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ara-mdk / platform / frameworks / av / 15e5c9f49ad92ce985c9d0c2fe2767df2833b12c / . / services / audioflinger / AudioFlinger.cpp
blob: 87eb6aacf432edf7985e5c8567b6a618948ccc61 [file] [log] [blame]
/*
**
** 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 <dirent.h>
#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 <utils/Trace.h>
#include <binder/Parcel.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 <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 <powermanager/PowerManager.h>
#include <common_time/cc_helper.h>
//#include <common_time/local_clock.h>
#include <media/IMediaLogService.h>
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
// ----------------------------------------------------------------------------
// Note: the following macro is used for extremely verbose logging message. In
// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
// 0; but one side effect of this is to turn all LOGV's as well. Some messages
// are so verbose that we want to suppress them even when we have ALOG_ASSERT
// turned on. Do not uncomment the #def below unless you really know what you
// are doing and want to see all of the extremely verbose messages.
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
namespace android {
static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
static const char kHardwareLockedString[] = "Hardware lock is taken\n";
nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
uint32_t AudioFlinger::mScreenState;
#ifdef TEE_SINK
bool AudioFlinger::mTeeSinkInputEnabled = false;
bool AudioFlinger::mTeeSinkOutputEnabled = false;
bool AudioFlinger::mTeeSinkTrackEnabled = false;
size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
#endif
// ----------------------------------------------------------------------------
static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
const hw_module_t *mod;
int rc;
rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc) {
goto out;
}
rc = audio_hw_device_open(mod, dev);
ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc) {
goto out;
}
if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
rc = BAD_VALUE;
goto out;
}
return 0;
out:
*dev = NULL;
return rc;
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mPrimaryHardwareDev(NULL),
mHardwareStatus(AUDIO_HW_IDLE),
mMasterVolume(1.0f),
mMasterMute(false),
mNextUniqueId(1),
mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false)
{
getpid_cached = getpid();
char value[PROPERTY_VALUE_MAX];
bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
if (doLog) {
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters");
}
#ifdef TEE_SINK
(void) property_get("ro.debuggable", value, "0");
int debuggable = atoi(value);
int teeEnabled = 0;
if (debuggable) {
(void) property_get("af.tee", value, "0");
teeEnabled = atoi(value);
}
if (teeEnabled & 1)
mTeeSinkInputEnabled = true;
if (teeEnabled & 2)
mTeeSinkOutputEnabled = true;
if (teeEnabled & 4)
mTeeSinkTrackEnabled = true;
#endif
}
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));
}
}
mMode = AUDIO_MODE_NORMAL;
}
AudioFlinger::~AudioFlinger()
{
while (!mRecordThreads.isEmpty()) {
// closeInput_nonvirtual() will remove specified entry from mRecordThreads
closeInput_nonvirtual(mRecordThreads.keyAt(0));
}
while (!mPlaybackThreads.isEmpty()) {
// closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
closeOutput_nonvirtual(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.valueAt(i)->hwDevice());
delete mAudioHwDevs.valueAt(i);
}
}
static const char * const audio_interfaces[] = {
AUDIO_HARDWARE_MODULE_ID_PRIMARY,
AUDIO_HARDWARE_MODULE_ID_A2DP,
AUDIO_HARDWARE_MODULE_ID_USB,
};
#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
audio_module_handle_t module,
audio_devices_t devices)
{
// if module is 0, the request comes from an old policy manager and we should load
// well known modules
if (module == 0) {
ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
loadHwModule_l(audio_interfaces[i]);
}
// then try to find a module supporting the requested device.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
audio_hw_device_t *dev = audioHwDevice->hwDevice();
if ((dev->get_supported_devices != NULL) &&
(dev->get_supported_devices(dev) & devices) == devices)
return audioHwDevice;
}
} else {
// check a match for the requested module handle
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
if (audioHwDevice != NULL) {
return audioHwDevice;
}
}
return NULL;
}
void 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 count\n");
for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
AudioSessionRef *r = mAudioSessionRefs[i];
snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt);
result.append(buffer);
}
write(fd, result.string(), result.size());
}
void 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());
}
void 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());
}
bool AudioFlinger::dumpTryLock(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 = dumpTryLock(mHardwareLock);
if (!hardwareLocked) {
