eclair snapshot
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index da7cc8a..ebd470f 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -24,10 +24,10 @@
#include <sys/time.h>
#include <sys/resource.h>
-#include <utils/IServiceManager.h>
+#include <binder/IServiceManager.h>
#include <utils/Log.h>
-#include <utils/Parcel.h>
-#include <utils/IPCThreadState.h>
+#include <binder/Parcel.h>
+#include <binder/IPCThreadState.h>
#include <utils/String16.h>
#include <utils/threads.h>
@@ -62,8 +62,6 @@
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
@@ -71,14 +69,10 @@
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;
+static const nsecs_t kWarningThrottle = seconds(5);
#define AUDIOFLINGER_SECURITY_ENABLED 1
@@ -121,132 +115,41 @@
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mAudioHardware(0), mA2dpAudioInterface(0), mA2dpEnabled(false), mNotifyA2dpChange(false),
- mForcedSpeakerCount(0), mA2dpDisableCount(0), mA2dpSuppressed(false), mForcedRoute(0),
- mRouteRestoreTime(0), mMusicMuteSaved(false)
+ mAudioHardware(0), mMasterVolume(1.0f), mMasterMute(false), mNextThreadId(0)
{
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 {
+ } else {
LOGE("Couldn't even initialize the stubbed audio hardware!");
}
}
AudioFlinger::~AudioFlinger()
{
- if (mAudioRecordThread != 0) {
- mAudioRecordThread->exit();
- mAudioRecordThread.clear();
+ while (!mRecordThreads.isEmpty()) {
+ // closeInput() will remove first entry from mRecordThreads
+ closeInput(mRecordThreads.keyAt(0));
}
- 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;
-
- LOGD_IF(enable, "set output to A2DP\n");
- LOGD_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();
+ while (!mPlaybackThreads.isEmpty()) {
+ // closeOutput() will remove first entry from mPlaybackThreads
+ closeOutput(mPlaybackThreads.keyAt(0));
}
- 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;
+ if (mAudioHardware) {
+ delete mAudioHardware;
}
}
-#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)
{
@@ -276,10 +179,7 @@
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());
@@ -337,13 +237,16 @@
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);
+ // dump playback threads
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ mPlaybackThreads.valueAt(i)->dump(fd, args);
+ }
+
+ // dump record threads
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ mRecordThreads.valueAt(i)->dump(fd, args);
+ }
if (mAudioHardware) {
mAudioHardware->dumpState(fd, args);
@@ -353,6 +256,7 @@
return NO_ERROR;
}
+
// IAudioFlinger interface
@@ -365,9 +269,10 @@
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
+ int output,
status_t *status)
{
- sp<MixerThread::Track> track;
+ sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
wp<Client> wclient;
@@ -381,6 +286,12 @@
{
Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGE("unknown output thread");
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
wclient = mClients.valueFor(pid);
@@ -390,20 +301,15 @@
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);
- }
+ track = thread->createTrack_l(client, streamType, sampleRate, format,
+ channelCount, frameCount, sharedBuffer, &lStatus);
}
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();
}
@@ -416,52 +322,57 @@
uint32_t AudioFlinger::sampleRate(int output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->sampleRate();
- }
-#endif
- return mHardwareMixerThread->sampleRate();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("sampleRate() unknown thread %d", output);
+ return 0;
+ }
+ return thread->sampleRate();
}
int AudioFlinger::channelCount(int output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->channelCount();
- }
-#endif
- return mHardwareMixerThread->channelCount();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("channelCount() unknown thread %d", output);
+ return 0;
+ }
+ return thread->channelCount();
}
int AudioFlinger::format(int output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->format();
- }
-#endif
- return mHardwareMixerThread->format();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("format() unknown thread %d", output);
+ return 0;
+ }
+ return thread->format();
}
size_t AudioFlinger::frameCount(int output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->frameCount();
- }
-#endif
- return mHardwareMixerThread->frameCount();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("frameCount() unknown thread %d", output);
+ return 0;
+ }
+ return thread->frameCount();
}
uint32_t AudioFlinger::latency(int output) const
{
-#ifdef WITH_A2DP
- if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
- return mA2dpMixerThread->latency();
- }
-#endif
- return mHardwareMixerThread->latency();
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ LOGW("latency() unknown thread %d", output);
+ return 0;
+ }
+ return thread->latency();
}
status_t AudioFlinger::setMasterVolume(float value)
@@ -478,96 +389,14 @@
value = 1.0f;
}
mHardwareStatus = AUDIO_HW_IDLE;
- mHardwareMixerThread->setMasterVolume(value);
-#ifdef WITH_A2DP
- mA2dpMixerThread->setMasterVolume(value);
-#endif
+
+ mMasterVolume = value;
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads.valueAt(i)->setMasterVolume(value);
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
@@ -586,15 +415,6 @@
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
@@ -624,60 +444,55 @@
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- mHardwareMixerThread->setMasterMute(muted);
-#ifdef WITH_A2DP
- mA2dpMixerThread->setMasterMute(muted);
-#endif
+
+ mMasterMute = muted;
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads.valueAt(i)->setMasterMute(muted);
+
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
- return mHardwareMixerThread->masterVolume();
+ return mMasterVolume;
}
bool AudioFlinger::masterMute() const
{
- return mHardwareMixerThread->masterMute();
+ return mMasterMute;
}
-status_t AudioFlinger::setStreamVolume(int stream, float value)
+status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
- uint32_t(stream) == AudioSystem::ENFORCED_AUDIBLE) {
+ if (stream < 0 || uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
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(mLock);
+ PlaybackThread *thread = NULL;
+ if (output) {
+ thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ return BAD_VALUE;
}
-
- 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
+ mStreamTypes[stream].volume = value;
- return ret;
+ if (thread == NULL) {
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+ mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
+ }
+ } else {
+ thread->setStreamVolume(stream, value);
+ }
+
+ return NO_ERROR;
}
status_t AudioFlinger::setStreamMute(int stream, bool muted)
@@ -687,82 +502,115 @@
return PERMISSION_DENIED;
}
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES ||
+ if (stream < 0 || 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);
- }
+ mStreamTypes[stream].mute = muted;
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
return NO_ERROR;
}
-float AudioFlinger::streamVolume(int stream) const
+float AudioFlinger::streamVolume(int stream, int output) const
{
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ if (stream < 0 || 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 ;
+
+ AutoMutex lock(mLock);
+ float volume;
+ if (output) {
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ return 0.0f;
+ }
+ volume = thread->streamVolume(stream);
+ } else {
+ volume = mStreamTypes[stream].volume;
}
-
+
return volume;
}
bool AudioFlinger::streamMute(int stream) const
{
- if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ if (stream < 0 || stream >= (int)AudioSystem::NUM_STREAM_TYPES) {
return true;
}
-
- if (stream == AudioSystem::MUSIC && mForcedRoute != 0)
- {
- return mMusicMuteSaved;
- }
- return mHardwareMixerThread->streamMute(stream);
+
+ return mStreamTypes[stream].mute;
}
bool AudioFlinger::isMusicActive() const
{
Mutex::Autolock _l(mLock);
- #ifdef WITH_A2DP
- if (isA2dpEnabled()) {
- return mA2dpMixerThread->isMusicActive_l();
- }
- #endif
- return mHardwareMixerThread->isMusicActive_l();
+ for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads.valueAt(i)->isMusicActive()) {
+ return true;
+ }
+ }
+ return false;
}
-status_t AudioFlinger::setParameter(const char* key, const char* value)
+status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
{
- 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;
+ status_t result;
+
+ LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
+ ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
}
- mHardwareStatus = AUDIO_HW_IDLE;
- return result;
+
+ // ioHandle == 0 means the parameters are global to the audio hardware interface
+ if (ioHandle == 0) {
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_PARAMETER;
+ result = mAudioHardware->setParameters(keyValuePairs);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return result;
+ }
+
+ // hold a strong ref on thread in case closeOutput() or closeInput() is called
+ // and the thread is exited once the lock is released
+ sp<ThreadBase> thread;
+ {
+ Mutex::Autolock _l(mLock);
+ thread = checkPlaybackThread_l(ioHandle);
+ if (thread == NULL) {
+ thread = checkRecordThread_l(ioHandle);
+ }
+ }
+ if (thread != NULL) {
+ return thread->setParameters(keyValuePairs);
+ }
+ return BAD_VALUE;
+}
+
+String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
+{
+// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
+// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
+
+ if (ioHandle == 0) {
+ return mAudioHardware->getParameters(keys);
+ }
+
+ Mutex::Autolock _l(mLock);
+
+ PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
+ if (playbackThread != NULL) {
+ return playbackThread->getParameters(keys);
+ }
+ RecordThread *recordThread = checkRecordThread_l(ioHandle);
+ if (recordThread != NULL) {
+ return recordThread->getParameters(keys);
+ }
+ return String8("");
}
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
@@ -770,9 +618,24 @@
return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
}
+status_t AudioFlinger::setVoiceVolume(float value)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+ status_t ret = mAudioHardware->setVoiceVolume(value);
+ mHardwareStatus = AUDIO_HW_IDLE;
+
+ return ret;
+}
+
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);
@@ -781,12 +644,21 @@
LOGV("Adding notification client %p", binder.get());
binder->linkToDeath(this);
mNotificationClients.add(binder);
- client->a2dpEnabledChanged(isA2dpEnabled());
+ }
+
+ // the config change is always sent from playback or record threads to avoid deadlock
+ // with AudioSystem::gLock
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED);
+ }
+
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED);
}
}
void AudioFlinger::binderDied(const wp<IBinder>& who) {
-
+
LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
@@ -801,156 +673,242 @@
}
}
-void AudioFlinger::removeClient(pid_t pid)
+// audioConfigChanged_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::audioConfigChanged_l(int event, const sp<ThreadBase>& thread, void *param2) {
+ int ioHandle = 0;
+
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads.valueAt(i) == thread) {
+ ioHandle = mPlaybackThreads.keyAt(i);
+ break;
+ }
+ }
+ if (ioHandle == 0) {
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ if (mRecordThreads.valueAt(i) == thread) {
