}
} else {
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (sampleRate > mSampleRate*2) {
LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
}
if (mOutput == 0) {
LOGE("Audio driver not initialized.");
lStatus = NO_INIT;
goto Exit;
}
{ // scope for mLock
Mutex::Autolock _l(mLock);
// 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((AudioSystem::stream_type)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((AudioSystem::stream_type)t->type())) {
lStatus = BAD_VALUE;
goto Exit;
}
}
}
track = new Track(this, client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, sessionId);
if (track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
mTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
LOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
chain->setStrategy(AudioSystem::getStrategyForStream((AudioSystem::stream_type)track->type()));
}
}
lStatus = NO_ERROR;
Exit:
if(status) {
*status = lStatus;
}
return track;
}
uint32_t AudioFlinger::PlaybackThread::latency() const
{
if (mOutput) {
return mOutput->latency();
}
else {
return 0;
}
}
status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setMasterVolume(audioOutputType, value);
}
#endif
mMasterVolume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setMasterMute(audioOutputType, muted);
}
#endif
mMasterMute = muted;
return NO_ERROR;
}
float AudioFlinger::PlaybackThread::masterVolume() const
{
return mMasterVolume;
}
bool AudioFlinger::PlaybackThread::masterMute() const
{
return mMasterMute;
}
status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setStreamVolume(audioOutputType, stream, value);
}
#endif
mStreamTypes[stream].volume = value;
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
{
#ifdef LVMX
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LifeVibes::setStreamMute(audioOutputType, stream, muted);
}
#endif
mStreamTypes[stream].mute = muted;
return NO_ERROR;
}
float AudioFlinger::PlaybackThread::streamVolume(int stream) const
{
return mStreamTypes[stream].volume;
}
bool AudioFlinger::PlaybackThread::streamMute(int stream) const
{
return mStreamTypes[stream].mute;
}
bool AudioFlinger::PlaybackThread::isStreamActive(int stream) const
{
Mutex::Autolock _l(mLock);
size_t count = mActiveTracks.size();
for (size_t i = 0 ; i < count ; ++i) {
sp<Track> t = mActiveTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
if (t->type() == stream)
return true;
}
return false;
}
// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
status_t status = ALREADY_EXISTS;
// set retry count for buffer fill
track->mRetryCount = kMaxTrackStartupRetries;
if (mActiveTracks.indexOf(track) < 0) {
// the track is newly added, make sure it fills up all its
// buffers before playing. This is to ensure the client will
// effectively get the latency it requested.
track->mFillingUpStatus = Track::FS_FILLING;
track->mResetDone = false;
mActiveTracks.add(track);
if (track->mainBuffer() != mMixBuffer) {
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
chain->startTrack();
}
}
status = NO_ERROR;
}
LOGV("mWaitWorkCV.broadcast");
mWaitWorkCV.broadcast();
return status;
}
// destroyTrack_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
{
track->mState = TrackBase::TERMINATED;
if (mActiveTracks.indexOf(track) < 0) {
mTracks.remove(track);
deleteTrackName_l(track->name());
}
}
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
return mOutput->getParameters(keys);
}
// destroyTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
switch (event) {
case AudioSystem::OUTPUT_OPENED:
case AudioSystem::OUTPUT_CONFIG_CHANGED:
desc.channels = mChannels;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = latency();
param2 = &desc;
break;
case AudioSystem::STREAM_CONFIG_CHANGED:
param2 = &param;
case AudioSystem::OUTPUT_CLOSED:
default:
break;
}
if (event != AudioSystem::A2DP_OUTPUT_STATE) {
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
else
{
mAudioFlinger->audioConfigChanged_l(event, param, NULL);
}
}
void AudioFlinger::PlaybackThread::readOutputParameters()
{
mSampleRate = mOutput->sampleRate();
mChannels = mOutput->channels();
mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
mFormat = mOutput->format();
mFrameSize = (uint16_t)mOutput->frameSize();
mFrameCount = mOutput->bufferSize() / mFrameSize;
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
if (mMixBuffer != NULL) delete[] mMixBuffer;
mMixBuffer = new int16_t[mFrameCount * 2];
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
// 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/mOutput->frameSize();
return mOutput->getRenderPosition(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 AudioSystem::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 == AudioSystem::SESSION_OUTPUT_MIX) {
return AudioSystem::getStrategyForStream(AudioSystem::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((AudioSystem::stream_type) track->type());
}
}
return AudioSystem::getStrategyForStream(AudioSystem::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;
#ifdef SRS_PROCESSING
LOGD("SRS_Processing - MixerThread - OutNotify_Init: %p TID %d\n", this, gettid());
SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, true);
#endif
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);
LOGV("MixerThread:: calling lockEffectChains_l()");
// 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)) {
LOGV("MixerThread:: calling mAudioMixer->process()");
// 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 ++) {
LOGV("MixerThread:: calling effectChains[%d]->process_l", i);
effectChains[i]->process_l();
}
LOGV("MixerThread:: calling unlockEffectChains()");
// enable changes in effect chain
unlockEffectChains(effectChains);
#ifdef SRS_PROCESSING
if (mFormat == AudioSystem::PCM_16_BIT) {
SRS_Processing::ProcessOut(SRS_Processing::AUTO, this, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount);
}
#endif
#ifdef LVMX
LOGV("MixerThread:: calling LifeVibes::getMixerType()");
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType)) {
LOGV("MixerThread:: calling LifeVibes::process()");
LifeVibes::process(audioOutputType, mMixBuffer, mixBufferSize);
}
#endif
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
nsecs_t now = systemTime();
nsecs_t delta = now - mLastWriteTime;
if (delta > maxPeriod) {
mNumDelayedWrites++;
if ((now - lastWarning) > kWarningThrottle) {
LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
}
if (mStandby) {
longStandbyExit = true;
}
}
mStandby = false;
} else {
// enable changes in effect chain
unlockEffectChains(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->standby();
}
#ifdef SRS_PROCESSING
LOGD("SRS_Processing - MixerThread - OutNotify_Exit: %p TID %d\n", this, gettid());
SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, false);
#endif
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;
}
#ifdef LVMX
bool tracksConnectedChanged = false;
bool stateChanged = false;
int audioOutputType = LifeVibes::getMixerType(mId, mType);
if (LifeVibes::audioOutputTypeIsLifeVibes(audioOutputType))
{
int activeTypes = 0;
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
int iTracktype=track->type();
activeTypes |= 1<<track->type();
}
LifeVibes::computeVolumes(audioOutputType, activeTypes, tracksConnectedChanged, stateChanged, masterVolume, masterMute);
}
#endif
// Delegate master volume control to effect in output mix effect chain if needed
sp<EffectChain> chain = getEffectChain_l(AudioSystem::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;
}
} 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;
#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;
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);
#ifdef LVMX
if ( tracksConnectedChanged || stateChanged )
{
// only do the ramp when the volume is changed by the user / application
param = AudioMixer::VOLUME;
}
#endif
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(track);
mAudioMixer->enable(AudioMixer::MIXING);
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::CHANNEL_COUNT, (void *)track->channelCount());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(cblk->sampleRate));
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
// reset retry count
track->mRetryCount = kMaxTrackRetries;
mixerStatus = MIXER_TRACKS_READY;
} else {
//LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
if (track->isStopped()) {
track->reset();
}
if (track->isTerminated() || track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
tracksToRemove->add(track);
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun
cblk->flags |= CBLK_DISABLED_ON;
} else if (mixerStatus != MIXER_TRACKS_READY) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
mAudioMixer->disable(AudioMixer::MIXING);
}
}
// remove all the tracks that need to be...
count = tracksToRemove->size();
if (UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove->itemAt(i);
mActiveTracks.remove(track);
if (track->mainBuffer() != mMixBuffer) {
chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
chain->stopTrack();
}
}
if (track->isTerminated()) {
mTracks.remove(track);
deleteTrackName_l(track->mName);
}
}
}
// mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to
// mix buffer and track effects will accumulate into it
if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
}
return mixerStatus;
}
void AudioFlinger::MixerThread::invalidateTracks(int streamType)
{
LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
this, streamType, mTracks.size());
Mutex::Autolock _l(mLock);
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
if (t->type() == streamType) {
t->mCblk->lock.lock();
t->mCblk->flags |= CBLK_INVALID_ON;
t->mCblk->cv.signal();
t->mCblk->lock.unlock();
}
}
}
// getTrackName_l() must be called with ThreadBase::mLock held
int AudioFlinger::MixerThread::getTrackName_l()
{
return mAudioMixer->getTrackName();
}
// deleteTrackName_l() must be called with ThreadBase::mLock held
void AudioFlinger::MixerThread::deleteTrackName_l(int name)
{
LOGD("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;
#ifdef SRS_PROCESSING
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR){
SRS_Processing::ProcessOutRoute(SRS_Processing::AUTO, this, value);
}
#endif
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
if (value != AudioSystem::PCM_16_BIT) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
if (value != AudioSystem::CHANNEL_OUT_STEREO) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
// forward device change to effects that have requested to be
// aware of attached audio device.
mDevice = (uint32_t)value;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevice_l(mDevice);
}
}
if (status == NO_ERROR) {
status = mOutput->setParameters(keyValuePair);
if (!mStandby && status == INVALID_OPERATION) {
mOutput->standby();
mStandby = true;
mBytesWritten = 0;
status = mOutput->setParameters(keyValuePair);
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
readOutputParameters();
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
int name = getTrackName_l();
if (name < 0) break;
mTracks[i]->mName = name;
// limit track sample rate to 2 x new output sample rate
if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
}
}
sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
}
}
mNewParameters.removeAt(0);
mParamStatus = status;
mParamCond.signal();
mWaitWorkCV.wait(mLock);
}
return reconfig;
}
status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
PlaybackThread::dumpInternals(fd, args);
snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
uint32_t AudioFlinger::MixerThread::activeSleepTimeUs()
{
return (uint32_t)(mOutput->latency() * 1000) / 2;
}
uint32_t AudioFlinger::MixerThread::idleSleepTimeUs()
{
return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 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 (!AudioSystem::isLinearPCM(mFormat)) {
return;
}
// convert to signed 16 bit before volume calculation
if (mFormat == AudioSystem::PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
uint8_t *src = (uint8_t *)mMixBuffer + count-1;
int16_t *dst = mMixBuffer + count-1;
while(count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
}
size_t frameCount = mFrameCount;
int16_t *out = mMixBuffer;
if (ramp) {
if (mChannelCount == 1) {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out++;
vl += vlInc;
} while (--frameCount);
} else {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
int32_t vrInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
int32_t vr = ((int32_t)mRightVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
out += 2;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
}
} else {
if (mChannelCount == 1) {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out++;
} while (--frameCount);
} else {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out[1] = clamp16(mul(out[1], rightVol) >> 12);
out += 2;
} while (--frameCount);
}
}
// convert back to unsigned 8 bit after volume calculation
if (mFormat == AudioSystem::PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
int16_t *src = mMixBuffer;
uint8_t *dst = (uint8_t *)mMixBuffer;
while(count--) {
*dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
}
}
mLeftVolShort = leftVol;
mRightVolShort = rightVol;
}
bool AudioFlinger::DirectOutputThread::threadLoop()
{
uint32_t mixerStatus = MIXER_IDLE;
sp<Track> trackToRemove;
sp<Track> activeTrack;
nsecs_t standbyTime = systemTime();
int8_t *curBuf;
size_t mixBufferSize = mFrameCount*mFrameSize;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
// use shorter standby delay as on normal output to release
// hardware resources as soon as possible
nsecs_t standbyDelay = microseconds(activeSleepTime*2);
while (!exitPending())
{
bool rampVolume;
uint16_t leftVol;
uint16_t rightVol;
Vector< sp<EffectChain> > effectChains;
processConfigEvents();
mixerStatus = MIXER_IDLE;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
if (checkForNewParameters_l()) {
mixBufferSize = mFrameCount*mFrameSize;
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
standbyDelay = microseconds(activeSleepTime*2);
}
// put audio hardware into standby after short delay
if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
mSuspended) {
// wait until we have something to do...
