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replicant-frameworks_native/libs/audioflinger/AudioFlinger.cpp
Nick Pelly 07c8a20f39 Standby A2DP audio hardware interface when disabling A2DP. 
Patch supplied on advice of partner. This causes us to send suspend_sink to
Bluez via socket interface, so we enter suspend on the A2DP link faster.
This is especially important when switching to SCO so that we come closer to
whitepaper recommendations to suspend A2DP before setting up SCO.

We have another patch set to add DBUS A2DP suspend and resume calls to Bluez
that will do a better job of following whitepaper recommendations for
A2DP -> SCO -> A2DP, but this small patch is still an improvement.
2009-07-15 12:22:55 -07:00

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/* //device/include/server/AudioFlinger/AudioFlinger.cpp
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <cutils/properties.h>
#include <media/AudioTrack.h>
#include <media/AudioRecord.h>
#include <private/media/AudioTrackShared.h>
#include <hardware_legacy/AudioHardwareInterface.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#ifdef WITH_A2DP
#include "A2dpAudioInterface.h"
#endif
// ----------------------------------------------------------------------------
// the sim build doesn't have gettid
#ifndef HAVE_GETTID
# define gettid getpid
#endif
// ----------------------------------------------------------------------------
namespace android {
static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
static const char* kHardwareLockedString = "Hardware lock is taken\n";
//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const unsigned long kBufferRecoveryInUsecs = 2000;
static const unsigned long kMaxBufferRecoveryInUsecs = 20000;
static const float MAX_GAIN = 4096.0f;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
static const int8_t kMaxTrackRetries = 50;
static const int8_t kMaxTrackStartupRetries = 50;
static const int kStartSleepTime = 30000;
static const int kStopSleepTime = 30000;
static const int kDumpLockRetries = 50;
static const int kDumpLockSleep = 20000;
// Maximum number of pending buffers allocated by OutputTrack::write()
static const uint8_t kMaxOutputTrackBuffers = 5;
#define AUDIOFLINGER_SECURITY_ENABLED 1
// ----------------------------------------------------------------------------
static bool recordingAllowed() {
#ifndef HAVE_ANDROID_OS
return true;
#endif
#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
return ok;
#else
if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
return true;
#endif
}
static bool settingsAllowed() {
#ifndef HAVE_ANDROID_OS
return true;
#endif
#if AUDIOFLINGER_SECURITY_ENABLED
if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
return ok;
#else
if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
return true;
#endif
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mAudioHardware(0), mA2dpAudioInterface(0), mA2dpEnabled(false), mNotifyA2dpChange(false),
mForcedSpeakerCount(0), mA2dpDisableCount(0), mA2dpSuppressed(false), mForcedRoute(0),
mRouteRestoreTime(0), mMusicMuteSaved(false)
{
mHardwareStatus = AUDIO_HW_IDLE;
mAudioHardware = AudioHardwareInterface::create();
mHardwareStatus = AUDIO_HW_INIT;
if (mAudioHardware->initCheck() == NO_ERROR) {
// open 16-bit output stream for s/w mixer
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
status_t status;
AudioStreamOut *hwOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
mHardwareStatus = AUDIO_HW_IDLE;
if (hwOutput) {
mHardwareMixerThread = new MixerThread(this, hwOutput, AudioSystem::AUDIO_OUTPUT_HARDWARE);
} else {
LOGE("Failed to initialize hardware output stream, status: %d", status);
}
#ifdef WITH_A2DP
// Create A2DP interface
mA2dpAudioInterface = new A2dpAudioInterface();
AudioStreamOut *a2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
if (a2dpOutput) {
mA2dpMixerThread = new MixerThread(this, a2dpOutput, AudioSystem::AUDIO_OUTPUT_A2DP);
if (hwOutput) {
uint32_t frameCount = ((a2dpOutput->bufferSize()/a2dpOutput->frameSize()) * hwOutput->sampleRate()) / a2dpOutput->sampleRate();
MixerThread::OutputTrack *a2dpOutTrack = new MixerThread::OutputTrack(mA2dpMixerThread,
hwOutput->sampleRate(),
AudioSystem::PCM_16_BIT,
hwOutput->channelCount(),
frameCount);
mHardwareMixerThread->setOuputTrack(a2dpOutTrack);
}
} else {
LOGE("Failed to initialize A2DP output stream, status: %d", status);
}
#endif
// FIXME - this should come from settings
setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
setMode(AudioSystem::MODE_NORMAL);
setMasterVolume(1.0f);
setMasterMute(false);
// Start record thread
mAudioRecordThread = new AudioRecordThread(mAudioHardware, this);
if (mAudioRecordThread != 0) {
mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
}
} else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
}
AudioFlinger::~AudioFlinger()
{
if (mAudioRecordThread != 0) {
mAudioRecordThread->exit();
mAudioRecordThread.clear();
}
mHardwareMixerThread.clear();
delete mAudioHardware;
// deleting mA2dpAudioInterface also deletes mA2dpOutput;
#ifdef WITH_A2DP
mA2dpMixerThread.clear();
delete mA2dpAudioInterface;
#endif
}
#ifdef WITH_A2DP
// setA2dpEnabled_l() must be called with AudioFlinger::mLock held
void AudioFlinger::setA2dpEnabled_l(bool enable)
{
SortedVector < sp<MixerThread::Track> > tracks;
SortedVector < wp<MixerThread::Track> > activeTracks;
LOGV_IF(enable, "set output to A2DP\n");
LOGV_IF(!enable, "set output to hardware audio\n");
// Transfer tracks playing on MUSIC stream from one mixer to the other
if (enable) {
mHardwareMixerThread->getTracks_l(tracks, activeTracks);
mA2dpMixerThread->putTracks_l(tracks, activeTracks);
} else {
