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-rw-r--r--services/audioflinger/AudioFlinger.cpp34
-rw-r--r--services/audioflinger/AudioFlinger.h13
-rw-r--r--services/audioflinger/AudioMixer.cpp111
-rw-r--r--services/audioflinger/Threads.cpp190
-rw-r--r--services/audioflinger/Threads.h31
5 files changed, 260 insertions, 119 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 93b1642..52fce34 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -181,7 +181,8 @@ AudioFlinger::AudioFlinger()
mIsLowRamDevice(true),
mIsDeviceTypeKnown(false),
mGlobalEffectEnableTime(0),
- mPrimaryOutputSampleRate(0)
+ mPrimaryOutputSampleRate(0),
+ mSystemReady(false)
{
getpid_cached = getpid();
char value[PROPERTY_VALUE_MAX];
@@ -1722,6 +1723,26 @@ audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId
return (audio_hw_sync_t)value;
}
+status_t AudioFlinger::systemReady()
+{
+ Mutex::Autolock _l(mLock);
+ ALOGI("%s", __FUNCTION__);
+ if (mSystemReady) {
+ ALOGW("%s called twice", __FUNCTION__);
+ return NO_ERROR;
+ }
+ mSystemReady = true;
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+ ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
+ thread->systemReady();
+ }
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
+ thread->systemReady();
+ }
+ return NO_ERROR;
+}
+
// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
{
@@ -1794,15 +1815,15 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_
PlaybackThread *thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
- thread = new OffloadThread(this, outputStream, *output, devices);
+ thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
- thread = new DirectOutputThread(this, outputStream, *output, devices);
+ thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
} else {
- thread = new MixerThread(this, outputStream, *output, devices);
+ thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
}
mPlaybackThreads.add(*output, thread);
@@ -1873,7 +1894,7 @@ audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
}
audio_io_handle_t id = nextUniqueId();
- DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
+ DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
@@ -2120,7 +2141,8 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m
inputStream,
*input,
primaryOutputDevice_l(),
- devices
+ devices,
+ mSystemReady
#ifdef TEE_SINK
, teeSink
#endif
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 51b2610..d087ced 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -257,6 +257,9 @@ public:
/* Get the HW synchronization source used for an audio session */
virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId);
+ /* Indicate JAVA services are ready (scheduling, power management ...) */
+ virtual status_t systemReady();
+
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
@@ -356,6 +359,15 @@ private:
// check that channelMask is the "canonical" one we expect for the channelCount.
return channelMask == audio_channel_out_mask_from_count(channelCount);
}
+ case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+ if (kEnableExtendedChannels) {
+ const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
+ if (channelCount >= FCC_2 // mono is not supported at this time
+ && channelCount <= AudioMixer::MAX_NUM_CHANNELS) {
+ return true;
+ }
+ }
+ return false;
default:
return false;
}
@@ -752,6 +764,7 @@ private:
uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none
// protected by mHardwareLock
+ bool mSystemReady;
};
#undef INCLUDING_FROM_AUDIOFLINGER_H
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 586c737..01efc53 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -66,6 +66,13 @@
#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
#endif
+// TODO: Move these macro/inlines to a header file.
+template <typename T>
+static inline
+T max(const T& x, const T& y) {
+ return x > y ? x : y;
+}
+
// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
// original code will be used for stereo sinks, the new mixer for multichannel.
static const bool kUseNewMixer = true;
@@ -499,41 +506,99 @@ void AudioMixer::disable(int name)
static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
+ // check floating point volume to see if it is identical to the previously
+ // set volume.
+ // We do not use a tolerance here (and reject changes too small)
+ // as it may be confusing to use a different value than the one set.
+ // If the resulting volume is too small to ramp, it is a direct set of the volume.
if (newVolume == *pSetVolume) {
return false;
}
- /* set the floating point volume variables */
- if (ramp != 0) {
- *pVolumeInc = (newVolume - *pSetVolume) / ramp;
- *pPrevVolume = *pSetVolume;
+ if (newVolume < 0) {
+ newVolume = 0; // should not have negative volumes
} else {
- *pVolumeInc = 0;
- *pPrevVolume = newVolume;
+ switch (fpclassify(newVolume)) {
+ case FP_SUBNORMAL:
+ case FP_NAN:
+ newVolume = 0;
+ break;
+ case FP_ZERO:
+ break; // zero volume is fine
+ case FP_INFINITE:
+ // Infinite volume could be handled consistently since
+ // floating point math saturates at infinities,
+ // but we limit volume to unity gain float.