if (!mStandby) {
LOGV("Audio hardware entering standby, mixer %p\n", this);
mOutput->standby();
mStandby = true;
mBytesWritten = 0;
}
if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) break;
LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
mWaitWorkCV.wait(mLock);
LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + standbyDelay;
sleepTime = idleSleepTime;
continue;
}
}
effectChains = mEffectChains;
// find out which tracks need to be processed
if (mActiveTracks.size() != 0) {
sp<Track> t = mActiveTracks[0].promote();
if (t == 0) continue;
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
if (cblk->framesReady() && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
//LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
mLeftVolFloat = mRightVolFloat = 0;
mLeftVolShort = mRightVolShort = 0;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
rampVolume = true;
}
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
rampVolume = true;
}
// compute volume for this track
float left, right;
if (track->isMuted() || mMasterMute || track->isPausing() ||
mStreamTypes[track->type()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
float typeVolume = mStreamTypes[track->type()].volume;
float v = mMasterVolume * typeVolume;
float v_clamped = v * cblk->volume[0];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = v_clamped/MAX_GAIN;
v_clamped = v * cblk->volume[1];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = v_clamped/MAX_GAIN;
}
if (left != mLeftVolFloat || right != mRightVolFloat) {
mLeftVolFloat = left;
mRightVolFloat = right;
// If audio HAL implements volume control,
// force software volume to nominal value
if (mOutput->setVolume(left, right) == NO_ERROR) {
left = 1.0f;
right = 1.0f;
}
// Convert volumes from float to 8.24
uint32_t vl = (uint32_t)(left * (1 << 24));
uint32_t vr = (uint32_t)(right * (1 << 24));
// Delegate volume control to effect in track effect chain if needed
// only one effect chain can be present on DirectOutputThread, so if
// there is one, the track is connected to it
if (!effectChains.isEmpty()) {
// Do not ramp volume 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]->stopTrack();
}
if (trackToRemove->isTerminated()) {
mTracks.remove(trackToRemove);
deleteTrackName_l(trackToRemove->mName);
}
}
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 && AudioSystem::isLinearPCM(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->write(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->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) / 2;
} else {
time = 10000;
}
return time;
}
uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
{
uint32_t time;
if (AudioSystem::isLinearPCM(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
} else {
time = 10000;
}
return time;
}
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
: MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
{
mType = PlaybackThread::DUPLICATING;
addOutputTrack(mainThread);
}
AudioFlinger::DuplicatingThread::~DuplicatingThread()
{
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
mOutputTracks.clear();
}
bool AudioFlinger::DuplicatingThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
uint32_t mixerStatus = MIXER_IDLE;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount*mFrameSize;
SortedVector< sp<OutputTrack> > outputTracks;
uint32_t writeFrames = 0;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
Vector< sp<EffectChain> > effectChains;
#ifdef SRS_PROCESSING
LOGD("SRS_Processing - DuplicatingThread - OutNotify_Init: %p TID %d\n", this, gettid());
SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, true);
#endif
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);
#ifdef SRS_PROCESSING
if (mFormat == AudioSystem::PCM_16_BIT) {
SRS_Processing::ProcessOut(SRS_Processing::AUTO, this, mMixBuffer, mixBufferSize, mSampleRate, mChannelCount);
}
#endif
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();
}
#ifdef SRS_PROCESSING
LOGD("SRS_Processing - DuplicatingThread - OutNotify_Exit: %p TID %d\n", this, gettid());
SRS_Processing::ProcessOutNotify(SRS_Processing::AUTO, this, false);
#endif
return false;
}
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
this,
mSampleRate,
mFormat,
mChannelCount,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
mOutputTracks.add(outputTrack);
LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
updateWaitTime();
}
}
void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mOutputTracks.size(); i++) {
if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
mOutputTracks[i]->destroy();
mOutputTracks.removeAt(i);
updateWaitTime();
return;
}
}
LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
}
void AudioFlinger::DuplicatingThread::updateWaitTime()
{
mWaitTimeMs = UINT_MAX;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
if (strong != NULL) {
uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
if (waitTimeMs < mWaitTimeMs) {
mWaitTimeMs = waitTimeMs;
}
}
}
}
bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks)
{
for (size_t i = 0; i < outputTracks.size(); i++) {
sp <ThreadBase> thread = outputTracks[i]->thread().promote();
if (thread == 0) {
LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
return false;
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->standby() && !playbackThread->isSuspended()) {
LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
return false;
}
}
return true;
}
uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs()
{
return (mWaitTimeMs * 1000) / 2;
}
// ----------------------------------------------------------------------------
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(0),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK),
mSessionId(sessionId)
{
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = 0;
if ( (format == AudioSystem::PCM_16_BIT) ||
(format == AudioSystem::PCM_8_BIT))
{
bufferSize = frameCount*channelCount*sizeof(int16_t);
}
else if (format == AudioSystem::AMR_NB)
{
bufferSize = frameCount*channelCount*32; // full rate frame size
}
else if (format == AudioSystem::EVRC)
{
bufferSize = frameCount*channelCount*23; // full rate frame size
}
else if (format == AudioSystem::QCELP)
{
bufferSize = frameCount*channelCount*35; // full rate frame size
}
else if (format == AudioSystem::AAC)
{
bufferSize = frameCount*2048; // full rate frame size
}
if (sharedBuffer == 0) {
size += bufferSize;
}
if (client != NULL) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
if (mCblk) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mCblk->channelCount = (uint8_t)channelCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
// Change for Codec type
if ( (format == AudioSystem::PCM_16_BIT) ||
(format == AudioSystem::PCM_8_BIT))
{
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
}
else if (format == AudioSystem::AMR_NB)
{
memset(mBuffer, 0, frameCount*channelCount*32); // full rate frame size
}
else if (format == AudioSystem::EVRC)
{
memset(mBuffer, 0, frameCount*channelCount*23); // full rate frame size
}
else if (format == AudioSystem::QCELP)
{
memset(mBuffer, 0, frameCount*channelCount*35); // full rate frame size
}
else if (format == AudioSystem::AAC)
{
memset(mBuffer, 0, frameCount*2048); // full rate frame size
}
// 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;
mCblk->channelCount = (uint8_t)channelCount;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer (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 (int)mCblk->channelCount;
}
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
(cblk->channelCount == 2 && ((unsigned long)bufferStart & 3))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d, channelCount %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);
return 0;
}
return bufferStart;
}
// ----------------------------------------------------------------------------
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
const wp<ThreadBase>& thread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
: TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
if (mCblk != NULL) {
sp<ThreadBase> baseThread = thread.promote();
if (baseThread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
mName = playbackThread->getTrackName_l();
mMainBuffer = playbackThread->mixBuffer();
}
LOGD("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,
mSessionId);
}
AudioSystem::releaseOutput(thread->id());
}
Mutex::Autolock _l(thread->mLock);
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->destroyTrack_l(this);
}
}
}
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
mFormat,
mCblk->channelCount,
mSessionId,
mFrameCount,
mState,
mMute,
mFillingUpStatus,
mCblk->sampleRate,
mCblk->volume[0],
mCblk->volume[1],
mCblk->server,
mCblk->user,
(int)mMainBuffer,
(int)mAuxBuffer);
}
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
uint32_t framesReq = buffer->frameCount;
LOGV("PlaybackThread::Track::getNextBuffer() mName %d framesReq %d", mName, framesReq);
// 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 || isStopped() || isPausing()) return true;
if (mCblk->framesReady() >= mCblk->frameCount ||
(mCblk->flags & CBLK_FORCEREADY_MSK)) {
mFillingUpStatus = FS_FILLED;
mCblk->flags &= ~CBLK_FORCEREADY_MSK;
return true;
}
return false;
}
status_t AudioFlinger::PlaybackThread::Track::start()
{
status_t status = NO_ERROR;
LOGD("start(%d), calling thread %d session %d",
mName, IPCThreadState::self()->getCallingPid(), mSessionId);
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,
mSessionId);
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()
{
LOGD("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,
mSessionId);
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,
mSessionId);
thread->mLock.lock();
}
}
}
}
void AudioFlinger::PlaybackThread::Track::flush()
{
LOGV("flush(%d)", mName);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
return;
}
// No point remaining in PAUSED state after a flush => go to
// STOPPED state
mState = STOPPED;
mCblk->lock.lock();
// NOTE: reset() will reset cblk->user and cblk->server with
// the risk that at the same time, the AudioMixer is trying to read
// data. In this case, getNextBuffer() would return a NULL pointer
// as audio buffer => the AudioMixer code MUST always test that pointer
// returned by getNextBuffer() is not NULL!
