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android_frameworks_base/services/audioflinger/AudioFlinger.cpp
Glenn Kasten 685c9ce3bd Avoid wp<>::unsafe_get() with a few exceptions 
Avoid using wp<>::unsafe_get() except in a log, and other specific cases
when it's known to be safe.

Use more specific subclass types for parameters to avoid down-casts.

When a constructor or method parameter is "this" of an object that is
currently being constructed, it's better to use a raw pointer rather
than either sp<> or wp<>.

Using the raw pointer is safe, provided either:
 - it is "this" of an object being constructed (which has sp<> refcount of 0),
 - or the caller already holds an sp<>

The raw pointer is simpler and faster, and it avoids the problem of the
sp<> reference count being incremented and then decremented to zero on
scope exit, which would cause the object's destructor to run while the
object is still being constructed.

Also removed some dead code per a review comment.

Change-Id: I7375f64da3aec11b928c33cb01faff186252ef5e
2012-02-22 13:19:26 -08:00

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/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <utils/Atomic.h>
#include <cutils/bitops.h>
#include <cutils/properties.h>
#include <cutils/compiler.h>
#include <media/IMediaPlayerService.h>
#include <media/IMediaDeathNotifier.h>
#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#include "ServiceUtilities.h"
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
#include <audio_effects/effect_ns.h>
#include <audio_effects/effect_aec.h>
#include <audio_utils/primitives.h>
#include <cpustats/ThreadCpuUsage.h>
#include <powermanager/PowerManager.h>
// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds
#include <common_time/cc_helper.h>
#include <common_time/local_clock.h>
// ----------------------------------------------------------------------------
namespace android {
static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
static const char kHardwareLockedString[] = "Hardware lock is taken\n";
static const float MAX_GAIN = 4096.0f;
static const uint32_t 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 kDumpLockSleepUs = 20000;
// don't warn about blocked writes or record buffer overflows more often than this
static const nsecs_t kWarningThrottleNs = seconds(5);
// RecordThread loop sleep time upon application overrun or audio HAL read error
static const int kRecordThreadSleepUs = 5000;
// maximum time to wait for setParameters to complete
static const nsecs_t kSetParametersTimeoutNs = seconds(2);
// minimum sleep time for the mixer thread loop when tracks are active but in underrun
static const uint32_t kMinThreadSleepTimeUs = 5000;
// maximum divider applied to the active sleep time in the mixer thread loop
static const uint32_t kMaxThreadSleepTimeShift = 2;
nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
// ----------------------------------------------------------------------------
// To collect the amplifier usage
static void addBatteryData(uint32_t params) {
sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
if (service == NULL) {
// it already logged
return;
}
service->addBatteryData(params);
}
static int load_audio_interface(const char *if_name, const hw_module_t **mod,
audio_hw_device_t **dev)
{
int rc;
rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod);
if (rc)
goto out;
rc = audio_hw_device_open(*mod, dev);
ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)",
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc)
goto out;
return 0;
out:
*mod = NULL;
*dev = NULL;
return rc;
}
static const char * const audio_interfaces[] = {
"primary",
"a2dp",
"usb",
};
#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mPrimaryHardwareDev(NULL),
mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef()
mMasterVolume(1.0f),
mMasterVolumeSupportLvl(MVS_NONE),
mMasterMute(false),
mNextUniqueId(1),
mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false)
{
}
void AudioFlinger::onFirstRef()
{
int rc = 0;
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
char val_str[PROPERTY_VALUE_MAX] = { 0 };
if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
uint32_t int_val;
if (1 == sscanf(val_str, "%u", &int_val)) {
mStandbyTimeInNsecs = milliseconds(int_val);
ALOGI("Using %u mSec as standby time.", int_val);
} else {
mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
ALOGI("Using default %u mSec as standby time.",
(uint32_t)(mStandbyTimeInNsecs / 1000000));
}
}
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
const hw_module_t *mod;
audio_hw_device_t *dev;
rc = load_audio_interface(audio_interfaces[i], &mod, &dev);
if (rc)
continue;
ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
mod->name, mod->id);
mAudioHwDevs.push(dev);
if (mPrimaryHardwareDev == NULL) {
mPrimaryHardwareDev = dev;
ALOGI("Using '%s' (%s.%s) as the primary audio interface",
mod->name, mod->id, audio_interfaces[i]);
}
}
if (mPrimaryHardwareDev == NULL) {
ALOGE("Primary audio interface not found");
// proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck()
}
// Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the
// primary HW dev is selected can change so these conditions might not always be equivalent.
// When that happens, re-visit all the code that assumes this.
AutoMutex lock(mHardwareLock);
// Determine the level of master volume support the primary audio HAL has,
// and set the initial master volume at the same time.
float initialVolume = 1.0;
mMasterVolumeSupportLvl = MVS_NONE;
if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) {
audio_hw_device_t *dev = mPrimaryHardwareDev;
mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
if ((NULL != dev->get_master_volume) &&
(NO_ERROR == dev->get_master_volume(dev, &initialVolume))) {
mMasterVolumeSupportLvl = MVS_FULL;
} else {
mMasterVolumeSupportLvl = MVS_SETONLY;
initialVolume = 1.0;
}
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if ((NULL == dev->set_master_volume) ||
(NO_ERROR != dev->set_master_volume(dev, initialVolume))) {
mMasterVolumeSupportLvl = MVS_NONE;
}
mHardwareStatus = AUDIO_HW_INIT;
}
// Set the mode for each audio HAL, and try to set the initial volume (if
// supported) for all of the non-primary audio HALs.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->init_check(dev);
mHardwareStatus = AUDIO_HW_IDLE;
if (rc == 0) {
mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value
mHardwareStatus = AUDIO_HW_SET_MODE;
dev->set_mode(dev, mMode);
if ((dev != mPrimaryHardwareDev) &&
(NULL != dev->set_master_volume)) {
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
dev->set_master_volume(dev, initialVolume);
}
mHardwareStatus = AUDIO_HW_INIT;
}
}
mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl)
? initialVolume
: 1.0;
mMasterVolume = initialVolume;
mHardwareStatus = AUDIO_HW_IDLE;
}
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));
}
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
// no mHardwareLock needed, as there are no other references to this
audio_hw_device_close(mAudioHwDevs[i]);
}
}
audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
{
/* first matching HW device is returned */
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
if ((dev->get_supported_devices(dev) & devices) == devices)
return dev;
}
return NULL;
}
status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
sp<Client> client = mClients.valueAt(i).promote();
if (client != 0) {
snprintf(buffer, SIZE, " pid: %d\n", client->pid());
result.append(buffer);
}
}
result.append("Global session refs:\n");
result.append(" session pid cnt\n");
for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
AudioSessionRef *r = mAudioSessionRefs[i];
snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt);
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;
hardware_call_state hardwareStatus = mHardwareStatus;
snprintf(buffer, SIZE, "Hardware status: %d\n"
"Standby Time mSec: %u\n",
hardwareStatus,
(uint32_t)(mStandbyTimeInNsecs / 1000000));
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(kDumpLockSleepUs);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
if (!dumpAllowed()) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
bool hardwareLocked = tryLock(mHardwareLock);
if (!hardwareLocked) {
String8 result(kHardwareLockedString);
write(fd, result.string(), result.size());
} else {
mHardwareLock.unlock();
}
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
String8 result(kDeadlockedString);
write(fd, result.string(), result.size());
}
dumpClients(fd, args);
dumpInternals(fd, args);
// dump playback threads
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->dump(fd, args);
}
// dump record threads
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->dump(fd, args);
}
// dump all hardware devs
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
dev->dump(dev, fd);
}
if (locked) mLock.unlock();
}
return NO_ERROR;
}
sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
{
// If pid is already in the mClients wp<> map, then use that entry
// (for which promote() is always != 0), otherwise create a new entry and Client.
sp<Client> client = mClients.valueFor(pid).promote();
if (client == 0) {
client = new Client(this, pid);
mClients.add(pid, client);
}
return client;
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
bool isTimed,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
status_t lStatus;
int lSessionId;
// client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
// but if someone uses binder directly they could bypass that and cause us to crash
if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
ALOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
PlaybackThread *effectThread = NULL;
if (thread == NULL) {
ALOGE("unknown output thread");
lStatus = BAD_VALUE;
goto Exit;
}
client = registerPid_l(pid);
ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output) {
// prevent same audio session on different output threads
uint32_t sessions = t->hasAudioSession(*sessionId);
if (sessions & PlaybackThread::TRACK_SESSION) {
ALOGE("createTrack() session ID %d already in use", *sessionId);
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;
}
}
ALOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
moveEffectChain_l(lSessionId, effectThread, thread, true);
}
}
if (lStatus == NO_ERROR) {
trackHandle = new TrackHandle(track);
} else {
// remove local strong reference to Client before deleting the Track so that the Client
// destructor is called by the TrackBase destructor with mLock held
client.clear();
track.clear();
}
Exit:
if(status) {
*status = lStatus;
}
return trackHandle;
}
uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("sampleRate() unknown thread %d", output);
return 0;
}
return thread->sampleRate();
}
int AudioFlinger::channelCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("channelCount() unknown thread %d", output);
return 0;
}
return thread->channelCount();
}
audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("format() unknown thread %d", output);
return AUDIO_FORMAT_INVALID;
}
return thread->format();
}
size_t AudioFlinger::frameCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("frameCount() unknown thread %d", output);
return 0;
}
return thread->frameCount();
}
uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("latency() unknown thread %d", output);
return 0;
}
return thread->latency();
}
status_t AudioFlinger::setMasterVolume(float value)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
float swmv = value;
// when hw supports master volume, don't scale in sw mixer
if (MVS_NONE != mMasterVolumeSupportLvl) {
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AutoMutex lock(mHardwareLock);
audio_hw_device_t *dev = mAudioHwDevs[i];
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (NULL != dev->set_master_volume) {
dev->set_master_volume(dev, value);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
swmv = 1.0;
}
Mutex::Autolock _l(mLock);
mMasterVolume = value;
mMasterVolumeSW = swmv;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
return NO_ERROR;
}
status_t AudioFlinger::setMode(audio_mode_t mode)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(mode) >= AUDIO_MODE_CNT) {
ALOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
{ // scope for the lock
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode);
mHardwareStatus = AUDIO_HW_IDLE;
}
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMode(mode);
}
return ret;
}
status_t AudioFlinger::setMicMute(bool state)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return false;
}
bool state = AUDIO_MODE_INVALID;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
// This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
mMasterMute = muted;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterMute(muted);
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
Mutex::Autolock _l(mLock);
return masterVolume_l();
}
float AudioFlinger::masterVolumeSW() const
{
Mutex::Autolock _l(mLock);
return masterVolumeSW_l();
}
bool AudioFlinger::masterMute() const
{
Mutex::Autolock _l(mLock);
return masterMute_l();
}
float AudioFlinger::masterVolume_l() const
{
if (MVS_FULL == mMasterVolumeSupportLvl) {
float ret_val;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
assert(NULL != mPrimaryHardwareDev);
assert(NULL != mPrimaryHardwareDev->get_master_volume);
mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val);
mHardwareStatus = AUDIO_HW_IDLE;
return ret_val;
}
return mMasterVolume;
}
status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
ALOGE("setStreamVolume() invalid stream %d", stream);
return BAD_VALUE;
}
AutoMutex lock(mLock);
PlaybackThread *thread = NULL;
if (output) {
thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
}
mStreamTypes[stream].volume = value;
if (thread == NULL) {
for (size_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(audio_stream_type_t stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
ALOGE("setStreamMute() invalid stream %d", stream);
return BAD_VALUE;
}
AutoMutex lock(mLock);
mStreamTypes[stream].mute = muted;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
return NO_ERROR;
}
float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return 0.0f;
}
AutoMutex lock(mLock);
float volume;
if (output) {
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return 0.0f;
}
volume = thread->streamVolume(stream);
} else {
volume = streamVolume_l(stream);
}
return volume;
}
bool AudioFlinger::streamMute(audio_stream_type_t stream) const
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return true;
}
AutoMutex lock(mLock);
return streamMute_l(stream);
}
status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
status_t result;
ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d",
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
// ioHandle == 0 means the parameters are global to the audio hardware interface
if (ioHandle == 0) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
status_t final_result = NO_ERROR;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
result = dev->set_parameters(dev, keyValuePairs.string());
final_result = result ?: final_result;
}
mHardwareStatus = AUDIO_HW_IDLE;
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
AudioParameter param = AudioParameter(keyValuePairs);
String8 value;
if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
Mutex::Autolock _l(mLock);
bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
if (mBtNrecIsOff != btNrecIsOff) {
for (size_t i = 0; i < mRecordThreads.size(); i++) {
sp<RecordThread> thread = mRecordThreads.valueAt(i);
RecordThread::RecordTrack *track = thread->track();
if (track != NULL) {
audio_devices_t device = (audio_devices_t)(
thread->device() & AUDIO_DEVICE_IN_ALL);
bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
thread->setEffectSuspended(FX_IID_AEC,
suspend,
track->sessionId());
thread->setEffectSuspended(FX_IID_NS,
suspend,
track->sessionId());
}
}
mBtNrecIsOff = btNrecIsOff;
}
}
return final_result;
}
// hold a strong ref on thread in case closeOutput() or closeInput() is called
// and the thread is exited once the lock is released
sp<ThreadBase> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(ioHandle);
if (thread == NULL) {
thread = checkRecordThread_l(ioHandle);
} else if (thread == primaryPlaybackThread_l()) {
// indicate output device change to all input threads for pre processing
AudioParameter param = AudioParameter(keyValuePairs);
int value;
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
}
}
}
}
if (thread != 0) {
return thread->setParameters(keyValuePairs);
}
return BAD_VALUE;
}
String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d",
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
if (ioHandle == 0) {
String8 out_s8;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
char *s = dev->get_parameters(dev, keys.string());
out_s8 += String8(s ? s : "");
free(s);
}
return out_s8;
}
Mutex::Autolock _l(mLock);
PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
if (playbackThread != NULL) {
return playbackThread->getParameters(keys);
}
RecordThread *recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getParameters(keys);
}
return String8("");
}
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
mHardwareStatus = AUDIO_HW_IDLE;
return size;
}
unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
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)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const
{
status_t status;
Mutex::Autolock _l(mLock);
PlaybackThread *playbackThread = checkPlaybackThread_l(output);
if (playbackThread != NULL) {
return playbackThread->getRenderPosition(halFrames, dspFrames);