+ // ramp = 0; break;
+ //
+ newVolume = AudioMixer::UNITY_GAIN_FLOAT;
+ break;
+ case FP_NORMAL:
+ default:
+ // Floating point does not have problems with overflow wrap
+ // that integer has. However, we limit the volume to
+ // unity gain here.
+ // TODO: Revisit the volume limitation and perhaps parameterize.
+ if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
+ newVolume = AudioMixer::UNITY_GAIN_FLOAT;
+ }
+ break;
+ }
+ }
+
+ // set floating point volume ramp
+ if (ramp != 0) {
+ // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
+ // is no computational mismatch; hence equality is checked here.
+ ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
+ " prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
+ const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
+ const float maxv = max(newVolume, *pPrevVolume); // could be inf, cannot be nan, subnormal
+
+ if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
+ && maxv + inc != maxv) { // inc must make forward progress
+ *pVolumeInc = inc;
+ // ramp is set now.
+ // Note: if newVolume is 0, then near the end of the ramp,
+ // it may be possible that the ramped volume may be subnormal or
+ // temporarily negative by a small amount or subnormal due to floating
+ // point inaccuracies.
+ } else {
+ ramp = 0; // ramp not allowed
+ }
}
- *pSetVolume = newVolume;
- /* set the legacy integer volume variables */
- int32_t intVolume = newVolume * AudioMixer::UNITY_GAIN_INT;
- if (intVolume > AudioMixer::UNITY_GAIN_INT) {
- intVolume = AudioMixer::UNITY_GAIN_INT;
- } else if (intVolume < 0) {
- ALOGE("negative volume %.7g", newVolume);
- intVolume = 0; // should never happen, but for safety check.
+ // compute and check integer volume, no need to check negative values
+ // The integer volume is limited to "unity_gain" to avoid wrapping and other
+ // audio artifacts, so it never reaches the range limit of U4.28.
+ // We safely use signed 16 and 32 bit integers here.
+ const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
+ const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
+ AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
+
+ // set integer volume ramp
+ if (ramp != 0) {
+ // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
+ // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
+ // is no computational mismatch; hence equality is checked here.
+ ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
+ " prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
+ const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
+
+ if (inc != 0) { // inc must make forward progress
+ *pIntVolumeInc = inc;
+ } else {
+ ramp = 0; // ramp not allowed
+ }
}
- if (intVolume == *pIntSetVolume) {
- *pIntVolumeInc = 0;
- /* TODO: integer/float workaround: ignore floating volume ramp */
+
+ // if no ramp, or ramp not allowed, then clear float and integer increments
+ if (ramp == 0) {
*pVolumeInc = 0;
*pPrevVolume = newVolume;
- return true;
- }
- if (ramp != 0) {
- *pIntVolumeInc = ((intVolume - *pIntSetVolume) << 16) / ramp;
- *pIntPrevVolume = (*pIntVolumeInc == 0 ? intVolume : *pIntSetVolume) << 16;
- } else {
*pIntVolumeInc = 0;
*pIntPrevVolume = intVolume << 16;
}
+ *pSetVolume = newVolume;
*pIntSetVolume = intVolume;
return true;
}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d9b4249..721f6af 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -487,7 +487,7 @@ const char *sourceToString(audio_source_t source)
}
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
+ audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady)
: Thread(false /*canCallJava*/),
mType(type),
mAudioFlinger(audioFlinger),
@@ -498,7 +498,8 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio
mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
// mName will be set by concrete (non-virtual) subclass
- mDeathRecipient(new PMDeathRecipient(this))
+ mDeathRecipient(new PMDeathRecipient(this)),
+ mSystemReady(systemReady)
{
memset(&mPatch, 0, sizeof(struct audio_patch));
}
@@ -567,6 +568,11 @@ status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
{
status_t status = NO_ERROR;
+ if (event->mRequiresSystemReady && !mSystemReady) {
+ event->mWaitStatus = false;
+ mPendingConfigEvents.add(event);
+ return status;
+ }
mConfigEvents.add(event);
ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType);
mWaitWorkCV.signal();
@@ -598,6 +604,12 @@ void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event)
sendConfigEvent_l(configEvent);
}
+void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio)
+{
+ Mutex::Autolock _l(mLock);
+ sendPrioConfigEvent_l(pid, tid, prio);
+}
+
// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio)
{
@@ -697,49 +709,62 @@ void AudioFlinger::ThreadBase::processConfigEvents_l()
String8 channelMaskToString(audio_channel_mask_t mask, bool output) {
String8 s;
- if (output) {
- if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
- if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
- if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
- if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
- if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
- if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
- if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
- if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
- if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
- if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
- if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
- if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
- } else {
- if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
- if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
- if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
- if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
- if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
- if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
- if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
- if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
- if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
- if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
- if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
- if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
- if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
- if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
- if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
- }
- int len = s.length();
- if (s.length() > 2) {
- char *str = s.lockBuffer(len);
- s.unlockBuffer(len - 2);
- }
- return s;
+ const audio_channel_representation_t representation = audio_channel_mask_get_representation(mask);
+
+ switch (representation) {
+ case AUDIO_CHANNEL_REPRESENTATION_POSITION: {
+ if (output) {
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, ");
+ if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
+ if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
+ if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
+ if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, ");
+ } else {
+ if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
+ if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, ");
+ if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, ");
+ if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, ");
+ if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, ");
+ if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, ");
+ if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
+ if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
+ if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
+ if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
+ if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
+ if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, ");
+ }
+ const int len = s.length();
+ if (len > 2) {
+ char *str = s.lockBuffer(len); // needed?