reset();
mCblk->lock.unlock();
}
}
void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
if (!mResetDone) {
TrackBase::reset();
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flags |= CBLK_UNDERRUN_ON;
mCblk->flags &= ~CBLK_FORCEREADY_MSK;
mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
}
void AudioFlinger::PlaybackThread::Track::mute(bool muted)
{
mMute = muted;
}
void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
{
mVolume[0] = left;
mVolume[1] = right;
}
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
{
status_t status = DEAD_OBJECT;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
status = playbackThread->attachAuxEffect(this, EffectId);
}
return status;
}
void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer)
{
mAuxEffectId = EffectId;
mAuxBuffer = buffer;
}
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
channelCount, frameCount, flags, 0, sessionId),
mOverflow(false)
{
if (mCblk != NULL) {
LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
if (format == AudioSystem::AMR_NB) {
mCblk->frameSize = channelCount * 32;
} else if (format == AudioSystem::EVRC) {
mCblk->frameSize = channelCount * 23;
} else if (format == AudioSystem::QCELP) {
mCblk->frameSize = channelCount * 35;
} else if (format == AudioSystem::AAC) {
mCblk->frameSize = 2048;
} else if (format == AudioSystem::PCM_16_BIT) {
mCblk->frameSize = channelCount * sizeof(int16_t);
} else if (format == AudioSystem::PCM_8_BIT) {
mCblk->frameSize = channelCount * sizeof(int8_t);
} else {
mCblk->frameSize = sizeof(int8_t);
}
}
}
AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
AudioSystem::releaseInput(thread->id());
}
}
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
LOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesAvail = cblk->framesAvailable_l();
if (LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
status_t AudioFlinger::RecordThread::RecordTrack::start()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->start(this);
} else {
return BAD_VALUE;
}
}
void AudioFlinger::RecordThread::RecordTrack::stop()
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
recordThread->stop(this);
TrackBase::reset();
// Force overerrun condition to avoid false overrun callback until first data is
// read from buffer
mCblk->flags |= CBLK_UNDERRUN_ON;
}
}
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n",
(mClient == NULL) ? getpid() : mClient->pid(),
mFormat,
mCblk->channelCount,
mSessionId,
mFrameCount,
mState,
mCblk->sampleRate,
mCblk->server,
mCblk->user);
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
const wp<ThreadBase>& thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount)
: Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);
} else {
LOGW("Error creating output track on thread %p", playbackThread);
}
}
AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
{
clearBufferQueue();
}
status_t AudioFlinger::PlaybackThread::OutputTrack::start()
{
status_t status = Track::start();
if (status != NO_ERROR) {
return status;
}
mActive = true;
mRetryCount = 127;
return status;
}
void AudioFlinger::PlaybackThread::OutputTrack::stop()
{
Track::stop();
clearBufferQueue();
mOutBuffer.frameCount = 0;
mActive = false;
}
bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
uint32_t channelCount = mCblk->channelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
start();
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
MixerThread *mixerThread = (MixerThread *)thread.get();
if (mCblk->frameCount > frames){
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
uint32_t startFrames = (mCblk->frameCount - frames);
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
pInBuffer->frameCount = startFrames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else {
LOGW ("OutputTrack::write() %p no more buffers in queue", this);
}
}
}
}
while (waitTimeLeftMs) {
// First write pending buffers, then new data
if (mBufferQueue.size()) {
pInBuffer = mBufferQueue.itemAt(0);
} else {
pInBuffer = &inBuffer;
}
if (pInBuffer->frameCount == 0) {
break;
}
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
outputBufferFull = true;
break;
}
uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
if (waitTimeLeftMs >= waitTimeMs) {
waitTimeLeftMs -= waitTimeMs;
} else {
waitTimeLeftMs = 0;
}
}
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
mCblk->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channelCount;
mOutBuffer.frameCount -= outFrames;
mOutBuffer.i16 += outFrames * channelCount;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
delete [] pInBuffer->mBuffer;
delete pInBuffer;
LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
} else {
break;
}
}
}
// If we could not write all frames, allocate a buffer and queue it for next time.
if (inBuffer.frameCount) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->i16 = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
} else {
LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
}
}
}
// Calling write() with a 0 length buffer, means that no more data will be written:
// If no more buffers are pending, fill output track buffer to make sure it is started
// by output mixer.
if (frames == 0 && mBufferQueue.size() == 0) {
if (mCblk->user < mCblk->frameCount) {
frames = mCblk->frameCount - mCblk->user;
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[frames * channelCount];
pInBuffer->frameCount = frames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else if (mActive) {
stop();
}
}
return outputBufferFull;
}
status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
int active;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = buffer->frameCount;
// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
buffer->frameCount = 0;
uint32_t framesAvail = cblk->framesAvailable();
if (framesAvail == 0) {
Mutex::Autolock _l(cblk->lock);
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
if (UNLIKELY(!active)) {
LOGV("Not active and NO_MORE_BUFFERS");
return AudioTrack::NO_MORE_BUFFERS;
}
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (result != NO_ERROR) {
return AudioTrack::NO_MORE_BUFFERS;
}
// read the server count again
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
}
// if (framesAvail < framesReq) {
// return AudioTrack::NO_MORE_BUFFERS;
// }
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (u + framesReq > bufferEnd) {
framesReq = bufferEnd - u;
}
buffer->frameCount = framesReq;
buffer->raw = (void *)cblk->buffer(u);
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
Buffer *pBuffer;
for (size_t i = 0; i < size; i++) {
pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
delete pBuffer;
}
mBufferQueue.clear();
}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
mPid(pid)
{
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}
// Client destructor must be called with AudioFlinger::mLock held
AudioFlinger::Client::~Client()
{
mAudioFlinger->removeClient_l(mPid);
}
const sp<MemoryDealer>& AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
sp<IBinder> binder)
: mAudioFlinger(audioFlinger), mBinder(binder), mClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
mClient.clear();
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
{
sp<NotificationClient> keep(this);
{
mAudioFlinger->removeNotificationClient(mBinder);
}
}
// ----------------------------------------------------------------------------
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
}
AudioFlinger::TrackHandle::~TrackHandle() {
// just stop the track on deletion, associated resources
// will be freed from the main thread once all pending buffers have
// been played. Unless it's not in the active track list, in which
// case we free everything now...
mTrack->destroy();
}
status_t AudioFlinger::TrackHandle::start() {
return mTrack->start();
}
void AudioFlinger::TrackHandle::stop() {
mTrack->stop();
}
void AudioFlinger::TrackHandle::flush() {
mTrack->flush();
}
void AudioFlinger::TrackHandle::mute(bool e) {
mTrack->mute(e);
}
void AudioFlinger::TrackHandle::pause() {
mTrack->pause();
}
void AudioFlinger::TrackHandle::setVolume(float left, float right) {
mTrack->setVolume(left, right);
}
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
return mTrack->getCblk();
}
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
}
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioTrack::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
int input,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
int *sessionId,
status_t *status)
{
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
wp<Client> wclient;
size_t inputBufferSize = 0;
status_t lStatus;
RecordThread *thread;
size_t inFrameCount;
int lSessionId;
// check calling permissions
if (!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// Check that audio input stream accepts requested audio parameters
inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
if (inputBufferSize == 0) {
lStatus = BAD_VALUE;
LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount);
goto Exit;
}
// add client to list
{ // scope for mLock
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
if (thread == NULL) {
lStatus = BAD_VALUE;
goto Exit;
}
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
// If no audio session id is provided, create one here
if (sessionId != NULL && *sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
lSessionId = *sessionId;
} else {
lSessionId = nextUniqueId();
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
// frameCount must be a multiple of input buffer size
// Change for Codec type
if ((format == AudioSystem::PCM_16_BIT) ||
(format == AudioSystem::PCM_8_BIT))
{
inFrameCount = inputBufferSize/channelCount/sizeof(short);
}
else if (format == AudioSystem::AMR_NB)
{
inFrameCount = inputBufferSize/channelCount/32;
}
else if (format == AudioSystem::EVRC)
{
inFrameCount = inputBufferSize/channelCount/23;
}
else if (format == AudioSystem::QCELP)
{
inFrameCount = inputBufferSize/channelCount/35;
}
else if (format == AudioSystem::AAC)
{
inFrameCount = inputBufferSize/2048;
}
frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
format, channelCount, frameCount, flags, lSessionId);
}
if (recordTrack->getCblk() == NULL) {
// remove local strong reference to Client before deleting the RecordTrack so that the Client
// destructor is called by the TrackBase destructor with mLock held
client.clear();
recordTrack.clear();
lStatus = NO_MEMORY;
goto Exit;
}
// return to handle to client
recordHandle = new RecordHandle(recordTrack);
lStatus = NO_ERROR;
Exit:
if (status) {
*status = lStatus;
}
return recordHandle;
}
// ----------------------------------------------------------------------------
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
}
AudioFlinger::RecordHandle::~RecordHandle() {
stop();
}
status_t AudioFlinger::RecordHandle::start() {
LOGV("RecordHandle::start()");
return mRecordTrack->start();
}
void AudioFlinger::RecordHandle::stop() {
LOGV("RecordHandle::stop()");
mRecordTrack->stop();
}
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
return mRecordTrack->getCblk();
}
status_t AudioFlinger::RecordHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioRecord::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) :
ThreadBase(audioFlinger, id),
mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
{
mReqChannelCount = getInputChannelCount(channels);
mReqSampleRate = sampleRate;
readInputParameters();
}
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
if (mResampler != 0) {
delete mResampler;
delete[] mRsmpOutBuffer;
}
}
void AudioFlinger::RecordThread::onFirstRef()
{
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "Record Thread %p", this);
run(buffer, PRIORITY_URGENT_AUDIO);
}
bool AudioFlinger::RecordThread::threadLoop()
{
AudioBufferProvider::Buffer buffer;
sp<RecordTrack> activeTrack;
nsecs_t lastWarning = 0;
// start recording
while (!exitPending()) {
processConfigEvents();
{ // scope for mLock
Mutex::Autolock _l(mLock);
checkForNewParameters_l();
if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
if (!mStandby) {
mInput->standby();
mStandby = true;
}
if (exitPending()) break;
LOGV("RecordThread: loop stopping");
// go to sleep
mWaitWorkCV.wait(mLock);
LOGV("RecordThread: loop starting");
continue;
}
if (mActiveTrack != 0) {
if (mActiveTrack->mState == TrackBase::PAUSING) {
if (!mStandby) {
mInput->standby();
mStandby = true;
}
mActiveTrack.clear();
mStartStopCond.broadcast();
} else if (mActiveTrack->mState == TrackBase::RESUMING) {
if (mReqChannelCount != mActiveTrack->channelCount()) {
mActiveTrack.clear();
mStartStopCond.broadcast();
} else if (mBytesRead != 0) {
// record start succeeds only if first read from audio input
// succeeds
if (mBytesRead > 0) {
mActiveTrack->mState = TrackBase::ACTIVE;
} else {
mActiveTrack.clear();
}
mStartStopCond.broadcast();
}
mStandby = false;
}
}
}
if (mActiveTrack != 0) {
if (mActiveTrack->mState != TrackBase::ACTIVE &&
mActiveTrack->mState != TrackBase::RESUMING) {
usleep(5000);
continue;
}
buffer.frameCount = mFrameCount;
if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
if (mResampler == 0) {
// no resampling
while (framesOut) {
size_t framesIn = mFrameCount - mRsmpInIndex;
if (framesIn) {
int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
if (framesIn > framesOut)
framesIn = framesOut;
mRsmpInIndex += framesIn;
framesOut -= framesIn;
if ((int)mChannelCount == mReqChannelCount ||
mFormat != AudioSystem::PCM_16_BIT) {
memcpy(dst, src, framesIn * mFrameSize);
} else {
int16_t *src16 = (int16_t *)src;
int16_t *dst16 = (int16_t *)dst;
if (mChannelCount == 1) {
while (framesIn--) {
*dst16++ = *src16;
*dst16++ = *src16++;
}
} else {
while (framesIn--) {
*dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
src16 += 2;
}
}
}
}
if (framesOut && mFrameCount == mRsmpInIndex) {
if (((int) framesOut != mFrameCount) &&
(mFormat != AudioSystem::PCM_16_BIT) ) {
mBytesRead = mInput->read(buffer.raw, buffer.frameCount * mFrameSize);
if(mBytesRead >= 0 ){
buffer.frameCount = mBytesRead/mFrameSize;
}
framesOut = 0;
} else
if (framesOut == mFrameCount &&
((int)mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
mBytesRead = mInput->read(buffer.raw, mInputBytes);
if( mBytesRead >= 0 ){
buffer.frameCount = mBytesRead/mFrameSize;
}
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()) {
nsecs_t now = systemTime();
if ((now - lastWarning) > kWarningThrottle) {
LOGW("RecordThread: buffer overflow");
lastWarning = now;
}
}
// 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)
{
LOGD("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) {
LOGD("RecordThread::stop");
sp <ThreadBase> strongMe = this;
{
AutoMutex lock(&mLock);
if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
mActiveTrack->mState = TrackBase::PAUSING;
// do not wait for mStartStopCond if exiting
if (mExiting) {
return;
}
mStartStopCond.wait(mLock);
// if we have been restarted, recordTrack == mActiveTrack.get() here
if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) {
mLock.unlock();
AudioSystem::stopInput(mId);
mLock.lock();
LOGV("Record stopped OK");
}
}
}
}
status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
pid_t pid = 0;
snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
result.append(buffer);
if (mActiveTrack != 0) {
result.append("Active Track:\n");
result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n");
mActiveTrack->dump(buffer, SIZE);
result.append(buffer);
snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
result.append(buffer);
snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
result.append(buffer);
snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
result.append(buffer);
} else {
result.append("No record client\n");
}
write(fd, result.string(), result.size());
dumpBase(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
size_t framesReq = buffer->frameCount;
size_t framesReady = mFrameCount - mRsmpInIndex;
int channelCount;
if (framesReady == 0) {
mBytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
if (mBytesRead < 0) {
LOGE("RecordThread::getNextBuffer() Error reading audio input");
if (mActiveTrack->mState == TrackBase::ACTIVE) {
// Force input into standby so that it tries to
// recover at next read attempt
mInput->standby();
usleep(5000);
}
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
mRsmpInIndex = 0;
framesReady = mFrameCount;
}
if (framesReq > framesReady) {
framesReq = framesReady;
}
if (mChannelCount == 1 && mReqChannelCount == 2) {
channelCount = 1;
} else {
channelCount = 2;
}
buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
buffer->frameCount = framesReq;
return NO_ERROR;
}
void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
mRsmpInIndex += buffer->frameCount;
buffer->frameCount = 0;
}
bool AudioFlinger::RecordThread::checkForNewParameters_l()
{
bool reconfig = false;
while (!mNewParameters.isEmpty()) {
status_t status = NO_ERROR;
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
int reqFormat = mFormat;
int reqSamplingRate = mReqSampleRate;
int reqChannelCount = mReqChannelCount;
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
reqSamplingRate = value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
reqFormat = value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
reqChannelCount = getInputChannelCount(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) &&
(getInputChannelCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
readInputParameters();
sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
}
}
}
mNewParameters.removeAt(0);
mParamStatus = status;
mParamCond.signal();
mWaitWorkCV.wait(mLock);
}
return reconfig;
}
String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
{
return mInput->getParameters(keys);
}
void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
switch (event) {
case AudioSystem::INPUT_OPENED:
case AudioSystem::INPUT_CONFIG_CHANGED:
desc.channels = mChannels;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = 0;
param2 = &desc;
break;
case AudioSystem::INPUT_CLOSED:
default:
break;
}
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
void AudioFlinger::RecordThread::readInputParameters()
{
if (mRsmpInBuffer) delete mRsmpInBuffer;
if (mRsmpOutBuffer) delete mRsmpOutBuffer;
if (mResampler) delete mResampler;
mResampler = 0;
mSampleRate = mInput->sampleRate();
mChannels = mInput->channels();
mChannelCount = getInputChannelCount(mInput->channels());
mFormat = mInput->format();
mFrameSize = (uint16_t)mInput->frameSize();
mInputBytes = mInput->bufferSize();
mFrameCount = mInputBytes / mFrameSize;
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
{
int channelCount;
// optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
// stereo to mono post process as the resampler always outputs stereo.