}
return BAD_VALUE;
}
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{
Mutex::Autolock _l(mLock);
pid_t pid = IPCThreadState::self()->getCallingPid();
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
pid);
ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
mNotificationClients.add(pid, notificationClient);
sp<IBinder> binder = client->asBinder();
binder->linkToDeath(notificationClient);
// the config change is always sent from playback or record threads to avoid deadlock
// with AudioSystem::gLock
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
}
}
}
void AudioFlinger::removeNotificationClient(pid_t pid)
{
Mutex::Autolock _l(mLock);
ssize_t index = mNotificationClients.indexOfKey(pid);
if (index >= 0) {
sp <NotificationClient> client = mNotificationClients.valueFor(pid);
ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
mNotificationClients.removeItem(pid);
}
ALOGV("%d died, releasing its sessions", pid);
size_t num = mAudioSessionRefs.size();
bool removed = false;
for (size_t i = 0; i< num; ) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
ALOGV(" pid %d @ %d", ref->pid, i);
if (ref->pid == pid) {
ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid);
mAudioSessionRefs.removeAt(i);
delete ref;
removed = true;
num--;
} else {
i++;
}
}
if (removed) {
purgeStaleEffects_l();
}
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2)
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
param2);
}
}
// removeClient_l() must be called with AudioFlinger::mLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// ----------------------------------------------------------------------------
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
uint32_t device, type_t type)
: Thread(false),
mType(type),
mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
// mChannelMask
mChannelCount(0),
mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
mParamStatus(NO_ERROR),
mStandby(false), mId(id),
mDevice(device),
mDeathRecipient(new PMDeathRecipient(this))
{
}
AudioFlinger::ThreadBase::~ThreadBase()
{
mParamCond.broadcast();
// do not lock the mutex in destructor
releaseWakeLock_l();
if (mPowerManager != 0) {
sp<IBinder> binder = mPowerManager->asBinder();
binder->unlinkToDeath(mDeathRecipient);
}
}
void AudioFlinger::ThreadBase::exit()
{
ALOGV("ThreadBase::exit");
{
// This lock prevents the following race in thread (uniprocessor for illustration):
// if (!exitPending()) {
// // context switch from here to exit()
// // exit() calls requestExit(), what exitPending() observes
// // exit() calls signal(), which is dropped since no waiters
// // context switch back from exit() to here
// mWaitWorkCV.wait(...);
// // now thread is hung
// }
AutoMutex lock(mLock);
requestExit();
mWaitWorkCV.signal();
}
// When Thread::requestExitAndWait is made virtual and this method is renamed to
// "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
requestExitAndWait();
}
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
status_t status;
ALOGV("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, kSetParametersTimeoutNs) == NO_ERROR) {
status = mParamStatus;
mWaitWorkCV.signal();
} else {
status = TIMED_OUT;
}
return status;
}
void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
{
Mutex::Autolock _l(mLock);
sendConfigEvent_l(event, param);
}
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
ConfigEvent configEvent;
configEvent.mEvent = event;
configEvent.mParam = param;
mConfigEvents.add(configEvent);
ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
}
void AudioFlinger::ThreadBase::processConfigEvents()
{
mLock.lock();
while(!mConfigEvents.isEmpty()) {
ALOGV("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();
mLock.lock();
}
mLock.unlock();
}
status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
bool locked = tryLock(mLock);
if (!locked) {
snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
write(fd, buffer, strlen(buffer));
}
snprintf(buffer, SIZE, "standby: %d\n", mStandby);
result.append(buffer);
snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
result.append(buffer);
snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
snprintf(buffer, SIZE, "Frame size: %u\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;
}
status_t AudioFlinger::ThreadBase::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;
}
void AudioFlinger::ThreadBase::acquireWakeLock()
{
Mutex::Autolock _l(mLock);
acquireWakeLock_l();
}
void AudioFlinger::ThreadBase::acquireWakeLock_l()
{
if (mPowerManager == 0) {
// use checkService() to avoid blocking if power service is not up yet
sp<IBinder> binder =
defaultServiceManager()->checkService(String16("power"));
if (binder == 0) {
ALOGW("Thread %s cannot connect to the power manager service", mName);
} else {
mPowerManager = interface_cast<IPowerManager>(binder);
binder->linkToDeath(mDeathRecipient);
}
}
if (mPowerManager != 0) {
sp<IBinder> binder = new BBinder();
status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK,
binder,
String16(mName));
if (status == NO_ERROR) {
mWakeLockToken = binder;
}
ALOGV("acquireWakeLock_l() %s status %d", mName, status);
}
}
void AudioFlinger::ThreadBase::releaseWakeLock()
{
Mutex::Autolock _l(mLock);
releaseWakeLock_l();
}
void AudioFlinger::ThreadBase::releaseWakeLock_l()
{
if (mWakeLockToken != 0) {
ALOGV("releaseWakeLock_l() %s", mName);
if (mPowerManager != 0) {
mPowerManager->releaseWakeLock(mWakeLockToken, 0);
}
mWakeLockToken.clear();
}
}
void AudioFlinger::ThreadBase::clearPowerManager()
{
Mutex::Autolock _l(mLock);
releaseWakeLock_l();
mPowerManager.clear();
}
void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
thread->clearPowerManager();
}
ALOGW("power manager service died !!!");
}
void AudioFlinger::ThreadBase::setEffectSuspended(
const effect_uuid_t *type, bool suspend, int sessionId)
{
Mutex::Autolock _l(mLock);
setEffectSuspended_l(type, suspend, sessionId);
}
void AudioFlinger::ThreadBase::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend, int sessionId)
{
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
if (type != NULL) {
chain->setEffectSuspended_l(type, suspend);
} else {
chain->setEffectSuspendedAll_l(suspend);
}
}
updateSuspendedSessions_l(type, suspend, sessionId);
}
void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
{
ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
if (index < 0) {
return;
}
KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects =
mSuspendedSessions.editValueAt(index);
for (size_t i = 0; i < sessionEffects.size(); i++) {
sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
for (int j = 0; j < desc->mRefCount; j++) {
if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
chain->setEffectSuspendedAll_l(true);
} else {
ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
desc->mType.timeLow);
chain->setEffectSuspended_l(&desc->mType, true);
}
}
}
}
void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
bool suspend,
int sessionId)
{
ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
if (suspend) {
if (index >= 0) {
sessionEffects = mSuspendedSessions.editValueAt(index);
} else {
mSuspendedSessions.add(sessionId, sessionEffects);
}
} else {
if (index < 0) {
return;
}
sessionEffects = mSuspendedSessions.editValueAt(index);
}
int key = EffectChain::kKeyForSuspendAll;
if (type != NULL) {
key = type->timeLow;
}
index = sessionEffects.indexOfKey(key);
sp <SuspendedSessionDesc> desc;
if (suspend) {
if (index >= 0) {
desc = sessionEffects.valueAt(index);
} else {
desc = new SuspendedSessionDesc();
if (type != NULL) {
memcpy(&desc->mType, type, sizeof(effect_uuid_t));
}
sessionEffects.add(key, desc);
ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key);
}
desc->mRefCount++;
} else {
if (index < 0) {
return;
}
desc = sessionEffects.valueAt(index);
if (--desc->mRefCount == 0) {
ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key);
sessionEffects.removeItemsAt(index);
if (sessionEffects.isEmpty()) {
ALOGV("updateSuspendedSessions_l() restore removing session %d",
sessionId);
mSuspendedSessions.removeItem(sessionId);
}
}
}
if (!sessionEffects.isEmpty()) {
mSuspendedSessions.replaceValueFor(sessionId, sessionEffects);
}
}
void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled,
int sessionId)
{
Mutex::Autolock _l(mLock);
checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
}
void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
bool enabled,
int sessionId)
{
if (mType != RECORD) {
// suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
// another session. This gives the priority to well behaved effect control panels
// and applications not using global effects.
if (sessionId != AUDIO_SESSION_OUTPUT_MIX) {
setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
}
}
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
chain->checkSuspendOnEffectEnabled(effect, enabled);
}
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
uint32_t device,
type_t type)
: ThreadBase(audioFlinger, id, device, type),
mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
// Assumes constructor is called by AudioFlinger with it's mLock held,
// but it would be safer to explicitly pass initial masterMute as parameter
mMasterMute(audioFlinger->masterMute_l()),
// mStreamTypes[] initialized in constructor body
mOutput(output),
// Assumes constructor is called by AudioFlinger with it's mLock held,
// but it would be safer to explicitly pass initial masterVolume as parameter
mMasterVolume(audioFlinger->masterVolumeSW_l()),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
snprintf(mName, kNameLength, "AudioOut_%d", id);
readOutputParameters();
// mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
// There is no AUDIO_STREAM_MIN, and ++ operator does not compile
for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
stream = (audio_stream_type_t) (stream + 1)) {
mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
// initialized by stream_type_t default constructor
// mStreamTypes[stream].valid = true;
}
// mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
// because mAudioFlinger doesn't have one to copy from
}
AudioFlinger::PlaybackThread::~PlaybackThread()
{
delete [] mMixBuffer;
}
status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
dumpEffectChains(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
sp<Track> track = mActiveTracks[i].promote();
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
result.append(buffer);
snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
result.append(buffer);
snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
result.append(buffer);
snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended);
result.append(buffer);
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()
{
status_t status = initCheck();
if (status == NO_ERROR) {
ALOGI("AudioFlinger's thread %p ready to run", this);
} else {
ALOGE("No working audio driver found.");
}
return status;
}
void AudioFlinger::PlaybackThread::onFirstRef()
{
run(mName, 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,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
bool isTimed,
status_t *status)
{
sp<Track> track;
status_t lStatus;
if (mType == DIRECT) {
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
"for output %p with format %d",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
goto Exit;
}
}
} else {
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (sampleRate > mSampleRate*2) {
ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
}
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGE("Audio driver not initialized.");
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(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0) {
uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
if (sessionId == t->sessionId() && strategy != actual) {
ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
strategy, actual);
lStatus = BAD_VALUE;
goto Exit;
}
}
}
if (!isTimed) {
track = new Track(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId);
} else {
track = TimedTrack::create(this, client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, sessionId);
}
if (track == NULL || track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
mTracks.add(track);
sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
// invalidate track immediately if the stream type was moved to another thread since
// createTrack() was called by the client process.
if (!mStreamTypes[streamType].valid) {
ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
this, streamType);
android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
}
}
lStatus = NO_ERROR;
Exit:
if(status) {
*status = lStatus;
}
return track;
}
uint32_t AudioFlinger::PlaybackThread::latency() const
{
Mutex::Autolock _l(mLock);
if (initCheck() == NO_ERROR) {
return mOutput->stream->get_latency(mOutput->stream);
} else {
return 0;
}
}
void AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
Mutex::Autolock _l(mLock);
mMasterVolume = value;
}
void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
Mutex::Autolock _l(mLock);
setMasterMute_l(muted);
}
void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
Mutex::Autolock _l(mLock);
mStreamTypes[stream].volume = value;
}
void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
Mutex::Autolock _l(mLock);
mStreamTypes[stream].mute = muted;
}
float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
{
Mutex::Autolock _l(mLock);
return mStreamTypes[stream].volume;
}
// 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) {
ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
chain->incActiveTrackCnt();
}
}
status = NO_ERROR;
}
ALOGV("mWaitWorkCV.broadcast");
mWaitWorkCV.broadcast();
return status;
}
// destroyTrack_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
{
track->mState = TrackBase::TERMINATED;
if (mActiveTracks.indexOf(track) < 0) {
removeTrack_l(track);
}
}
void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track)
{
mTracks.remove(track);
deleteTrackName_l(track->name());
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
chain->decTrackCnt();
}
}
String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
String8 out_s8 = String8("");
char *s;
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
return out_s8;
}
s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string());
out_s8 = String8(s);
free(s);
return out_s8;
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = NULL;
ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
switch (event) {
case AudioSystem::OUTPUT_OPENED:
case AudioSystem::OUTPUT_CONFIG_CHANGED:
desc.channels = mChannelMask;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
desc.frameCount = mFrameCount;
desc.latency = latency();
param2 = &desc;
break;
case AudioSystem::STREAM_CONFIG_CHANGED:
param2 = &param;
case AudioSystem::OUTPUT_CLOSED:
default:
break;
}
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
void AudioFlinger::PlaybackThread::readOutputParameters()
{
mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common);
mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
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 == NULL || dspFrames == NULL) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
return INVALID_OPERATION;
}
*halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common);
return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
}
uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId)
{
Mutex::Autolock _l(mLock);
uint32_t result = 0;
if (getEffectChain_l(sessionId) != 0) {
result = EFFECT_SESSION;
}
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
result |= TRACK_SESSION;
break;
}
}
return result;
}
uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId)
{
// session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
// it is moved to correct output by audio policy manager when A2DP is connected or disconnected
if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
for (size_t i = 0; i < mTracks.size(); i++) {
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
return AudioSystem::getStrategyForStream(track->streamType());
}
}
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
{
Mutex::Autolock _l(mLock);
return mOutput;
}
AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
{
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mOutput;
mOutput = NULL;
return output;
}
// this method must always be called either with ThreadBase mLock held or inside the thread loop
audio_stream_t* AudioFlinger::PlaybackThread::stream()
{
if (mOutput == NULL) {
return NULL;
}
return &mOutput->stream->common;
}
uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
{
// A2DP output latency is not due only to buffering capacity. It also reflects encoding,
// decoding and transfer time. So sleeping for half of the latency would likely cause
// underruns
if (audio_is_a2dp_device((audio_devices_t)mDevice)) {
return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000);
} else {
return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2;
}
}
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, uint32_t device, type_t type)
: PlaybackThread(audioFlinger, output, id, device, type),
mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)),
mPrevMixerStatus(MIXER_IDLE)
{
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
ALOGE("Invalid audio hardware channel count");
}
}
AudioFlinger::MixerThread::~MixerThread()
{
delete mAudioMixer;
}
bool AudioFlinger::MixerThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
mixer_state 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
// increase threshold again due to low power audio mode. The way this warning threshold is
// calculated and its usefulness should be reconsidered anyway.