+ s.unlockBuffer(len - 2); // remove trailing ", "
+ }
+ return s;
+ }
+ case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+ s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask));
+ return s;
+ default:
+ s.appendFormat("unknown mask, representation:%d bits:%#x",
+ representation, audio_channel_mask_get_bits(mask));
+ return s;
+ }
}
void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
@@ -880,8 +905,7 @@ void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids)
}
void AudioFlinger::ThreadBase::getPowerManager_l() {
-
- if (mPowerManager == 0) {
+ if (mSystemReady && mPowerManager == 0) {
// use checkService() to avoid blocking if power service is not up yet
sp<IBinder> binder =
defaultServiceManager()->checkService(String16("power"));
@@ -895,7 +919,6 @@ void AudioFlinger::ThreadBase::getPowerManager_l() {
}
void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
-
getPowerManager_l();
if (mWakeLockToken == NULL) {
ALOGE("no wake lock to update!");
@@ -1337,6 +1360,20 @@ void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *conf
AUDIO_PORT_CONFIG_FORMAT;
}
+void AudioFlinger::ThreadBase::systemReady()
+{
+ Mutex::Autolock _l(mLock);
+ if (mSystemReady) {
+ return;
+ }
+ mSystemReady = true;
+
+ for (size_t i = 0; i < mPendingConfigEvents.size(); i++) {
+ sendConfigEvent_l(mPendingConfigEvents.editItemAt(i));
+ }
+ mPendingConfigEvents.clear();
+}
+
// ----------------------------------------------------------------------------
// Playback
@@ -1346,8 +1383,9 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
AudioStreamOut* output,
audio_io_handle_t id,
audio_devices_t device,
- type_t type)
- : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
+ type_t type,
+ bool systemReady)
+ : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady),
mNormalFrameCount(0), mSinkBuffer(NULL),
mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
mMixerBuffer(NULL),
@@ -1572,10 +1610,12 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
) &&
// PCM data
audio_is_linear_pcm(format) &&
- // identical channel mask to sink, or mono in and stereo sink
+ // TODO: extract as a data library function that checks that a computationally
+ // expensive downmixer is not required: isFastOutputChannelConversion()
(channelMask == mChannelMask ||
- (channelMask == AUDIO_CHANNEL_OUT_MONO &&
- mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) &&
+ mChannelMask != AUDIO_CHANNEL_OUT_STEREO ||
+ (channelMask == AUDIO_CHANNEL_OUT_MONO
+ /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) &&
// hardware sample rate
(sampleRate == mSampleRate) &&
// normal mixer has an associated fast mixer
@@ -3118,8 +3158,8 @@ void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device, type_t type)
- : PlaybackThread(audioFlinger, output, id, device, type),
+ audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type)
+ : PlaybackThread(audioFlinger, output, id, device, type, systemReady),
// mAudioMixer below
// mFastMixer below
mFastMixerFutex(0)
@@ -3252,11 +3292,7 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
// start the fast mixer
mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
pid_t tid = mFastMixer->getTid();
- int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
- kPriorityFastMixer, getpid_cached, tid, err);
- }
+ sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
#ifdef AUDIO_WATCHDOG
// create and start the watchdog
@@ -3264,11 +3300,7 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
mAudioWatchdog->setDump(&mAudioWatchdogDump);
mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
tid = mAudioWatchdog->getTid();
- err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
- kPriorityFastMixer, getpid_cached, tid, err);
- }
+ sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
#endif
}
@@ -4294,16 +4326,16 @@ void AudioFlinger::MixerThread::cacheParameters_l()
// ----------------------------------------------------------------------------
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device)
- : PlaybackThread(audioFlinger, output, id, device, DIRECT)
+ AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady)
+ : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady)
// mLeftVolFloat, mRightVolFloat
{
}
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output, audio_io_handle_t id, uint32_t device,
- ThreadBase::type_t type)
- : PlaybackThread(audioFlinger, output, id, device, type)
+ ThreadBase::type_t type, bool systemReady)
+ : PlaybackThread(audioFlinger, output, id, device, type, systemReady)
// mLeftVolFloat, mRightVolFloat
{
}