if (mChannelCount == 1 && mReqChannelCount == 2) {
channelCount = 1;
} else {
channelCount = 2;
}
mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
mResampler->setSampleRate(mSampleRate);
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
mRsmpOutBuffer = new int32_t[mFrameCount * 2];
// optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
if (mChannelCount == 1 && mReqChannelCount == 1) {
mFrameCount >>= 1;
}
}
mRsmpInIndex = mFrameCount;
}
unsigned int AudioFlinger::RecordThread::getInputFramesLost()
{
return mInput->getInputFramesLost();
}
// ----------------------------------------------------------------------------
int AudioFlinger::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
uint32_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags)
{
status_t status;
PlaybackThread *thread = NULL;
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
uint32_t format = pFormat ? *pFormat : 0;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
pDevices ? *pDevices : 0,
samplingRate,
format,
channels,
flags);
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
(int *)&format,
&channels,
&samplingRate,
&status);
LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
output,
samplingRate,
format,
channels,
status);
mHardwareStatus = AUDIO_HW_IDLE;
if (output != 0) {
int id = nextUniqueId();
if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
(format != AudioSystem::PCM_16_BIT) ||
(channels != AudioSystem::CHANNEL_OUT_STEREO)) {
thread = new DirectOutputThread(this, output, id, *pDevices);
LOGV("openOutput() created direct output: ID %d thread %p", id, thread);
} else {
thread = new MixerThread(this, output, id, *pDevices);
LOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
#ifdef LVMX
unsigned bitsPerSample =
(format == AudioSystem::PCM_16_BIT) ? 16 :
((format == AudioSystem::PCM_8_BIT) ? 8 : 0);
unsigned channelCount = (channels == AudioSystem::CHANNEL_OUT_STEREO) ? 2 : 1;
int audioOutputType = LifeVibes::threadIdToAudioOutputType(thread->id());
LifeVibes::init_aot(audioOutputType, samplingRate, bitsPerSample, channelCount);
LifeVibes::setDevice(audioOutputType, *pDevices);
#endif
}
mPlaybackThreads.add(id, thread);
if (pSamplingRate) *pSamplingRate = samplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = channels;
if (pLatencyMs) *pLatencyMs = thread->latency();
// if the device is a A2DP, then this is an A2DP Output
if ( true == AudioSystem::isA2dpDevice((AudioSystem::audio_devices) *pDevices) )
{
mA2DPHandle = id;
LOGV("A2DP device activated. The handle is set to %d", mA2DPHandle);
}
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return id;
}
return 0;
}
int AudioFlinger::openSession(uint32_t *pDevices,
uint32_t *pFormat,
uint32_t flags,
int32_t streamType,
int32_t sessionId)
{
status_t status;
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
uint32_t format = pFormat ? *pFormat : 0;
LOGV("openSession(), Device %x, Format %d, flags %x sessionId %x",
pDevices ? *pDevices : 0,
format,
flags,
sessionId);
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mAudioHardware->openOutputSession(*pDevices,
(int *)&format,
&status,sessionId);
LOGV("openSession() openOutputSession returned output %p, Format %d, status %d",
output,
format,
status);
mHardwareStatus = AUDIO_HW_IDLE;
if (output != 0) {
int id = nextUniqueId();
mLPAOutput = output;
mLPAHandle = id;
mLPAStreamType = streamType;
mLPAStreamIsActive = true;
if (pFormat) *pFormat = format;
return id;
}
return 0;
}
status_t AudioFlinger::pauseSession(int output, int32_t streamType)
{
if (output == mLPAHandle && streamType == mLPAStreamType ) {
mLPAStreamIsActive = false;
}
return NO_ERROR;
}
status_t AudioFlinger::resumeSession(int output, int32_t streamType)
{
if (output == mLPAHandle && streamType == mLPAStreamType ) {
mLPAStreamIsActive = true;
}
return NO_ERROR;
}
status_t AudioFlinger::closeSession(int output)
{
Mutex::Autolock _l(mLock);
LOGV("closeSession() %d", output);
// Is this required?
//AudioSystem::stopOutput(output, (AudioSystem::stream_type)mStreamType);
// Delete the Audio session
if(mLPAOutput) {
mLPAOutput->standby();
delete mLPAOutput;
mLPAOutput = NULL;
mLPAHandle = -1;
mLPAStreamIsActive = false;
}
return NO_ERROR;
}
int AudioFlinger::openDuplicateOutput(int output1, int output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
MixerThread *thread2 = checkMixerThread_l(output2);
if (thread1 == NULL || thread2 == NULL) {
LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
return 0;
}
int id = nextUniqueId();
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return id;
}
status_t AudioFlinger::closeOutput(int output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
sp <PlaybackThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
LOGV("closeOutput() %d", output);
if (thread->type() == PlaybackThread::MIXER) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) {
DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
dupThread->removeOutputTrack((MixerThread *)thread.get());
}
}
}
void *param2 = 0;
audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
mPlaybackThreads.removeItem(output);
if (mA2DPHandle == output)
{
mA2DPHandle = -1;
LOGV("A2DP OutputClosed Notifying Client");
audioConfigChanged_l(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle, param2);
}
}
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) &&
(getInputChannelCount(channels) < 3) && (getInputChannelCount(reqChannels) < 3)) {
LOGV("openInput() reopening with proposed sampling rate and channels");
input = mAudioHardware->openInputStream(*pDevices,
(int *)&format,
&channels,
&samplingRate,
&status,
(AudioSystem::audio_in_acoustics)acoustics);
}
if (input != 0) {
int id = nextUniqueId();
// Start record thread
thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id);
mRecordThreads.add(id, thread);
LOGV("openInput() created record thread: ID %d thread %p", id, thread);
if (pSamplingRate) *pSamplingRate = reqSamplingRate;
if (pFormat) *pFormat = format;
if (pChannels) *pChannels = reqChannels;
input->standby();
// notify client processes of the new input creation
thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
return id;
}
return 0;
}
status_t AudioFlinger::closeInput(int input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
sp <RecordThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
if (thread == NULL) {
return BAD_VALUE;
}
LOGV("closeInput() %d", input);
void *param2 = 0;
audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
mRecordThreads.removeItem(input);
}
thread->exit();
mAudioHardware->closeInputStream(thread->getInput());
return NO_ERROR;
}
status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
{
Mutex::Autolock _l(mLock);
MixerThread *dstThread = checkMixerThread_l(output);
if (dstThread == NULL) {
LOGW("setStreamOutput() bad output id %d", output);
return BAD_VALUE;
}
LOGV("setStreamOutput() stream %d to output %d", stream, output);
audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if (thread != dstThread &&
thread->type() != PlaybackThread::DIRECT) {
MixerThread *srcThread = (MixerThread *)thread;
srcThread->invalidateTracks(stream);
}
}
if ( mA2DPHandle == output ) {
LOGV("A2DP Activated and hence notifying the client");
dstThread->sendConfigEvent(AudioSystem::A2DP_OUTPUT_STATE, mA2DPHandle);
}
return NO_ERROR;
}
int AudioFlinger::newAudioSessionId()
{
return nextUniqueId();
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
{
PlaybackThread *thread = NULL;
if (mPlaybackThreads.indexOfKey(output) >= 0) {
thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
}
return thread;
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread != NULL) {
if (thread->type() == PlaybackThread::DIRECT) {
thread = NULL;
}
}
return (MixerThread *)thread;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
{
RecordThread *thread = NULL;
if (mRecordThreads.indexOfKey(input) >= 0) {
thread = (RecordThread *)mRecordThreads.valueFor(input).get();
}
return thread;
}
int AudioFlinger::nextUniqueId()
{
return android_atomic_inc(&mNextUniqueId);
}
// ----------------------------------------------------------------------------
// Effect management
// ----------------------------------------------------------------------------
status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
// only allow libraries loaded from /system/lib/soundfx for now
if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
return EffectLoadLibrary(libPath, handle);
}
status_t AudioFlinger::unloadEffectLibrary(int handle)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
return EffectUnloadLibrary(handle);
}
status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
{
Mutex::Autolock _l(mLock);
return EffectQueryNumberEffects(numEffects);
}
status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
{
Mutex::Autolock _l(mLock);
return EffectQueryEffect(index, descriptor);
}
status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
{
Mutex::Autolock _l(mLock);
return EffectGetDescriptor(pUuid, descriptor);
}
// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp
static const effect_uuid_t VISUALIZATION_UUID_ =
{0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}};
sp<IEffect> AudioFlinger::createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
int output,
int sessionId,
status_t *status,
int *id,
int *enabled)
{
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
effect_interface_t itfe;
effect_descriptor_t desc;
sp<Client> client;
wp<Client> wclient;
LOGV("createEffect %s pid %d, client %p, priority %d, sessionId %d, output %d",
pDesc->name, pid, effectClient.get(), priority, sessionId, output);
if (pDesc == NULL) {
lStatus = BAD_VALUE;
goto Exit;
}
// check audio settings permission for global effects
if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && !settingsAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// Session AudioSystem::SESSION_OUTPUT_STAGE is reserved for output stage effects
// that can only be created by audio policy manager (running in same process)
if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE && getpid() != pid) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// check recording permission for visualizer
if ((memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 ||
memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) &&
!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
if (output == 0) {
if (sessionId == AudioSystem::SESSION_OUTPUT_STAGE) {
// output must be specified by AudioPolicyManager when using session
// AudioSystem::SESSION_OUTPUT_STAGE
lStatus = BAD_VALUE;
goto Exit;
} else if (sessionId == AudioSystem::SESSION_OUTPUT_MIX) {
// if the output returned by getOutputForEffect() is removed before we lock the
// mutex below, the call to checkPlaybackThread_l(output) below will detect it
// and we will exit safely
output = AudioSystem::getOutputForEffect(&desc);
}
}
{
Mutex::Autolock _l(mLock);
if (!EffectIsNullUuid(&pDesc->uuid)) {
// if uuid is specified, request effect descriptor
lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
if (lStatus < 0) {
LOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
goto Exit;
}
} else {
// if uuid is not specified, look for an available implementation
// of the required type in effect factory
if (EffectIsNullUuid(&pDesc->type)) {