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
nsecs_t lastWarning = 0;
bool longStandbyExit = false;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
uint32_t sleepTime = idleSleepTime;
uint32_t sleepTimeShift = 0;
Vector< sp<EffectChain> > effectChains;
#ifdef DEBUG_CPU_USAGE
ThreadCpuUsage cpu;
const CentralTendencyStatistics& stats = cpu.statistics();
#endif
acquireWakeLock();
while (!exitPending())
{
#ifdef DEBUG_CPU_USAGE
cpu.sampleAndEnable();
unsigned n = stats.n();
// cpu.elapsed() is expensive, so don't call it every loop
if ((n & 127) == 1) {
long long elapsed = cpu.elapsed();
if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
double perLoop = elapsed / (double) n;
double perLoop100 = perLoop * 0.01;
double mean = stats.mean();
double stddev = stats.stddev();
double minimum = stats.minimum();
double maximum = stats.maximum();
cpu.resetStatistics();
ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
elapsed * .000000001, n, perLoop * .000001,
mean * .001,
stddev * .001,
minimum * .001,
maximum * .001,
mean / perLoop100,
stddev / perLoop100,
minimum / perLoop100,
maximum / perLoop100);
}
}
#endif
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
// increase threshold again due to low power audio mode. The way this warning
// threshold is calculated and its usefulness should be reconsidered anyway.
maxPeriod = seconds(mFrameCount) / mSampleRate * 15;
activeSleepTime = activeSleepTimeUs();
idleSleepTime = idleSleepTimeUs();
}
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
mSuspended)) {
if (!mStandby) {
ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
}
if (!activeTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) break;
releaseWakeLock_l();
// wait until we have something to do...
ALOGV("MixerThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
ALOGV("MixerThread %p TID %d waking up", this, gettid());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
ALOGD("Silence is golden");
setMasterMute_l(true);
}
}
standbyTime = systemTime() + mStandbyTimeInNsecs;
sleepTime = idleSleepTime;
sleepTimeShift = 0;
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 (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// obtain the presentation timestamp of the next output buffer
int64_t pts;
status_t status = INVALID_OPERATION;
if (NULL != mOutput->stream->get_next_write_timestamp) {
status = mOutput->stream->get_next_write_timestamp(
mOutput->stream, &pts);
}
if (status != NO_ERROR) {
pts = AudioBufferProvider::kInvalidPTS;
}
// mix buffers...
mAudioMixer->process(pts);
// increase sleep time progressively when application underrun condition clears.
// Only increase sleep time if the mixer is ready for two consecutive times to avoid
// that a steady state of alternating ready/not ready conditions keeps the sleep time
// such that we would underrun the audio HAL.
if ((sleepTime == 0) && (sleepTimeShift > 0)) {
sleepTimeShift--;
}
sleepTime = 0;
standbyTime = systemTime() + mStandbyTimeInNsecs;
//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 >> sleepTimeShift;
if (sleepTime < kMinThreadSleepTimeUs) {
sleepTime = kMinThreadSleepTimeUs;
}
// reduce sleep time in case of consecutive application underruns to avoid
// starving the audio HAL. As activeSleepTimeUs() is larger than a buffer
// duration we would end up writing less data than needed by the audio HAL if
// the condition persists.
if (sleepTimeShift < kMaxThreadSleepTimeShift) {
sleepTimeShift++;
}
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 ||
(mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) {
memset (mMixBuffer, 0, mixBufferSize);
sleepTime = 0;
ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start");
}
// TODO add standby time extension fct of effect tail
}
if (mSuspended) {
sleepTime = suspendSleepTimeUs();
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
// enable changes in effect chain
unlockEffectChains(effectChains);
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
nsecs_t now = systemTime();
nsecs_t delta = now - mLastWriteTime;
if (!mStandby && delta > maxPeriod) {
mNumDelayedWrites++;
if ((now - lastWarning) > kWarningThrottleNs) {
ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
}
if (mStandby) {
longStandbyExit = true;
}
}
mStandby = false;
} else {
// enable changes in effect chain
unlockEffectChains(effectChains);
usleep(sleepTime);
}
// finally let go of all our tracks, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
tracksToRemove.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
if (!mStandby) {
mOutput->stream->common.standby(&mOutput->stream->common);
}
releaseWakeLock();
ALOGV("MixerThread %p exiting", this);
return false;
}
// prepareTracks_l() must be called with ThreadBase::mLock held
AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
{
mixer_state mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
size_t count = activeTracks.size();
size_t mixedTracks = 0;
size_t tracksWithEffect = 0;
float masterVolume = mMasterVolume;
bool masterMute = mMasterMute;
if (masterMute) {
masterVolume = 0;
}
// Delegate master volume control to effect in output mix effect chain if needed
sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (chain != 0) {
uint32_t v = (uint32_t)(masterVolume * (1 << 24));
chain->setVolume_l(&v, &v);
masterVolume = (float)((v + (1 << 23)) >> 24);
chain.clear();
}
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
// this const just means the local variable doesn't change
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
int name = track->name();
// make sure that we have enough frames to mix one full buffer.
// enforce this condition only once to enable draining the buffer in case the client
// app does not call stop() and relies on underrun to stop:
// hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
// during last round
uint32_t minFrames = 1;
if (!track->isStopped() && !track->isPausing() &&
(mPrevMixerStatus == MIXER_TRACKS_READY)) {
if (t->sampleRate() == (int)mSampleRate) {
minFrames = mFrameCount;
} else {
// +1 for rounding and +1 for additional sample needed for interpolation
minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
// add frames already consumed but not yet released by the resampler
// because cblk->framesReady() will include these frames
minFrames += mAudioMixer->getUnreleasedFrames(track->name());
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
// of unreleased frames after each pass, but just in case...
ALOG_ASSERT(minFrames <= cblk->frameCount);
}
}
if ((track->framesReady() >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
//ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", 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 {
ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
name, track->sessionId());
}
}
int param = AudioMixer::VOLUME;
if (track->mFillingUpStatus == Track::FS_FILLED) {
// no ramp for the first volume setting
track->mFillingUpStatus = Track::FS_ACTIVE;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
}
mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
param = AudioMixer::RAMP_VOLUME;
}
// compute volume for this track
uint32_t vl, vr, va;
if (track->isMuted() || track->isPausing() ||
mStreamTypes[track->streamType()].mute) {
vl = vr = va = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
// read original volumes with volume control
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
uint32_t vlr = cblk->getVolumeLR();
vl = vlr & 0xFFFF;
vr = vlr >> 16;
// track volumes come from shared memory, so can't be trusted and must be clamped
if (vl > MAX_GAIN_INT) {
ALOGV("Track left volume out of range: %04X", vl);
vl = MAX_GAIN_INT;
}
if (vr > MAX_GAIN_INT) {
ALOGV("Track right volume out of range: %04X", vr);
vr = MAX_GAIN_INT;
}
// now apply the master volume and stream type volume
vl = (uint32_t)(v * vl) << 12;
vr = (uint32_t)(v * vr) << 12;
// assuming master volume and stream type volume each go up to 1.0,
// vl and vr are now in 8.24 format
uint16_t sendLevel = cblk->getSendLevel_U4_12();
// send level comes from shared memory and so may be corrupt
if (sendLevel > MAX_GAIN_INT) {
ALOGV("Track send level out of range: %04X", sendLevel);
sendLevel = MAX_GAIN_INT;
}
va = (uint32_t)(v * 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
// This additional clamping is needed in case chain->setVolume_l() overshot
vl = (vl + (1 << 11)) >> 12;
if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT;
vr = (vr + (1 << 11)) >> 12;
if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT;
if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for -
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(name, track);
mAudioMixer->enable(name);
mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
mAudioMixer->setParameter(
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(cblk->sampleRate));
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
// reset retry count
track->mRetryCount = kMaxTrackRetries;
// If one track is ready, set the mixer ready if:
// - the mixer was not ready during previous round OR
// - no other track is not ready
if (mPrevMixerStatus != MIXER_TRACKS_READY ||
mixerStatus != MIXER_TRACKS_ENABLED) {
mixerStatus = MIXER_TRACKS_READY;
}
} else {
//ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", 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) {
ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun
android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
// If one track is not ready, mark the mixer also not ready if:
// - the mixer was ready during previous round OR
// - no other track is ready
} else if (mPrevMixerStatus == MIXER_TRACKS_READY ||
mixerStatus != MIXER_TRACKS_READY) {
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
mAudioMixer->disable(name);
}
}
// remove all the tracks that need to be...
count = tracksToRemove->size();
if (CC_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) {
ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
chain->decActiveTrackCnt();
}
}
if (track->isTerminated()) {
removeTrack_l(track);
}
}
}
// mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to
// mix buffer and track effects will accumulate into it
if (mixedTracks != 0 && mixedTracks == tracksWithEffect) {
memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t));
}
mPrevMixerStatus = mixerStatus;
return mixerStatus;
}
void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
{
ALOGV ("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->streamType() == streamType) {
android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
t->mCblk->cv.signal();
}
}
}
void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
{
ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
this, streamType, valid);
Mutex::Autolock _l(mLock);
mStreamTypes[streamType].valid = valid;
}
// 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)
{
ALOGV("remove track (%d) and delete from mixer", name);
mAudioMixer->deleteTrackName(name);
}
// checkForNewParameters_l() must be called with ThreadBase::mLock held
bool AudioFlinger::MixerThread::checkForNewParameters_l()
{
bool reconfig = false;
while (!mNewParameters.isEmpty()) {
status_t status = NO_ERROR;
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
if (value != AUDIO_CHANNEL_OUT_STEREO) {
status = BAD_VALUE;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
// when changing the audio output device, call addBatteryData to notify
// the change
if ((int)mDevice != value) {
uint32_t params = 0;
// check whether speaker is on
if (value & AUDIO_DEVICE_OUT_SPEAKER) {
params |= IMediaPlayerService::kBatteryDataSpeakerOn;
}
int deviceWithoutSpeaker
= AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
// check if any other device (except speaker) is on
if (value & deviceWithoutSpeaker ) {
params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
}
if (params != 0) {
addBatteryData(params);
}
}
// forward device change to effects that have requested to be
// aware of attached audio device.
mDevice = (uint32_t)value;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevice_l(mDevice);
}
}
if (status == NO_ERROR) {
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
// for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
mAudioMixer = NULL;
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();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
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::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, audio_io_handle_t id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device, DIRECT)
// mLeftVolFloat, mRightVolFloat
// mLeftVolShort, mRightVolShort
{
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
{
// Do not apply volume on compressed audio
if (!audio_is_linear_pcm(mFormat)) {
return;
}
// convert to signed 16 bit before volume calculation
if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
uint8_t *src = (uint8_t *)mMixBuffer + count-1;
int16_t *dst = mMixBuffer + count-1;
while(count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
}
size_t frameCount = mFrameCount;
int16_t *out = mMixBuffer;
if (ramp) {
if (mChannelCount == 1) {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out++;
vl += vlInc;
} while (--frameCount);
} else {
int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16;
int32_t vlInc = d / (int32_t)frameCount;
d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16;
int32_t vrInc = d / (int32_t)frameCount;
int32_t vl = ((int32_t)mLeftVolShort << 16);
int32_t vr = ((int32_t)mRightVolShort << 16);
do {
out[0] = clamp16(mul(out[0], vl >> 16) >> 12);
out[1] = clamp16(mul(out[1], vr >> 16) >> 12);
out += 2;
vl += vlInc;
vr += vrInc;
} while (--frameCount);
}
} else {
if (mChannelCount == 1) {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out++;
} while (--frameCount);
} else {
do {
out[0] = clamp16(mul(out[0], leftVol) >> 12);
out[1] = clamp16(mul(out[1], rightVol) >> 12);
out += 2;
} while (--frameCount);
}
}
// convert back to unsigned 8 bit after volume calculation
if (mFormat == AUDIO_FORMAT_PCM_8_BIT) {
size_t count = mFrameCount * mChannelCount;
int16_t *src = mMixBuffer;
uint8_t *dst = (uint8_t *)mMixBuffer;
while(count--) {
*dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
}
}
mLeftVolShort = leftVol;
mRightVolShort = rightVol;
}
bool AudioFlinger::DirectOutputThread::threadLoop()
{
mixer_state mixerStatus = MIXER_IDLE;
sp<Track> trackToRemove;
sp<Track> activeTrack;
nsecs_t standbyTime = systemTime();
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);
acquireWakeLock();
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 (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
mSuspended)) {
// wait until we have something to do...