+ ioHandle = mRecordThreads.keyAt(i);
+ break;
+ }
+ }
+ }
+
+ if (ioHandle != 0) {
+ size_t size = mNotificationClients.size();
+ for (size_t i = 0; i < size; i++) {
+ sp<IBinder> binder = mNotificationClients.itemAt(i);
+ LOGV("audioConfigChanged_l() Notifying change to client %p", binder.get());
+ sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
+ client->ioConfigChanged(event, ioHandle, param2);
+ }
+ }
+}
+
+// removeClient_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::removeClient_l(pid_t pid)
{
- LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
- Mutex::Autolock _l(mLock);
+ LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
-bool AudioFlinger::isA2dpEnabled() const
+// ----------------------------------------------------------------------------
+
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger)
+ : Thread(false),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
+ mFormat(0), mFrameSize(1), mStandby(false)
{
- return mA2dpEnabled;
}
-void AudioFlinger::handleForcedSpeakerRoute(int command)
+AudioFlinger::ThreadBase::~ThreadBase()
{
- 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;
- }
+ mParamCond.broadcast();
+ mNewParameters.clear();
}
-#ifdef WITH_A2DP
-// handleRouteDisablesA2dp_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
+void AudioFlinger::ThreadBase::exit()
{
- 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");
- }
+ // keep a strong ref on ourself so that we wont get
+ // destroyed in the middle of requestExitAndWait()
+ sp <ThreadBase> strongMe = this;
+
+ LOGV("ThreadBase::exit");
+ {
+ AutoMutex lock(&mLock);
+ requestExit();
+ mWaitWorkCV.signal();
}
+ requestExitAndWait();
}
-#endif
+
+uint32_t AudioFlinger::ThreadBase::sampleRate() const
+{
+ return mSampleRate;
+}
+
+int AudioFlinger::ThreadBase::channelCount() const
+{
+ return mChannelCount;
+}
+
+int AudioFlinger::ThreadBase::format() const
+{
+ return mFormat;
+}
+
+size_t AudioFlinger::ThreadBase::frameCount() const
+{
+ return mFrameCount;
+}
+
+status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+{
+ status_t status;
+
+ LOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
+ Mutex::Autolock _l(mLock);
+
+ mNewParameters.add(keyValuePairs);
+ mWaitWorkCV.signal();
+ // wait condition with timeout in case the thread loop has exited
+ // before the request could be processed
+ if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) {
+ status = mParamStatus;
+ mWaitWorkCV.signal();
+ } else {
+ status = TIMED_OUT;
+ }
+ return status;
+}
+
+void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
+{
+ Mutex::Autolock _l(mLock);
+ sendConfigEvent_l(event, param);
+}
+
+// sendConfigEvent_l() must be called with ThreadBase::mLock held
+void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
+{
+ ConfigEvent *configEvent = new ConfigEvent();
+ configEvent->mEvent = event;
+ configEvent->mParam = param;
+ mConfigEvents.add(configEvent);
+ LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
+ mWaitWorkCV.signal();
+}
+
+void AudioFlinger::ThreadBase::processConfigEvents()
+{
+ mLock.lock();
+ while(!mConfigEvents.isEmpty()) {
+ LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
+ ConfigEvent *configEvent = mConfigEvents[0];
+ mConfigEvents.removeAt(0);
+ // release mLock because audioConfigChanged() will lock AudioFlinger mLock
+ // before calling Audioflinger::audioConfigChanged_l() thus creating
+ // potential cross deadlock between AudioFlinger::mLock and mLock
+ mLock.unlock();
+ audioConfigChanged(configEvent->mEvent, configEvent->mParam);
+ delete configEvent;
+ mLock.lock();
+ }
+ mLock.unlock();
+}
+
+status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ bool locked = tryLock(mLock);
+ if (!locked) {
+ snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this);
+ write(fd, buffer, strlen(buffer));
+ }
+
+ snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Format: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
+ result.append(buffer);
+
+ snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
+ result.append(buffer);
+ result.append(" Index Command");
+ for (size_t i = 0; i < mNewParameters.size(); ++i) {
+ snprintf(buffer, SIZE, "\n %02d ", i);
+ result.append(buffer);
+ result.append(mNewParameters[i]);
+ }
+
+ snprintf(buffer, SIZE, "\n\nPending config events: \n");
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Index event param\n");
+ result.append(buffer);
+ for (size_t i = 0; i < mConfigEvents.size(); i++) {
+ snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
+ result.append(buffer);
+ }
+ result.append("\n");
+
+ write(fd, result.string(), result.size());
+
+ if (locked) {
+ mLock.unlock();
+ }
+ return NO_ERROR;
+}
+
// ----------------------------------------------------------------------------
-AudioFlinger::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)
+AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+ : ThreadBase(audioFlinger),
+ mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
+ mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
- mSampleRate = output->sampleRate();
- mChannelCount = output->channelCount();
+ readOutputParameters();
- // FIXME - Current mixer implementation only supports stereo output
- if (mChannelCount == 1) {
- LOGE("Invalid audio hardware channel count");
+ mMasterVolume = mAudioFlinger->masterVolume();
+ mMasterMute = mAudioFlinger->masterMute();
+
+ for (int stream = 0; stream < AudioSystem::NUM_STREAM_TYPES; stream++) {
+ mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
+ mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
}
-
- 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));
+ // notify client processes that a new input has been opened
+ sendConfigEvent(AudioSystem::OUTPUT_OPENED);
}
-AudioFlinger::MixerThread::~MixerThread()
+AudioFlinger::PlaybackThread::~PlaybackThread()
{
delete [] mMixBuffer;
- delete mAudioMixer;
}
-status_t AudioFlinger::MixerThread::dump(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::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)
+status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType);
+ snprintf(buffer, SIZE, "Output thread %p tracks\n", this);
result.append(buffer);
- result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+ 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) {
@@ -959,9 +917,9 @@
}
}
- snprintf(buffer, SIZE, "Output %d mixer thread active tracks\n", mOutputType);
+ snprintf(buffer, SIZE, "Output thread %p active tracks\n", this);
result.append(buffer);
- result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+ 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) {
@@ -976,15 +934,13 @@
return NO_ERROR;
}
-status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+status_t AudioFlinger::PlaybackThread::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());
+ snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
@@ -994,282 +950,36 @@
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());
+
+ dumpBase(fd, args);
+
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()
+status_t AudioFlinger::PlaybackThread::readyToRun()
{
if (mSampleRate == 0) {
LOGE("No working audio driver found.");
return NO_INIT;
}
- LOGI("AudioFlinger's thread ready to run for output %d", mOutputType);
+ LOGI("AudioFlinger's thread %p ready to run", this);
return NO_ERROR;
}
-void AudioFlinger::MixerThread::onFirstRef()
+void AudioFlinger::PlaybackThread::onFirstRef()
{
const size_t SIZE = 256;
char buffer[SIZE];
- snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType);
+ snprintf(buffer, SIZE, "Playback Thread %p", this);
run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
}
-// MixerThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack_l(
+// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
int streamType,
uint32_t sampleRate,
@@ -1281,28 +991,39 @@
{
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 (mType == DIRECT) {
+ if (sampleRate != mSampleRate || format != mFormat || channelCount != mChannelCount) {
+ LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
+ sampleRate, format, channelCount, mOutput);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+ } else {
+ // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+ if (sampleRate > mSampleRate*2) {
+ LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
}
-
- if (mSampleRate == 0) {
+ if (mOutput == 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;
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelCount, frameCount, sharedBuffer);
+ if (track->getCblk() == NULL || track->name() < 0) {
+ lStatus = NO_MEMORY;
+ goto Exit;
+ }
+ mTracks.add(track);
}
- mTracks.add(track);
lStatus = NO_ERROR;
Exit:
@@ -1312,87 +1033,7 @@
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
+uint32_t AudioFlinger::PlaybackThread::latency() const
{
if (mOutput) {
return mOutput->latency();
@@ -1402,66 +1043,66 @@
}
}
-status_t AudioFlinger::MixerThread::setMasterVolume(float value)
+status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
mMasterVolume = value;
return NO_ERROR;
}
-status_t AudioFlinger::MixerThread::setMasterMute(bool muted)
+status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
mMasterMute = muted;
return NO_ERROR;
}
-float AudioFlinger::MixerThread::masterVolume() const
+float AudioFlinger::PlaybackThread::masterVolume() const
{
return mMasterVolume;
}
-bool AudioFlinger::MixerThread::masterMute() const
+bool AudioFlinger::PlaybackThread::masterMute() const
{
return mMasterMute;
}
-status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value)
+status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
{
mStreamTypes[stream].volume = value;
return NO_ERROR;
}
-status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted)
+status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
{
mStreamTypes[stream].mute = muted;
return NO_ERROR;
}
-float AudioFlinger::MixerThread::streamVolume(int stream) const
+float AudioFlinger::PlaybackThread::streamVolume(int stream) const
{
return mStreamTypes[stream].volume;
}
-bool AudioFlinger::MixerThread::streamMute(int stream) const
+bool AudioFlinger::PlaybackThread::streamMute(int stream) const
{
return mStreamTypes[stream].mute;
}
-// isMusicActive_l() must be called with AudioFlinger::mLock held
-bool AudioFlinger::MixerThread::isMusicActive_l() const
+bool AudioFlinger::PlaybackThread::isMusicActive() const
{
+ Mutex::Autolock _l(mLock);
size_t count = mActiveTracks.size();
for (size_t i = 0 ; i < count ; ++i) {
sp<Track> t = mActiveTracks[i].promote();
if (t == 0) continue;
Track* const track = t.get();
- if (t->mStreamType == AudioSystem::MUSIC)
+ if (t->type() == 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)
+// addTrack_l() must be called with ThreadBase::mLock held
+status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
status_t status = ALREADY_EXISTS;
@@ -1469,10 +1110,10 @@
// in both cases "unstop" the track
if (track->isPaused()) {
track->mState = TrackBase::RESUMING;
- LOGV("PAUSED => RESUMING (%d)", track->name());
+ LOGV("PAUSED => RESUMING (%d) on thread %p", track->name(), this);
} else {
track->mState = TrackBase::ACTIVE;
- LOGV("? => ACTIVE (%d)", track->name());
+ LOGV("? => ACTIVE (%d) on thread %p", track->name(), this);
}
// set retry count for buffer fill
track->mRetryCount = kMaxTrackStartupRetries;
@@ -1482,18 +1123,18 @@
// effectively get the latency it requested.