LOGW("createEffect() no effect type");
lStatus = BAD_VALUE;
goto Exit;
}
uint32_t numEffects = 0;
effect_descriptor_t d;
bool found = false;
lStatus = EffectQueryNumberEffects(&numEffects);
if (lStatus < 0) {
LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
goto Exit;
}
for (uint32_t i = 0; i < numEffects; i++) {
lStatus = EffectQueryEffect(i, &desc);
if (lStatus < 0) {
LOGW("createEffect() error %d from EffectQueryEffect", lStatus);
continue;
}
if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
// If matching type found save effect descriptor. If the session is
// 0 and the effect is not auxiliary, continue enumeration in case
// an auxiliary version of this effect type is available
found = true;
memcpy(&d, &desc, sizeof(effect_descriptor_t));
if (sessionId != AudioSystem::SESSION_OUTPUT_MIX ||
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
break;
}
}
}
if (!found) {
lStatus = BAD_VALUE;
LOGW("createEffect() effect not found");
goto Exit;
}
// For same effect type, chose auxiliary version over insert version if
// connect to output mix (Compliance to OpenSL ES)
if (sessionId == AudioSystem::SESSION_OUTPUT_MIX &&
(d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
memcpy(&desc, &d, sizeof(effect_descriptor_t));
}
}
// Do not allow auxiliary effects on a session different from 0 (output mix)
if (sessionId != AudioSystem::SESSION_OUTPUT_MIX &&
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
lStatus = INVALID_OPERATION;
goto Exit;
}
// return effect descriptor
memcpy(pDesc, &desc, sizeof(effect_descriptor_t));
// If output is not specified try to find a matching audio session ID in one of the
// output threads.
// If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
// because of code checking output when entering the function.
if (output == 0) {
// look for the thread where the specified audio session is present
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
output = mPlaybackThreads.keyAt(i);
break;
}
}
// If no output thread contains the requested session ID, default to
// first output. The effect chain will be moved to the correct output
// thread when a track with the same session ID is created
if (output == 0 && mPlaybackThreads.size()) {
output = mPlaybackThreads.keyAt(0);
}
}
LOGV("createEffect() got output %d for effect %s", output, desc.name);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
LOGE("createEffect() unknown output thread");
lStatus = BAD_VALUE;
goto Exit;
}
// TODO: allow attachment of effect to inputs
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
// create effect on selected output trhead
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
&desc, enabled, &lStatus);
if (handle != 0 && id != NULL) {
*id = handle->id();
}
}
Exit:
if(status) {
*status = lStatus;
}
return handle;
}
status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput)
{
LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
session, srcOutput, dstOutput);
Mutex::Autolock _l(mLock);
if (srcOutput == dstOutput) {
LOGW("moveEffects() same dst and src outputs %d", dstOutput);
return NO_ERROR;
}
PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
if (srcThread == NULL) {
LOGW("moveEffects() bad srcOutput %d", srcOutput);
return BAD_VALUE;
}
PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
if (dstThread == NULL) {
LOGW("moveEffects() bad dstOutput %d", dstOutput);
return BAD_VALUE;
}
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(srcThread->mLock);
moveEffectChain_l(session, srcThread, dstThread, false);
return NO_ERROR;
}
// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held
status_t AudioFlinger::moveEffectChain_l(int session,
AudioFlinger::PlaybackThread *srcThread,
AudioFlinger::PlaybackThread *dstThread,
bool reRegister)
{
LOGV("moveEffectChain_l() session %d from thread %p to thread %p",
session, srcThread, dstThread);
sp<EffectChain> chain = srcThread->getEffectChain_l(session);
if (chain == 0) {
LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
session, srcThread);
return INVALID_OPERATION;
}
// remove chain first. This is useful only if reconfiguring effect chain on same output thread,
// so that a new chain is created with correct parameters when first effect is added. This is
// otherwise unecessary as removeEffect_l() will remove the chain when last effect is
// removed.
srcThread->removeEffectChain_l(chain);
// transfer all effects one by one so that new effect chain is created on new thread with
// correct buffer sizes and audio parameters and effect engines reconfigured accordingly
int dstOutput = dstThread->id();
sp<EffectChain> dstChain;
uint32_t strategy;
sp<EffectModule> effect = chain->getEffectFromId_l(0);
while (effect != 0) {
srcThread->removeEffect_l(effect);
dstThread->addEffect_l(effect);
// if the move request is not received from audio policy manager, the effect must be
// re-registered with the new strategy and output
if (dstChain == 0) {
dstChain = effect->chain().promote();
if (dstChain == 0) {
LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
srcThread->addEffect_l(effect);
return NO_INIT;
}
strategy = dstChain->strategy();
}
if (reRegister) {
AudioSystem::unregisterEffect(effect->id());
AudioSystem::registerEffect(&effect->desc(),
dstOutput,
strategy,
session,
effect->id());
}
effect = chain->getEffectFromId_l(0);
}
return NO_ERROR;
}
// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l(
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority,
int sessionId,
effect_descriptor_t *desc,
int *enabled,
status_t *status
)
{
sp<EffectModule> effect;
sp<EffectHandle> handle;
status_t lStatus;
sp<Track> track;
sp<EffectChain> chain;
bool chainCreated = false;
bool effectCreated = false;
bool effectRegistered = false;
if (mOutput == 0) {
LOGW("createEffect_l() Audio driver not initialized.");
lStatus = NO_INIT;
goto Exit;
}
// Do not allow auxiliary effect on session other than 0
if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY &&
sessionId != AudioSystem::SESSION_OUTPUT_MIX) {
LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
desc->name, sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
// Do not allow effects with session ID 0 on direct output or duplicating threads
// TODO: add rule for hw accelerated effects on direct outputs with non PCM format
if (sessionId == AudioSystem::SESSION_OUTPUT_MIX && mType != MIXER) {
LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
desc->name, sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId);
{ // scope for mLock
Mutex::Autolock _l(mLock);
// check for existing effect chain with the requested audio session
chain = getEffectChain_l(sessionId);
if (chain == 0) {
// create a new chain for this session
LOGV("createEffect_l() new effect chain for session %d", sessionId);
chain = new EffectChain(this, sessionId);
addEffectChain_l(chain);
chain->setStrategy(getStrategyForSession_l(sessionId));
chainCreated = true;
if(sessionId == mAudioFlinger->mLPASessionId) {
// Clear reference to previous effect chain if any
if(mAudioFlinger->mLPAEffectChain.get()) {
mAudioFlinger->mLPAEffectChain.clear();
}
LOGV("New EffectChain is created for LPA session ID %d", sessionId);
mAudioFlinger->mLPAEffectChain = chain;
chain->setLPAFlag(true);
// For LPA, the volume will be applied in DSP. No need for volume
// control in the Effect chain, so setting it to unity.
uint32_t volume = 0x1000000; // Equals to 1.0 in 8.24 format
chain->setVolume_l(&volume,&volume);
}
} else {
effect = chain->getEffectFromDesc_l(desc);
}
LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
if (effect == 0) {
int id = mAudioFlinger->nextUniqueId();
// Check CPU and memory usage
lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id);
if (lStatus != NO_ERROR) {
goto Exit;
}
effectRegistered = true;
// create a new effect module if none present in the chain
effect = new EffectModule(this, chain, desc, id, sessionId);
lStatus = effect->status();
if (lStatus != NO_ERROR) {
goto Exit;
}
lStatus = chain->addEffect_l(effect);
if (lStatus != NO_ERROR) {
goto Exit;
}
effectCreated = true;
effect->setDevice(mDevice);
effect->setMode(mAudioFlinger->getMode());
if(chain == mAudioFlinger->mLPAEffectChain) {
effect->setLPAFlag(true);
}
}
// create effect handle and connect it to effect module
handle = new EffectHandle(effect, client, effectClient, priority);
lStatus = effect->addHandle(handle);
if (enabled) {
*enabled = (int)effect->isEnabled();
}
}
Exit:
if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
Mutex::Autolock _l(mLock);
if (effectCreated) {
chain->removeEffect_l(effect);
}
if (effectRegistered) {
AudioSystem::unregisterEffect(effect->id());
}
if (chainCreated) {
removeEffectChain_l(chain);
}
handle.clear();
}
if(status) {
*status = lStatus;
}
return handle;
}
// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
// PlaybackThread::mLock held
status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect)
{
// check for existing effect chain with the requested audio session
int sessionId = effect->sessionId();
sp<EffectChain> chain = getEffectChain_l(sessionId);
bool chainCreated = false;
if (chain == 0) {
// create a new chain for this session
LOGV("addEffect_l() new effect chain for session %d", sessionId);
chain = new EffectChain(this, sessionId);
addEffectChain_l(chain);
chain->setStrategy(getStrategyForSession_l(sessionId));
chainCreated = true;
}
LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
if (chain->getEffectFromId_l(effect->id()) != 0) {
LOGW("addEffect_l() %p effect %s already present in chain %p",
this, effect->desc().name, chain.get());
return BAD_VALUE;
}
status_t status = chain->addEffect_l(effect);
if (status != NO_ERROR) {
if (chainCreated) {
removeEffectChain_l(chain);
}
return status;
}
effect->setDevice(mDevice);
effect->setMode(mAudioFlinger->getMode());
return NO_ERROR;
}
void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) {
LOGV("removeEffect_l() %p effect %p", this, effect.get());
effect_descriptor_t desc = effect->desc();
if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
detachAuxEffect_l(effect->id());
}
sp<EffectChain> chain = effect->chain().promote();
if (chain != 0) {
// remove effect chain if removing last effect
if (chain->removeEffect_l(effect) == 0) {
removeEffectChain_l(chain);
}
} else {
LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
}
}
void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect,
const wp<EffectHandle>& handle) {
Mutex::Autolock _l(mLock);
LOGV("disconnectEffect() %p effect %p", this, effect.get());
// delete the effect module if removing last handle on it
if (effect->removeHandle(handle) == 0) {
removeEffect_l(effect);
AudioSystem::unregisterEffect(effect->id());
}
}
status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
int16_t *buffer = mMixBuffer;
bool ownsBuffer = false;
LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
if (session > 0) {
// Only one effect chain can be present in direct output thread and it uses
// the mix buffer as input
if (mType != DIRECT) {
size_t numSamples = mFrameCount * mChannelCount;
buffer = new int16_t[numSamples];
memset(buffer, 0, numSamples * sizeof(int16_t));
LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session);
ownsBuffer = true;
}
// Attach all tracks with same session ID to this chain.