if (!mStandby) {
ALOGV("Audio hardware entering standby, mixer %p", this);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
}
if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
if (exitPending()) break;
releaseWakeLock_l();
ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid());
acquireWakeLock_l();
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
ALOGD("Silence is golden");
setMasterMute_l(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())
{
//ALOGV("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->streamType()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
uint32_t vlr = cblk->getVolumeLR();
float v_clamped = v * (vlr & 0xFFFF);
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = v_clamped/MAX_GAIN;
v_clamped = v * (vlr >> 16);
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = v_clamped/MAX_GAIN;
}
if (left != mLeftVolFloat || right != mRightVolFloat) {
mLeftVolFloat = left;
mRightVolFloat = right;
// If audio HAL implements volume control,
// force software volume to nominal value
if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) {
left = 1.0f;
right = 1.0f;
}
// Convert volumes from float to 8.24
uint32_t vl = (uint32_t)(left * (1 << 24));
uint32_t vr = (uint32_t)(right * (1 << 24));
// Delegate volume control to effect in track effect chain if needed
// only one effect chain can be present on DirectOutputThread, so if
// there is one, the track is connected to it
if (!effectChains.isEmpty()) {
// Do not ramp volume if volume is controlled by effect
if(effectChains[0]->setVolume_l(&vl, &vr)) {
rampVolume = false;
}
}
// Convert volumes from 8.24 to 4.12 format
uint32_t v_clamped = (vl + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
leftVol = (uint16_t)v_clamped;
v_clamped = (vr + (1 << 11)) >> 12;
if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
rightVol = (uint16_t)v_clamped;
} else {
leftVol = mLeftVolShort;
rightVol = mRightVolShort;
rampVolume = false;
}
// reset retry count
track->mRetryCount = kMaxTrackRetriesDirect;
activeTrack = t;
mixerStatus = MIXER_TRACKS_READY;
} else {
//ALOGV("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) {
ALOGV("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 (CC_UNLIKELY(trackToRemove != 0)) {
mActiveTracks.remove(trackToRemove);
if (!effectChains.isEmpty()) {
ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
trackToRemove->sessionId());
effectChains[0]->decActiveTrackCnt();
}
if (trackToRemove->isTerminated()) {
removeTrack_l(trackToRemove);
}
}
lockEffectChains_l(effectChains);
}
if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
AudioBufferProvider::Buffer buffer;
size_t frameCount = mFrameCount;
int8_t *curBuf = (int8_t *)mMixBuffer;
// output audio to hardware
while (frameCount) {
buffer.frameCount = frameCount;
activeTrack->getNextBuffer(&buffer,
AudioBufferProvider::kInvalidPTS);
if (CC_UNLIKELY(buffer.raw == NULL)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
frameCount -= buffer.frameCount;
curBuf += buffer.frameCount * mFrameSize;
activeTrack->releaseBuffer(&buffer);
}
sleepTime = 0;
standbyTime = systemTime() + standbyDelay;
} else {
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_ENABLED) {
sleepTime = activeSleepTime;
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) {
memset (mMixBuffer, 0, mFrameCount * mFrameSize);
sleepTime = 0;
}
}
if (mSuspended) {
sleepTime = suspendSleepTimeUs();
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_READY) {
applyVolume(leftVol, rightVol, rampVolume);
}
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
unlockEffectChains(effectChains);
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize);
if (bytesWritten < 0) mBytesWritten -= mixBufferSize;
mNumWrites++;
mInWrite = false;
mStandby = false;
} else {
unlockEffectChains(effectChains);
usleep(sleepTime);
}
// finally let go of removed track, without the lock held
// since we can't guarantee the destructors won't acquire that
// same lock.
trackToRemove.clear();
activeTrack.clear();
// Effect chains will be actually deleted here if they were removed from
// mEffectChains list during mixing or effects processing
effectChains.clear();
}
if (!mStandby) {
mOutput->stream->common.standby(&mOutput->stream->common);
}
releaseWakeLock();
ALOGV("DirectOutputThread %p exiting", this);
return false;
}
// getTrackName_l() must be called with ThreadBase::mLock held
int AudioFlinger::DirectOutputThread::getTrackName_l()
{
return 0;
}
// deleteTrackName_l() must be called with ThreadBase::mLock held
void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
{
}
// checkForNewParameters_l() must be called with ThreadBase::mLock held
bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
{
bool reconfig = false;
while (!mNewParameters.isEmpty()) {
status_t status = NO_ERROR;
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be garantied
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
} else {
reconfig = true;
}
}
if (status == NO_ERROR) {
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
readOutputParameters();
sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
}
}
mNewParameters.removeAt(0);
mParamStatus = status;
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs()
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
time = PlaybackThread::activeSleepTimeUs();
} else {
time = 10000;
}
return time;
}
uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs()
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2;
} else {
time = 10000;
}
return time;
}
uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs()
{
uint32_t time;
if (audio_is_linear_pcm(mFormat)) {
time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000);
} else {
time = 10000;
}
return time;
}
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
: MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
mWaitTimeMs(UINT_MAX)
{
addOutputTrack(mainThread);
}
AudioFlinger::DuplicatingThread::~DuplicatingThread()
{
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
}
bool AudioFlinger::DuplicatingThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
mixer_state 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;
acquireWakeLock();
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 (CC_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;
releaseWakeLock_l();
ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
ALOGV("DuplicatingThread %p TID %d waking up", this, gettid());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
ALOGD("Silence is golden");
setMasterMute_l(true);
}
}
standbyTime = systemTime() + mStandbyTimeInNsecs;
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 (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
if (outputsReady(outputTracks)) {
mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
} else {
memset(mMixBuffer, 0, mixBufferSize);
}
sleepTime = 0;
writeFrames = mFrameCount;
} else {
if (sleepTime == 0) {
if (mixerStatus == MIXER_TRACKS_ENABLED) {
sleepTime = activeSleepTime;
} else {
sleepTime = idleSleepTime;
}
} else if (mBytesWritten != 0) {
// flush remaining overflow buffers in output tracks
for (size_t i = 0; i < outputTracks.size(); i++) {
if (outputTracks[i]->isActive()) {
sleepTime = 0;
writeFrames = 0;
memset(mMixBuffer, 0, mixBufferSize);
break;
}
}
}
}
if (mSuspended) {
sleepTime = suspendSleepTimeUs();
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
// enable changes in effect chain
unlockEffectChains(effectChains);
standbyTime = systemTime() + mStandbyTimeInNsecs;
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();
}
releaseWakeLock();
return false;
}
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
// FIXME explain this formula
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
OutputTrack *outputTrack = new OutputTrack(thread,
this,
mSampleRate,
mFormat,
mChannelMask,
frameCount);
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
mOutputTracks.add(outputTrack);
ALOGV("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() == thread) {
mOutputTracks[i]->destroy();
mOutputTracks.removeAt(i);
updateWaitTime();
return;
}
}
ALOGV("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 != 0) {
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) {
ALOGW("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()) {
ALOGV("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(
ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId)
: RefBase(),
mThread(thread),
mClient(client),
mCblk(NULL),
// mBuffer
// mBufferEnd
mFrameCount(0),
mState(IDLE),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK),
mSessionId(sessionId)
// mChannelCount
// mChannelMask
{
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
uint8_t channelCount = popcount(channelMask);
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
if (sharedBuffer == 0) {
size += bufferSize;
}
if (client != NULL) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
if (mCblk != NULL) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mChannelCount = channelCount;
mChannelMask = channelMask;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer (other flags are cleared)
mCblk->flags = CBLK_UNDERRUN_ON;
} else {
mBuffer = sharedBuffer->pointer();
}
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
} else {
ALOGE("not enough memory for AudioTrack size=%u", size);
client->heap()->dump("AudioTrack");
return;
}
} else {
mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
// construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mChannelCount = channelCount;
mChannelMask = channelMask;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer (other flags are cleared)
mCblk->flags = CBLK_UNDERRUN_ON;
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
if (mCblk != NULL) {
if (mClient == 0) {
delete mCblk;
} else {
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
}
}
mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
if (mClient != 0) {
// Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
// If the client's reference count drops to zero, the associated destructor
// must run with AudioFlinger lock held. Thus the explicit clear() rather than
// relying on the automatic clear() at end of scope.
mClient.clear();
}
}
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = NULL;
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) {
ALOGV("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);
ALOGV("TrackBase::reset");
}
int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
size_t frameSize = cblk->frameSize;
int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
int8_t *bufferEnd = bufferStart + frames * frameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
return NULL;
}
return bufferStart;
}
// ----------------------------------------------------------------------------
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
: TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
if (mCblk != NULL) {
if (thread != NULL) {
mName = thread->getTrackName_l();
mMainBuffer = thread->mixBuffer();
}
ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
if (mName < 0) {
ALOGE("no more track names available");
}
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 = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
}
}
AudioFlinger::PlaybackThread::Track::~Track()
{
ALOGV("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
// destructor 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(), mStreamType, mSessionId);
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
}
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)
{
uint32_t vlr = mCblk->getVolumeLR();
snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
(mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
mChannelMask,
mSessionId,
mFrameCount,
mState,
mMute,
mFillingUpStatus,
mCblk->sampleRate,
vlr & 0xFFFF,
vlr >> 16,
mCblk->server,
mCblk->user,
(int)mMainBuffer,
(int)mAuxBuffer);
}
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
AudioBufferProvider::Buffer* buffer, int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
ALOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesReady = cblk->framesReady();
if (CC_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 == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = NULL;
buffer->frameCount = 0;
ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
return NOT_ENOUGH_DATA;
}
uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{
return mCblk->framesReady();
}
bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
if (framesReady() >= mCblk->frameCount ||
(mCblk->flags & CBLK_FORCEREADY_MSK)) {
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
return true;
}
return false;
}
status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid)
{
status_t status = NO_ERROR;
ALOGV("start(%d), calling pid %d session %d tid %d",
mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
track_state state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (mState == PAUSED) {
mState = TrackBase::RESUMING;
ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
} else {
mState = TrackBase::ACTIVE;
ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
}
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
if (status == NO_ERROR) {
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart);
}
}
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()
{
ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
track_state 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();
}
ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
}
}
}
void AudioFlinger::PlaybackThread::Track::pause()
{
ALOGV("pause(%d), calling pid %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;
ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
}
}
}
}
void AudioFlinger::PlaybackThread::Track::flush()
{
ALOGV("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;
// do not reset the track if it is still in the process of being stopped or paused.
// this will be done by prepareTracks_l() when the track is stopped.