track->mFillingUpStatus = Track::FS_FILLING;
track->mResetDone = false;
- addActiveTrack_l(track);
+ mActiveTracks.add(track);
status = NO_ERROR;
}
-
+
LOGV("mWaitWorkCV.broadcast");
- mAudioFlinger->mWaitWorkCV.broadcast();
+ mWaitWorkCV.broadcast();
return status;
}
-// destroyTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::destroyTrack_l(const sp<Track>& track)
+// destroyTrack_l() must be called with ThreadBase::mLock held
+void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track)
{
track->mState = TrackBase::TERMINATED;
if (mActiveTracks.indexOf(track) < 0) {
@@ -1503,62 +1144,956 @@
}
}
-// addActiveTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::addActiveTrack_l(const wp<Track>& t)
+String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
{
- mActiveTracks.add(t);
+ return mOutput->getParameters(keys);
+}
- // 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);
- }
+void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
+ AudioSystem::OutputDescriptor desc;
+ void *param2 = 0;
+
+ LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param);
+
+ switch (event) {
+ case AudioSystem::OUTPUT_OPENED:
+ case AudioSystem::OUTPUT_CONFIG_CHANGED:
+ desc.channels = mChannelCount;
+ desc.samplingRate = mSampleRate;
+ desc.format = mFormat;
+ desc.frameCount = mFrameCount;
+ desc.latency = latency();
+ param2 = &desc;
+ break;
+
+ case AudioSystem::STREAM_CONFIG_CHANGED:
+ param2 = ¶m;
+ case AudioSystem::OUTPUT_CLOSED:
+ default:
+ break;
+ }
+ Mutex::Autolock _l(mAudioFlinger->mLock);
+ mAudioFlinger->audioConfigChanged_l(event, this, param2);
+}
+
+void AudioFlinger::PlaybackThread::readOutputParameters()
+{
+ mSampleRate = mOutput->sampleRate();
+ mChannelCount = AudioSystem::popCount(mOutput->channels());
+
+ mFormat = mOutput->format();
+ mFrameSize = mOutput->frameSize();
+ mFrameCount = mOutput->bufferSize() / mFrameSize;
+
+ mMinBytesToWrite = (mOutput->latency() * mSampleRate * mFrameSize) / 1000;
+ // FIXME - Current mixer implementation only supports stereo output: Always
+ // Allocate a stereo buffer even if HW output is mono.
+ if (mMixBuffer != NULL) delete mMixBuffer;
+ mMixBuffer = new int16_t[mFrameCount * 2];
+ memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+ : PlaybackThread(audioFlinger, output),
+ mAudioMixer(0)
+{
+ mType = PlaybackThread::MIXER;
+ mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+
+ // FIXME - Current mixer implementation only supports stereo output
+ if (mChannelCount == 1) {
+ LOGE("Invalid audio hardware channel count");
}
}
-// removeActiveTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::removeActiveTrack_l(const wp<Track>& t)
+AudioFlinger::MixerThread::~MixerThread()
{
- mActiveTracks.remove(t);
+ delete mAudioMixer;
+}
- // 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;
+bool AudioFlinger::MixerThread::threadLoop()
+{
+ uint32_t sleepTime = 1000;
+ uint32_t maxBufferRecoveryInUsecs = getMaxBufferRecoveryInUsecs();
+ int16_t* curBuf = mMixBuffer;
+ Vector< sp<Track> > tracksToRemove;
+ size_t enabledTracks = 0;
+ nsecs_t standbyTime = systemTime();
+ size_t mixBufferSize = mFrameCount * mFrameSize;
+ // FIXME: Relaxed timing because of a certain device that can't meet latency
+ // Should be reduced to 2x after the vendor fixes the driver issue
+ nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
+ nsecs_t lastWarning = 0;
- if (streamForcedToSpeaker(track->type())) {
- mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED);
+ while (!exitPending())
+ {
+ processConfigEvents();
+
+ enabledTracks = 0;
+ { // scope for mLock
+
+ Mutex::Autolock _l(mLock);
+
+ if (checkForNewParameters_l()) {
+ mixBufferSize = mFrameCount * mFrameSize;
+ // FIXME: Relaxed timing because of a certain device that can't meet latency
+ // Should be reduced to 2x after the vendor fixes the driver issue
+ maxPeriod = seconds(mFrameCount) / mSampleRate * 3;
+ maxBufferRecoveryInUsecs = getMaxBufferRecoveryInUsecs();
+ }
+
+ const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+ // put audio hardware into standby after short delay
+ if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
+ if (!mStandby) {
+ LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ }
+
+ if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+
+ if (exitPending()) break;
+
+ // wait until we have something to do...
+ LOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("MixerThread %p TID %d waking up\n", this, gettid());
+
+ if (mMasterMute == false) {
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.audio.silent", value, "0");
+ if (atoi(value)) {
+ LOGD("Silence is golden");
+ setMasterMute(true);
+ }
+ }
+
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ sleepTime = 1000;
+ continue;
+ }
+ }
+
+ enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
+ }
+
+ if (LIKELY(enabledTracks)) {
+ // mix buffers...
+ mAudioMixer->process(curBuf);
+ sleepTime = 0;
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ } else {
+ // If no tracks are ready, sleep once for the duration of an output
+ // buffer size, then write 0s to the output
+ if (sleepTime == 0) {
+ sleepTime = maxBufferRecoveryInUsecs;
+ } else if (mBytesWritten != 0) {
+ memset (curBuf, 0, mixBufferSize);
+ sleepTime = 0;
+ }
+ }
+
+ if (mSuspended) {
+ sleepTime = maxBufferRecoveryInUsecs;
+ }
+ // sleepTime == 0 means we must write to audio hardware
+ if (sleepTime == 0) {
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ int bytesWritten = (int)mOutput->write(curBuf, mixBufferSize);
+ if (bytesWritten > 0) mBytesWritten += bytesWritten;
+ mNumWrites++;
+ mInWrite = false;
+ mStandby = false;
+ nsecs_t now = systemTime();
+ nsecs_t delta = now - mLastWriteTime;
+ if (delta > maxPeriod) {
+ mNumDelayedWrites++;
+ if ((now - lastWarning) > kWarningThrottle) {
+ LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
+ ns2ms(delta), mNumDelayedWrites, this);
+ lastWarning = now;
+ }
+ }
+ } else {
+ usleep(sleepTime);
+ }
+
+ // finally let go of all our tracks, without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock.
+ tracksToRemove.clear();
+ }
+
+ if (!mStandby) {
+ mOutput->standby();
+ }
+
+ LOGV("MixerThread %p exiting", this);
+ return false;
+}
+
+// prepareTracks_l() must be called with ThreadBase::mLock held
+size_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
+{
+
+ size_t enabledTracks = 0;
+ // 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() ||
+ mStreamTypes[track->type()].mute) {
+ left = right = 0;
+ if (track->isPausing()) {
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->type()].volume;
+ float v = mMasterVolume * typeVolume;
+ float v_clamped = v * cblk->volume[0];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ left = 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 = AudioMixer::VOLUME;
+ if (track->mFillingUpStatus == Track::FS_FILLED) {
+ // no ramp for the first volume setting
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ param = AudioMixer::RAMP_VOLUME;
+ }
+ } else if (cblk->server != 0) {
+ // If the track is stopped before the first frame was mixed,
+ // do not apply ramp
+ param = AudioMixer::RAMP_VOLUME;
+ }
+
+ 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.
+ tracksToRemove->add(track);
+ mAudioMixer->disable(AudioMixer::MIXING);
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
+ tracksToRemove->add(track);
+ }
+ // For tracks using static shared memory buffer, make sure that we have
+ // written enough data to audio hardware before disabling the track
+ // NOTE: this condition with arrive before track->mRetryCount <= 0 so we
+ // don't care about code removing track from active list above.
+ if ((track->mSharedBuffer == 0) || (mBytesWritten >= mMinBytesToWrite)) {
+ mAudioMixer->disable(AudioMixer::MIXING);
+ } else {
+ enabledTracks++;
+ }
+ }
+ }
+ }
+
+ // remove all the tracks that need to be...