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
track->setMainBuffer(buffer);
}
}
// indicate all active tracks in the chain
for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) {
sp<Track> track = mActiveTracks[i].promote();
if (track == 0) continue;
if (session == track->sessionId()) {
LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
chain->startTrack();
}
}
}
chain->setInBuffer(buffer, ownsBuffer);
chain->setOutBuffer(mMixBuffer);
// Effect chain for session AudioSystem::SESSION_OUTPUT_STAGE is inserted at end of effect
// chains list in order to be processed last as it contains output stage effects
// Effect chain for session AudioSystem::SESSION_OUTPUT_MIX is inserted before
// session AudioSystem::SESSION_OUTPUT_STAGE to be processed
// after track specific effects and before output stage
// It is therefore mandatory that AudioSystem::SESSION_OUTPUT_MIX == 0 and
// that AudioSystem::SESSION_OUTPUT_STAGE < AudioSystem::SESSION_OUTPUT_MIX
// Effect chain for other sessions are inserted at beginning of effect
// chains list to be processed before output mix effects. Relative order between other
// sessions is not important
size_t size = mEffectChains.size();
size_t i = 0;
for (i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() < session) break;
}
mEffectChains.insertAt(chain, i);
return NO_ERROR;
}
size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session);
for (size_t i = 0; i < mEffectChains.size(); i++) {
if (chain == mEffectChains[i]) {
mEffectChains.removeAt(i);
// detach all tracks with same session ID from this chain
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
track->setMainBuffer(mMixBuffer);
}
}
break;
}
}
return mEffectChains.size();
}
void AudioFlinger::PlaybackThread::lockEffectChains_l(
Vector<sp <AudioFlinger::EffectChain> >& effectChains)
{
effectChains = mEffectChains;
for (size_t i = 0; i < mEffectChains.size(); i++) {
// Do not LPA playback track's effect chain
if(mEffectChains[i] != mAudioFlinger->mLPAEffectChain) {
mEffectChains[i]->lock();
}
}
}
void AudioFlinger::PlaybackThread::unlockEffectChains(
Vector<sp <AudioFlinger::EffectChain> >& effectChains)
{
for (size_t i = 0; i < effectChains.size(); i++) {
// LPA playback track's effect chain is not locked - do not unlock
if(mEffectChains[i] != mAudioFlinger->mLPAEffectChain) {
effectChains[i]->unlock();
}
}
}
sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId)
{
sp<EffectModule> effect;
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
effect = chain->getEffectFromId_l(effectId);
}
return effect;
}
status_t AudioFlinger::PlaybackThread::attachAuxEffect(
const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
{
Mutex::Autolock _l(mLock);
return attachAuxEffect_l(track, EffectId);
}
status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
{
status_t status = NO_ERROR;
LOGV("PlaybackThread::attachAuxEffect_l: EffectId %d", EffectId);
if (EffectId == 0) {
track->setAuxBuffer(0, NULL);
} else {
// Auxiliary effects are always in audio session AudioSystem::SESSION_OUTPUT_MIX
sp<EffectModule> effect = getEffect_l(AudioSystem::SESSION_OUTPUT_MIX, EffectId);
if (effect != 0) {
if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer());
} else {
status = INVALID_OPERATION;
}
} else {
status = BAD_VALUE;
}
}
return status;
}
void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
{
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track->auxEffectId() == effectId) {
attachAuxEffect_l(track, 0);
}
}
}
// ----------------------------------------------------------------------------
// EffectModule implementation
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectModule"
AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId)
: mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
mStatus(NO_INIT), mState(IDLE), mIsForLPA(false)
{
LOGV("Constructor %p sessionId %d", this, sessionId);
int lStatus;
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return;
}
PlaybackThread *p = (PlaybackThread *)thread.get();
memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
// create effect engine from effect factory
mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface);
if (mStatus != NO_ERROR) {
return;
}
lStatus = init();
if (lStatus < 0) {
mStatus = lStatus;
goto Error;
}
LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
return;
Error:
EffectRelease(mEffectInterface);
mEffectInterface = NULL;
LOGV("Constructor Error %d", mStatus);
}
AudioFlinger::EffectModule::~EffectModule()
{
LOGV("Destructor %p", this);
if (mEffectInterface != NULL) {
// release effect engine
EffectRelease(mEffectInterface);
}
}
status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
{
status_t status;
Mutex::Autolock _l(mLock);
// First handle in mHandles has highest priority and controls the effect module
int priority = handle->priority();
size_t size = mHandles.size();
sp<EffectHandle> h;
size_t i;
for (i = 0; i < size; i++) {
h = mHandles[i].promote();
if (h == 0) continue;
if (h->priority() <= priority) break;
}
// if inserted in first place, move effect control from previous owner to this handle
if (i == 0) {
if (h != 0) {
h->setControl(false, true);
}
handle->setControl(true, false);
status = NO_ERROR;
} else {
status = ALREADY_EXISTS;
}
mHandles.insertAt(handle, i);
return status;
}
size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle)
{
Mutex::Autolock _l(mLock);
size_t size = mHandles.size();
size_t i;
for (i = 0; i < size; i++) {
if (mHandles[i] == handle) break;
}
if (i == size) {
return size;
}
mHandles.removeAt(i);
size = mHandles.size();
// if removed from first place, move effect control from this handle to next in line
if (i == 0 && size != 0) {
sp<EffectHandle> h = mHandles[0].promote();
if (h != 0) {
h->setControl(true, true);
}
}
// Release effect engine here so that it is done immediately. Otherwise it will be released
// by the destructor when the last strong reference on the this object is released which can
// happen after next process is called on this effect.
if (size == 0 && mEffectInterface != NULL) {
// release effect engine
EffectRelease(mEffectInterface);
mEffectInterface = NULL;
}
return size;
}
void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle)
{
setEnabled(false);
// keep a strong reference on this EffectModule to avoid calling the
// destructor before we exit
sp<EffectModule> keep(this);
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
playbackThread->disconnectEffect(keep, handle);
}
}
}
void AudioFlinger::EffectModule::updateState() {
Mutex::Autolock _l(mLock);
switch (mState) {
case RESTART:
reset_l();
// FALL THROUGH
case STARTING:
// clear auxiliary effect input buffer for next accumulation
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
memset(mConfig.inputCfg.buffer.raw,
0,
mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
}
start_l();
mState = ACTIVE;
break;
case STOPPING:
stop_l();
mDisableWaitCnt = mMaxDisableWaitCnt;
mState = STOPPED;
break;
case STOPPED:
// mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the
// turn off sequence.
if (--mDisableWaitCnt == 0) {
reset_l();
mState = IDLE;
}
break;
default: //IDLE , ACTIVE
break;
}
}
void AudioFlinger::EffectModule::process()
{
Mutex::Autolock _l(mLock);
LOGV("EffectModule::process()");
if (mEffectInterface == NULL ||
mConfig.inputCfg.buffer.raw == NULL ||
mConfig.outputCfg.buffer.raw == NULL) {
return;
}
if (isProcessEnabled()) {
// do 32 bit to 16 bit conversion for auxiliary effect input buffer
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount/2);
}
// do the actual processing in the effect engine
int ret = (*mEffectInterface)->process(mEffectInterface,
&mConfig.inputCfg.buffer,
&mConfig.outputCfg.buffer);
// force transition to IDLE state when engine is ready
if (mState == STOPPED && ret == -ENODATA) {
mDisableWaitCnt = 1;
}
// clear auxiliary effect input buffer for next accumulation
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
memset(mConfig.inputCfg.buffer.raw, 0,
mConfig.inputCfg.buffer.frameCount*sizeof(int32_t));
}
} else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&
mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
// If an insert effect is idle and input buffer is different from output buffer,
// accumulate input onto output
sp<EffectChain> chain = mChain.promote();
if (chain != 0 && chain->activeTracks() != 0) {
size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here
int16_t *in = mConfig.inputCfg.buffer.s16;
int16_t *out = mConfig.outputCfg.buffer.s16;
for (size_t i = 0; i < frameCnt; i++) {
out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]);
}
}
}
}
void AudioFlinger::EffectModule::reset_l()
{
if (mEffectInterface == NULL) {
return;
}
(*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
}
status_t AudioFlinger::EffectModule::configure(bool isForLPA, int sampleRate, int channelCount, int frameCount)
{
uint32_t channels;
// Acquire lock here to make sure that any other thread does not delete
// the effect handle and release the effect module.
Mutex::Autolock _l(mLock);
if (mEffectInterface == NULL) {
return NO_INIT;
}
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return DEAD_OBJECT;
}
// TODO: handle configuration of effects replacing track process
mIsForLPA = isForLPA;
if(isForLPA) {
if (channelCount == 1) {
channels = CHANNEL_MONO;
} else {
channels = CHANNEL_STEREO;
}
LOGV("%s: LPA ON - channels %d", __func__, channels);
} else {
if (thread->channelCount() == 1) {
channels = CHANNEL_MONO;
} else {
channels = CHANNEL_STEREO;
}
}
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
mConfig.inputCfg.channels = CHANNEL_MONO;
} else {
mConfig.inputCfg.channels = channels;
}
mConfig.outputCfg.channels = channels;
mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15;
mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15;
if(isForLPA){
mConfig.inputCfg.samplingRate = sampleRate;
LOGV("%s: LPA ON - sampleRate %d", __func__, sampleRate);
} else {
mConfig.inputCfg.samplingRate = thread->sampleRate();
}
mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
mConfig.inputCfg.bufferProvider.cookie = NULL;
mConfig.inputCfg.bufferProvider.getBuffer = NULL;
mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
mConfig.outputCfg.bufferProvider.cookie = NULL;
mConfig.outputCfg.bufferProvider.getBuffer = NULL;
mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
// Insert effect:
// - in session AudioSystem::SESSION_OUTPUT_MIX or AudioSystem::SESSION_OUTPUT_STAGE,
// always overwrites output buffer: input buffer == output buffer
// - in other sessions:
// last effect in the chain accumulates in output buffer: input buffer != output buffer
// other effect: overwrites output buffer: input buffer == output buffer
// Auxiliary effect:
// accumulates in output buffer: input buffer != output buffer
// Therefore: accumulate <=> input buffer != output buffer
if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
} else {
mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
}
mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
if(isForLPA) {
mConfig.inputCfg.buffer.frameCount = frameCount;
LOGV("%s: LPA ON - frameCount %d", __func__, frameCount);
} else {
mConfig.inputCfg.buffer.frameCount = thread->frameCount();
}
mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
LOGV("configure() %p thread %p buffer %p framecount %d",
this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
status_t cmdStatus;
uint32_t size = sizeof(int);
status_t status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_CONFIGURE,
sizeof(effect_config_t),
&mConfig,
&size,
&cmdStatus);
if (status == 0) {
status = cmdStatus;
}
mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) /
(1000 * mConfig.outputCfg.buffer.frameCount);
return status;
}
status_t AudioFlinger::EffectModule::init()
{
Mutex::Autolock _l(mLock);
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status_t status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_INIT,
0,
NULL,
&size,
&cmdStatus);
if (status == 0) {
status = cmdStatus;
}
return status;
}
status_t AudioFlinger::EffectModule::start_l()
{
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status_t status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_ENABLE,
0,
NULL,
&size,
&cmdStatus);
if (status == 0) {
if(cmdStatus == 0) {
// Some of the effects are applied based on the output device.