mA2dpMixerThread->getTracks_l(tracks, activeTracks);
mHardwareMixerThread->putTracks_l(tracks, activeTracks);
mA2dpMixerThread->mOutput->standby();
}
mA2dpEnabled = enable;
mNotifyA2dpChange = true;
mWaitWorkCV.broadcast();
}
// checkA2dpEnabledChange_l() must be called with AudioFlinger::mLock held
void AudioFlinger::checkA2dpEnabledChange_l()
{
if (mNotifyA2dpChange) {
// Notify AudioSystem of the A2DP activation/deactivation
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
sp<IBinder> binder = mNotificationClients.itemAt(i).promote();
if (binder != NULL) {
LOGV("Notifying output change to client %p", binder.get());
sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
client->a2dpEnabledChanged(mA2dpEnabled);
}
}
mNotifyA2dpChange = false;
}
}
#endif // WITH_A2DP
bool AudioFlinger::streamForcedToSpeaker(int streamType)
{
// NOTE that streams listed here must not be routed to A2DP by default:
// AudioSystem::routedToA2dpOutput(streamType) == false
return (streamType == AudioSystem::RING ||
streamType == AudioSystem::ALARM ||
streamType == AudioSystem::NOTIFICATION ||
streamType == AudioSystem::ENFORCED_AUDIBLE);
}
status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
wp<Client> wClient = mClients.valueAt(i);
if (wClient != 0) {
sp<Client> client = wClient.promote();
if (client != 0) {
snprintf(buffer, SIZE, " pid: %d\n", client->pid());
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
int hardwareStatus = mHardwareStatus;
if (hardwareStatus == AUDIO_HW_IDLE && mHardwareMixerThread->mStandby) {
hardwareStatus = AUDIO_HW_STANDBY;
}
snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Permission Denial: "
"can't dump AudioFlinger from pid=%d, uid=%d\n",
IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
static bool tryLock(Mutex& mutex)
{
bool locked = false;
for (int i = 0; i < kDumpLockRetries; ++i) {
if (mutex.tryLock() == NO_ERROR) {
locked = true;
break;
}
usleep(kDumpLockSleep);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
bool hardwareLocked = tryLock(mHardwareLock);
if (!hardwareLocked) {
String8 result(kHardwareLockedString);
write(fd, result.string(), result.size());
} else {
mHardwareLock.unlock();
}
bool locked = tryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
String8 result(kDeadlockedString);
write(fd, result.string(), result.size());
}
dumpClients(fd, args);
dumpInternals(fd, args);
mHardwareMixerThread->dump(fd, args);
#ifdef WITH_A2DP
mA2dpMixerThread->dump(fd, args);
#endif
// dump record client
if (mAudioRecordThread != 0) mAudioRecordThread->dump(fd, args);
if (mAudioHardware) {
mAudioHardware->dumpState(fd, args);
}
if (locked) mLock.unlock();
}
return NO_ERROR;
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
status_t *status)
{
sp<MixerThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
status_t lStatus;
if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
#ifdef WITH_A2DP
if (isA2dpEnabled() && AudioSystem::routedToA2dpOutput(streamType)) {
track = mA2dpMixerThread->createTrack_l(client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, &lStatus);
} else
#endif
{
track = mHardwareMixerThread->createTrack_l(client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer, &lStatus);
}
}
if (lStatus == NO_ERROR) {
trackHandle = new TrackHandle(track);
} else {
track.clear();
}
Exit:
if(status) {
*status = lStatus;
}
return trackHandle;
}
uint32_t AudioFlinger::sampleRate(int output) const
{
#ifdef WITH_A2DP
if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
return mA2dpMixerThread->sampleRate();
}
#endif
return mHardwareMixerThread->sampleRate();
}
int AudioFlinger::channelCount(int output) const
{
#ifdef WITH_A2DP
if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
return mA2dpMixerThread->channelCount();
}
#endif
return mHardwareMixerThread->channelCount();
}
int AudioFlinger::format(int output) const
{
#ifdef WITH_A2DP
if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
return mA2dpMixerThread->format();
}
#endif
return mHardwareMixerThread->format();
}
size_t AudioFlinger::frameCount(int output) const
{
#ifdef WITH_A2DP
if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
return mA2dpMixerThread->frameCount();
}
#endif
return mHardwareMixerThread->frameCount();
}
uint32_t AudioFlinger::latency(int output) const
{
#ifdef WITH_A2DP
if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
return mA2dpMixerThread->latency();
}
#endif
return mHardwareMixerThread->latency();
}
status_t AudioFlinger::setMasterVolume(float value)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
// when hw supports master volume, don't scale in sw mixer
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
value = 1.0f;
}
mHardwareStatus = AUDIO_HW_IDLE;
mHardwareMixerThread->setMasterVolume(value);
#ifdef WITH_A2DP
mA2dpMixerThread->setMasterVolume(value);
#endif
return NO_ERROR;
}
status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
{
status_t err = NO_ERROR;
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
return BAD_VALUE;
}
#ifdef WITH_A2DP
LOGV("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(),
IPCThreadState::self()->getCallingPid());
if (mode == AudioSystem::MODE_NORMAL &&
(mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
AutoMutex lock(&mLock);
bool enableA2dp = false;
if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
enableA2dp = true;
}
if (mA2dpDisableCount > 0) {
mA2dpSuppressed = enableA2dp;
} else {
setA2dpEnabled_l(enableA2dp);
}
LOGV("setOutput done\n");
}
// setRouting() is always called at least for mode == AudioSystem::MODE_IN_CALL when
// SCO is enabled, whatever current mode is so we can safely handle A2DP disabling only
// in this case to avoid doing it several times.