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (playbackThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
}
}
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
android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
}
void AudioFlinger::PlaybackThread::Track::mute(bool muted)
{
mMute = muted;
}
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;
}
// timed audio tracks
sp<AudioFlinger::PlaybackThread::TimedTrack>
AudioFlinger::PlaybackThread::TimedTrack::create(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId) {
if (!client->reserveTimedTrack())
return NULL;
sp<TimedTrack> track = new TimedTrack(
thread, client, streamType, sampleRate, format, channelMask, frameCount,
sharedBuffer, sessionId);
if (track == NULL) {
client->releaseTimedTrack();
return NULL;
}
return track;
}
AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
PlaybackThread *thread,
const sp<Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
: Track(thread, client, streamType, sampleRate, format, channelMask,
frameCount, sharedBuffer, sessionId),
mTimedSilenceBuffer(NULL),
mTimedSilenceBufferSize(0),
mTimedAudioOutputOnTime(false),
mMediaTimeTransformValid(false)
{
LocalClock lc;
mLocalTimeFreq = lc.getLocalFreq();
mLocalTimeToSampleTransform.a_zero = 0;
mLocalTimeToSampleTransform.b_zero = 0;
mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
&mLocalTimeToSampleTransform.a_to_b_denom);
}
AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
mClient->releaseTimedTrack();
delete [] mTimedSilenceBuffer;
}
status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
size_t size, sp<IMemory>* buffer) {
Mutex::Autolock _l(mTimedBufferQueueLock);
trimTimedBufferQueue_l();
// lazily initialize the shared memory heap for timed buffers
if (mTimedMemoryDealer == NULL) {
const int kTimedBufferHeapSize = 512 << 10;
mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
"AudioFlingerTimed");
if (mTimedMemoryDealer == NULL)
return NO_MEMORY;
}
sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
if (newBuffer == NULL) {
newBuffer = mTimedMemoryDealer->allocate(size);
if (newBuffer == NULL)
return NO_MEMORY;
}
*buffer = newBuffer;
return NO_ERROR;
}
// caller must hold mTimedBufferQueueLock
void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
int64_t mediaTimeNow;
{
Mutex::Autolock mttLock(mMediaTimeTransformLock);
if (!mMediaTimeTransformValid)
return;
int64_t targetTimeNow;
status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
? mCCHelper.getCommonTime(&targetTimeNow)
: mCCHelper.getLocalTime(&targetTimeNow);
if (OK != res)
return;
if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
&mediaTimeNow)) {
return;
}
}
size_t trimIndex;
for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) {
if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow)
break;
}
if (trimIndex) {
mTimedBufferQueue.removeItemsAt(0, trimIndex);
}
}
status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
const sp<IMemory>& buffer, int64_t pts) {
{
Mutex::Autolock mttLock(mMediaTimeTransformLock);
if (!mMediaTimeTransformValid)
return INVALID_OPERATION;
}
Mutex::Autolock _l(mTimedBufferQueueLock);
mTimedBufferQueue.add(TimedBuffer(buffer, pts));
return NO_ERROR;
}
status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
target);
if (!(target == TimedAudioTrack::LOCAL_TIME ||
target == TimedAudioTrack::COMMON_TIME)) {
return BAD_VALUE;
}
Mutex::Autolock lock(mMediaTimeTransformLock);
mMediaTimeTransform = xform;
mMediaTimeTransformTarget = target;
mMediaTimeTransformValid = true;
return NO_ERROR;
}
#define min(a, b) ((a) < (b) ? (a) : (b))
// implementation of getNextBuffer for tracks whose buffers have timestamps
status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
AudioBufferProvider::Buffer* buffer, int64_t pts)
{
if (pts == AudioBufferProvider::kInvalidPTS) {
buffer->raw = 0;
buffer->frameCount = 0;
return INVALID_OPERATION;
}
Mutex::Autolock _l(mTimedBufferQueueLock);
while (true) {
// if we have no timed buffers, then fail
if (mTimedBufferQueue.isEmpty()) {
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
// calculate the PTS of the head of the timed buffer queue expressed in
// local time
int64_t headLocalPTS;
{
Mutex::Autolock mttLock(mMediaTimeTransformLock);
assert(mMediaTimeTransformValid);
if (mMediaTimeTransform.a_to_b_denom == 0) {
// the transform represents a pause, so yield silence
timedYieldSilence(buffer->frameCount, buffer);
return NO_ERROR;
}
int64_t transformedPTS;
if (!mMediaTimeTransform.doForwardTransform(head.pts(),
&transformedPTS)) {
// the transform failed. this shouldn't happen, but if it does
// then just drop this buffer
ALOGW("timedGetNextBuffer transform failed");
buffer->raw = 0;
buffer->frameCount = 0;
mTimedBufferQueue.removeAt(0);
return NO_ERROR;
}
if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
&headLocalPTS)) {
buffer->raw = 0;
buffer->frameCount = 0;
return INVALID_OPERATION;
}
} else {
headLocalPTS = transformedPTS;
}
}
// adjust the head buffer's PTS to reflect the portion of the head buffer
// that has already been consumed
int64_t effectivePTS = headLocalPTS +
((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
// Calculate the delta in samples between the head of the input buffer
// queue and the start of the next output buffer that will be written.
// If the transformation fails because of over or underflow, it means
// that the sample's position in the output stream is so far out of
// whack that it should just be dropped.
int64_t sampleDelta;
if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
ALOGV("*** head buffer is too far from PTS: dropped buffer");
mTimedBufferQueue.removeAt(0);
continue;
}
if (!mLocalTimeToSampleTransform.doForwardTransform(
(effectivePTS - pts) << 32, &sampleDelta)) {
ALOGV("*** too late during sample rate transform: dropped buffer");
mTimedBufferQueue.removeAt(0);
continue;
}
ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
__PRETTY_FUNCTION__, head.pts(), head.position(), pts,
static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
// if the delta between the ideal placement for the next input sample and
// the current output position is within this threshold, then we will
// concatenate the next input samples to the previous output
const int64_t kSampleContinuityThreshold =
(static_cast<int64_t>(sampleRate()) << 32) / 10;
// if this is the first buffer of audio that we're emitting from this track
// then it should be almost exactly on time.
const int64_t kSampleStartupThreshold = 1LL << 32;
if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
(!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
// the next input is close enough to being on time, so concatenate it
// with the last output
timedYieldSamples(buffer);
ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
return NO_ERROR;
} else if (sampleDelta > 0) {
// the gap between the current output position and the proper start of
// the next input sample is too big, so fill it with silence
uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
timedYieldSilence(framesUntilNextInput, buffer);
ALOGV("*** silence: frameCount=%u", buffer->frameCount);
return NO_ERROR;
} else {
// the next input sample is late
uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
size_t onTimeSamplePosition =
head.position() + lateFrames * mCblk->frameSize;
if (onTimeSamplePosition > head.buffer()->size()) {
// all the remaining samples in the head are too late, so
// drop it and move on
ALOGV("*** too late: dropped buffer");
mTimedBufferQueue.removeAt(0);
continue;
} else {
// skip over the late samples
head.setPosition(onTimeSamplePosition);
// yield the available samples
timedYieldSamples(buffer);
ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
return NO_ERROR;
}
}
}
}
// Yield samples from the timed buffer queue head up to the given output
// buffer's capacity.
//
// Caller must hold mTimedBufferQueueLock
void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples(
AudioBufferProvider::Buffer* buffer) {
const TimedBuffer& head = mTimedBufferQueue[0];
buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
head.position());
uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
mCblk->frameSize);
size_t framesRequested = buffer->frameCount;
buffer->frameCount = min(framesLeftInHead, framesRequested);
mTimedAudioOutputOnTime = true;
}
// Yield samples of silence up to the given output buffer's capacity
//
// Caller must hold mTimedBufferQueueLock
void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence(
uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
// lazily allocate a buffer filled with silence
if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
delete [] mTimedSilenceBuffer;
mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
}
buffer->raw = mTimedSilenceBuffer;
size_t framesRequested = buffer->frameCount;
buffer->frameCount = min(numFrames, framesRequested);
mTimedAudioOutputOnTime = false;
}
void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
AudioBufferProvider::Buffer* buffer) {
Mutex::Autolock _l(mTimedBufferQueueLock);
// If the buffer which was just released is part of the buffer at the head
// of the queue, be sure to update the amt of the buffer which has been
// consumed. If the buffer being returned is not part of the head of the
// queue, its either because the buffer is part of the silence buffer, or
// because the head of the timed queue was trimmed after the mixer called
// getNextBuffer but before the mixer called releaseBuffer.
if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) {
TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
void* start = head.buffer()->pointer();
void* end = head.buffer()->pointer() + head.buffer()->size();
if ((buffer->raw >= start) && (buffer->raw <= end)) {
head.setPosition(head.position() +
(buffer->frameCount * mCblk->frameSize));
if (static_cast<size_t>(head.position()) >= head.buffer()->size()) {
mTimedBufferQueue.removeAt(0);
}
}
}
buffer->raw = 0;
buffer->frameCount = 0;
}
uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
Mutex::Autolock _l(mTimedBufferQueueLock);
uint32_t frames = 0;
for (size_t i = 0; i < mTimedBufferQueue.size(); i++) {
const TimedBuffer& tb = mTimedBufferQueue[i];
frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize;
}
return frames;
}
AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
: mPTS(0), mPosition(0) {}
AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
const sp<IMemory>& buffer, int64_t pts)
: mBuffer(buffer), mPTS(pts), mPosition(0) {}
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
channelMask, frameCount, flags, 0, sessionId),
mOverflow(false)
{
if (mCblk != NULL) {
ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
if (format == AUDIO_FORMAT_PCM_16_BIT) {
mCblk->frameSize = mChannelCount * sizeof(int16_t);
} else if (format == AUDIO_FORMAT_PCM_8_BIT) {
mCblk->frameSize = mChannelCount * 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, int64_t pts)
{
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;
ALOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesAvail = cblk->framesAvailable_l();
if (CC_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 == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
return recordThread->start(this, tid);
} 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
android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
}
}
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
mSessionId,
mFrameCount,
mState,
mCblk->sampleRate,
mCblk->server,
mCblk->user);
}
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
PlaybackThread *playbackThread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount)
: Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
"mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p",
mCblk, mBuffer, mCblk->buffers,
mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd);
} else {
ALOGW("Error creating output track on thread %p", playbackThread);
}
}
AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
{
clearBufferQueue();
}
status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid)
{
status_t status = Track::start(tid);
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 = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
start(0);
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 {
ALOGW ("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)NO_MORE_BUFFERS) {
ALOGV ("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;
ALOGV("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);
ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
} else {
ALOGW("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;
// ALOGV("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 (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
return NO_MORE_BUFFERS;
}
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (result != NO_ERROR) {
return NO_MORE_BUFFERS;
}
// read the server count again
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
}
// if (framesAvail < framesReq) {
// return 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();
for (size_t i = 0; i < size; i++) {
Buffer *pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
delete pBuffer;
}
mBufferQueue.clear();
}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
// FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
mPid(pid),
mTimedTrackCount(0)
{
// 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);
}
sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
// Reserve one of the limited slots for a timed audio track associated
// with this client
bool AudioFlinger::Client::reserveTimedTrack()
{
const int kMaxTimedTracksPerClient = 4;
Mutex::Autolock _l(mTimedTrackLock);
if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
ALOGW("can not create timed track - pid %d has exceeded the limit",
mPid);
return false;
}
mTimedTrackCount++;
return true;
}
// Release a slot for a timed audio track
void AudioFlinger::Client::releaseTimedTrack()
{
Mutex::Autolock _l(mTimedTrackLock);
mTimedTrackCount--;
}
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid)
: mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
{
sp<NotificationClient> keep(this);
mAudioFlinger->removeNotificationClient(mPid);
}
// ----------------------------------------------------------------------------
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
}
AudioFlinger::TrackHandle::~TrackHandle() {
// just stop the track on deletion, associated resources
// will be freed from the main thread once all pending buffers have
// been played. Unless it's not in the active track list, in which
// case we free everything now...
mTrack->destroy();
}
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
return mTrack->getCblk();
}
status_t AudioFlinger::TrackHandle::start(pid_t tid) {
return mTrack->start(tid);
}
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();
}
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
}
status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
sp<IMemory>* buffer) {
if (!mTrack->isTimedTrack())
return INVALID_OPERATION;
PlaybackThread::TimedTrack* tt =
reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
return tt->allocateTimedBuffer(size, buffer);
}
status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
int64_t pts) {
if (!mTrack->isTimedTrack())
return INVALID_OPERATION;
PlaybackThread::TimedTrack* tt =
reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
return tt->queueTimedBuffer(buffer, pts);
}
status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
const LinearTransform& xform, int target) {
if (!mTrack->isTimedTrack())
return INVALID_OPERATION;
PlaybackThread::TimedTrack* tt =
reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
return tt->setMediaTimeTransform(
xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
}
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,
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
int *sessionId,
status_t *status)
{
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
status_t lStatus;
RecordThread *thread;
size_t inFrameCount;
int lSessionId;
// check calling permissions
if (!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// add client to list
{ // scope for mLock
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
if (thread == NULL) {
lStatus = BAD_VALUE;
goto Exit;
}
client = registerPid_l(pid);
// If no audio session id is provided, create one here
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
lSessionId = *sessionId;
} else {
lSessionId = nextUniqueId();
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
recordTrack = thread->createRecordTrack_l(client,
sampleRate,
format,
channelMask,
frameCount,
flags,
lSessionId,
&lStatus);
}
if (lStatus != NO_ERROR) {
// 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();
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();
}
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
return mRecordTrack->getCblk();
}
status_t AudioFlinger::RecordHandle::start(pid_t tid) {
ALOGV("RecordHandle::start()");
return mRecordTrack->start(tid);
}
void AudioFlinger::RecordHandle::stop() {
ALOGV("RecordHandle::stop()");
mRecordTrack->stop();
}
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,
audio_io_handle_t id,
uint32_t device) :
ThreadBase(audioFlinger, id, device, RECORD),
mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
// mRsmpInIndex and mInputBytes set by readInputParameters()
mReqChannelCount(popcount(channels)),
mReqSampleRate(sampleRate)
// mBytesRead is only meaningful while active, and so is cleared in start()
// (but might be better to also clear here for dump?)