// For example, BASS_BOOST will not be enabled if output device is Speaker.
// If the command to set device is called first and then the command
// to enable the effect, it is not checking the previously set device.
// The below command to set device after enabling the effect, ensures that
// output device is considered to enable/disable the effect.
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_DEVICE,
sizeof(uint32_t),
&mDevice,
&size,
&cmdStatus);
if(status == 0) {
status = cmdStatus;
}
} else {
status = cmdStatus;
}
}
return status;
}
status_t AudioFlinger::EffectModule::stop_l()
{
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status_t status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_DISABLE,
0,
NULL,
&size,
&cmdStatus);
if (status == 0) {
status = cmdStatus;
}
return status;
}
status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t *replySize,
void *pReplyData)
{
Mutex::Autolock _l(mLock);
LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
if (mEffectInterface == NULL) {
return NO_INIT;
}
status_t status = (*mEffectInterface)->command(mEffectInterface,
cmdCode,
cmdSize,
pCmdData,
replySize,
pReplyData);
if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) {
uint32_t size = (replySize == NULL) ? 0 : *replySize;
for (size_t i = 1; i < mHandles.size(); i++) {
sp<EffectHandle> h = mHandles[i].promote();
if (h != 0) {
h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData);
}
}
}
return status;
}
status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
{
bool effectStateChanged = false;
{
Mutex::Autolock _l(mLock);
LOGV("setEnabled %p enabled %d", this, enabled);
if (enabled != isEnabled()) {
effectStateChanged = true;
switch (mState) {
// going from disabled to enabled
case IDLE:
mState = STARTING;
break;
case STOPPED:
mState = RESTART;
break;
case STOPPING:
mState = ACTIVE;
break;
// going from enabled to disabled
case RESTART:
mState = STOPPED;
break;
case STARTING:
mState = IDLE;
break;
case ACTIVE:
mState = STOPPING;
break;
}
for (size_t i = 1; i < mHandles.size(); i++) {
sp<EffectHandle> h = mHandles[i].promote();
if (h != 0) {
h->setEnabled(enabled);
}
}
}
}
/*
Send notification event to LPA Player when an effect for
LPA output is enabled or disabled.
*/
if (effectStateChanged && mIsForLPA) {
sp<ThreadBase> thread = mThread.promote();
thread->effectConfigChanged();
}
return NO_ERROR;
}
bool AudioFlinger::EffectModule::isEnabled()
{
switch (mState) {
case RESTART:
case STARTING:
case ACTIVE:
return true;
case IDLE:
case STOPPING:
case STOPPED:
default:
return false;
}
}
bool AudioFlinger::EffectModule::isProcessEnabled()
{
switch (mState) {
case RESTART:
case ACTIVE:
case STOPPING:
case STOPPED:
return true;
case IDLE:
case STARTING:
default:
return false;
}
}
status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
// Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume
// if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set)
if (isProcessEnabled() &&
((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL ||
(mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) {
status_t cmdStatus;
uint32_t volume[2];
uint32_t *pVolume = NULL;
uint32_t size = sizeof(volume);
volume[0] = *left;
volume[1] = *right;
if (controller) {
pVolume = volume;
}
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_VOLUME,
size,
volume,
&size,
pVolume);
if (controller && status == NO_ERROR && size == sizeof(volume)) {
*left = volume[0];
*right = volume[1];
}
}
return status;
}
status_t AudioFlinger::EffectModule::setDevice(uint32_t device)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
// convert device bit field from AudioSystem to EffectApi format.
device = deviceAudioSystemToEffectApi(device);
if (device == 0) {
return BAD_VALUE;
}
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_DEVICE,
sizeof(uint32_t),
&device,
&size,
&cmdStatus);
mDevice = device;
if (status == NO_ERROR) {
status = cmdStatus;
}
}
return status;
}
status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
// convert audio mode from AudioSystem to EffectApi format.
int effectMode = modeAudioSystemToEffectApi(mode);
if (effectMode < 0) {
return BAD_VALUE;
}
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_AUDIO_MODE,
sizeof(int),
&effectMode,
&size,
&cmdStatus);
if (status == NO_ERROR) {
status = cmdStatus;
}
}
return status;
}
// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified
const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = {
DEVICE_EARPIECE, // AudioSystem::DEVICE_OUT_EARPIECE
DEVICE_SPEAKER, // AudioSystem::DEVICE_OUT_SPEAKER
DEVICE_WIRED_HEADSET, // case AudioSystem::DEVICE_OUT_WIRED_HEADSET
DEVICE_WIRED_HEADPHONE, // AudioSystem::DEVICE_OUT_WIRED_HEADPHONE
DEVICE_BLUETOOTH_SCO, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO
DEVICE_BLUETOOTH_SCO_HEADSET, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_HEADSET
DEVICE_BLUETOOTH_SCO_CARKIT, // AudioSystem::DEVICE_OUT_BLUETOOTH_SCO_CARKIT
DEVICE_BLUETOOTH_A2DP, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP
DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES
DEVICE_BLUETOOTH_A2DP_SPEAKER, // AudioSystem::DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER
DEVICE_AUX_DIGITAL, // AudioSystem::DEVICE_OUT_AUX_DIGITAL
DEVICE_AUX_HDMI,
DEVICE_FM,
DEVICE_ANC_HEADSET,
DEVICE_ANC_HEADPHONE,
DEVICE_FM_TX,
};
uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device)
{
uint32_t deviceOut = 0;
while (device) {
const uint32_t i = 31 - __builtin_clz(device);
device &= ~(1 << i);
if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) {
LOGE("device convertion error for AudioSystem device 0x%08x", device);
return 0;
}
deviceOut |= (uint32_t)sDeviceConvTable[i];
}
return deviceOut;
}
// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified
const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = {
AUDIO_MODE_NORMAL, // AudioSystem::MODE_NORMAL
AUDIO_MODE_RINGTONE, // AudioSystem::MODE_RINGTONE
AUDIO_MODE_IN_CALL, // AudioSystem::MODE_IN_CALL
AUDIO_MODE_IN_CALL // AudioSystem::MODE_IN_COMMUNICATION, same conversion as for MODE_IN_CALL
};
int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode)
{
int modeOut = -1;
if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) {
modeOut = (int)sModeConvTable[mode];
}
return modeOut;
}
status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId);
result.append(buffer);
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
result.append("\t\tCould not lock Fx mutex:\n");
}
result.append("\t\tSession Status State Engine:\n");
snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n",
mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
result.append(buffer);
result.append("\t\tDescriptor:\n");
snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n",
mDescriptor.apiVersion,
mDescriptor.flags);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- name: %s\n",
mDescriptor.name);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- implementor: %s\n",
mDescriptor.implementor);
result.append(buffer);
result.append("\t\t- Input configuration:\n");
result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n",
(uint32_t)mConfig.inputCfg.buffer.raw,
mConfig.inputCfg.buffer.frameCount,
mConfig.inputCfg.samplingRate,
mConfig.inputCfg.channels,
mConfig.inputCfg.format);
result.append(buffer);
result.append("\t\t- Output configuration:\n");
result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n",
(uint32_t)mConfig.outputCfg.buffer.raw,
mConfig.outputCfg.buffer.frameCount,
mConfig.outputCfg.samplingRate,
mConfig.outputCfg.channels,
mConfig.outputCfg.format);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
result.append(buffer);
result.append("\t\t\tPid Priority Ctrl Locked client server\n");
for (size_t i = 0; i < mHandles.size(); ++i) {
sp<EffectHandle> handle = mHandles[i].promote();
if (handle != 0) {
handle->dump(buffer, SIZE);
result.append(buffer);
}
}
result.append("\n");
write(fd, result.string(), result.length());
if (locked) {
mLock.unlock();
}
return NO_ERROR;
}
// ----------------------------------------------------------------------------
// EffectHandle implementation
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectHandle"
AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority)
: BnEffect(),
mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false)
{
LOGV("constructor %p", this);
int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int);
mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
if (mCblkMemory != 0) {
mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
if (mCblk) {
new(mCblk) effect_param_cblk_t();
mBuffer = (uint8_t *)mCblk + bufOffset;
}
} else {
LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
return;
}
}
AudioFlinger::EffectHandle::~EffectHandle()
{
LOGV("Destructor %p", this);
disconnect();
}
status_t AudioFlinger::EffectHandle::enable()
{
if (!mHasControl) return INVALID_OPERATION;
if (mEffect == 0) return DEAD_OBJECT;
return mEffect->setEnabled(true);
}
status_t AudioFlinger::EffectHandle::disable()
{
if (!mHasControl) return INVALID_OPERATION;
if (mEffect == NULL) return DEAD_OBJECT;
return mEffect->setEnabled(false);
}
void AudioFlinger::EffectHandle::disconnect()
{
if (mEffect == 0) {
return;
}
mEffect->disconnect(this);
// release sp on module => module destructor can be called now
mEffect.clear();
if (mCblk) {
mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
}
mCblkMemory.clear(); // and free the shared memory
if (mClient != 0) {
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
}
status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t *replySize,
void *pReplyData)
{
LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p",
cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get());
// only get parameter command is permitted for applications not controlling the effect
if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
return INVALID_OPERATION;
}
if (mEffect == 0) return DEAD_OBJECT;
// handle commands that are not forwarded transparently to effect engine
if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
// No need to trylock() here as this function is executed in the binder thread serving a particular client process:
// no risk to block the whole media server process or mixer threads is we are stuck here
Mutex::Autolock _l(mCblk->lock);
if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
mCblk->serverIndex = 0;
mCblk->clientIndex = 0;
return BAD_VALUE;
}
status_t status = NO_ERROR;
while (mCblk->serverIndex < mCblk->clientIndex) {
int reply;
uint32_t rsize = sizeof(int);
int *p = (int *)(mBuffer + mCblk->serverIndex);
int size = *p++;
if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) {
LOGW("command(): invalid parameter block size");
break;
}
effect_param_t *param = (effect_param_t *)p;
if (param->psize == 0 || param->vsize == 0) {
LOGW("command(): null parameter or value size");
mCblk->serverIndex += size;
continue;
}
uint32_t psize = sizeof(effect_param_t) +
((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
param->vsize;
status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM,
psize,
p,
&rsize,
&reply);
// stop at first error encountered
if (ret != NO_ERROR) {
status = ret;
*(int *)pReplyData = reply;
break;
} else if (reply != NO_ERROR) {
*(int *)pReplyData = reply;
break;
}
mCblk->serverIndex += size;
}
mCblk->serverIndex = 0;
mCblk->clientIndex = 0;