String8 result(kHardwareLockedString);
write(fd, result.string(), result.size());
} else {
mHardwareLock.unlock();
}
bool locked = dumpTryLock(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.valueAt(i)->hwDevice();
dev->dump(dev, fd);
}
#ifdef TEE_SINK
// dump the serially shared record tee sink
if (mRecordTeeSource != 0) {
dumpTee(fd, mRecordTeeSource);
}
#endif
if (locked) {
mLock.unlock();
}
// append a copy of media.log here by forwarding fd to it, but don't attempt
// to lookup the service if it's not running, as it will block for a second
if (mLogMemoryDealer != 0) {
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
if (binder != 0) {
fdprintf(fd, "\nmedia.log:\n");
Vector<String16> args;
binder->dump(fd, args);
}
}
}
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;
}
sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
{
if (mLogMemoryDealer == 0) {
return new NBLog::Writer();
}
sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
sp<NBLog::Writer> writer = new NBLog::Writer(size, shared);
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
if (binder != 0) {
interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name);
}
return writer;
}
void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
{
if (writer == 0) {
return;
}
sp<IMemory> iMemory(writer->getIMemory());
if (iMemory == 0) {
return;
}
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
if (binder != 0) {
interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory);
// Now the media.log remote reference to IMemory is gone.
// When our last local reference to IMemory also drops to zero,
// the IMemory destructor will deallocate the region from mMemoryDealer.
}
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
pid_t tid,
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;
}
// client is responsible for conversion of 8-bit PCM to 16-bit PCM,
// and we don't yet support 8.24 or 32-bit PCM
if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) {
ALOGE("createTrack() invalid format %d", format);
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
PlaybackThread *effectThread = NULL;
if (thread == NULL) {
ALOGE("no playback thread found for output handle %d", output);
lStatus = BAD_VALUE;
goto Exit;
}
pid_t pid = IPCThreadState::self()->getCallingPid();
client = registerPid_l(pid);
ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
// check if an effect chain with the same session ID is present on another
// output thread and move it here.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output) {
uint32_t sessions = t->hasAudioSession(*sessionId);
if (sessions & PlaybackThread::EFFECT_SESSION) {
effectThread = t.get();
break;
}
}
}
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, flags, tid, &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);
}
// Look for sync events awaiting for a session to be used.
for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) {
if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
if (lStatus == NO_ERROR) {
(void) track->setSyncEvent(mPendingSyncEvents[i]);
} else {
mPendingSyncEvents[i]->cancel();
}
mPendingSyncEvents.removeAt(i);
i--;
}
}
}
}
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 != NULL) {
*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;
}
// FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
// should examine all callers and fix them to handle smaller counts
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(): no playback thread found for output handle %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;
}
Mutex::Autolock _l(mLock);
mMasterVolume = value;
// Set master volume in the HALs which support it.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (dev->canSetMasterVolume()) {
dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
// Now set the master volume in each playback thread. Playback threads
// assigned to HALs which do not have master volume support will apply
// master volume during the mix operation. Threads with HALs which do
// support master volume will simply ignore the setting.
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterVolume(value);
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);
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_MODE;
ret = dev->set_mode(dev, 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);
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
ret = dev->set_mic_mute(dev, 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);
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
dev->get_mic_mute(dev, &state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
mMasterMute = muted;
// Set master mute in the HALs which support it.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if (dev->canSetMasterMute()) {
dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
// Now set the master mute in each playback thread. Playback threads
// assigned to HALs which do not have master mute support will apply master
// mute during the mix operation. Threads with HALs which do support master
// mute will simply ignore the setting.