if (mode == AudioSystem::MODE_IN_CALL &&
(mask & AudioSystem::ROUTE_BLUETOOTH_SCO)) {
AutoMutex lock(&mLock);
handleRouteDisablesA2dp_l(routes);
}
#endif
// do nothing if only A2DP routing is affected
mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
if (mask) {
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_ROUTING;
uint32_t r;
err = mAudioHardware->getRouting(mode, &r);
if (err == NO_ERROR) {
r = (r & ~mask) | (routes & mask);
if (mode == AudioSystem::MODE_NORMAL ||
(mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
mSavedRoute = r;
r |= mForcedRoute;
LOGV("setRouting mSavedRoute %08x mForcedRoute %08x\n", mSavedRoute, mForcedRoute);
}
mHardwareStatus = AUDIO_HW_SET_ROUTING;
err = mAudioHardware->setRouting(mode, r);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
return err;
}
uint32_t AudioFlinger::getRouting(int mode) const
{
uint32_t routes = 0;
if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
if (mode == AudioSystem::MODE_NORMAL ||
(mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
routes = mSavedRoute;
} else {
mHardwareStatus = AUDIO_HW_GET_ROUTING;
mAudioHardware->getRouting(mode, &routes);
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
LOGW("Illegal value: getRouting(%d)", mode);
}
return routes;
}
status_t AudioFlinger::setMode(int mode)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
LOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
status_t ret = mAudioHardware->setMode(mode);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
int AudioFlinger::getMode() const
{
int mode = AudioSystem::MODE_INVALID;
mHardwareStatus = AUDIO_HW_SET_MODE;
mAudioHardware->getMode(&mode);
mHardwareStatus = AUDIO_HW_IDLE;
return mode;
}
status_t AudioFlinger::setMicMute(bool state)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
status_t ret = mAudioHardware->setMicMute(state);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
bool state = AudioSystem::MODE_INVALID;
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mAudioHardware->getMicMute(&state);
mHardwareStatus = AUDIO_HW_IDLE;
return state;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
mHardwareMixerThread->setMasterMute(muted);
#ifdef WITH_A2DP
mA2dpMixerThread->setMasterMute(muted);
#endif
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
return mHardwareMixerThread->masterVolume();
}
bool AudioFlinger::masterMute() const
{
return mHardwareMixerThread->masterMute();
}
status_t AudioFlinger::setStreamVolume(int stream, float value)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
return BAD_VALUE;
}
status_t ret = NO_ERROR;
if (stream == AudioSystem::VOICE_CALL ||
stream == AudioSystem::BLUETOOTH_SCO) {
float hwValue;
if (stream == AudioSystem::VOICE_CALL) {
hwValue = (float)AudioSystem::logToLinear(value)/100.0f;
// offset value to reflect actual hardware volume that never reaches 0
// 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
value = 0.01 + 0.99 * value;
} else { // (type == AudioSystem::BLUETOOTH_SCO)
hwValue = 1.0f;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
ret = mAudioHardware->setVoiceVolume(hwValue);
mHardwareStatus = AUDIO_HW_IDLE;
}
mHardwareMixerThread->setStreamVolume(stream, value);
#ifdef WITH_A2DP
mA2dpMixerThread->setStreamVolume(stream, value);
#endif
mHardwareMixerThread->setStreamVolume(stream, value);
#ifdef WITH_A2DP
mA2dpMixerThread->setStreamVolume(stream, value);
#endif
return ret;
}
status_t AudioFlinger::setStreamMute(int stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
return BAD_VALUE;
}
#ifdef WITH_A2DP
mA2dpMixerThread->setStreamMute(stream, muted);
#endif
if (stream == AudioSystem::MUSIC)
{
AutoMutex lock(&mHardwareLock);
if (mForcedRoute != 0)
mMusicMuteSaved = muted;
else
mHardwareMixerThread->setStreamMute(stream, muted);
} else {
mHardwareMixerThread->setStreamMute(stream, muted);
}
return NO_ERROR;
}
float AudioFlinger::streamVolume(int stream) const
{
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return 0.0f;
}
float volume = mHardwareMixerThread->streamVolume(stream);
// remove correction applied by setStreamVolume()
if (stream == AudioSystem::VOICE_CALL) {
volume = (volume - 0.01) / 0.99 ;
}
return volume;
}
bool AudioFlinger::streamMute(int stream) const
{
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return true;
}
if (stream == AudioSystem::MUSIC && mForcedRoute != 0)
{
return mMusicMuteSaved;
}
return mHardwareMixerThread->streamMute(stream);
}
bool AudioFlinger::isMusicActive() const
{
Mutex::Autolock _l(mLock);
#ifdef WITH_A2DP
if (isA2dpEnabled()) {
return mA2dpMixerThread->isMusicActive_l();
}
#endif
return mHardwareMixerThread->isMusicActive_l();
}
status_t AudioFlinger::setParameter(const char* key, const char* value)
{
status_t result, result2;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_SET_PARAMETER;
LOGV("setParameter() key %s, value %s, tid %d, calling tid %d", key, value, gettid(), IPCThreadState::self()->getCallingPid());
result = mAudioHardware->setParameter(key, value);
if (mA2dpAudioInterface) {
result2 = mA2dpAudioInterface->setParameter(key, value);
if (result2)
result = result2;
}
mHardwareStatus = AUDIO_HW_IDLE;
return result;
}
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
{
return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
}
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{
LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
sp<IBinder> binder = client->asBinder();
if (mNotificationClients.indexOf(binder) < 0) {
LOGV("Adding notification client %p", binder.get());
binder->linkToDeath(this);
mNotificationClients.add(binder);
client->a2dpEnabledChanged(isA2dpEnabled());
}
}
void AudioFlinger::binderDied(const wp<IBinder>& who) {
LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
IBinder *binder = who.unsafe_get();
if (binder != NULL) {
int index = mNotificationClients.indexOf(binder);
if (index >= 0) {
LOGV("Removing notification client %p", binder);
mNotificationClients.removeAt(index);
}
}
}
void AudioFlinger::removeClient(pid_t pid)
{
LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
mClients.removeItem(pid);
}
bool AudioFlinger::isA2dpEnabled() const
{
return mA2dpEnabled;
}
void AudioFlinger::handleForcedSpeakerRoute(int command)
{
switch(command) {
case ACTIVE_TRACK_ADDED:
{
AutoMutex lock(mHardwareLock);
if (mForcedSpeakerCount++ == 0) {
if (mForcedRoute == 0) {
mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
LOGV("++mForcedSpeakerCount == 0, mMusicMuteSaved = %d, mRouteRestoreTime = %d", mMusicMuteSaved, mRouteRestoreTime);
if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
usleep(mHardwareMixerThread->latency()*1000);
mHardwareStatus = AUDIO_HW_SET_ROUTING;
mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
mHardwareStatus = AUDIO_HW_IDLE;
// delay track start so that audio hardware has time to siwtch routes
usleep(kStartSleepTime);
}
}
mForcedRoute = AudioSystem::ROUTE_SPEAKER;
mRouteRestoreTime = 0;
}
LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