+ count = tracksToRemove->size();
+ if (UNLIKELY(count)) {
+ for (size_t i=0 ; i<count ; i++) {
+ const sp<Track>& track = tracksToRemove->itemAt(i);
+ mActiveTracks.remove(track);
+ if (track->isTerminated()) {
+ mTracks.remove(track);
+ deleteTrackName_l(track->mName);
+ }
+ }
+ }
+
+ return enabledTracks;
+}
+
+void AudioFlinger::MixerThread::getTracks(
+ SortedVector < sp<Track> >& tracks,
+ SortedVector < wp<Track> >& activeTracks,
+ int streamType)
+{
+ LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this, mTracks.size(), mActiveTracks.size());
+ Mutex::Autolock _l(mLock);
+ size_t size = mTracks.size();
+ for (size_t i = 0; i < size; i++) {
+ sp<Track> t = mTracks[i];
+ if (t->type() == streamType) {
+ 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++) {
+ mActiveTracks.remove(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());
+ }
+}
+
+void AudioFlinger::MixerThread::putTracks(
+ SortedVector < sp<Track> >& tracks,
+ SortedVector < wp<Track> >& activeTracks)
+{
+ LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this, tracks.size(), activeTracks.size());
+ Mutex::Autolock _l(mLock);
+ 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->mThread = this;
+ mTracks.add(t);
+
+ int j = activeTracks.indexOf(t);
+ if (j >= 0) {
+ mActiveTracks.add(t);
+ // force buffer refilling and no ramp volume when the track is mixed for the first time
+ t->mFillingUpStatus = Track::FS_FILLING;
}
}
}
-// getTrackName_l() must be called with AudioFlinger::mLock held
+// getTrackName_l() must be called with ThreadBase::mLock held
int AudioFlinger::MixerThread::getTrackName_l()
{
return mAudioMixer->getTrackName();
}
-// deleteTrackName_l() must be called with AudioFlinger::mLock held
+// deleteTrackName_l() must be called with ThreadBase::mLock held
void AudioFlinger::MixerThread::deleteTrackName_l(int name)
{
mAudioMixer->deleteTrackName(name);
}
-size_t AudioFlinger::MixerThread::getOutputFrameCount()
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::MixerThread::checkForNewParameters_l()
{
- return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t);
+ bool reconfig = false;
+
+ while (!mNewParameters.isEmpty()) {
+ status_t status = NO_ERROR;
+ String8 keyValuePair = mNewParameters[0];
+ AudioParameter param = AudioParameter(keyValuePair);
+ int value;
+
+ if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+ if (value != AudioSystem::PCM_16_BIT) {
+ status = BAD_VALUE;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+ if (value != AudioSystem::CHANNEL_OUT_STEREO) {
+ status = BAD_VALUE;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be garantied
+ // if frame count is changed after track creation
+ if (!mTracks.isEmpty()) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (status == NO_ERROR) {
+ status = mOutput->setParameters(keyValuePair);
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->setParameters(keyValuePair);
+ }
+ if (status == NO_ERROR && reconfig) {
+ delete mAudioMixer;
+ readOutputParameters();
+ mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
+ for (size_t i = 0; i < mTracks.size() ; i++) {
+ int name = getTrackName_l();
+ if (name < 0) break;
+ mTracks[i]->mName = name;
+ // limit track sample rate to 2 x new output sample rate
+ if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) {
+ mTracks[i]->mCblk->sampleRate = 2 * sampleRate();
+ }
+ }
+ sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ mWaitWorkCV.wait(mLock);
+ }
+ return reconfig;
+}
+
+status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ PlaybackThread::dumpInternals(fd, args);
+
+ snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+uint32_t AudioFlinger::MixerThread::getMaxBufferRecoveryInUsecs()
+{
+ uint32_t time = ((mFrameCount * 1000) / mSampleRate) * 1000;
+ // Add some margin with regard to scheduling precision
+ if (time > 10000) {
+ time -= 10000;
+ }
+ return time;
+}
+
+// ----------------------------------------------------------------------------
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output)
+ : PlaybackThread(audioFlinger, output),
+ mLeftVolume (1.0), mRightVolume(1.0)
+{
+ mType = PlaybackThread::DIRECT;
+}
+
+AudioFlinger::DirectOutputThread::~DirectOutputThread()
+{
+}
+
+
+bool AudioFlinger::DirectOutputThread::threadLoop()
+{
+ uint32_t sleepTime = 1000;
+ uint32_t maxBufferRecoveryInUsecs = getMaxBufferRecoveryInUsecs();
+ sp<Track> trackToRemove;
+ sp<Track> activeTrack;
+ nsecs_t standbyTime = systemTime();
+ int8_t *curBuf;
+ size_t mixBufferSize = mFrameCount*mFrameSize;
+
+ while (!exitPending())
+ {
+ processConfigEvents();
+
+ { // scope for the mLock
+
+ Mutex::Autolock _l(mLock);
+
+ if (checkForNewParameters_l()) {
+ mixBufferSize = mFrameCount*mFrameSize;
+ maxBufferRecoveryInUsecs = getMaxBufferRecoveryInUsecs();
+ }
+
+ // put audio hardware into standby after short delay
+ if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
+ // wait until we have something to do...
+ if (!mStandby) {
+ LOGV("Audio hardware entering standby, mixer %p\n", this);
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ }
+
+ if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+
+ if (exitPending()) break;
+
+ LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+
+ if (mMasterMute == false) {
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.audio.silent", value, "0");
+ if (atoi(value)) {
+ LOGD("Silence is golden");
+ setMasterMute(true);
+ }
+ }
+
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ sleepTime = 1000;
+ continue;
+ }
+ }
+
+ // find out which tracks need to be processed
+ if (mActiveTracks.size() != 0) {
+ sp<Track> t = mActiveTracks[0].promote();
+ if (t == 0) continue;
+
+ Track* const track = t.get();
+ audio_track_cblk_t* cblk = track->cblk();
+
+ // The first time a track is added we wait
+ // for all its buffers to be filled before processing it
+ if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+ !track->isPaused())
+ {
+ //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+ // compute volume for this track
+ float left, right;
+ if (track->isMuted() || mMasterMute || track->isPausing() ||
+ mStreamTypes[track->type()].mute) {
+ left = right = 0;
+ if (track->isPausing()) {
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->type()].volume;
+ float v = mMasterVolume * typeVolume;
+ float v_clamped = v * cblk->volume[0];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ left = v_clamped/MAX_GAIN;
+ v_clamped = v * cblk->volume[1];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ right = v_clamped/MAX_GAIN;
+ }
+
+ if (left != mLeftVolume || right != mRightVolume) {
+ mOutput->setVolume(left, right);
+ left = mLeftVolume;
+ right = mRightVolume;
+ }
+
+ if (track->mFillingUpStatus == Track::FS_FILLED) {
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ }
+ }
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetries;
+ activeTrack = t;
+ } else {
+ //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+ if (track->isStopped()) {
+ track->reset();
+ }
+ if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ trackToRemove = track;
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+ trackToRemove = track;
+ }
+
+ // For tracks using static shared memry buffer, make sure that we have
+ // written enough data to audio hardware before disabling the track
+ // NOTE: this condition with arrive before track->mRetryCount <= 0 so we
+ // don't care about code removing track from active list above.
+ if ((track->mSharedBuffer != 0) && (mBytesWritten < mMinBytesToWrite)) {
+ activeTrack = t;
+ }
+ }
+ }
+ }
+
+ // remove all the tracks that need to be...
+ if (UNLIKELY(trackToRemove != 0)) {
+ mActiveTracks.remove(trackToRemove);
+ if (trackToRemove->isTerminated()) {
+ mTracks.remove(trackToRemove);
+ deleteTrackName_l(trackToRemove->mName);
+ }
+ }
+ }
+
+ if (activeTrack != 0) {
+ AudioBufferProvider::Buffer buffer;
+ size_t frameCount = mFrameCount;
+ curBuf = (int8_t *)mMixBuffer;
+ // output audio to hardware
+ while(frameCount) {
+ buffer.frameCount = frameCount;
+ activeTrack->getNextBuffer(&buffer);
+ if (UNLIKELY(buffer.raw == 0)) {
+ memset(curBuf, 0, frameCount * mFrameSize);
+ break;
+ }
+ memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize);
+ frameCount -= buffer.frameCount;
+ curBuf += buffer.frameCount * mFrameSize;
+ activeTrack->releaseBuffer(&buffer);
+ }
+ sleepTime = 0;
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ } else {
+ if (sleepTime == 0) {
+ sleepTime = maxBufferRecoveryInUsecs;
+ } else if (mBytesWritten != 0 && AudioSystem::isLinearPCM(mFormat)) {
+ memset (mMixBuffer, 0, mFrameCount * mFrameSize);
+ sleepTime = 0;
+ }
+ }
+
+ if (mSuspended) {
+ sleepTime = maxBufferRecoveryInUsecs;
+ }
+ // sleepTime == 0 means we must write to audio hardware
+ if (sleepTime == 0) {
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ int bytesWritten = (int)mOutput->write(mMixBuffer, mixBufferSize);
+ if (bytesWritten) mBytesWritten += bytesWritten;
+ mNumWrites++;
+ mInWrite = false;
+ mStandby = false;
+ } else {
+ usleep(sleepTime);
+ }
+
+ // finally let go of removed track, without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock.
+ trackToRemove.clear();
+ activeTrack.clear();
+ }
+
+ if (!mStandby) {
+ mOutput->standby();
+ }
+
+ LOGV("DirectOutputThread %p exiting", this);
+ return false;
+}
+
+// getTrackName_l() must be called with ThreadBase::mLock held
+int AudioFlinger::DirectOutputThread::getTrackName_l()
+{
+ return 0;
+}
+
+// deleteTrackName_l() must be called with ThreadBase::mLock held
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+{
+}
+
+// checkForNewParameters_l() must be called with ThreadBase::mLock held
+bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
+{
+ bool reconfig = false;
+
+ while (!mNewParameters.isEmpty()) {
+ status_t status = NO_ERROR;
+ String8 keyValuePair = mNewParameters[0];
+ AudioParameter param = AudioParameter(keyValuePair);
+ int value;
+
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be garantied
+ // if frame count is changed after track creation
+ if (!mTracks.isEmpty()) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (status == NO_ERROR) {
+ status = mOutput->setParameters(keyValuePair);
+ if (!mStandby && status == INVALID_OPERATION) {
+ mOutput->standby();
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->setParameters(keyValuePair);
+ }
+ if (status == NO_ERROR && reconfig) {
+ readOutputParameters();
+ sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED);
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ mWaitWorkCV.wait(mLock);
+ }
+ return reconfig;
+}
+
+uint32_t AudioFlinger::DirectOutputThread::getMaxBufferRecoveryInUsecs()
+{
+ uint32_t time;
+ if (AudioSystem::isLinearPCM(mFormat)) {
+ time = ((mFrameCount * 1000) / mSampleRate) * 1000;
+ // Add some margin with regard to scheduling precision
+ if (time > 10000) {
+ time -= 10000;
+ }
+ } else {
+ time = 10000;
+ }
+ return time;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread)
+ : MixerThread(audioFlinger, mainThread->getOutput())
+{
+ mType = PlaybackThread::DUPLICATING;
+ addOutputTrack(mainThread);
+}
+
+AudioFlinger::DuplicatingThread::~DuplicatingThread()
+{
+ mOutputTracks.clear();
+}
+
+bool AudioFlinger::DuplicatingThread::threadLoop()
+{
+ uint32_t sleepTime = 1000;
+ uint32_t maxBufferRecoveryInUsecs = getMaxBufferRecoveryInUsecs();
+ int16_t* curBuf = mMixBuffer;
+ Vector< sp<Track> > tracksToRemove;
+ size_t enabledTracks = 0;
+ nsecs_t standbyTime = systemTime();
+ size_t mixBufferSize = mFrameCount*mFrameSize;
+ SortedVector< sp<OutputTrack> > outputTracks;
+ uint32_t writeFrames = 0;
+
+ while (!exitPending())
+ {
+ processConfigEvents();
+
+ enabledTracks = 0;
+ { // scope for the mLock
+
+ Mutex::Autolock _l(mLock);
+
+ if (checkForNewParameters_l()) {
+ mixBufferSize = mFrameCount*mFrameSize;
+ maxBufferRecoveryInUsecs = getMaxBufferRecoveryInUsecs();
+ }
+
+ const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ outputTracks.add(mOutputTracks[i]);
+ }
+
+ // put audio hardware into standby after short delay
+ if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended) {
+ if (!mStandby) {
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ outputTracks[i]->stop();
+ }
+ mStandby = true;
+ mBytesWritten = 0;
+ }
+
+ if (!activeTracks.size() && mConfigEvents.isEmpty()) {
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+ outputTracks.clear();
+
+ if (exitPending()) break;
+
+ LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+ mWaitWorkCV.wait(mLock);
+ LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+ if (mMasterMute == false) {
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.audio.silent", value, "0");
+ if (atoi(value)) {
+ LOGD("Silence is golden");
+ setMasterMute(true);
+ }
+ }
+
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ sleepTime = 1000;
+ continue;
+ }
+ }
+
+ enabledTracks = prepareTracks_l(activeTracks, &tracksToRemove);
+ }
+
+ if (LIKELY(enabledTracks)) {
+ // mix buffers...