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();
}
bool AudioFlinger::masterMute() const
{
Mutex::Autolock _l(mLock);
return masterMute_l();
}
float AudioFlinger::masterVolume_l() const
{
return mMasterVolume;
}
bool AudioFlinger::masterMute_l() const
{
return mMasterMute;
}
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)
{
ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
ioHandle, keyValuePairs.string(), 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) {
Mutex::Autolock _l(mLock);
status_t final_result = NO_ERROR;
{
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_PARAMETER;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t 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) {
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);
audio_devices_t device = thread->inDevice();
bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
// collect all of the thread's session IDs
KeyedVector<int, bool> ids = thread->sessionIds();
// suspend effects associated with those session IDs
for (size_t j = 0; j < ids.size(); ++j) {
int sessionId = ids.keyAt(j);
thread->setEffectSuspended(FX_IID_AEC,
suspend,
sessionId);
thread->setEffectSuspended(FX_IID_NS,
suspend,
sessionId);
}
}
mBtNrecIsOff = btNrecIsOff;
}
}
String8 screenState;
if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
bool isOff = screenState == "off";
if (isOff != (AudioFlinger::mScreenState & 1)) {
AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
}
}
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 == 0) {
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) &&
(value != 0)) {
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
{
ALOGVV("getParameters() io %d, keys %s, calling pid %d",
ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
if (ioHandle == 0) {
String8 out_s8;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
char *s;
{
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_PARAMETER;
audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
s = dev->get_parameters(dev, keys.string());
mHardwareStatus = AUDIO_HW_IDLE;
}
out_s8 += String8(s ? s : "");
free(s);
}
return out_s8;
}
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,
audio_channel_mask_t channelMask) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
struct audio_config config = {
sample_rate: sampleRate,
channel_mask: channelMask,
format: format,
};
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
size_t size = dev->get_input_buffer_size(dev, &config);
mHardwareStatus = AUDIO_HW_IDLE;
return size;
}
unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
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);
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
ret = dev->set_voice_volume(dev, value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_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)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED);
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED);
}
}
}
void AudioFlinger::removeNotificationClient(pid_t pid)
{
Mutex::Autolock _l(mLock);
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->mPid, i);
if (ref->mPid == pid) {
ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
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, const 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, calling pid %d", pid,
IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// getEffectThread_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId)
{
sp<PlaybackThread> thread;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
ALOG_ASSERT(thread == 0);
thread = mPlaybackThreads.valueAt(i);
}
}
return thread;
}
// ----------------------------------------------------------------------------
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);
}
// ----------------------------------------------------------------------------
sp<IAudioRecord> AudioFlinger::openRecord(
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
IAudioFlinger::track_flags_t flags,
pid_t tid,
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;
}
pid_t pid = IPCThreadState::self()->getCallingPid();
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, lSessionId, flags, tid, &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;
}
// ----------------------------------------------------------------------------
audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
{
if (!settingsAllowed()) {
return 0;
}
Mutex::Autolock _l(mLock);
return loadHwModule_l(name);
}
// loadHwModule_l() must be called with AudioFlinger::mLock held
audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
{
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
ALOGW("loadHwModule() module %s already loaded", name);
return mAudioHwDevs.keyAt(i);
}
}
audio_hw_device_t *dev;
int rc = load_audio_interface(name, &dev);
if (rc) {
ALOGI("loadHwModule() error %d loading module %s ", rc, name);
return 0;
}
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->init_check(dev);
mHardwareStatus = AUDIO_HW_IDLE;
if (rc) {
ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
return 0;
}
// Check and cache this HAL's level of support for master mute and master
// volume. If this is the first HAL opened, and it supports the get
// methods, use the initial values provided by the HAL as the current
// master mute and volume settings.
AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
{ // scope for auto-lock pattern
AutoMutex lock(mHardwareLock);
if (0 == mAudioHwDevs.size()) {
mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
if (NULL != dev->get_master_volume) {
float mv;
if (OK == dev->get_master_volume(dev, &mv)) {
mMasterVolume = mv;
}
}
mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
if (NULL != dev->get_master_mute) {
bool mm;
if (OK == dev->get_master_mute(dev, &mm)) {
mMasterMute = mm;
}
}
}
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if ((NULL != dev->set_master_volume) &&
(OK == dev->set_master_volume(dev, mMasterVolume))) {
flags = static_cast<AudioHwDevice::Flags>(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
}
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if ((NULL != dev->set_master_mute) &&
(OK == dev->set_master_mute(dev, mMasterMute))) {
flags = static_cast<AudioHwDevice::Flags>(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
audio_module_handle_t handle = nextUniqueId();
mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags));
ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
name, dev->common.module->name, dev->common.module->id, handle);
return handle;
}
// ----------------------------------------------------------------------------
uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = primaryPlaybackThread_l();
return thread != NULL ? thread->sampleRate() : 0;
}
size_t AudioFlinger::getPrimaryOutputFrameCount()
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = primaryPlaybackThread_l();
return thread != NULL ? thread->frameCountHAL() : 0;
}
// ----------------------------------------------------------------------------
audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
uint32_t *pLatencyMs,
audio_output_flags_t flags)
{
status_t status;
PlaybackThread *thread = NULL;
struct audio_config config = {
sample_rate: pSamplingRate ? *pSamplingRate : 0,
channel_mask: pChannelMask ? *pChannelMask : 0,
format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT,
};
audio_stream_out_t *outStream = NULL;
AudioHwDevice *outHwDev;
ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
module,
(pDevices != NULL) ? *pDevices : 0,
config.sample_rate,
config.format,
config.channel_mask,
flags);
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
outHwDev = findSuitableHwDev_l(module, *pDevices);
if (outHwDev == NULL)
return 0;
audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
audio_io_handle_t id = nextUniqueId();
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
status = hwDevHal->open_output_stream(hwDevHal,
id,
*pDevices,
(audio_output_flags_t)flags,
&config,
&outStream);
mHardwareStatus = AUDIO_HW_IDLE;
ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, "
"Channels %x, status %d",
outStream,
config.sample_rate,
config.format,
config.channel_mask,
status);
if (status == NO_ERROR && outStream != NULL) {
AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
(config.format != AUDIO_FORMAT_PCM_16_BIT) ||
(config.channel_mask != 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 = config.sample_rate;
if (pFormat != NULL) *pFormat = config.format;
if (pChannelMask != NULL) *pChannelMask = config.channel_mask;
if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
// the first primary output opened designates the primary hw device
if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
ALOGI("Using module %d has the primary audio interface", module);
mPrimaryHardwareDev = outHwDev;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
hwDevHal->set_mode(hwDevHal, mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
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)
{
return closeOutput_nonvirtual(output);
}
status_t AudioFlinger::closeOutput_nonvirtual(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();
ALOG_ASSERT(out != NULL, "out shouldn't be 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(audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask)
{
status_t status;
RecordThread *thread = NULL;
struct audio_config config = {
sample_rate: pSamplingRate ? *pSamplingRate : 0,
channel_mask: pChannelMask ? *pChannelMask : 0,
format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT,
};
uint32_t reqSamplingRate = config.sample_rate;
audio_format_t reqFormat = config.format;
audio_channel_mask_t reqChannels = config.channel_mask;
audio_stream_in_t *inStream = NULL;
AudioHwDevice *inHwDev;
if (pDevices == NULL || *pDevices == 0) {
return 0;
}
Mutex::Autolock _l(mLock);
inHwDev = findSuitableHwDev_l(module, *pDevices);
if (inHwDev == NULL)
return 0;
audio_hw_device_t *inHwHal = inHwDev->hwDevice();
audio_io_handle_t id = nextUniqueId();
status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
&inStream);
ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
"status %d",
inStream,
config.sample_rate,
config.format,
config.channel_mask,
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 (status == BAD_VALUE &&
reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
(config.sample_rate <= 2 * reqSamplingRate) &&
(popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
ALOGV("openInput() reopening with proposed sampling rate and channel mask");
inStream = NULL;