{
snprintf(mName, kNameLength, "AudioIn_%d", id);
readInputParameters();
}
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
delete mResampler;
delete[] mRsmpOutBuffer;
}
void AudioFlinger::RecordThread::onFirstRef()
{
run(mName, PRIORITY_URGENT_AUDIO);
}
status_t AudioFlinger::RecordThread::readyToRun()
{
status_t status = initCheck();
ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
return status;
}
bool AudioFlinger::RecordThread::threadLoop()
{
AudioBufferProvider::Buffer buffer;
sp<RecordTrack> activeTrack;
Vector< sp<EffectChain> > effectChains;
nsecs_t lastWarning = 0;
acquireWakeLock();
// start recording
while (!exitPending()) {
processConfigEvents();
{ // scope for mLock
Mutex::Autolock _l(mLock);
checkForNewParameters_l();
if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
if (!mStandby) {
mInput->stream->common.standby(&mInput->stream->common);
mStandby = true;
}
if (exitPending()) break;
releaseWakeLock_l();
ALOGV("RecordThread: loop stopping");
// go to sleep
mWaitWorkCV.wait(mLock);
ALOGV("RecordThread: loop starting");
acquireWakeLock_l();
continue;
}
if (mActiveTrack != 0) {
if (mActiveTrack->mState == TrackBase::PAUSING) {
if (!mStandby) {
mInput->stream->common.standby(&mInput->stream->common);
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;
}
}
lockEffectChains_l(effectChains);
}
if (mActiveTrack != 0) {
if (mActiveTrack->mState != TrackBase::ACTIVE &&
mActiveTrack->mState != TrackBase::RESUMING) {
unlockEffectChains(effectChains);
usleep(kRecordThreadSleepUs);
continue;
}
for (size_t i = 0; i < effectChains.size(); i ++) {
effectChains[i]->process_l();
}
buffer.frameCount = mFrameCount;
if (CC_LIKELY(mActiveTrack->getNextBuffer(
&buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
if (mResampler == NULL) {
// 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 != AUDIO_FORMAT_PCM_16_BIT) {
memcpy(dst, src, framesIn * mFrameSize);
} else {
int16_t *src16 = (int16_t *)src;
int16_t *dst16 = (int16_t *)dst;
if (mChannelCount == 1) {
while (framesIn--) {
*dst16++ = *src16;
*dst16++ = *src16++;
}
} else {
while (framesIn--) {
*dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
src16 += 2;
}
}
}
}
if (framesOut && mFrameCount == mRsmpInIndex) {
if (framesOut == mFrameCount &&
((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes);
framesOut = 0;
} else {
mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
mRsmpInIndex = 0;
}
if (mBytesRead < 0) {
ALOGE("Error reading audio input");
if (mActiveTrack->mState == TrackBase::ACTIVE) {
// Force input into standby so that it tries to
// recover at next read attempt
mInput->stream->common.standby(&mInput->stream->common);
usleep(kRecordThreadSleepUs);
}
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) {
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 {
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) > kWarningThrottleNs) {
ALOGW("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(kRecordThreadSleepUs);
}
}
// enable changes in effect chain
unlockEffectChains(effectChains);
effectChains.clear();
}
if (!mStandby) {
mInput->stream->common.standby(&mInput->stream->common);
}
mActiveTrack.clear();
mStartStopCond.broadcast();
releaseWakeLock();
ALOGV("RecordThread %p exiting", this);
return false;
}
sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
int channelMask,
int frameCount,
uint32_t flags,
int sessionId,
status_t *status)
{
sp<RecordTrack> track;
status_t lStatus;
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGE("Audio driver not initialized.");
goto Exit;
}
{ // scope for mLock
Mutex::Autolock _l(mLock);
track = new RecordTrack(this, client, sampleRate,
format, channelMask, frameCount, flags, sessionId);
if (track->getCblk() == 0) {
lStatus = NO_MEMORY;
goto Exit;
}
mTrack = track.get();
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
bool suspend = audio_is_bluetooth_sco_device(
(audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff();
setEffectSuspended_l(FX_IID_AEC, suspend, sessionId);
setEffectSuspended_l(FX_IID_NS, suspend, sessionId);
}
lStatus = NO_ERROR;
Exit:
if (status) {
*status = lStatus;
}
return track;
}
status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid)
{
ALOGV("RecordThread::start tid=%d", tid);
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;
}
mRsmpInIndex = mFrameCount;
mBytesRead = 0;
if (mResampler != NULL) {
mResampler->reset();
}
mActiveTrack->mState = TrackBase::RESUMING;
// signal thread to start
ALOGV("Signal record thread");
mWaitWorkCV.signal();
// do not wait for mStartStopCond if exiting
if (exitPending()) {
mActiveTrack.clear();
status = INVALID_OPERATION;
goto startError;
}
mStartStopCond.wait(mLock);
if (mActiveTrack == 0) {
ALOGV("Record failed to start");
status = BAD_VALUE;
goto startError;
}
ALOGV("Record started OK");
return status;
}
startError:
AudioSystem::stopInput(mId);
return status;
}
void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
ALOGV("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 (exitPending()) {
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();
ALOGV("Record stopped OK");
}
}
}
}
status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
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 mask 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 != NULL));
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);
dumpEffectChains(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
{
size_t framesReq = buffer->frameCount;
size_t framesReady = mFrameCount - mRsmpInIndex;
int channelCount;
if (framesReady == 0) {
mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes);
if (mBytesRead < 0) {
ALOGE("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->stream->common.standby(&mInput->stream->common);
usleep(kRecordThreadSleepUs);
}
buffer->raw = NULL;
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;
audio_format_t 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 = (audio_format_t) value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
reqChannelCount = popcount(value);
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
// size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (mActiveTrack != 0) {
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.
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->setDevice_l(value);
}
// store input device and output device but do not forward output device to audio HAL.
// Note that status is ignored by the caller for output device
// (see AudioFlinger::setParameters()
if (value & AUDIO_DEVICE_OUT_ALL) {
mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL);
status = BAD_VALUE;
} else {
mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL);
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
if (mTrack != NULL) {
bool suspend = audio_is_bluetooth_sco_device(
(audio_devices_t)value) && mAudioFlinger->btNrecIsOff();
setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId());
setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId());
}
}
mDevice |= (uint32_t)value;
}
if (status == NO_ERROR) {
status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
if (status == INVALID_OPERATION) {
mInput->stream->common.standby(&mInput->stream->common);
status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
}
if (reconfig) {
if (status == BAD_VALUE &&
reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
(popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) &&
(reqChannelCount < 3)) {
status = NO_ERROR;
}
if (status == NO_ERROR) {
readInputParameters();
sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
}
}
}
mNewParameters.removeAt(0);
mParamStatus = status;
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
{
char *s;
String8 out_s8 = String8();
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
return out_s8;
}
s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string());
out_s8 = String8(s);
free(s);
return out_s8;
}
void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = NULL;
switch (event) {
case AudioSystem::INPUT_OPENED:
case AudioSystem::INPUT_CONFIG_CHANGED:
desc.channels = mChannelMask;
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()
{
delete mRsmpInBuffer;
// mRsmpInBuffer is always assigned a new[] below
delete mRsmpOutBuffer;
mRsmpOutBuffer = NULL;
delete mResampler;
mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
mFrameSize = audio_stream_frame_size(&mInput->stream->common);
mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
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()
{
Mutex::Autolock _l(mLock);
if (initCheck() != NO_ERROR) {
return 0;
}
return mInput->stream->get_input_frames_lost(mInput->stream);
}
uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId)
{
Mutex::Autolock _l(mLock);
uint32_t result = 0;
if (getEffectChain_l(sessionId) != 0) {
result = EFFECT_SESSION;
}
if (mTrack != NULL && sessionId == mTrack->sessionId()) {
result |= TRACK_SESSION;
}
return result;
}
AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track()
{
Mutex::Autolock _l(mLock);
return mTrack;
}
AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const
{
Mutex::Autolock _l(mLock);
return mInput;
}
AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput()
{
Mutex::Autolock _l(mLock);
AudioStreamIn *input = mInput;
mInput = NULL;
return input;
}
// this method must always be called either with ThreadBase mLock held or inside the thread loop
audio_stream_t* AudioFlinger::RecordThread::stream()
{
if (mInput == NULL) {
return NULL;
}
return &mInput->stream->common;
}
// ----------------------------------------------------------------------------
audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_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;
audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
audio_stream_out_t *outStream;
audio_hw_device_t *outHwDev;
ALOGV("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);
outHwDev = findSuitableHwDev_l(*pDevices);
if (outHwDev == NULL)
return 0;
status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
&channels, &samplingRate, &outStream);
ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
outStream,
samplingRate,
format,
channels,
status);
mHardwareStatus = AUDIO_HW_IDLE;
if (outStream != NULL) {
AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
audio_io_handle_t id = nextUniqueId();
if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
(format != AUDIO_FORMAT_PCM_16_BIT) ||
(channels != AUDIO_CHANNEL_OUT_STEREO)) {
thread = new DirectOutputThread(this, output, id, *pDevices);
ALOGV("openOutput() created direct output: ID %d thread %p", id, thread);
} else {
thread = new MixerThread(this, output, id, *pDevices);
ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread);
}
mPlaybackThreads.add(id, thread);
if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
if (pFormat != NULL) *pFormat = format;
if (pChannels != NULL) *pChannels = channels;
if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return id;
}
return 0;
}
audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
MixerThread *thread2 = checkMixerThread_l(output2);
if (thread1 == NULL || thread2 == NULL) {
ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
return 0;
}
audio_io_handle_t 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(audio_io_handle_t 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;
}
ALOGV("closeOutput() %d", output);
if (thread->type() == ThreadBase::MIXER) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
dupThread->removeOutputTrack((MixerThread *)thread.get());
}
}
}
audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
mPlaybackThreads.removeItem(output);
}
thread->exit();
// The thread entity (active unit of execution) is no longer running here,
// but the ThreadBase container still exists.
if (thread->type() != ThreadBase::DUPLICATING) {
AudioStreamOut *out = thread->clearOutput();
assert(out != NULL);
// from now on thread->mOutput is NULL
out->hwDev->close_output_stream(out->hwDev, out->stream);
delete out;
}
return NO_ERROR;
}
status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
ALOGV("suspendOutput() %d", output);
thread->suspend();
return NO_ERROR;
}
status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
ALOGV("restoreOutput() %d", output);
thread->restore();
return NO_ERROR;
}
audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
audio_in_acoustics_t acoustics)
{
status_t status;
RecordThread *thread = NULL;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t reqSamplingRate = samplingRate;
audio_format_t reqFormat = format;
uint32_t reqChannels = channels;
audio_stream_in_t *inStream;
audio_hw_device_t *inHwDev;
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
inHwDev = findSuitableHwDev_l(*pDevices);
if (inHwDev == NULL)
return 0;
status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
acoustics,
&inStream);
ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
inStream,
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 (inStream == NULL && status == BAD_VALUE &&
reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT &&
(samplingRate <= 2 * reqSamplingRate) &&
(popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
ALOGV("openInput() reopening with proposed sampling rate and channels");
status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
acoustics,
&inStream);
}
if (inStream != NULL) {
AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
audio_io_handle_t id = nextUniqueId();
// Start record thread
// RecorThread require both input and output device indication to forward to audio
// pre processing modules
uint32_t device = (*pDevices) | primaryOutputDevice_l();
thread = new RecordThread(this,
input,
reqSamplingRate,
reqChannels,
id,
device);
mRecordThreads.add(id, thread);
ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
if (pFormat != NULL) *pFormat = format;
if (pChannels != NULL) *pChannels = reqChannels;
input->stream->common.standby(&input->stream->common);
// notify client processes of the new input creation
thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
return id;
}
return 0;
}
status_t AudioFlinger::closeInput(audio_io_handle_t 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;
}
ALOGV("closeInput() %d", input);
audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
mRecordThreads.removeItem(input);
}
thread->exit();
// The thread entity (active unit of execution) is no longer running here,
// but the ThreadBase container still exists.