+ mAudioMixer->process(curBuf);
+ sleepTime = 0;
+ writeFrames = mFrameCount;
+ } else {
+ if (sleepTime == 0) {
+ sleepTime = maxBufferRecoveryInUsecs;
+ } else if (mBytesWritten != 0) {
+ // flush remaining overflow buffers in output tracks
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ if (outputTracks[i]->isActive()) {
+ sleepTime = 0;
+ writeFrames = 0;
+ break;
+ }
+ }
+ }
+ }
+
+ if (mSuspended) {
+ sleepTime = maxBufferRecoveryInUsecs;
+ }
+ // sleepTime == 0 means we must write to audio hardware
+ if (sleepTime == 0) {
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ for (size_t i = 0; i < outputTracks.size(); i++) {
+ outputTracks[i]->write(curBuf, writeFrames);
+ }
+ mStandby = false;
+ mBytesWritten += mixBufferSize;
+ } else {
+ usleep(sleepTime);
+ }
+
+ // finally let go of all our tracks, without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock.
+ tracksToRemove.clear();
+ outputTracks.clear();
+ }
+
+ { // scope for the mLock
+
+ Mutex::Autolock _l(mLock);
+ if (!mStandby) {
+ LOGV("DuplicatingThread() exiting out of standby");
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ mOutputTracks[i]->destroy();
+ }
+ }
+ }
+
+ return false;
+}
+
+void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+{
+ int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
+ OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
+ mSampleRate,
+ mFormat,
+ mChannelCount,
+ frameCount);
+ if (outputTrack->cblk() != NULL) {
+ thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f);
+ mOutputTracks.add(outputTrack);
+ LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+ }
+}
+
+void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+{
+ Mutex::Autolock _l(mLock);
+ for (size_t i = 0; i < mOutputTracks.size(); i++) {
+ if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
+ mOutputTracks[i]->destroy();
+ mOutputTracks.removeAt(i);
+ return;
+ }
+ }
+ LOGV("removeOutputTrack(): unkonwn thread: %p", thread);
}
// ----------------------------------------------------------------------------
// TrackBase constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::TrackBase::TrackBase(
- const sp<MixerThread>& mixerThread,
+AudioFlinger::ThreadBase::TrackBase::TrackBase(
+ const wp<ThreadBase>& thread,
const sp<Client>& client,
uint32_t sampleRate,
int format,
@@ -1567,21 +2102,15 @@
uint32_t flags,
const sp<IMemory>& sharedBuffer)
: RefBase(),
- mMixerThread(mixerThread),
+ mThread(thread),
mClient(client),
+ mCblk(0),
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);
@@ -1635,16 +2164,22 @@
}
}
-AudioFlinger::MixerThread::TrackBase::~TrackBase()
+AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
if (mCblk) {
- mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
+ if (mClient == NULL) {
+ delete mCblk;
+ }
}
mCblkMemory.clear(); // and free the shared memory
- mClient.clear();
+ if (mClient != NULL) {
+ Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+ mClient.clear();
+ }
}
-void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
mFrameCount = buffer->frameCount;
@@ -1652,7 +2187,7 @@
buffer->frameCount = 0;
}
-bool AudioFlinger::MixerThread::TrackBase::step() {
+bool AudioFlinger::ThreadBase::TrackBase::step() {
bool result;
audio_track_cblk_t* cblk = this->cblk();
@@ -1664,7 +2199,7 @@
return result;
}
-void AudioFlinger::MixerThread::TrackBase::reset() {
+void AudioFlinger::ThreadBase::TrackBase::reset() {
audio_track_cblk_t* cblk = this->cblk();
cblk->user = 0;
@@ -1675,27 +2210,27 @@
LOGV("TrackBase::reset");
}
-sp<IMemory> AudioFlinger::MixerThread::TrackBase::getCblk() const
+sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
{
return mCblkMemory;
}
-int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
+int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
-int AudioFlinger::MixerThread::TrackBase::channelCount() const {
+int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
return (int)mCblk->channels;
}
-void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+void* AudioFlinger::ThreadBase::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;
+ int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
+ int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
- if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
- (cblk->channels == 2 && ((unsigned long)bufferStart & 3))) {
+ if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
+ ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
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,
@@ -1708,9 +2243,9 @@
// ----------------------------------------------------------------------------
-// Track constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::Track::Track(
- const sp<MixerThread>& mixerThread,
+// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
+AudioFlinger::PlaybackThread::Track::Track(
+ const wp<ThreadBase>& thread,
const sp<Client>& client,
int streamType,
uint32_t sampleRate,
@@ -1718,42 +2253,62 @@
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer)
- : TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+ : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer),
+ mMute(false), mSharedBuffer(sharedBuffer), mName(-1)
{
- 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);
+ if (mCblk != NULL) {
+ sp<ThreadBase> baseThread = thread.promote();
+ if (baseThread != 0) {
+ PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
+ mName = playbackThread->getTrackName_l();
+ }
+ LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ if (mName < 0) {
+ LOGE("no more track names available");
+ }
+ mVolume[0] = 1.0f;
+ mVolume[1] = 1.0f;
+ mStreamType = streamType;
+ // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
+ // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
+ mCblk->frameSize = AudioSystem::isLinearPCM(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t);
}
}
-void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size)
+AudioFlinger::PlaybackThread::Track::~Track()
{
- snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
+ LOGV("PlaybackThread::Track destructor");
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ mState = TERMINATED;
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::destroy()
+{
+ // NOTE: destroyTrack_l() can remove a strong reference to this Track
+ // by removing it from mTracks vector, so there is a risk that this Tracks's
+ // desctructor is called. As the destructor needs to lock mLock,
+ // we must acquire a strong reference on this Track before locking mLock
+ // here so that the destructor is called only when exiting this function.
+ // On the other hand, as long as Track::destroy() is only called by
+ // TrackHandle destructor, the TrackHandle still holds a strong ref on
+ // this Track with its member mTrack.
+ sp<Track> keep(this);
+ { // scope for mLock
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ playbackThread->destroyTrack_l(this);
+ }
+ }
+}
+
+void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %5d %5d %3u %3u %3u %04u %1d %1d %1d %5u %5u %5u %08x %08x\n",
mName - AudioMixer::TRACK0,
(mClient == NULL) ? getpid() : mClient->pid(),
mStreamType,
@@ -1770,7 +2325,7 @@
mCblk->user);
}
-status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
@@ -1807,76 +2362,90 @@
getNextBuffer_exit:
buffer->raw = 0;
buffer->frameCount = 0;
+ LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
return NOT_ENOUGH_DATA;
}
-bool AudioFlinger::MixerThread::Track::isReady() const {
+bool AudioFlinger::PlaybackThread::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()
+status_t AudioFlinger::PlaybackThread::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);
+ LOGV("start(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ playbackThread->addTrack_l(this);
+ }
return NO_ERROR;
}
-void AudioFlinger::MixerThread::Track::stop()
+void AudioFlinger::PlaybackThread::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("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState > STOPPED) {
+ mState = STOPPED;
+ // If the track is not active (PAUSED and buffers full), flush buffers
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+ reset();
+ }
+ LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread);
}
- LOGV("(> STOPPED) => STOPPED (%d)", mName);
}
}
-void AudioFlinger::MixerThread::Track::pause()
+void AudioFlinger::PlaybackThread::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);
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState == ACTIVE || mState == RESUMING) {
+ mState = PAUSING;
+ LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
+ }
}
}
-void AudioFlinger::MixerThread::Track::flush()
+void AudioFlinger::PlaybackThread::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;
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ Mutex::Autolock _l(thread->mLock);
+ if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+ return;
+ }
+ // No point remaining in PAUSED state after a flush => go to
+ // STOPPED state
+ mState = STOPPED;
- mCblk->lock.lock();
- // NOTE: reset() will reset cblk->user and cblk->server with
- // the risk that at the same time, the AudioMixer is trying to read
- // data. In this case, getNextBuffer() would return a NULL pointer
- // as audio buffer => the AudioMixer code MUST always test that pointer
- // returned by getNextBuffer() is not NULL!
- reset();
- mCblk->lock.unlock();
+ 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()
+void AudioFlinger::PlaybackThread::Track::reset()
{
// Do not reset twice to avoid discarding data written just after a flush and before
// the audioflinger thread detects the track is stopped.