status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
}
if (status == NO_ERROR && inStream != NULL) {
#ifdef TEE_SINK
// Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
// or (re-)create if current Pipe is idle and does not match the new format
sp<NBAIO_Sink> teeSink;
enum {
TEE_SINK_NO, // don't copy input
TEE_SINK_NEW, // copy input using a new pipe
TEE_SINK_OLD, // copy input using an existing pipe
} kind;
NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
popcount(inStream->common.get_channels(&inStream->common)));
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
} else if (format == Format_Invalid) {
kind = TEE_SINK_NO;
} else if (mRecordTeeSink == 0) {
kind = TEE_SINK_NEW;
} else if (mRecordTeeSink->getStrongCount() != 1) {
kind = TEE_SINK_NO;
} else if (format == mRecordTeeSink->format()) {
kind = TEE_SINK_OLD;
} else {
kind = TEE_SINK_NEW;
}
switch (kind) {
case TEE_SINK_NEW: {
Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {format};
ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
PipeReader *pipeReader = new PipeReader(*pipe);
numCounterOffers = 0;
index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
mRecordTeeSink = pipe;
mRecordTeeSource = pipeReader;
teeSink = pipe;
}
break;
case TEE_SINK_OLD:
teeSink = mRecordTeeSink;
break;
case TEE_SINK_NO:
default:
break;
}
#endif
AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
// Start record thread
// RecorThread require both input and output device indication to forward to audio
// pre processing modules
thread = new RecordThread(this,
input,
reqSamplingRate,
reqChannels,
id,
primaryOutputDevice_l(),
*pDevices
#ifdef TEE_SINK
, teeSink
#endif
);
mRecordThreads.add(id, thread);
ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
if (pFormat != NULL) *pFormat = config.format;
if (pChannelMask != NULL) *pChannelMask = reqChannels;
// 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)
{
return closeInput_nonvirtual(input);
}
status_t AudioFlinger::closeInput_nonvirtual(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 == 0) {
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();
ALOG_ASSERT(in != NULL, "in shouldn't be 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);
ALOGV("setStreamOutput() stream %d to output %d", stream, output);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
thread->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->mSessionid == audioSession && ref->mPid == caller) {
ref->mCnt++;
ALOGV(" incremented refcount to %d", ref->mCnt);
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->mSessionid == audioSession && ref->mPid == caller) {
ref->mCnt--;
ALOGV(" decremented refcount to %d", ref->mCnt);
if (ref->mCnt == 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->mSessionid == sessionid) {
ALOGV(" session %d still exists for %d with %d refs",
sessionid, ref->mPid, ref->mCnt);
found = true;
break;
}
}
if (!found) {
Mutex::Autolock _l (t->mLock);
// remove all effects from the chain
while (ec->mEffects.size()) {
sp<EffectModule> effect = ec->mEffects[0];
effect->unPin();
t->removeEffect_l(effect);
if (effect->purgeHandles()) {
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() const
{
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
AudioStreamOut *output = thread->getOutput();
if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
return thread;
}
}
return NULL;
}
audio_devices_t AudioFlinger::primaryOutputDevice_l() const
{
PlaybackThread *thread = primaryPlaybackThread_l();
if (thread == NULL) {
return 0;
}
return thread->outDevice();
}
sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
int triggerSession,
int listenerSession,
sync_event_callback_t callBack,
void *cookie)
{
Mutex::Autolock _l(mLock);
sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
status_t playStatus = NAME_NOT_FOUND;
status_t recStatus = NAME_NOT_FOUND;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
if (playStatus == NO_ERROR) {
return event;
}
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
if (recStatus == NO_ERROR) {
return event;
}
}
if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
mPendingSyncEvents.add(event);
} else {
ALOGV("createSyncEvent() invalid event %d", event->type());
event.clear();
}
return event;
}
// ----------------------------------------------------------------------------
// 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(
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;
pid_t pid = IPCThreadState::self()->getCallingPid();
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;
d = desc;
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) {
desc = d;
}
}
// 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
*pDesc = desc;
// 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 != NULL) {
*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;
}
struct Entry {
#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav
char mName[MAX_NAME];
};
int comparEntry(const void *p1, const void *p2)
{
return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName);
}
#ifdef TEE_SINK
void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
{
NBAIO_Source *teeSource = source.get();
if (teeSource != NULL) {
// .wav rotation
// There is a benign race condition if 2 threads call this simultaneously.