}
break;
case ACTIVE_TRACK_REMOVED:
{
AutoMutex lock(mHardwareLock);
if (mForcedSpeakerCount > 0){
if (--mForcedSpeakerCount == 0) {
mRouteRestoreTime = systemTime() + milliseconds(kStopSleepTime/1000);
}
LOGV("mForcedSpeakerCount decremented to %d", mForcedSpeakerCount);
} else {
LOGE("mForcedSpeakerCount is already zero");
}
}
break;
case CHECK_ROUTE_RESTORE_TIME:
case FORCE_ROUTE_RESTORE:
if (mRouteRestoreTime) {
AutoMutex lock(mHardwareLock);
if (mRouteRestoreTime &&
(systemTime() > mRouteRestoreTime || command == FORCE_ROUTE_RESTORE)) {
mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, mMusicMuteSaved);
mForcedRoute = 0;
if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
mHardwareStatus = AUDIO_HW_SET_ROUTING;
mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute);
mHardwareStatus = AUDIO_HW_IDLE;
LOGV("Route forced to Speaker OFF %08x", mSavedRoute);
}
mRouteRestoreTime = 0;
}
}
break;
}
}
#ifdef WITH_A2DP
// handleRouteDisablesA2dp_l() must be called with AudioFlinger::mLock held
void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
{
if (routes & AudioSystem::ROUTE_BLUETOOTH_SCO) {
if (mA2dpDisableCount++ == 0) {
if (mA2dpEnabled) {
setA2dpEnabled_l(false);
mA2dpSuppressed = true;
}
}
LOGV("mA2dpDisableCount incremented to %d", mA2dpDisableCount);
} else {
if (mA2dpDisableCount > 0) {
if (--mA2dpDisableCount == 0) {
if (mA2dpSuppressed) {
setA2dpEnabled_l(true);
mA2dpSuppressed = false;
}
}
LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
} else {
LOGV("mA2dpDisableCount is already zero");
}
}
}
#endif
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType)
: Thread(false),
mAudioFlinger(audioFlinger), mAudioMixer(0), mOutput(output), mOutputType(outputType),
mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
mInWrite(false)
{
mSampleRate = output->sampleRate();
mChannelCount = output->channelCount();
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
LOGE("Invalid audio hardware channel count");
}
mFormat = output->format();
mFrameCount = output->bufferSize() / output->channelCount() / sizeof(int16_t);
mAudioMixer = new AudioMixer(mFrameCount, output->sampleRate());
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
mMixBuffer = new int16_t[mFrameCount * 2];
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
}
AudioFlinger::MixerThread::~MixerThread()
{
delete [] mMixBuffer;
delete mAudioMixer;
}
status_t AudioFlinger::MixerThread::dump(int fd, const Vector<String16>& args)
{
dumpInternals(fd, args);
dumpTracks(fd, args);
return NO_ERROR;
}
status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
snprintf(buffer, SIZE, "Output %d mixer thread active tracks\n", mOutputType);
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
wp<Track> wTrack = mActiveTracks[i];
if (wTrack != 0) {
sp<Track> track = wTrack.promote();
if (track != 0) {
track->dump(buffer, SIZE);
result.append(buffer);
}
}
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Output %d mixer thread internals\n", mOutputType);
result.append(buffer);
snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
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, "standby: %d\n", mStandby);
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
}
// Thread virtuals
bool AudioFlinger::MixerThread::threadLoop()
{
unsigned long sleepTime = kBufferRecoveryInUsecs;
int16_t* curBuf = mMixBuffer;
Vector< sp<Track> > tracksToRemove;
size_t enabledTracks = 0;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
#ifdef WITH_A2DP
bool outputTrackActive = false;
#endif
do {
enabledTracks = 0;
{ // scope for the AudioFlinger::mLock
Mutex::Autolock _l(mAudioFlinger->mLock);
#ifdef WITH_A2DP
if (mOutputTrack != NULL && !mAudioFlinger->isA2dpEnabled()) {
if (outputTrackActive) {
mAudioFlinger->mLock.unlock();
mOutputTrack->stop();
mAudioFlinger->mLock.lock();
outputTrackActive = false;
}
}
mAudioFlinger->checkA2dpEnabledChange_l();
#endif
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
// wait until we have something to do...
LOGV("Audio hardware entering standby, output %d\n", mOutputType);
if (!mStandby) {
mOutput->standby();
mStandby = true;
}
#ifdef WITH_A2DP
if (outputTrackActive) {
mAudioFlinger->mLock.unlock();
mOutputTrack->stop();
mAudioFlinger->mLock.lock();
outputTrackActive = false;
}
#endif
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
mAudioFlinger->handleForcedSpeakerRoute(FORCE_ROUTE_RESTORE);
}
// we're about to wait, flush the binder command buffer
IPCThreadState::self()->flushCommands();
mAudioFlinger->mWaitWorkCV.wait(mAudioFlinger->mLock);
LOGV("Audio hardware exiting standby, output %d\n", mOutputType);
if (mMasterMute == false) {
char value[PROPERTY_VALUE_MAX];
property_get("ro.audio.silent", value, "0");
if (atoi(value)) {
LOGD("Silence is golden");
setMasterMute(true);
}
}
standbyTime = systemTime() + kStandbyTimeInNsecs;
continue;
}
// Forced route to speaker is handled by hardware mixer thread
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
mAudioFlinger->handleForcedSpeakerRoute(CHECK_ROUTE_RESTORE_TIME);
}
// find out which tracks need to be processed
size_t count = activeTracks.size();
for (size_t i=0 ; i<count ; i++) {
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
mAudioMixer->setActiveTrack(track->name());
if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
!track->isPaused())
{
//LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
// compute volume for this track
int16_t left, right;
if (track->isMuted() || mMasterMute || track->isPausing()) {
left = right = 0;
if (track->isPausing()) {
LOGV("paused(%d)", track->name());
track->setPaused();
}
} else {
float typeVolume = mStreamTypes[track->type()].volume;
float v = mMasterVolume * typeVolume;
float v_clamped = v * cblk->volume[0];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = int16_t(v_clamped);
v_clamped = v * cblk->volume[1];
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = int16_t(v_clamped);
}
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(track);
mAudioMixer->enable(AudioMixer::MIXING);
int param;
if ( track->mFillingUpStatus == Track::FS_FILLED) {
// no ramp for the first volume setting
track->mFillingUpStatus = Track::FS_ACTIVE;
if (track->mState == TrackBase::RESUMING) {
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
} else {
param = AudioMixer::VOLUME;
}
} else {
param = AudioMixer::RAMP_VOLUME;
}
mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::FORMAT, track->format());
mAudioMixer->setParameter(
AudioMixer::TRACK,
AudioMixer::CHANNEL_COUNT, track->channelCount());
mAudioMixer->setParameter(
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
int(cblk->sampleRate));
// reset retry count
track->mRetryCount = kMaxTrackRetries;
enabledTracks++;
} else {
//LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
if (track->isStopped()) {
track->reset();
}
if (track->isTerminated() || track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
// Remove it from the list of active tracks.