return status;
} else if (cmdCode == EFFECT_CMD_ENABLE) {
*(int *)pReplyData = NO_ERROR;
return enable();
} else if (cmdCode == EFFECT_CMD_DISABLE) {
*(int *)pReplyData = NO_ERROR;
return disable();
}
LOGV("EffectHandle::command: isOnLPA %d", mEffect->isOnLPA());
if(mEffect->isOnLPA() &&
((cmdCode == EFFECT_CMD_SET_PARAM) || (cmdCode == EFFECT_CMD_SET_PARAM_DEFERRED) ||
(cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) || (cmdCode == EFFECT_CMD_SET_DEVICE) ||
(cmdCode == EFFECT_CMD_SET_VOLUME) || (cmdCode == EFFECT_CMD_SET_AUDIO_MODE)) ) {
// Notify LPA Player for the change in Effect module
// TODO: check if it is required to send mLPAHandle
LOGV("Notifying LPA player for the change in effect config");
mClient->audioFlinger()->audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
}
return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
return mCblkMemory;
}
void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal)
{
LOGV("setControl %p control %d", this, hasControl);
mHasControl = hasControl;
if (signal && mEffectClient != 0) {
mEffectClient->controlStatusChanged(hasControl);
}
}
void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t replySize,
void *pReplyData)
{
if (mEffectClient != 0) {
mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
}
void AudioFlinger::EffectHandle::setEnabled(bool enabled)
{
if (mEffectClient != 0) {
mEffectClient->enableStatusChanged(enabled);
}
}
status_t AudioFlinger::EffectHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnEffect::onTransact(code, data, reply, flags);
}
void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
{
bool locked = tryLock(mCblk->lock);
snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
(mClient == NULL) ? getpid() : mClient->pid(),
mPriority,
mHasControl,
!locked,
mCblk->clientIndex,
mCblk->serverIndex
);
if (locked) {
mCblk->lock.unlock();
}
}
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectChain"
AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
int sessionId)
: mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false),
mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX), mIsForLPATrack(false)
{
LOGV("EffectChain::ctor sessionId %d", sessionId);
mStrategy = AudioSystem::getStrategyForStream(AudioSystem::MUSIC);
}
AudioFlinger::EffectChain::~EffectChain()
{
LOGV("EffectChain::dtor mOwnInBuffer %d", mOwnInBuffer);
if (mOwnInBuffer) {
delete mInBuffer;
}
}
// getEffectFromDesc_l() must be called with PlaybackThread::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
{
sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
effect = mEffects[i];
break;
}
}
return effect;
}
// getEffectFromId_l() must be called with PlaybackThread::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
{
sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
// by convention, return first effect if id provided is 0 (0 is never a valid id)
if (id == 0 || mEffects[i]->id() == id) {
effect = mEffects[i];
break;
}
}
return effect;
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromIndex_l(int idx)
{
sp<EffectModule> effect = NULL;
if(idx < 0 || idx >= mEffects.size()) {
LOGE("EffectChain::getEffectFromIndex_l: invalid index %d", idx);
}
if(mEffects.size() > 0){
effect = mEffects[idx];
}
return effect;
}
// Must be called with EffectChain::mLock locked
void AudioFlinger::EffectChain::process_l()
{
LOGV("EffectChain::process_l()");
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
LOGW("process_l(): cannot promote mixer thread");
return;
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
bool isGlobalSession = (mSessionId == AudioSystem::SESSION_OUTPUT_MIX) ||
(mSessionId == AudioSystem::SESSION_OUTPUT_STAGE);
bool tracksOnSession = false;
if (!isGlobalSession && !isForLPATrack()) {
tracksOnSession =
playbackThread->hasAudioSession(mSessionId) & PlaybackThread::TRACK_SESSION;
}
size_t size = mEffects.size();
LOGV("process_l(): isGlobalSession %d tracksOnSession %d isForLPA %d numEffects %d",
isGlobalSession, tracksOnSession, isForLPATrack(), size);
// do not process effect if no track is present in same audio session
if (isGlobalSession || tracksOnSession || isForLPATrack()) {
for (size_t i = 0; i < size; i++) {
mEffects[i]->process();
}
}
for (size_t i = 0; i < size; i++) {
mEffects[i]->updateState();
}
// if no track is active, input buffer must be cleared here as the mixer process
// will not do it
if (tracksOnSession &&
activeTracks() == 0) {
size_t numSamples = playbackThread->frameCount() * playbackThread->channelCount();
memset(mInBuffer, 0, numSamples * sizeof(int16_t));
}
}
// addEffect_l() must be called with PlaybackThread::mLock held
status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
{
effect_descriptor_t desc = effect->desc();
uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
Mutex::Autolock _l(mLock);
effect->setChain(this);
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return NO_INIT;
}
effect->setThread(thread);
if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
// Auxiliary effects are inserted at the beginning of mEffects vector as
// they are processed first and accumulated in chain input buffer
mEffects.insertAt(effect, 0);
// the input buffer for auxiliary effect contains mono samples in
// 32 bit format. This is to avoid saturation in AudoMixer
// accumulation stage. Saturation is done in EffectModule::process() before
// calling the process in effect engine
size_t numSamples = thread->frameCount();
int32_t *buffer = new int32_t[numSamples];
memset(buffer, 0, numSamples * sizeof(int32_t));
effect->setInBuffer((int16_t *)buffer);
// auxiliary effects output samples to chain input buffer for further processing
// by insert effects
effect->setOutBuffer(mInBuffer);
} else {
// Insert effects are inserted at the end of mEffects vector as they are processed
// after track and auxiliary effects.
// Insert effect order as a function of indicated preference:
// if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if
// another effect is present
// else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the
// last effect claiming first position
// else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the
// first effect claiming last position
// else if EFFECT_FLAG_INSERT_ANY insert after first or before last
// Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
// already present
int size = (int)mEffects.size();
int idx_insert = size;
int idx_insert_first = -1;
int idx_insert_last = -1;
for (int i = 0; i < size; i++) {
effect_descriptor_t d = mEffects[i]->desc();
uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
if (iMode == EFFECT_FLAG_TYPE_INSERT) {
// check invalid effect chaining combinations
if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
return INVALID_OPERATION;
}
// remember position of first insert effect and by default
// select this as insert position for new effect
if (idx_insert == size) {
idx_insert = i;
}
// remember position of last insert effect claiming
// first position
if (iPref == EFFECT_FLAG_INSERT_FIRST) {
idx_insert_first = i;
}
// remember position of first insert effect claiming
// last position
if (iPref == EFFECT_FLAG_INSERT_LAST &&
idx_insert_last == -1) {
idx_insert_last = i;
}
}
}
// modify idx_insert from first position if needed
if (insertPref == EFFECT_FLAG_INSERT_LAST) {
if (idx_insert_last != -1) {
idx_insert = idx_insert_last;
} else {
idx_insert = size;
}
} else {
if (idx_insert_first != -1) {
idx_insert = idx_insert_first + 1;
}
}
// always read samples from chain input buffer
effect->setInBuffer(mInBuffer);
// if last effect in the chain, output samples to chain
// output buffer, otherwise to chain input buffer
if (idx_insert == size) {
if (idx_insert != 0) {
mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
mEffects[idx_insert-1]->configure();
}
effect->setOutBuffer(mOutBuffer);
} else {
effect->setOutBuffer(mInBuffer);
}
mEffects.insertAt(effect, idx_insert);
LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
}
effect->configure();
return NO_ERROR;
}
// removeEffect_l() must be called with PlaybackThread::mLock held
size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
{
Mutex::Autolock _l(mLock);
int size = (int)mEffects.size();
int i;
uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
for (i = 0; i < size; i++) {
if (effect == mEffects[i]) {
if (type == EFFECT_FLAG_TYPE_AUXILIARY) {
delete[] effect->inBuffer();
} else {
if (i == size - 1 && i != 0) {
mEffects[i - 1]->setOutBuffer(mOutBuffer);
mEffects[i - 1]->configure();
}
}
mEffects.removeAt(i);
LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
break;
}
}
return mEffects.size();
}
// setDevice_l() must be called with PlaybackThread::mLock held
void AudioFlinger::EffectChain::setDevice_l(uint32_t device)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
mEffects[i]->setDevice(device);
}
}
// setMode_l() must be called with PlaybackThread::mLock held
void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
mEffects[i]->setMode(mode);
}
}
// setVolume_l() must be called with PlaybackThread::mLock held
bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right)
{
uint32_t newLeft = *left;
uint32_t newRight = *right;
bool hasControl = false;
int ctrlIdx = -1;
size_t size = mEffects.size();
// first update volume controller
for (size_t i = size; i > 0; i--) {
if (mEffects[i - 1]->isProcessEnabled() &&
(mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) {
ctrlIdx = i - 1;
hasControl = true;
break;
}
}
if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) {
if (hasControl) {
*left = mNewLeftVolume;
*right = mNewRightVolume;
}
return hasControl;
}
mVolumeCtrlIdx = ctrlIdx;
mLeftVolume = newLeft;
mRightVolume = newRight;
// second get volume update from volume controller
if (ctrlIdx >= 0) {
mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true);
mNewLeftVolume = newLeft;
mNewRightVolume = newRight;
}
// then indicate volume to all other effects in chain.
// Pass altered volume to effects before volume controller
// and requested volume to effects after controller
uint32_t lVol = newLeft;
uint32_t rVol = newRight;
for (size_t i = 0; i < size; i++) {
if ((int)i == ctrlIdx) continue;
// this also works for ctrlIdx == -1 when there is no volume controller
if ((int)i > ctrlIdx) {
lVol = *left;
rVol = *right;
}
mEffects[i]->setVolume(&lVol, &rVol, false);
}
*left = newLeft;
*right = newRight;
return hasControl;
}
status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId);
result.append(buffer);
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
result.append("\tCould not lock mutex:\n");
}
result.append("\tNum fx In buffer Out buffer Active tracks:\n");
snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n",
mEffects.size(),
(uint32_t)mInBuffer,
(uint32_t)mOutBuffer,
mActiveTrackCnt);
result.append(buffer);
write(fd, result.string(), result.size());
for (size_t i = 0; i < mEffects.size(); ++i) {
sp<EffectModule> effect = mEffects[i];
if (effect != 0) {
effect->dump(fd, args);
}
}
if (locked) {
mLock.unlock();
}
return NO_ERROR;
}
#undef LOG_TAG
#define LOG_TAG "AudioFlinger"
// ----------------------------------------------------------------------------
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioFlinger::onTransact(code, data, reply, flags);
}
}; // namespace android
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