// They would both traverse the directory, but the result would simply be
// failures at unlink() which are ignored. It's also unlikely since
// normally dumpsys is only done by bugreport or from the command line.
char teePath[32+256];
strcpy(teePath, "/data/misc/media");
size_t teePathLen = strlen(teePath);
DIR *dir = opendir(teePath);
teePath[teePathLen++] = '/';
if (dir != NULL) {
#define MAX_SORT 20 // number of entries to sort
#define MAX_KEEP 10 // number of entries to keep
struct Entry entries[MAX_SORT];
size_t entryCount = 0;
while (entryCount < MAX_SORT) {
struct dirent de;
struct dirent *result = NULL;
int rc = readdir_r(dir, &de, &result);
if (rc != 0) {
ALOGW("readdir_r failed %d", rc);
break;
}
if (result == NULL) {
break;
}
if (result != &de) {
ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
break;
}
// ignore non .wav file entries
size_t nameLen = strlen(de.d_name);
if (nameLen <= 4 || nameLen >= MAX_NAME ||
strcmp(&de.d_name[nameLen - 4], ".wav")) {
continue;
}
strcpy(entries[entryCount++].mName, de.d_name);
}
(void) closedir(dir);
if (entryCount > MAX_KEEP) {
qsort(entries, entryCount, sizeof(Entry), comparEntry);
for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) {
strcpy(&teePath[teePathLen], entries[i].mName);
(void) unlink(teePath);
}
}
} else {
if (fd >= 0) {
fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
}
}
char teeTime[16];
struct timeval tv;
gettimeofday(&tv, NULL);
struct tm tm;
localtime_r(&tv.tv_sec, &tm);
strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
// if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
if (teeFd >= 0) {
char wavHeader[44];
memcpy(wavHeader,
"RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
sizeof(wavHeader));
NBAIO_Format format = teeSource->format();
unsigned channelCount = Format_channelCount(format);
ALOG_ASSERT(channelCount <= FCC_2);
uint32_t sampleRate = Format_sampleRate(format);
wavHeader[22] = channelCount; // number of channels
wavHeader[24] = sampleRate; // sample rate
wavHeader[25] = sampleRate >> 8;
wavHeader[32] = channelCount * 2; // block alignment
write(teeFd, wavHeader, sizeof(wavHeader));
size_t total = 0;
bool firstRead = true;
for (;;) {
#define TEE_SINK_READ 1024
short buffer[TEE_SINK_READ * FCC_2];
size_t count = TEE_SINK_READ;
ssize_t actual = teeSource->read(buffer, count,
AudioBufferProvider::kInvalidPTS);
bool wasFirstRead = firstRead;
firstRead = false;
if (actual <= 0) {
if (actual == (ssize_t) OVERRUN && wasFirstRead) {
continue;
}
break;
}
ALOG_ASSERT(actual <= (ssize_t)count);
write(teeFd, buffer, actual * channelCount * sizeof(short));
total += actual;
}
lseek(teeFd, (off_t) 4, SEEK_SET);
uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
write(teeFd, &temp, sizeof(temp));
lseek(teeFd, (off_t) 40, SEEK_SET);
temp = total * channelCount * sizeof(short);
write(teeFd, &temp, sizeof(temp));
close(teeFd);
if (fd >= 0) {
fdprintf(fd, "tee copied to %s\n", teePath);
}
} else {
if (fd >= 0) {
fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
}
}
}
}
#endif
// ----------------------------------------------------------------------------
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioFlinger::onTransact(code, data, reply, flags);
}
}; // namespace android