LOGV("remove(%d) from active list", track->name());
tracksToRemove.add(track);
} else {
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
tracksToRemove.add(track);
}
}
// LOGV("disable(%d)", track->name());
mAudioMixer->disable(AudioMixer::MIXING);
}
}
// remove all the tracks that need to be...
count = tracksToRemove.size();
if (UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove[i];
removeActiveTrack_l(track);
if (track->isTerminated()) {
mTracks.remove(track);
deleteTrackName_l(track->mName);
}
}
}
}
if (LIKELY(enabledTracks)) {
// mix buffers...
mAudioMixer->process(curBuf);
#ifdef WITH_A2DP
if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
if (!outputTrackActive) {
LOGV("starting output track in mixer for output %d", mOutputType);
mOutputTrack->start();
outputTrackActive = true;
}
mOutputTrack->write(curBuf, mFrameCount);
}
#endif
// output audio to hardware
mLastWriteTime = systemTime();
mInWrite = true;
mOutput->write(curBuf, mixBufferSize);
mNumWrites++;
mInWrite = false;
mStandby = false;
nsecs_t temp = systemTime();
standbyTime = temp + kStandbyTimeInNsecs;
nsecs_t delta = temp - mLastWriteTime;
if (delta > maxPeriod) {
LOGW("write blocked for %llu msecs", ns2ms(delta));
mNumDelayedWrites++;
}
sleepTime = kBufferRecoveryInUsecs;
} else {
#ifdef WITH_A2DP
if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
if (outputTrackActive) {
mOutputTrack->write(curBuf, 0);
if (mOutputTrack->bufferQueueEmpty()) {
mOutputTrack->stop();
outputTrackActive = false;
} else {
standbyTime = systemTime() + kStandbyTimeInNsecs;
}
}
}
#endif
// There was nothing to mix this round, which means all
// active tracks were late. Sleep a little bit to give
// them another chance. If we're too late, the audio
// hardware will zero-fill for us.
//LOGV("no buffers - usleep(%lu)", sleepTime);
usleep(sleepTime);
if (sleepTime < kMaxBufferRecoveryInUsecs) {
sleepTime += kBufferRecoveryInUsecs;
}
}
// 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();
} while (true);
return false;
}
status_t AudioFlinger::MixerThread::readyToRun()
{
if (mSampleRate == 0) {
LOGE("No working audio driver found.");
return NO_INIT;
}
LOGI("AudioFlinger's thread ready to run for output %d", mOutputType);
return NO_ERROR;
}
void AudioFlinger::MixerThread::onFirstRef()
{
const size_t SIZE = 256;
char buffer[SIZE];
snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType);
run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
}
// MixerThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer,
status_t *status)
{
sp<Track> track;
status_t lStatus;
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (sampleRate > mSampleRate*2) {
LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
if (mSampleRate == 0) {
LOGE("Audio driver not initialized.");
lStatus = NO_INIT;
goto Exit;
}
track = new Track(this, client, streamType, sampleRate, format,
channelCount, frameCount, sharedBuffer);
if (track->getCblk() == NULL) {
lStatus = NO_MEMORY;
goto Exit;
}
mTracks.add(track);
lStatus = NO_ERROR;
Exit:
if(status) {
*status = lStatus;
}
return track;
}
// getTracks_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::getTracks_l(
SortedVector < sp<Track> >& tracks,
SortedVector < wp<Track> >& activeTracks)
{
size_t size = mTracks.size();
LOGV ("MixerThread::getTracks_l() for output %d, mTracks.size %d, mActiveTracks.size %d", mOutputType, mTracks.size(), mActiveTracks.size());
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
if (AudioSystem::routedToA2dpOutput(t->mStreamType)) {
tracks.add(t);
int j = mActiveTracks.indexOf(t);
if (j >= 0) {
t = mActiveTracks[j].promote();
if (t != NULL) {
activeTracks.add(t);
}
}
}
}
size = activeTracks.size();
for (size_t i = 0; i < size; i++) {
removeActiveTrack_l(activeTracks[i]);
}
size = tracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = tracks[i];
mTracks.remove(t);
deleteTrackName_l(t->name());
}
}
// putTracks_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::putTracks_l(
SortedVector < sp<Track> >& tracks,
SortedVector < wp<Track> >& activeTracks)
{
LOGV ("MixerThread::putTracks_l() for output %d, tracks.size %d, activeTracks.size %d", mOutputType, tracks.size(), activeTracks.size());
size_t size = tracks.size();
for (size_t i = 0; i < size ; i++) {
sp<Track> t = tracks[i];
int name = getTrackName_l();
if (name < 0) return;
t->mName = name;
t->mMixerThread = this;
mTracks.add(t);
int j = activeTracks.indexOf(t);
if (j >= 0) {
addActiveTrack_l(t);
}
}
}
uint32_t AudioFlinger::MixerThread::sampleRate() const
{
return mSampleRate;
}
int AudioFlinger::MixerThread::channelCount() const
{
return mChannelCount;
}
int AudioFlinger::MixerThread::format() const
{
return mFormat;
}
size_t AudioFlinger::MixerThread::frameCount() const
{
return mFrameCount;
}
uint32_t AudioFlinger::MixerThread::latency() const
{
if (mOutput) {
return mOutput->latency();
}
else {
return 0;
}
}
status_t AudioFlinger::MixerThread::setMasterVolume(float value)
{
mMasterVolume = value;
return NO_ERROR;
}
status_t AudioFlinger::MixerThread::setMasterMute(bool muted)
{
mMasterMute = muted;
return NO_ERROR;
}
float AudioFlinger::MixerThread::masterVolume() const
{
return mMasterVolume;
}
bool AudioFlinger::MixerThread::masterMute() const
{
return mMasterMute;
}
status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value)
{
mStreamTypes[stream].volume = value;
return NO_ERROR;
}
status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted)
{
mStreamTypes[stream].mute = muted;
return NO_ERROR;
}
float AudioFlinger::MixerThread::streamVolume(int stream) const
{
return mStreamTypes[stream].volume;
}
bool AudioFlinger::MixerThread::streamMute(int stream) const
{
return mStreamTypes[stream].mute;
}
// isMusicActive_l() must be called with AudioFlinger::mLock held
bool AudioFlinger::MixerThread::isMusicActive_l() const
{
size_t count = mActiveTracks.size();
for (size_t i = 0 ; i < count ; ++i) {
sp<Track> t = mActiveTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
if (t->mStreamType == AudioSystem::MUSIC)
return true;
}
return false;
}
// addTrack_l() must be called with AudioFlinger::mLock held
status_t AudioFlinger::MixerThread::addTrack_l(const sp<Track>& track)
{
status_t status = ALREADY_EXISTS;
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (track->isPaused()) {
track->mState = TrackBase::RESUMING;
LOGV("PAUSED => RESUMING (%d)", track->name());
} else {
track->mState = TrackBase::ACTIVE;
LOGV("? => ACTIVE (%d)", track->name());
}
// 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;
addActiveTrack_l(track);
status = NO_ERROR;
}
LOGV("mWaitWorkCV.broadcast");
mAudioFlinger->mWaitWorkCV.broadcast();
return status;
}
// destroyTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::destroyTrack_l(const sp<Track>& track)
{
track->mState = TrackBase::TERMINATED;
if (mActiveTracks.indexOf(track) < 0) {
LOGV("remove track (%d) and delete from mixer", track->name());
mTracks.remove(track);
deleteTrackName_l(track->name());
}
}
// addActiveTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::addActiveTrack_l(const wp<Track>& t)
{
mActiveTracks.add(t);
// Force routing to speaker for certain stream types
// The forced routing to speaker is managed by hardware mixer