AudioStreamIn *in = thread->clearInput();
assert(in != NULL);
// from now on thread->mInput is NULL
in->hwDev->close_input_stream(in->hwDev, in->stream);
delete in;
return NO_ERROR;
}
status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
MixerThread *dstThread = checkMixerThread_l(output);
if (dstThread == NULL) {
ALOGW("setStreamOutput() bad output id %d", output);
return BAD_VALUE;
}
ALOGV("setStreamOutput() stream %d to output %d", stream, output);
audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
dstThread->setStreamValid(stream, true);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if (thread != dstThread && thread->type() != ThreadBase::DIRECT) {
MixerThread *srcThread = (MixerThread *)thread;
srcThread->setStreamValid(stream, false);
srcThread->invalidateTracks(stream);
}
}
return NO_ERROR;
}
int AudioFlinger::newAudioSessionId()
{
return nextUniqueId();
}
void AudioFlinger::acquireAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("acquiring %d from %d", audioSession, caller);
size_t num = mAudioSessionRefs.size();
for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
if (ref->sessionid == audioSession && ref->pid == caller) {
ref->cnt++;
ALOGV(" incremented refcount to %d", ref->cnt);
return;
}
}
mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
ALOGV(" added new entry for %d", audioSession);
}
void AudioFlinger::releaseAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("releasing %d from %d", audioSession, caller);
size_t num = mAudioSessionRefs.size();
for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
if (ref->sessionid == audioSession && ref->pid == caller) {
ref->cnt--;
ALOGV(" decremented refcount to %d", ref->cnt);
if (ref->cnt == 0) {
mAudioSessionRefs.removeAt(i);
delete ref;
purgeStaleEffects_l();
}
return;
}
}
ALOGW("session id %d not found for pid %d", audioSession, caller);
}
void AudioFlinger::purgeStaleEffects_l() {
ALOGV("purging stale effects");
Vector< sp<EffectChain> > chains;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
chains.push(ec);
}
}
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
sp<RecordThread> t = mRecordThreads.valueAt(i);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
chains.push(ec);
}
}
for (size_t i = 0; i < chains.size(); i++) {
sp<EffectChain> ec = chains[i];
int sessionid = ec->sessionId();
sp<ThreadBase> t = ec->mThread.promote();
if (t == 0) {
continue;
}
size_t numsessionrefs = mAudioSessionRefs.size();
bool found = false;
for (size_t k = 0; k < numsessionrefs; k++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
if (ref->sessionid == sessionid) {
ALOGV(" session %d still exists for %d with %d refs",
sessionid, ref->pid, ref->cnt);
found = true;
break;
}
}
if (!found) {
// remove all effects from the chain
while (ec->mEffects.size()) {
sp<EffectModule> effect = ec->mEffects[0];
effect->unPin();
Mutex::Autolock _l (t->mLock);
t->removeEffect_l(effect);
for (size_t j = 0; j < effect->mHandles.size(); j++) {
sp<EffectHandle> handle = effect->mHandles[j].promote();
if (handle != 0) {
handle->mEffect.clear();
if (handle->mHasControl && handle->mEnabled) {
t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
}
}
}
AudioSystem::unregisterEffect(effect->id());
}
}
}
return;
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
return mPlaybackThreads.valueFor(output).get();
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
{
return mRecordThreads.valueFor(input).get();
}
uint32_t AudioFlinger::nextUniqueId()
{
return android_atomic_inc(&mNextUniqueId);
}
AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l()
{
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
AudioStreamOut *output = thread->getOutput();
if (output != NULL && output->hwDev == mPrimaryHardwareDev) {
return thread;
}
}
return NULL;
}
uint32_t AudioFlinger::primaryOutputDevice_l()
{
PlaybackThread *thread = primaryPlaybackThread_l();
if (thread == NULL) {
return 0;
}
return thread->device();
}
// ----------------------------------------------------------------------------
// Effect management
// ----------------------------------------------------------------------------
status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
{
Mutex::Autolock _l(mLock);
return EffectQueryNumberEffects(numEffects);
}
status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectQueryEffect(index, descriptor);
}
status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectGetDescriptor(pUuid, descriptor);
}
sp<IEffect> AudioFlinger::createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
audio_io_handle_t io,
int sessionId,
status_t *status,
int *id,
int *enabled)
{
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
effect_descriptor_t desc;
ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
pid, effectClient.get(), priority, sessionId, io);
if (pDesc == NULL) {
lStatus = BAD_VALUE;
goto Exit;
}
// check audio settings permission for global effects
if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
// that can only be created by audio policy manager (running in same process)
if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
if (io == 0) {
if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
// output must be specified by AudioPolicyManager when using session
// AUDIO_SESSION_OUTPUT_STAGE
lStatus = BAD_VALUE;
goto Exit;
} else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// if the output returned by getOutputForEffect() is removed before we lock the
// mutex below, the call to checkPlaybackThread_l(io) below will detect it
// and we will exit safely
io = 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) {
ALOGW("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)) {
ALOGW("createEffect() no effect type");
lStatus = BAD_VALUE;
goto Exit;
}
uint32_t numEffects = 0;
effect_descriptor_t d;
d.flags = 0; // prevent compiler warning
bool found = false;
lStatus = EffectQueryNumberEffects(&numEffects);
if (lStatus < 0) {
ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
goto Exit;
}
for (uint32_t i = 0; i < numEffects; i++) {
lStatus = EffectQueryEffect(i, &desc);
if (lStatus < 0) {
ALOGW("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 != AUDIO_SESSION_OUTPUT_MIX ||
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
break;
}
}
}
if (!found) {
lStatus = BAD_VALUE;
ALOGW("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 == AUDIO_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 != AUDIO_SESSION_OUTPUT_MIX &&
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
lStatus = INVALID_OPERATION;
goto Exit;
}
// check recording permission for visualizer
if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
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.
// Note: io is never 0 when creating an effect on an input
if (io == 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) {
io = mPlaybackThreads.keyAt(i);
break;
}
}
if (io == 0) {
for (size_t i = 0; i < mRecordThreads.size(); i++) {
if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
io = mRecordThreads.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 (io == 0 && mPlaybackThreads.size()) {
io = mPlaybackThreads.keyAt(0);
}
ALOGV("createEffect() got io %d for effect %s", io, desc.name);
}
ThreadBase *thread = checkRecordThread_l(io);
if (thread == NULL) {
thread = checkPlaybackThread_l(io);
if (thread == NULL) {
ALOGE("createEffect() unknown output thread");
lStatus = BAD_VALUE;
goto Exit;
}
}
sp<Client> client = registerPid_l(pid);
// create effect on selected output thread
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 sessionId, audio_io_handle_t srcOutput,
audio_io_handle_t dstOutput)
{
ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
sessionId, srcOutput, dstOutput);
Mutex::Autolock _l(mLock);
if (srcOutput == dstOutput) {
ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
return NO_ERROR;
}
PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
if (srcThread == NULL) {
ALOGW("moveEffects() bad srcOutput %d", srcOutput);
return BAD_VALUE;
}
PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
if (dstThread == NULL) {
ALOGW("moveEffects() bad dstOutput %d", dstOutput);
return BAD_VALUE;
}
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(srcThread->mLock);
moveEffectChain_l(sessionId, srcThread, dstThread, false);
return NO_ERROR;
}
// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
status_t AudioFlinger::moveEffectChain_l(int sessionId,
AudioFlinger::PlaybackThread *srcThread,
AudioFlinger::PlaybackThread *dstThread,
bool reRegister)
{
ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
sessionId, srcThread, dstThread);
sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
if (chain == 0) {
ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
sessionId, 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 unnecessary 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
audio_io_handle_t dstOutput = dstThread->id();
sp<EffectChain> dstChain;
uint32_t strategy = 0; // prevent compiler warning
sp<EffectModule> effect = chain->getEffectFromId_l(0);
while (effect != 0) {
srcThread->removeEffect_l(effect);
dstThread->addEffect_l(effect);
// removeEffect_l() has stopped the effect if it was active so it must be restarted
if (effect->state() == EffectModule::ACTIVE ||
effect->state() == EffectModule::STOPPING) {
effect->start();
}
// 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) {
ALOGW("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,
sessionId,
effect->id());
}
effect = chain->getEffectFromId_l(0);
}
return NO_ERROR;
}
// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::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<EffectChain> chain;
bool chainCreated = false;
bool effectCreated = false;
bool effectRegistered = false;
lStatus = initCheck();
if (lStatus != NO_ERROR) {
ALOGW("createEffect_l() Audio driver not initialized.");
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 == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) {
ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d",
desc->name, sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
// Only Pre processor effects are allowed on input threads and only on input threads
if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
desc->name, desc->flags, mType);
lStatus = BAD_VALUE;
goto Exit;
}
ALOGV("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
ALOGV("createEffect_l() new effect chain for session %d", sessionId);
chain = new EffectChain(this, sessionId);
addEffectChain_l(chain);
chain->setStrategy(getStrategyForSession_l(sessionId));
chainCreated = true;
} else {
effect = chain->getEffectFromDesc_l(desc);
}
ALOGV("createEffect_l() got effect %p on chain %p", 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());
}
// create effect handle and connect it to effect module
handle = new EffectHandle(effect, client, effectClient, priority);
lStatus = effect->addHandle(handle);
if (enabled != NULL) {
*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;
}
sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
{
sp<EffectChain> chain = getEffectChain_l(sessionId);
return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
}
// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
// PlaybackThread::mLock held
status_t AudioFlinger::ThreadBase::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
ALOGV("addEffect_l() new effect chain for session %d", sessionId);
chain = new EffectChain(this, sessionId);
addEffectChain_l(chain);
chain->setStrategy(getStrategyForSession_l(sessionId));
chainCreated = true;
}
ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get());
if (chain->getEffectFromId_l(effect->id()) != 0) {
ALOGW("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::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
ALOGV("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 {
ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get());
}
}
void AudioFlinger::ThreadBase::lockEffectChains_l(
Vector<sp <AudioFlinger::EffectChain> >& effectChains)
{
effectChains = mEffectChains;
for (size_t i = 0; i < mEffectChains.size(); i++) {
mEffectChains[i]->lock();
}
}
void AudioFlinger::ThreadBase::unlockEffectChains(
Vector<sp <AudioFlinger::EffectChain> >& effectChains)
{
for (size_t i = 0; i < effectChains.size(); i++) {
effectChains[i]->unlock();
}
}
sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId)
{
Mutex::Autolock _l(mLock);
return getEffectChain_l(sessionId);
}
sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
{
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() == sessionId) {
return mEffectChains[i];
}
}
return 0;
}
void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
mEffectChains[i]->setMode_l(mode);
}
}
void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
const wp<EffectHandle>& handle,
bool unpinIfLast) {
Mutex::Autolock _l(mLock);
ALOGV("disconnectEffect() %p effect %p", this, effect.get());
// delete the effect module if removing last handle on it
if (effect->removeHandle(handle) == 0) {
if (!effect->isPinned() || unpinIfLast) {
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;
ALOGV("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));
ALOGV("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()) {
ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
track->setMainBuffer(buffer);
chain->incTrackCnt();
}
}
// 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()) {
ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session);
chain->incActiveTrackCnt();
}
}
}
chain->setInBuffer(buffer, ownsBuffer);
chain->setOutBuffer(mMixBuffer);
// Effect chain for session AUDIO_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 AUDIO_SESSION_OUTPUT_MIX is inserted before
// session AUDIO_SESSION_OUTPUT_STAGE to be processed
// after track specific effects and before output stage
// It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and
// that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_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);
checkSuspendOnAddEffectChain_l(chain);
return NO_ERROR;
}
size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain)
{
int session = chain->sessionId();
ALOGV("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 active tracks from 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()) {
ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d",
chain.get(), session);
chain->decActiveTrackCnt();
}
}
// 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);
chain->decTrackCnt();
}
}
break;
}
}
return mEffectChains.size();
}
status_t AudioFlinger::PlaybackThread::attachAuxEffect(
const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
{
Mutex::Autolock _l(mLock);
return attachAuxEffect_l(track, EffectId);
}
status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId)
{
status_t status = NO_ERROR;
if (EffectId == 0) {
track->setAuxBuffer(0, NULL);
} else {
// Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX
sp<EffectModule> effect = getEffect_l(AUDIO_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);
}
}
}
status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
{
// only one chain per input thread
if (mEffectChains.size() != 0) {
return INVALID_OPERATION;
}
ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
chain->setInBuffer(NULL);
chain->setOutBuffer(NULL);
checkSuspendOnAddEffectChain_l(chain);
mEffectChains.add(chain);
return NO_ERROR;
}
size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain)
{
ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
ALOGW_IF(mEffectChains.size() != 1,
"removeEffectChain_l() %p invalid chain size %d on thread %p",
chain.get(), mEffectChains.size(), this);
if (mEffectChains.size() == 1) {
mEffectChains.removeAt(0);
}
return 0;
}
// ----------------------------------------------------------------------------
// EffectModule implementation
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectModule"
AudioFlinger::EffectModule::EffectModule(ThreadBase *thread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId)
: mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
mStatus(NO_INIT), mState(IDLE), mSuspended(false)
{
ALOGV("Constructor %p", this);
int lStatus;
if (thread == NULL) {
return;
}
memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t));
// create effect engine from effect factory
mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface);
if (mStatus != NO_ERROR) {
return;
}
lStatus = init();
if (lStatus < 0) {
mStatus = lStatus;
goto Error;
}
if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
mPinned = true;
}
ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface);
return;
Error:
EffectRelease(mEffectInterface);
mEffectInterface = NULL;
ALOGV("Constructor Error %d", mStatus);
}
AudioFlinger::EffectModule::~EffectModule()
{
ALOGV("Destructor %p", this);
if (mEffectInterface != NULL) {
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
audio_stream_t *stream = thread->stream();
if (stream != NULL) {
stream->remove_audio_effect(stream, mEffectInterface);
}
}
}
// release effect engine
EffectRelease(mEffectInterface);
}
}
status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
{
status_t status;
Mutex::Autolock _l(mLock);
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) {
bool enabled = false;
if (h != 0) {
enabled = h->enabled();
h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/);
}
handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/);
status = NO_ERROR;
} else {
status = ALREADY_EXISTS;
}
ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i);
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;
}
ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i);
bool enabled = false;
EffectHandle *hdl = handle.unsafe_get();
if (hdl != NULL) {
ALOGV("removeHandle() unsafe_get OK");
enabled = hdl->enabled();
}
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 /*hasControl*/, true /*signal*/ , enabled /*enabled*/);
}
}
// Prevent calls to process() and other functions on effect interface from now on.