@@ -1886,17 +2455,17 @@
// written to buffer
mCblk->flowControlFlag = 1;
mCblk->forceReady = 0;
- mFillingUpStatus = FS_FILLING;
+ mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
}
-void AudioFlinger::MixerThread::Track::mute(bool muted)
+void AudioFlinger::PlaybackThread::Track::mute(bool muted)
{
mMute = muted;
}
-void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
+void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
{
mVolume[0] = left;
mVolume[1] = right;
@@ -1905,28 +2474,35 @@
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread::RecordTrack::RecordTrack(
- const sp<MixerThread>& mixerThread,
+AudioFlinger::RecordThread::RecordTrack::RecordTrack(
+ const wp<ThreadBase>& thread,
const sp<Client>& client,
- int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags)
- : TrackBase(mixerThread, client, sampleRate, format,
+ : TrackBase(thread, client, sampleRate, format,
channelCount, frameCount, flags, 0),
- mOverflow(false), mInputSource(inputSource)
+ mOverflow(false)
+{
+ if (mCblk != NULL) {
+ LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+ if (format == AudioSystem::PCM_16_BIT) {
+ mCblk->frameSize = channelCount * sizeof(int16_t);
+ } else if (format == AudioSystem::PCM_8_BIT) {
+ mCblk->frameSize = channelCount * sizeof(int8_t);
+ } else {
+ mCblk->frameSize = sizeof(int8_t);
+ }
+ }
+}
+
+AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
{
}
-AudioFlinger::MixerThread::RecordTrack::~RecordTrack()
-{
- Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
- mMixerThread->deleteTrackName_l(mName);
-}
-
-status_t AudioFlinger::MixerThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
@@ -1965,180 +2541,247 @@
return NOT_ENOUGH_DATA;
}
-status_t AudioFlinger::MixerThread::RecordTrack::start()
+status_t AudioFlinger::RecordThread::RecordTrack::start()
{
- return mMixerThread->mAudioFlinger->startRecord(this);
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ return recordThread->start(this);
+ }
+ return NO_INIT;
}
-void AudioFlinger::MixerThread::RecordTrack::stop()
+void AudioFlinger::RecordThread::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;
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ recordThread->stop(this);
+ TrackBase::reset();
+ // Force overerrun condition to avoid false overrun callback until first data is
+ // read from buffer
+ mCblk->flowControlFlag = 1;
+ }
+}
+
+void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %05d %03u %03u %04u %01d %05u %08x %08x\n",
+ (mClient == NULL) ? getpid() : mClient->pid(),
+ mFormat,
+ mCblk->channels,
+ mFrameCount,
+ mState,
+ mCblk->sampleRate,
+ mCblk->server,
+ mCblk->user);
}
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::OutputTrack::OutputTrack(
- const sp<MixerThread>& mixerThread,
+AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
+ const wp<ThreadBase>& thread,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount)
- : Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL),
- mOutputMixerThread(mixerThread)
+ : Track(thread, NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL),
+ mActive(false)
{
-
- 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);
-
+
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
+ if (mCblk != NULL) {
+ mCblk->out = 1;
+ mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ mCblk->volume[0] = mCblk->volume[1] = 0x1000;
+ mOutBuffer.frameCount = 0;
+ mWaitTimeMs = (playbackThread->frameCount() * 2 * 1000) / playbackThread->sampleRate();
+ playbackThread->mTracks.add(this);
+ LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p mWaitTimeMs %d",
+ mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd, mWaitTimeMs);
+ } else {
+ LOGW("Error creating output track on thread %p", playbackThread);
+ }
}
-AudioFlinger::MixerThread::OutputTrack::~OutputTrack()
+AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack()
{
- stop();
+ clearBufferQueue();
}
-status_t AudioFlinger::MixerThread::OutputTrack::start()
+status_t AudioFlinger::PlaybackThread::OutputTrack::start()
{
status_t status = Track::start();
-
+ if (status != NO_ERROR) {
+ return status;
+ }
+
+ mActive = true;
mRetryCount = 127;
return status;
}
-void AudioFlinger::MixerThread::OutputTrack::stop()
+void AudioFlinger::PlaybackThread::OutputTrack::stop()
{
Track::stop();
clearBufferQueue();
mOutBuffer.frameCount = 0;
+ mActive = false;
}
-void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames)
+bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
uint32_t channels = mCblk->channels;
-
+ bool outputBufferFull = false;
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");
+
+ uint32_t waitTimeLeftMs = mWaitTimeMs;
+
+ if (!mActive && frames != 0) {
+ start();
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ MixerThread *mixerThread = (MixerThread *)thread.get();
+ if (mCblk->frameCount > frames){
+ if (mBufferQueue.size() < kMaxOverFlowBuffers) {
+ uint32_t startFrames = (mCblk->frameCount - frames);
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[startFrames * channels];
+ pInBuffer->frameCount = startFrames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else {
+ LOGW ("OutputTrack::write() %p no more buffers in queue", this);
+ }
}
- }
+ }
}
- while (1) {
+ while (waitTimeLeftMs) {
// First write pending buffers, then new data
if (mBufferQueue.size()) {
pInBuffer = mBufferQueue.itemAt(0);
} else {
pInBuffer = &inBuffer;
}
-
+
if (pInBuffer->frameCount == 0) {
break;
}
-
+
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
- if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+ nsecs_t startTime = systemTime();
+ if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+ LOGV ("OutputTrack::write() %p no more output buffers", this);
+ outputBufferFull = true;
break;
}
+ uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
+ LOGV("OutputTrack::write() to thread %p waitTimeMs %d waitTimeLeftMs %d", mThread.unsafe_get(), waitTimeMs, waitTimeLeftMs);
+ if (waitTimeLeftMs >= waitTimeMs) {
+ waitTimeLeftMs -= waitTimeMs;
+ } else {
+ waitTimeLeftMs = 0;
+ }
}
-
+
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
mCblk->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
pInBuffer->i16 += outFrames * channels;
mOutBuffer.frameCount -= outFrames;
- mOutBuffer.i16 += outFrames * channels;
-
+ mOutBuffer.i16 += outFrames * channels;
+
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
delete [] pInBuffer->mBuffer;
delete pInBuffer;
+ LOGV("OutputTrack::write() %p released overflow buffer %d", this, mBufferQueue.size());
} 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) {
+ if (mBufferQueue.size() < kMaxOverFlowBuffers) {
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);
+ LOGV("OutputTrack::write() %p adding overflow buffer %d", this, mBufferQueue.size());
} else {
- LOGW("OutputTrack::write() no more buffers");
+ LOGW("OutputTrack::write() %p no more overflow buffers", this);
}
}
-
+
// Calling write() with a 0 length buffer, means that no more data will be written:
- // If no more buffers are pending, fill output track buffer to make sure it is started
+ // 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);
+ if (frames == 0 && mBufferQueue.size() == 0) {
+ if (mCblk->user < mCblk->frameCount) {
+ frames = mCblk->frameCount - mCblk->user;
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[frames * channels];
+ pInBuffer->frameCount = frames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else if (mActive) {
+ stop();
+ }
}
+ return outputBufferFull;
}
-status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
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);
+// 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;
+ Mutex::Autolock _l(cblk->lock);
+ goto start_loop_here;
+ while (framesAvail == 0) {
+ active = mActive;
+ if (UNLIKELY(!active)) {
+ LOGV("Not active and NO_MORE_BUFFERS");
+ return AudioTrack::NO_MORE_BUFFERS;
+ }
+ result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+ if (result != NO_ERROR) {
+ return AudioTrack::NO_MORE_BUFFERS;
+ }
+ // read the server count again
+ start_loop_here:
+ framesAvail = cblk->framesAvailable_l();
+ }
}
+// if (framesAvail < framesReq) {
+// return AudioTrack::NO_MORE_BUFFERS;
+// }
+
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
@@ -2156,11 +2799,11 @@
}
-void AudioFlinger::MixerThread::OutputTrack::clearBufferQueue()
+void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
Buffer *pBuffer;
-
+
for (size_t i = 0; i < size; i++) {
pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
@@ -2180,9 +2823,10 @@
// 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(mPid);
+ mAudioFlinger->removeClient_l(mPid);
}
const sp<MemoryDealer>& AudioFlinger::Client::heap() const
@@ -2192,7 +2836,7 @@
// ----------------------------------------------------------------------------
-AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::MixerThread::Track>& track)
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
: BnAudioTrack(),
mTrack(track)
{
@@ -2244,7 +2888,7 @@
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int inputSource,
+ int input,
uint32_t sampleRate,
int format,
int channelCount,
@@ -2252,14 +2896,13 @@
uint32_t flags,
status_t *status)
{
- sp<MixerThread::RecordTrack> recordTrack;
+ sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
wp<Client> wclient;
- AudioStreamIn* input = 0;
- int inFrameCount;
- size_t inputBufferSize;
status_t lStatus;
+ RecordThread *thread;
+ size_t inFrameCount;
// check calling permissions
if (!recordingAllowed()) {
@@ -2267,30 +2910,15 @@
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);
+ thread = checkRecordThread_l(input);
+ if (thread == NULL) {
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
wclient = mClients.valueFor(pid);
if (wclient != NULL) {
client = wclient.promote();
@@ -2299,15 +2927,14 @@
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,
+ recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate,
format, channelCount, frameCount, flags);
}
if (recordTrack->getCblk() == NULL) {
+ // remove local strong reference to Client before deleting the RecordTrack so that the Client
+ // destructor is called by the TrackBase destructor with mLock held
+ client.clear();
recordTrack.clear();
lStatus = NO_MEMORY;
goto Exit;
@@ -2324,22 +2951,9 @@
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)
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
@@ -2371,86 +2985,164 @@
// ----------------------------------------------------------------------------
-AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware,
- const sp<AudioFlinger>& audioFlinger) :
- mAudioHardware(audioHardware),
- mAudioFlinger(audioFlinger),
- mActive(false)
+AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels) :
+ ThreadBase(audioFlinger),
+ mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
{
+ mReqChannelCount = AudioSystem::popCount(channels);
+ mReqSampleRate = sampleRate;
+ readInputParameters();
+ sendConfigEvent(AudioSystem::INPUT_OPENED);
}
-AudioFlinger::AudioRecordThread::~AudioRecordThread()
+
+AudioFlinger::RecordThread::~RecordThread()
{
+ delete[] mRsmpInBuffer;
+ if (mResampler != 0) {
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
+ }
}
-bool AudioFlinger::AudioRecordThread::threadLoop()
+void AudioFlinger::RecordThread::onFirstRef()
{
- LOGV("AudioRecordThread: start record loop");
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "Record Thread %p", this);
+
+ run(buffer, PRIORITY_URGENT_AUDIO);
+}
+bool AudioFlinger::RecordThread::threadLoop()
+{
AudioBufferProvider::Buffer buffer;
- int inBufferSize = 0;
- int inFrameCount = 0;
- AudioStreamIn* input = 0;
+ sp<RecordTrack> activeTrack;
- 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();
+ processConfigEvents();
+
+ { // scope for mLock
+ Mutex::Autolock _l(mLock);
+ checkForNewParameters_l();
+ if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
+ if (!mStandby) {
+ mInput->standby();
+ mStandby = true;
+ }
+
+ if (exitPending()) break;
+
+ LOGV("RecordThread: loop stopping");
+ // go to sleep
mWaitWorkCV.wait(mLock);
-
- LOGV("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();
+ LOGV("RecordThread: loop starting");
+ continue;
+ }
+ if (mActiveTrack != 0) {
+ if (mActiveTrack->mState == TrackBase::PAUSING) {
+ mActiveTrack.clear();
+ mStartStopCond.broadcast();
+ } else if (mActiveTrack->mState == TrackBase::RESUMING) {
+ mRsmpInIndex = mFrameCount;
+ if (mReqChannelCount != mActiveTrack->channelCount()) {
+ mActiveTrack.clear();
+ } else {
+ mActiveTrack->mState = TrackBase::ACTIVE;
+ }
+ mStartStopCond.broadcast();
+ }
+ mStandby = false;
+ }
+ }
+
+ if (mActiveTrack != 0) {
+ buffer.frameCount = mFrameCount;
+ if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ size_t framesOut = buffer.frameCount;
+ if (mResampler == 0) {
+ // no resampling
+ while (framesOut) {
+ size_t framesIn = mFrameCount - mRsmpInIndex;
+ if (framesIn) {
+ int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
+ int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
+ if (framesIn > framesOut)
+ framesIn = framesOut;
+ mRsmpInIndex += framesIn;
+ framesOut -= framesIn;
+ if (mChannelCount == mReqChannelCount ||
+ mFormat != AudioSystem::PCM_16_BIT) {
+ memcpy(dst, src, framesIn * mFrameSize);
+ } else {
+ int16_t *src16 = (int16_t *)src;
+ int16_t *dst16 = (int16_t *)dst;
+ if (mChannelCount == 1) {
+ while (framesIn--) {
+ *dst16++ = *src16;
+ *dst16++ = *src16++;
+ }
+ } else {
+ while (framesIn--) {
+ *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1);
+ src16 += 2;
+ }
+ }
+ }
+ }
+ if (framesOut && mFrameCount == mRsmpInIndex) {
+ ssize_t bytesRead;
+ if (framesOut == mFrameCount &&
+ (mChannelCount == mReqChannelCount || mFormat != AudioSystem::PCM_16_BIT)) {
+ bytesRead = mInput->read(buffer.raw, mInputBytes);
+ framesOut = 0;
+ } else {
+ bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+ mRsmpInIndex = 0;
+ }
+ if (bytesRead < 0) {
+ LOGE("Error reading audio input");
+ sleep(1);
+ mRsmpInIndex = mFrameCount;
+ framesOut = 0;
+ buffer.frameCount = 0;
+ }
+ }
}
} 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) {
+ // resampling
- 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();
- }
+ memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));
+ // alter output frame count as if we were expecting stereo samples
+ if (mChannelCount == 1 && mReqChannelCount == 1) {
+ framesOut >>= 1;
+ }
+ mResampler->resample(mRsmpOutBuffer, framesOut, this);
+ // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
+ // are 32 bit aligned which should be always true.
+ if (mChannelCount == 2 && mReqChannelCount == 1) {
+ AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+ // the resampler always outputs stereo samples: do post stereo to mono conversion
+ int16_t *src = (int16_t *)mRsmpOutBuffer;
+ int16_t *dst = buffer.i16;
+ while (framesOut--) {
+ *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1);
+ src += 2;
+ }
+ } else {
+ AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ }
+ }
+ mActiveTrack->releaseBuffer(&buffer);
+ mActiveTrack->overflow();
+ }
// client isn't retrieving buffers fast enough
else {
- if (!mRecordTrack->setOverflow())
- LOGW("AudioRecordThread: buffer overflow");
+ if (!mActiveTrack->setOverflow())
+ LOGW("RecordThread: 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.
@@ -2459,73 +3151,559 @@
}
}
-
- if (input) {
- delete input;
+ if (!mStandby) {
+ mInput->standby();
}
- mRecordTrack.clear();
-
+ mActiveTrack.clear();
+
+ LOGV("RecordThread %p exiting", this);
return false;
}
-status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack)
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
{
- LOGV("AudioRecordThread::start");
+ LOGV("RecordThread::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;
+ if (mActiveTrack != 0) {
+ if (recordTrack != mActiveTrack.get()) return -EBUSY;
- mRecordTrack = recordTrack;
+ if (mActiveTrack->mState == TrackBase::PAUSING) mActiveTrack->mState = TrackBase::RESUMING;
+ return NO_ERROR;
+ }
+
+ mActiveTrack = recordTrack;
+ mActiveTrack->mState = TrackBase::RESUMING;
// signal thread to start
LOGV("Signal record thread");
mWaitWorkCV.signal();
- mWaitWorkCV.wait(mLock);
- LOGV("Record started, status %d", mStartStatus);
- return mStartStatus;
+ mStartStopCond.wait(mLock);
+ if (mActiveTrack != 0) {
+ LOGV("Record started OK");
+ return NO_ERROR;
+ } else {
+ LOGV("Record failed to start");
+ return BAD_VALUE;
+ }
}
-void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) {
- LOGV("AudioRecordThread::stop");
+void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
+ LOGV("RecordThread::stop");
AutoMutex lock(&mLock);
- if (mActive && (recordTrack == mRecordTrack.get())) {
- mActive = false;
- mStopped.wait(mLock);
+ if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
+ mActiveTrack->mState = TrackBase::PAUSING;
+ mStartStopCond.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)
+status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
pid_t pid = 0;
- if (mRecordTrack != 0 && mRecordTrack->mClient != 0) {
- snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid());
+ snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
+ result.append(buffer);
+
+ if (mActiveTrack != 0) {
+ result.append("Active Track:\n");
+ result.append(" Clien Fmt Chn Buf S SRate Serv User\n");
+ mActiveTrack->dump(buffer, SIZE);
result.append(buffer);
+
+ snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate);
+ result.append(buffer);
+
+
} else {
result.append("No record client\n");
}
write(fd, result.string(), result.size());
+
+ dumpBase(fd, args);
+
return NO_ERROR;
}
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ size_t framesReq = buffer->frameCount;
+ size_t framesReady = mFrameCount - mRsmpInIndex;
+ int channelCount;
+
+ if (framesReady == 0) {
+ ssize_t bytesRead = mInput->read(mRsmpInBuffer, mInputBytes);
+ if (bytesRead < 0) {
+ LOGE("RecordThread::getNextBuffer() Error reading audio input");
+ sleep(1);
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+ }
+ mRsmpInIndex = 0;
+ framesReady = mFrameCount;
+ }
+
+ if (framesReq > framesReady) {
+ framesReq = framesReady;
+ }
+
+ if (mChannelCount == 1 && mReqChannelCount == 2) {
+ channelCount = 1;
+ } else {
+ channelCount = 2;
+ }
+ buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount;
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ mRsmpInIndex += buffer->frameCount;
+ buffer->frameCount = 0;
+}
+
+bool AudioFlinger::RecordThread::checkForNewParameters_l()
+{
+ bool reconfig = false;
+
+ while (!mNewParameters.isEmpty()) {
+ status_t status = NO_ERROR;
+ String8 keyValuePair = mNewParameters[0];
+ AudioParameter param = AudioParameter(keyValuePair);
+ int value;
+ int reqFormat = mFormat;
+ int reqSamplingRate = mReqSampleRate;
+ int reqChannelCount = mReqChannelCount;
+
+ if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
+ reqSamplingRate = value;
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
+ reqFormat = value;
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
+ reqChannelCount = AudioSystem::popCount(value);
+ reconfig = true;
+ }
+ if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
+ // do not accept frame count changes if tracks are open as the track buffer
+ // size depends on frame count and correct behavior would not be garantied
+ // if frame count is changed after track creation
+ if (mActiveTrack != 0) {
+ status = INVALID_OPERATION;
+ } else {
+ reconfig = true;
+ }
+ }
+ if (status == NO_ERROR) {
+ status = mInput->setParameters(keyValuePair);
+ if (status == INVALID_OPERATION) {
+ mInput->standby();
+ status = mInput->setParameters(keyValuePair);
+ }
+ if (reconfig) {
+ if (status == BAD_VALUE &&
+ reqFormat == mInput->format() && reqFormat == AudioSystem::PCM_16_BIT &&
+ ((int)mInput->sampleRate() <= 2 * reqSamplingRate) &&
+ (AudioSystem::popCount(mInput->channels()) < 3) && (reqChannelCount < 3)) {
+ status = NO_ERROR;
+ }
+ if (status == NO_ERROR) {
+ readInputParameters();
+ sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED);
+ }
+ }
+ }
+
+ mNewParameters.removeAt(0);
+
+ mParamStatus = status;
+ mParamCond.signal();
+ mWaitWorkCV.wait(mLock);
+ }
+ return reconfig;
+}
+
+String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+{
+ return mInput->getParameters(keys);
+}
+
+void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) {
+ AudioSystem::OutputDescriptor desc;
+ void *param2 = 0;
+
+ switch (event) {
+ case AudioSystem::INPUT_OPENED:
+ case AudioSystem::INPUT_CONFIG_CHANGED:
+ desc.channels = mChannelCount;
+ desc.samplingRate = mSampleRate;
+ desc.format = mFormat;
+ desc.frameCount = mFrameCount;
+ desc.latency = 0;
+ param2 = &desc;
+ break;
+
+ case AudioSystem::INPUT_CLOSED:
+ default:
+ break;
+ }
+ Mutex::Autolock _l(mAudioFlinger->mLock);
+ mAudioFlinger->audioConfigChanged_l(event, this, param2);
+}
+
+void AudioFlinger::RecordThread::readInputParameters()
+{
+ if (mRsmpInBuffer) delete mRsmpInBuffer;
+ if (mRsmpOutBuffer) delete mRsmpOutBuffer;
+ if (mResampler) delete mResampler;
+ mResampler = 0;
+
+ mSampleRate = mInput->sampleRate();
+ mChannelCount = AudioSystem::popCount(mInput->channels());
+ mFormat = mInput->format();
+ mFrameSize = mInput->frameSize();
+ mInputBytes = mInput->bufferSize();
+ mFrameCount = mInputBytes / mFrameSize;
+ mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
+
+ if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3)
+ {
+ int channelCount;
+ // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+ // stereo to mono post process as the resampler always outputs stereo.