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
sp<Track> track = t.promote();
if (track == NULL) return;
if (streamForcedToSpeaker(track->type())) {
mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_ADDED);
}
}
}
// removeActiveTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::removeActiveTrack_l(const wp<Track>& t)
{
mActiveTracks.remove(t);
// Force routing to speaker for certain stream types
// The forced routing to speaker is managed by hardware mixer
if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
sp<Track> track = t.promote();
if (track == NULL) return;
if (streamForcedToSpeaker(track->type())) {
mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED);
}
}
}
// getTrackName_l() must be called with AudioFlinger::mLock held
int AudioFlinger::MixerThread::getTrackName_l()
{
return mAudioMixer->getTrackName();
}
// deleteTrackName_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::deleteTrackName_l(int name)
{
mAudioMixer->deleteTrackName(name);
}
size_t AudioFlinger::MixerThread::getOutputFrameCount()
{
return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t);
}
// ----------------------------------------------------------------------------
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread::TrackBase::TrackBase(
const sp<MixerThread>& mixerThread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer)
: RefBase(),
mMixerThread(mixerThread),
mClient(client),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK)
{
mName = mixerThread->getTrackName_l();
LOGV("TrackBase contructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
if (mName < 0) {
LOGE("no more track names availlable");
return;
}
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
if (sharedBuffer == 0) {
size += bufferSize;
}
if (client != NULL) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
if (mCblk) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mCblk->channels = (uint8_t)channelCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flowControlFlag = 1;
} else {
mBuffer = sharedBuffer->pointer();
}
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
} else {
LOGE("not enough memory for AudioTrack size=%u", size);
client->heap()->dump("AudioTrack");
return;
}
} else {
mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
if (mCblk) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
mCblk->sampleRate = sampleRate;
mCblk->channels = (uint8_t)channelCount;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flowControlFlag = 1;
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
}
}
AudioFlinger::MixerThread::TrackBase::~TrackBase()
{
if (mCblk) {
mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
}
mCblkMemory.clear(); // and free the shared memory
mClient.clear();
}
void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
}
bool AudioFlinger::MixerThread::TrackBase::step() {
bool result;
audio_track_cblk_t* cblk = this->cblk();
result = cblk->stepServer(mFrameCount);
if (!result) {
LOGV("stepServer failed acquiring cblk mutex");
mFlags |= STEPSERVER_FAILED;
}
return result;
}
void AudioFlinger::MixerThread::TrackBase::reset() {
audio_track_cblk_t* cblk = this->cblk();
cblk->user = 0;
cblk->server = 0;
cblk->userBase = 0;
cblk->serverBase = 0;
mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
LOGV("TrackBase::reset");
}
sp<IMemory> AudioFlinger::MixerThread::TrackBase::getCblk() const
{
return mCblkMemory;
}
int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
int AudioFlinger::MixerThread::TrackBase::channelCount() const {
return (int)mCblk->channels;
}
void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
int16_t *bufferEnd = bufferStart + frames * cblk->channels;
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
(cblk->channels == 2 && ((unsigned long)bufferStart & 3))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d, channels %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels);
return 0;
}
return bufferStart;
}
// ----------------------------------------------------------------------------
// Track constructor must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread::Track::Track(
const sp<MixerThread>& mixerThread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer)
: TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
{
mVolume[0] = 1.0f;
mVolume[1] = 1.0f;
mMute = false;
mSharedBuffer = sharedBuffer;
mStreamType = streamType;
}
AudioFlinger::MixerThread::Track::~Track()
{
wp<Track> weak(this); // never create a strong ref from the dtor
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
mState = TERMINATED;
}
void AudioFlinger::MixerThread::Track::destroy()
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
// desctructor is called. As the destructor needs to lock AudioFlinger::mLock,
// we must acquire a strong reference on this Track before locking AudioFlinger::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 AudioFlinger::mLock
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
mMixerThread->destroyTrack_l(this);
}
}
void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
mFormat,
mCblk->channels,
mFrameCount,
mState,
mMute,
mFillingUpStatus,
mCblk->sampleRate,
mCblk->volume[0],
mCblk->volume[1],
mCblk->server,
mCblk->user);
}
status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
LOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesReady = cblk->framesReady();
if (LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
if (framesReq > framesReady) {
framesReq = framesReady;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
bool AudioFlinger::MixerThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING) return true;
if (mCblk->framesReady() >= mCblk->frameCount ||
mCblk->forceReady) {
mFillingUpStatus = FS_FILLED;
mCblk->forceReady = 0;
LOGV("Track::isReady() track %d for output %d", mName, mMixerThread->mOutputType);
return true;
}
return false;
}
status_t AudioFlinger::MixerThread::Track::start()
{
LOGV("start(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
mMixerThread->addTrack_l(this);
return NO_ERROR;
}
void AudioFlinger::MixerThread::Track::stop()
{
LOGV("stop(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
if (mState > STOPPED) {
mState = STOPPED;
// If the track is not active (PAUSED and buffers full), flush buffers
if (mMixerThread->mActiveTracks.indexOf(this) < 0) {
reset();
}
LOGV("(> STOPPED) => STOPPED (%d)", mName);
}
}
void AudioFlinger::MixerThread::Track::pause()
{
LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
if (mState == ACTIVE || mState == RESUMING) {
mState = PAUSING;
LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
}
}
void AudioFlinger::MixerThread::Track::flush()
{
LOGV("flush(%d)", mName);
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
return;
}
// No point remaining in PAUSED state after a flush => go to
// STOPPED state
mState = STOPPED;
mCblk->lock.lock();
// NOTE: reset() will reset cblk->user and cblk->server with
// the risk that at the same time, the AudioMixer is trying to read
// data. In this case, getNextBuffer() would return a NULL pointer
// as audio buffer => the AudioMixer code MUST always test that pointer
// returned by getNextBuffer() is not NULL!