// The effect engine will be released by the destructor when the last strong reference on
// this object is released which can happen after next process is called.
if (size == 0 && !mPinned) {
mState = DESTROYED;
}
return size;
}
sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
{
Mutex::Autolock _l(mLock);
return mHandles.size() != 0 ? mHandles[0].promote() : 0;
}
void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast)
{
ALOGV("disconnect() %p handle %p", this, handle.unsafe_get());
// 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) {
thread->disconnectEffect(keep, handle, unpinIfLast);
}
}
}
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, DESTROYED
break;
}
}
void AudioFlinger::EffectModule::process()
{
Mutex::Autolock _l(mLock);
if (mState == DESTROYED || 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) {
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->activeTrackCnt() != 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()
{
uint32_t channels;
if (mEffectInterface == NULL) {
return NO_INIT;
}
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
return DEAD_OBJECT;
}
// TODO: handle configuration of effects replacing track process
if (thread->channelCount() == 1) {
channels = AUDIO_CHANNEL_OUT_MONO;
} else {
channels = AUDIO_CHANNEL_OUT_STEREO;
}
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
} else {
mConfig.inputCfg.channels = channels;
}
mConfig.outputCfg.channels = channels;
mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
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 AUDIO_SESSION_OUTPUT_MIX or AUDIO_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;
mConfig.inputCfg.buffer.frameCount = thread->frameCount();
mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
ALOGV("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_SET_CONFIG,
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()
{
Mutex::Autolock _l(mLock);
return start_l();
}
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) {
status = cmdStatus;
}
if (status == 0 &&
((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
audio_stream_t *stream = thread->stream();
if (stream != NULL) {
stream->add_audio_effect(stream, mEffectInterface);
}
}
}
return status;
}
status_t AudioFlinger::EffectModule::stop()
{
Mutex::Autolock _l(mLock);
return stop_l();
}
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;
}
if (status == 0 &&
((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) {
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
audio_stream_t *stream = thread->stream();
if (stream != NULL) {
stream->remove_audio_effect(stream, mEffectInterface);
}
}
}
return status;
}
status_t AudioFlinger::EffectModule::command(uint32_t cmdCode,
uint32_t cmdSize,
void *pCmdData,
uint32_t *replySize,
void *pReplyData)
{
Mutex::Autolock _l(mLock);
// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface);
if (mState == DESTROYED || 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)
{
Mutex::Autolock _l(mLock);
ALOGV("setEnabled %p enabled %d", this, enabled);
if (enabled != isEnabled()) {
status_t status = AudioSystem::setEffectEnabled(mId, enabled);
if (enabled && status != NO_ERROR) {
return status;
}
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;
case DESTROYED:
return NO_ERROR; // simply ignore as we are being destroyed
}
for (size_t i = 1; i < mHandles.size(); i++) {
sp<EffectHandle> h = mHandles[i].promote();
if (h != 0) {
h->setEnabled(enabled);
}
}
}
return NO_ERROR;
}
bool AudioFlinger::EffectModule::isEnabled() const
{
switch (mState) {
case RESTART:
case STARTING:
case ACTIVE:
return true;
case IDLE:
case STOPPING:
case STOPPED:
case DESTROYED:
default:
return false;
}
}
bool AudioFlinger::EffectModule::isProcessEnabled() const
{
switch (mState) {
case RESTART:
case ACTIVE:
case STOPPING:
case STOPPED:
return true;
case IDLE:
case STARTING:
case DESTROYED:
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 (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
// audio pre processing modules on RecordThread can receive both output and
// input device indication in the same call
uint32_t dev = device & AUDIO_DEVICE_OUT_ALL;
if (dev) {
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_DEVICE,
sizeof(uint32_t),
&dev,
&size,
&cmdStatus);
if (status == NO_ERROR) {
status = cmdStatus;
}
}
dev = device & AUDIO_DEVICE_IN_ALL;
if (dev) {
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status_t status2 = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_INPUT_DEVICE,
sizeof(uint32_t),
&dev,
&size,
&cmdStatus);
if (status2 == NO_ERROR) {
status2 = cmdStatus;
}
if (status == NO_ERROR) {
status = status2;
}
}
}
return status;
}
status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) {
status_t cmdStatus;
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_AUDIO_MODE,
sizeof(audio_mode_t),
&mode,
&size,
&cmdStatus);
if (status == NO_ERROR) {
status = cmdStatus;
}
}
return status;
}
void AudioFlinger::EffectModule::setSuspended(bool suspended)
{
Mutex::Autolock _l(mLock);
mSuspended = suspended;
}
bool AudioFlinger::EffectModule::suspended() const
{
Mutex::Autolock _l(mLock);
return mSuspended;
}
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: %08X\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), mCblk(NULL),
mPriority(priority), mHasControl(false), mEnabled(false)
{
ALOGV("constructor %p", this);
if (client == 0) {
return;
}
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 != NULL) {
new(mCblk) effect_param_cblk_t();
mBuffer = (uint8_t *)mCblk + bufOffset;
}
} else {
ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
return;
}
}
AudioFlinger::EffectHandle::~EffectHandle()
{
ALOGV("Destructor %p", this);
disconnect(false);
ALOGV("Destructor DONE %p", this);
}
status_t AudioFlinger::EffectHandle::enable()
{
ALOGV("enable %p", this);
if (!mHasControl) return INVALID_OPERATION;
if (mEffect == 0) return DEAD_OBJECT;
if (mEnabled) {
return NO_ERROR;
}
mEnabled = true;
sp<ThreadBase> thread = mEffect->thread().promote();
if (thread != 0) {
thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId());
}
// checkSuspendOnEffectEnabled() can suspend this same effect when enabled
if (mEffect->suspended()) {
return NO_ERROR;
}
status_t status = mEffect->setEnabled(true);
if (status != NO_ERROR) {
if (thread != 0) {
thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
}
mEnabled = false;
}
return status;
}
status_t AudioFlinger::EffectHandle::disable()
{
ALOGV("disable %p", this);
if (!mHasControl) return INVALID_OPERATION;
if (mEffect == 0) return DEAD_OBJECT;
if (!mEnabled) {
return NO_ERROR;
}
mEnabled = false;
if (mEffect->suspended()) {
return NO_ERROR;
}
status_t status = mEffect->setEnabled(false);
sp<ThreadBase> thread = mEffect->thread().promote();
if (thread != 0) {
thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
}
return status;
}
void AudioFlinger::EffectHandle::disconnect()
{
disconnect(true);
}
void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
{
ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false");
if (mEffect == 0) {
return;
}
mEffect->disconnect(this, unpinIfLast);
if (mHasControl && mEnabled) {
sp<ThreadBase> thread = mEffect->thread().promote();
if (thread != 0) {
thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId());
}
}
// release sp on module => module destructor can be called now
mEffect.clear();
if (mClient != 0) {
if (mCblk != NULL) {
// unlike ~TrackBase(), mCblk is never a local new, so don't delete
mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
}
mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
// Client destructor must run with AudioFlinger mutex locked
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)
{
// ALOGV("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;
if (mClient == 0) return INVALID_OPERATION;
// 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) {
ALOGW("command(): invalid parameter block size");
break;
}
effect_param_t *param = (effect_param_t *)p;
if (param->psize == 0 || param->vsize == 0) {
ALOGW("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();
}
return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
{
ALOGV("setControl %p control %d", this, hasControl);
mHasControl = hasControl;
mEnabled = enabled;
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 = mCblk != NULL && tryLock(mCblk->lock);
snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mPriority,
mHasControl,
!locked,
mCblk ? mCblk->clientIndex : 0,
mCblk ? mCblk->serverIndex : 0
);
if (locked) {
mCblk->lock.unlock();
}
}
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectChain"
AudioFlinger::EffectChain::EffectChain(ThreadBase *thread,
int sessionId)
: mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
{
mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
if (thread == NULL) {
return;
}
mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
thread->frameCount();
}
AudioFlinger::EffectChain::~EffectChain()
{
if (mOwnInBuffer) {
delete mInBuffer;
}
}
// getEffectFromDesc_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
{
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) {
return mEffects[i];
}
}
return 0;
}
// getEffectFromId_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
{
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) {
return mEffects[i];
}
}
return 0;
}
// getEffectFromType_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
const effect_uuid_t *type)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
return mEffects[i];
}
}
return 0;
}
// Must be called with EffectChain::mLock locked
void AudioFlinger::EffectChain::process_l()
{
sp<ThreadBase> thread = mThread.promote();
if (thread == 0) {
ALOGW("process_l(): cannot promote mixer thread");
return;
}
bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
(mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
// always process effects unless no more tracks are on the session and the effect tail
// has been rendered
bool doProcess = true;
if (!isGlobalSession) {
bool tracksOnSession = (trackCnt() != 0);
if (!tracksOnSession && mTailBufferCount == 0) {
doProcess = false;
}
if (activeTrackCnt() == 0) {
// if no track is active and the effect tail has not been rendered,
// the input buffer must be cleared here as the mixer process will not do it
if (tracksOnSession || mTailBufferCount > 0) {
size_t numSamples = thread->frameCount() * thread->channelCount();
memset(mInBuffer, 0, numSamples * sizeof(int16_t));
if (mTailBufferCount > 0) {
mTailBufferCount--;
}
}
}
}
size_t size = mEffects.size();
if (doProcess) {
for (size_t i = 0; i < size; i++) {
mEffects[i]->process();
}
}
for (size_t i = 0; i < size; i++) {
mEffects[i]->updateState();
}
}
// 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
size_t size = mEffects.size();
size_t idx_insert = size;
ssize_t idx_insert_first = -1;
ssize_t idx_insert_last = -1;
for (size_t 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) {
ALOGW("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);
ALOGV("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);
size_t size = mEffects.size();
uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
for (size_t i = 0; i < size; i++) {
if (effect == mEffects[i]) {
// calling stop here will remove pre-processing effect from the audio HAL.
// This is safe as we hold the EffectChain mutex which guarantees that we are not in
// the middle of a read from audio HAL
if (mEffects[i]->state() == EffectModule::ACTIVE ||
mEffects[i]->state() == EffectModule::STOPPING) {
mEffects[i]->stop();
}
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);
ALOGV("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(audio_mode_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;
}
// must be called with ThreadBase::mLock held
void AudioFlinger::EffectChain::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend)
{
sp<SuspendedEffectDesc> desc;
// use effect type UUID timelow as key as there is no real risk of identical
// timeLow fields among effect type UUIDs.
ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow);
if (suspend) {
if (index >= 0) {
desc = mSuspendedEffects.valueAt(index);
} else {
desc = new SuspendedEffectDesc();
memcpy(&desc->mType, type, sizeof(effect_uuid_t));
mSuspendedEffects.add(type->timeLow, desc);
ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow);
}
if (desc->mRefCount++ == 0) {
sp<EffectModule> effect = getEffectIfEnabled(type);
if (effect != 0) {
desc->mEffect = effect;
effect->setSuspended(true);
effect->setEnabled(false);
}
}
} else {
if (index < 0) {
return;
}
desc = mSuspendedEffects.valueAt(index);
if (desc->mRefCount <= 0) {
ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount);
desc->mRefCount = 1;
}
if (--desc->mRefCount == 0) {
ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
if (desc->mEffect != 0) {
sp<EffectModule> effect = desc->mEffect.promote();
if (effect != 0) {
effect->setSuspended(false);
sp<EffectHandle> handle = effect->controlHandle();
if (handle != 0) {
effect->setEnabled(handle->enabled());
}
}
desc->mEffect.clear();
}
mSuspendedEffects.removeItemsAt(index);
}
}
}
// must be called with ThreadBase::mLock held
void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend)
{
sp<SuspendedEffectDesc> desc;
ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
if (suspend) {
if (index >= 0) {
desc = mSuspendedEffects.valueAt(index);
} else {
desc = new SuspendedEffectDesc();
mSuspendedEffects.add((int)kKeyForSuspendAll, desc);
ALOGV("setEffectSuspendedAll_l() add entry for 0");
}
if (desc->mRefCount++ == 0) {
Vector< sp<EffectModule> > effects;
getSuspendEligibleEffects(effects);
for (size_t i = 0; i < effects.size(); i++) {
setEffectSuspended_l(&effects[i]->desc().type, true);
}
}
} else {
if (index < 0) {
return;
}
desc = mSuspendedEffects.valueAt(index);
if (desc->mRefCount <= 0) {
ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount);
desc->mRefCount = 1;
}
if (--desc->mRefCount == 0) {
Vector<const effect_uuid_t *> types;
for (size_t i = 0; i < mSuspendedEffects.size(); i++) {
if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) {
continue;
}
types.add(&mSuspendedEffects.valueAt(i)->mType);
}
for (size_t i = 0; i < types.size(); i++) {
setEffectSuspended_l(types[i], false);
}
ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
}
}
}
// The volume effect is used for automated tests only
#ifndef OPENSL_ES_H_
static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
{ 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } };
const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_;
#endif //OPENSL_ES_H_
bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc)
{
// auxiliary effects and visualizer are never suspended on output mix
if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
(((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
(memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
(memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
return false;
}
return true;
}
void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
{
effects.clear();
for (size_t i = 0; i < mEffects.size(); i++) {
if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
effects.add(mEffects[i]);
}
}
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
const effect_uuid_t *type)
{
sp<EffectModule> effect = getEffectFromType_l(type);
return effect != 0 && effect->isEnabled() ? effect : 0;
}
void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled)
{
ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
if (enabled) {
if (index < 0) {
// if the effect is not suspend check if all effects are suspended
index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
if (index < 0) {
return;
}
if (!isEffectEligibleForSuspend(effect->desc())) {
return;
}
setEffectSuspended_l(&effect->desc().type, enabled);
index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
if (index < 0) {
ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!");
return;
}
}
ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
effect->desc().type.timeLow);
sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
// if effect is requested to suspended but was not yet enabled, supend it now.
if (desc->mEffect == 0) {
desc->mEffect = effect;
effect->setEnabled(false);
effect->setSuspended(true);
}
} else {
if (index < 0) {
return;
}
ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
effect->desc().type.timeLow);
sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
desc->mEffect.clear();
effect->setSuspended(false);
}
}
#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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