+ if (mChannelCount == 1 && mReqChannelCount == 2) {
+ channelCount = 1;
+ } else {
+ channelCount = 2;
+ }
+ mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);
+ mResampler->setSampleRate(mSampleRate);
+ mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
+ mRsmpOutBuffer = new int32_t[mFrameCount * 2];
+
+ // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
+ if (mChannelCount == 1 && mReqChannelCount == 1) {
+ mFrameCount >>= 1;
+ }
+
+ }
+ mRsmpInIndex = mFrameCount;
+}
+
+// ----------------------------------------------------------------------------
+
+int AudioFlinger::openOutput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pLatencyMs,
+ uint32_t flags)
+{
+ status_t status;
+ PlaybackThread *thread = NULL;
+ mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+ uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+ uint32_t format = pFormat ? *pFormat : 0;
+ uint32_t channels = pChannels ? *pChannels : 0;
+ uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
+
+ LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x",
+ pDevices ? *pDevices : 0,
+ samplingRate,
+ format,
+ channels,
+ flags);
+
+ if (pDevices == NULL || *pDevices == 0) {
+ return 0;
+ }
+ Mutex::Autolock _l(mLock);
+
+ AudioStreamOut *output = mAudioHardware->openOutputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status);
+ LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
+ output,
+ samplingRate,
+ format,
+ channels,
+ status);
+
+ mHardwareStatus = AUDIO_HW_IDLE;
+ if (output != 0) {
+ if ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
+ (format != AudioSystem::PCM_16_BIT) ||
+ (channels != AudioSystem::CHANNEL_OUT_STEREO)) {
+ thread = new DirectOutputThread(this, output);
+ LOGV("openOutput() created direct output: ID %d thread %p", (mNextThreadId + 1), thread);
+ } else {
+ thread = new MixerThread(this, output);
+ LOGV("openOutput() created mixer output: ID %d thread %p", (mNextThreadId + 1), thread);
+ }
+ mPlaybackThreads.add(++mNextThreadId, thread);
+
+ if (pSamplingRate) *pSamplingRate = samplingRate;
+ if (pFormat) *pFormat = format;
+ if (pChannels) *pChannels = channels;
+ if (pLatencyMs) *pLatencyMs = thread->latency();
+ }
+
+ return mNextThreadId;
+}
+
+int AudioFlinger::openDuplicateOutput(int output1, int output2)
+{
+ Mutex::Autolock _l(mLock);
+ MixerThread *thread1 = checkMixerThread_l(output1);
+ MixerThread *thread2 = checkMixerThread_l(output2);
+
+ if (thread1 == NULL || thread2 == NULL) {
+ LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
+ return 0;
+ }
+
+
+ DuplicatingThread *thread = new DuplicatingThread(this, thread1);
+ thread->addOutputTrack(thread2);
+ mPlaybackThreads.add(++mNextThreadId, thread);
+ return mNextThreadId;
+}
+
+status_t AudioFlinger::closeOutput(int output)
+{
+ // keep strong reference on the playback thread so that
+ // it is not destroyed while exit() is executed
+ sp <PlaybackThread> thread;
+ {
+ Mutex::Autolock _l(mLock);
+ thread = checkPlaybackThread_l(output);
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("closeOutput() %d", output);
+
+ if (thread->type() == PlaybackThread::MIXER) {
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) {
+ DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
+ dupThread->removeOutputTrack((MixerThread *)thread.get());
+ }
+ }
+ }
+ void *param2 = 0;
+ audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, thread, param2);
+ mPlaybackThreads.removeItem(output);
+ }
+ thread->exit();
+
+ if (thread->type() != PlaybackThread::DUPLICATING) {
+ mAudioHardware->closeOutputStream(thread->getOutput());
+ }
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::suspendOutput(int output)
+{
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("suspendOutput() %d", output);
+ thread->suspend();
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::restoreOutput(int output)
+{
+ Mutex::Autolock _l(mLock);
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("restoreOutput() %d", output);
+
+ thread->restore();
+
+ return NO_ERROR;
+}
+
+int AudioFlinger::openInput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ uint32_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t acoustics)
+{
+ status_t status;
+ RecordThread *thread = NULL;
+ uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
+ uint32_t format = pFormat ? *pFormat : 0;
+ uint32_t channels = pChannels ? *pChannels : 0;
+ uint32_t reqSamplingRate = samplingRate;
+ uint32_t reqFormat = format;
+ uint32_t reqChannels = channels;
+
+ if (pDevices == NULL || *pDevices == 0) {
+ return 0;
+ }
+ Mutex::Autolock _l(mLock);
+
+ AudioStreamIn *input = mAudioHardware->openInputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status,
+ (AudioSystem::audio_in_acoustics)acoustics);
+ LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
+ input,
+ samplingRate,
+ format,
+ channels,
+ acoustics,
+ status);
+
+ // If the input could not be opened with the requested parameters and we can handle the conversion internally,
+ // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
+ // or stereo to mono conversions on 16 bit PCM inputs.
+ if (input == 0 && status == BAD_VALUE &&
+ reqFormat == format && format == AudioSystem::PCM_16_BIT &&
+ (samplingRate <= 2 * reqSamplingRate) &&
+ (AudioSystem::popCount(channels) < 3) && (AudioSystem::popCount(reqChannels) < 3)) {
+ LOGV("openInput() reopening with proposed sampling rate and channels");
+ input = mAudioHardware->openInputStream(*pDevices,
+ (int *)&format,
+ &channels,
+ &samplingRate,
+ &status,
+ (AudioSystem::audio_in_acoustics)acoustics);
+ }
+
+ if (input != 0) {
+ // Start record thread
+ thread = new RecordThread(this, input, reqSamplingRate, reqChannels);
+ mRecordThreads.add(++mNextThreadId, thread);
+ LOGV("openInput() created record thread: ID %d thread %p", mNextThreadId, thread);
+ if (pSamplingRate) *pSamplingRate = reqSamplingRate;
+ if (pFormat) *pFormat = format;
+ if (pChannels) *pChannels = reqChannels;
+
+ input->standby();
+ }
+
+ return mNextThreadId;
+}
+
+status_t AudioFlinger::closeInput(int input)
+{
+ // keep strong reference on the record thread so that
+ // it is not destroyed while exit() is executed
+ sp <RecordThread> thread;
+ {
+ Mutex::Autolock _l(mLock);
+ thread = checkRecordThread_l(input);
+ if (thread == NULL) {
+ return BAD_VALUE;
+ }
+
+ LOGV("closeInput() %d", input);
+ void *param2 = 0;
+ audioConfigChanged_l(AudioSystem::INPUT_CLOSED, thread, param2);
+ mRecordThreads.removeItem(input);
+ }
+ thread->exit();
+
+ mAudioHardware->closeInputStream(thread->getInput());
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
+{
+ Mutex::Autolock _l(mLock);
+ MixerThread *dstThread = checkMixerThread_l(output);
+ if (dstThread == NULL) {
+ LOGW("setStreamOutput() bad output id %d", output);
+ return BAD_VALUE;
+ }
+
+ LOGV("setStreamOutput() stream %d to output %d", stream, output);
+
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ if (thread != dstThread &&
+ thread->type() != PlaybackThread::DIRECT) {
+ MixerThread *srcThread = (MixerThread *)thread;
+ SortedVector < sp<MixerThread::Track> > tracks;
+ SortedVector < wp<MixerThread::Track> > activeTracks;
+ srcThread->getTracks(tracks, activeTracks, stream);
+ if (tracks.size()) {
+ dstThread->putTracks(tracks, activeTracks);
+ }
+ }
+ }
+
+ dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream);
+
+ return NO_ERROR;
+}
+
+// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
+{
+ PlaybackThread *thread = NULL;
+ if (mPlaybackThreads.indexOfKey(output) >= 0) {
+ thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
+ }
+ return thread;
+}
+
+// checkMixerThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
+{
+ PlaybackThread *thread = checkPlaybackThread_l(output);
+ if (thread != NULL) {
+ if (thread->type() == PlaybackThread::DIRECT) {
+ thread = NULL;
+ }
+ }
+ return (MixerThread *)thread;
+}
+
+// checkRecordThread_l() must be called with AudioFlinger::mLock held
+AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
+{
+ RecordThread *thread = NULL;
+ if (mRecordThreads.indexOfKey(input) >= 0) {
+ thread = (RecordThread *)mRecordThreads.valueFor(input).get();
+ }
+ return thread;
+}
+
+// ----------------------------------------------------------------------------
+
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -2533,6 +3711,7 @@
}
// ----------------------------------------------------------------------------
+
void AudioFlinger::instantiate() {
defaultServiceManager()->addService(
String16("media.audio_flinger"), new AudioFlinger());