reset();
mCblk->lock.unlock();
}
void AudioFlinger::MixerThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
if (!mResetDone) {
TrackBase::reset();
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
mCblk->flowControlFlag = 1;
mCblk->forceReady = 0;
mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
}
void AudioFlinger::MixerThread::Track::mute(bool muted)
{
mMute = muted;
}
void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
{
mVolume[0] = left;
mVolume[1] = right;
}
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread::RecordTrack::RecordTrack(
const sp<MixerThread>& mixerThread,
const sp<Client>& client,
int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags)
: TrackBase(mixerThread, client, sampleRate, format,
channelCount, frameCount, flags, 0),
mOverflow(false), mInputSource(inputSource)
{
}
AudioFlinger::MixerThread::RecordTrack::~RecordTrack()
{
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
mMixerThread->deleteTrackName_l(mName);
}
status_t AudioFlinger::MixerThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
if (mFlags & TrackBase::STEPSERVER_FAILED) {
if (!step()) goto getNextBuffer_exit;
LOGV("stepServer recovered");
mFlags &= ~TrackBase::STEPSERVER_FAILED;
}
framesAvail = cblk->framesAvailable_l();
if (LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
if (s + framesReq > bufferEnd) {
framesReq = bufferEnd - s;
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
status_t AudioFlinger::MixerThread::RecordTrack::start()
{
return mMixerThread->mAudioFlinger->startRecord(this);
}
void AudioFlinger::MixerThread::RecordTrack::stop()
{
mMixerThread->mAudioFlinger->stopRecord(this);
TrackBase::reset();
// Force overerrun condition to avoid false overrun callback until first data is
// read from buffer
mCblk->flowControlFlag = 1;
}
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::OutputTrack::OutputTrack(
const sp<MixerThread>& mixerThread,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount)
: Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL),
mOutputMixerThread(mixerThread)
{
mCblk->out = 1;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
mCblk->bufferTimeoutMs = 10;
LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p",
mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);
}
AudioFlinger::MixerThread::OutputTrack::~OutputTrack()
{
stop();
}
status_t AudioFlinger::MixerThread::OutputTrack::start()
{
status_t status = Track::start();
mRetryCount = 127;
return status;
}
void AudioFlinger::MixerThread::OutputTrack::stop()
{
Track::stop();
clearBufferQueue();
mOutBuffer.frameCount = 0;
}
void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
uint32_t channels = mCblk->channels;
inBuffer.frameCount = frames;
inBuffer.i16 = data;
if (mCblk->user == 0) {
mOutputMixerThread->mAudioFlinger->mLock.lock();
bool isMusicActive = mOutputMixerThread->isMusicActive_l();
mOutputMixerThread->mAudioFlinger->mLock.unlock();
if (isMusicActive) {
mCblk->forceReady = 1;
LOGV("OutputTrack::start() force ready");
} else if (mCblk->frameCount > frames){
if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
uint32_t startFrames = (mCblk->frameCount - frames);
LOGV("OutputTrack::start() write %d frames", startFrames);
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[startFrames * channels];
pInBuffer->frameCount = startFrames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else {
LOGW ("OutputTrack::write() no more buffers");
}
}
}
while (1) {
// 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;
if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
break;
}
}
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
mCblk->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channels;
mOutBuffer.frameCount -= outFrames;
mOutBuffer.i16 += outFrames * channels;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
delete [] pInBuffer->mBuffer;
delete pInBuffer;
} else {
break;
}
}
}
// If we could not write all frames, allocate a buffer and queue it for next time.
if (inBuffer.frameCount) {
if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->i16 = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
} else {
LOGW("OutputTrack::write() no more buffers");
}
}
// 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 && mCblk->user < mCblk->frameCount) {
frames = mCblk->frameCount - mCblk->user;
pInBuffer = new Buffer;
pInBuffer->mBuffer = new int16_t[frames * channels];
pInBuffer->frameCount = frames;
pInBuffer->i16 = pInBuffer->mBuffer;
memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
mBufferQueue.add(pInBuffer);
}
}
status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer)
{
int active;
int timeout = 0;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = buffer->frameCount;
LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
buffer->frameCount = 0;
uint32_t framesAvail = cblk->framesAvailable();
if (framesAvail == 0) {
return AudioTrack::NO_MORE_BUFFERS;
}
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (u + framesReq > bufferEnd) {
framesReq = bufferEnd - u;
}
buffer->frameCount = framesReq;
buffer->raw = (void *)cblk->buffer(u);
return NO_ERROR;
}
void AudioFlinger::MixerThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
Buffer *pBuffer;
for (size_t i = 0; i < size; i++) {
pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
delete pBuffer;
}
mBufferQueue.clear();
}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mMemoryDealer(new MemoryDealer(1024*1024)),
mPid(pid)
{
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}
AudioFlinger::Client::~Client()
{
mAudioFlinger->removeClient(mPid);
}
const sp<MemoryDealer>& AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
// ----------------------------------------------------------------------------
AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::MixerThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
}
AudioFlinger::TrackHandle::~TrackHandle() {
// just stop the track on deletion, associated resources
// will be freed from the main thread once all pending buffers have
// been played. Unless it's not in the active track list, in which
// case we free everything now...
mTrack->destroy();
}
status_t AudioFlinger::TrackHandle::start() {
return mTrack->start();
}
void AudioFlinger::TrackHandle::stop() {
mTrack->stop();
}
void AudioFlinger::TrackHandle::flush() {
mTrack->flush();
}
void AudioFlinger::TrackHandle::mute(bool e) {
mTrack->mute(e);
}
void AudioFlinger::TrackHandle::pause() {
mTrack->pause();
}
void AudioFlinger::TrackHandle::setVolume(float left, float right) {
mTrack->setVolume(left, right);
}
sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
return mTrack->getCblk();
}
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioTrack::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags,
status_t *status)
{
sp<MixerThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
wp<Client> wclient;
AudioStreamIn* input = 0;
int inFrameCount;
size_t inputBufferSize;
status_t lStatus;
// check calling permissions
if (!recordingAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
if (uint32_t(inputSource) >= AudioRecord::NUM_INPUT_SOURCES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
if (mAudioRecordThread == 0) {
LOGE("Audio record thread not started");
lStatus = NO_INIT;
goto Exit;
}
// Check that audio input stream accepts requested audio parameters
inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
if (inputBufferSize == 0) {
lStatus = BAD_VALUE;
LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount);
goto Exit;
}
// add client to list
{ // scope for mLock
Mutex::Autolock _l(mLock);
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
} else {
client = new Client(this, pid);
mClients.add(pid, client);
}
// frameCount must be a multiple of input buffer size
inFrameCount = inputBufferSize/channelCount/sizeof(short);
frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, inputSource, sampleRate,
format, channelCount, frameCount, flags);
}
if (recordTrack->getCblk() == NULL) {
recordTrack.clear();
lStatus = NO_MEMORY;
goto Exit;
}
// return to handle to client
recordHandle = new RecordHandle(recordTrack);
lStatus = NO_ERROR;
Exit:
if (status) {
*status = lStatus;
}
return recordHandle;
}
status_t AudioFlinger::startRecord(MixerThread::RecordTrack* recordTrack) {
if (mAudioRecordThread != 0) {
return mAudioRecordThread->start(recordTrack);
}
return NO_INIT;
}
void AudioFlinger::stopRecord(MixerThread::RecordTrack* recordTrack) {
if (mAudioRecordThread != 0) {
mAudioRecordThread->stop(recordTrack);
}
}
// ----------------------------------------------------------------------------
AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::MixerThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
}
AudioFlinger::RecordHandle::~RecordHandle() {
stop();
}
status_t AudioFlinger::RecordHandle::start() {
LOGV("RecordHandle::start()");
return mRecordTrack->start();
}
void AudioFlinger::RecordHandle::stop() {
LOGV("RecordHandle::stop()");
mRecordTrack->stop();
}
sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
return mRecordTrack->getCblk();
}
status_t AudioFlinger::RecordHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioRecord::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware,
const sp<AudioFlinger>& audioFlinger) :
mAudioHardware(audioHardware),
mAudioFlinger(audioFlinger),
mActive(false)
{
}
AudioFlinger::AudioRecordThread::~AudioRecordThread()
{
}
bool AudioFlinger::AudioRecordThread::threadLoop()
{
LOGV("AudioRecordThread: start record loop");
AudioBufferProvider::Buffer buffer;
int inBufferSize = 0;
int inFrameCount = 0;
AudioStreamIn* input = 0;
mActive = 0;
// start recording
while (!exitPending()) {
if (!mActive) {
mLock.lock();
if (!mActive && !exitPending()) {
LOGV("AudioRecordThread: loop stopping");
if (input) {
delete input;
input = 0;
}
mRecordTrack.clear();
mStopped.signal();
mWaitWorkCV.wait(mLock);
LOGV("AudioRecordThread: loop starting");
if (mRecordTrack != 0) {
input = mAudioHardware->openInputStream(
mRecordTrack->inputSource(),
mRecordTrack->format(),
mRecordTrack->channelCount(),
mRecordTrack->sampleRate(),
&mStartStatus,
(AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16));
if (input != 0) {
inBufferSize = input->bufferSize();
inFrameCount = inBufferSize/input->frameSize();
}
} else {
mStartStatus = NO_INIT;
}
if (mStartStatus !=NO_ERROR) {
LOGW("record start failed, status %d", mStartStatus);
mActive = false;
mRecordTrack.clear();
}
mWaitWorkCV.signal();
}
mLock.unlock();
} else if (mRecordTrack != 0) {
buffer.frameCount = inFrameCount;
if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR &&
(int)buffer.frameCount == inFrameCount)) {
LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
ssize_t bytesRead = input->read(buffer.raw, inBufferSize);
if (bytesRead < 0) {
LOGE("Error reading audio input");
sleep(1);
}
mRecordTrack->releaseBuffer(&buffer);
mRecordTrack->overflow();
}
// client isn't retrieving buffers fast enough
else {
if (!mRecordTrack->setOverflow())
LOGW("AudioRecordThread: buffer overflow");
// Release the processor for a while before asking for a new buffer.
// This will give the application more chance to read from the buffer and
// clear the overflow.
usleep(5000);
}
}
}
if (input) {
delete input;
}
mRecordTrack.clear();
return false;
}
status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack)
{
LOGV("AudioRecordThread::start");
AutoMutex lock(&mLock);
mActive = true;
// If starting the active track, just reset mActive in case a stop
// was pending and exit
if (recordTrack == mRecordTrack.get()) return NO_ERROR;
if (mRecordTrack != 0) return -EBUSY;
mRecordTrack = recordTrack;
// signal thread to start
LOGV("Signal record thread");
mWaitWorkCV.signal();
mWaitWorkCV.wait(mLock);
LOGV("Record started, status %d", mStartStatus);
return mStartStatus;
}
void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) {
LOGV("AudioRecordThread::stop");
AutoMutex lock(&mLock);
if (mActive && (recordTrack == mRecordTrack.get())) {
mActive = false;
mStopped.wait(mLock);
}
}
void AudioFlinger::AudioRecordThread::exit()
{
LOGV("AudioRecordThread::exit");
{
AutoMutex lock(&mLock);
requestExit();
mWaitWorkCV.signal();
}
requestExitAndWait();
}
status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
pid_t pid = 0;
if (mRecordTrack != 0 && mRecordTrack->mClient != 0) {
snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid());
result.append(buffer);
} else {
result.append("No record client\n");
}
write(fd, result.string(), result.size());
return NO_ERROR;
}
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioFlinger::onTransact(code, data, reply, flags);
}
// ----------------------------------------------------------------------------
void AudioFlinger::instantiate() {
defaultServiceManager()->addService(
String16("media.audio_flinger"), new AudioFlinger());
}
}; // namespace android
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