@@ -4832,8 +4864,8 @@ void AudioFlinger::AsyncCallbackThread::resetDraining()
// ----------------------------------------------------------------------------
AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
- : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD),
+ AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady)
+ : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady),
mPausedBytesRemaining(0)
{
//FIXME: mStandby should be set to true by ThreadBase constructor
@@ -5110,9 +5142,9 @@ void AudioFlinger::OffloadThread::onAddNewTrack_l()
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
- AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
+ AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady)
: MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
- DUPLICATING),
+ systemReady, DUPLICATING),
mWaitTimeMs(UINT_MAX)
{
addOutputTrack(mainThread);
@@ -5292,12 +5324,13 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
audio_io_handle_t id,
audio_devices_t outDevice,
- audio_devices_t inDevice
+ audio_devices_t inDevice,
+ bool systemReady
#ifdef TEE_SINK
, const sp<NBAIO_Sink>& teeSink
#endif
) :
- ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
+ ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady),
mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL),
// mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l()
mRsmpInRear(0)
@@ -5416,12 +5449,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
// start the fast capture
mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
pid_t tid = mFastCapture->getTid();
- int err = requestPriority(getpid_cached, tid, kPriorityFastMixer);
- if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
- kPriorityFastCapture, getpid_cached, tid, err);
- }
-
+ sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer);
#ifdef AUDIO_WATCHDOG
// FIXME
#endif
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 37bacae..aa20525 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -35,7 +35,8 @@ public:
static const char *threadTypeToString(type_t type);
ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
+ audio_devices_t outDevice, audio_devices_t inDevice, type_t type,
+ bool systemReady);
virtual ~ThreadBase();
virtual status_t readyToRun();
@@ -92,10 +93,13 @@ public:
Condition mCond; // condition for status return
status_t mStatus; // status communicated to sender
bool mWaitStatus; // true if sender is waiting for status
+ bool mRequiresSystemReady; // true if must wait for system ready to enter event queue
sp<ConfigEventData> mData; // event specific parameter data
protected:
- ConfigEvent(int type) : mType(type), mStatus(NO_ERROR), mWaitStatus(false), mData(NULL) {}
+ ConfigEvent(int type, bool requiresSystemReady = false) :
+ mType(type), mStatus(NO_ERROR), mWaitStatus(false),
+ mRequiresSystemReady(requiresSystemReady), mData(NULL) {}
};
class IoConfigEventData : public ConfigEventData {
@@ -136,7 +140,7 @@ public:
class PrioConfigEvent : public ConfigEvent {
public:
PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) :
- ConfigEvent(CFG_EVENT_PRIO) {
+ ConfigEvent(CFG_EVENT_PRIO, true) {
mData = new PrioConfigEventData(pid, tid, prio);
}
virtual ~PrioConfigEvent() {}
@@ -258,6 +262,7 @@ public:
status_t sendConfigEvent_l(sp<ConfigEvent>& event);
void sendIoConfigEvent(audio_io_config_event event);
void sendIoConfigEvent_l(audio_io_config_event event);
+ void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio);
void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
status_t sendSetParameterConfigEvent_l(const String8& keyValuePair);
status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch,
@@ -359,6 +364,8 @@ public:
virtual sp<IMemory> pipeMemory() const { return 0; }
+ void systemReady();
+
mutable Mutex mLock;
protected:
@@ -418,6 +425,7 @@ protected:
size_t mBufferSize; // HAL buffer size for read() or write()
Vector< sp<ConfigEvent> > mConfigEvents;
+ Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready
// These fields are written and read by thread itself without lock or barrier,
// and read by other threads without lock or barrier via standby(), outDevice()
@@ -445,6 +453,7 @@ protected:
mSuspendedSessions;
static const size_t kLogSize = 4 * 1024;
sp<NBLog::Writer> mNBLogWriter;
+ bool mSystemReady;
};
// --- PlaybackThread ---
@@ -470,7 +479,7 @@ public:
static const int8_t kMaxTrackRetriesOffload = 20;
PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device, type_t type);
+ audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady);
virtual ~PlaybackThread();
void dump(int fd, const Vector<String16>& args);
@@ -842,6 +851,7 @@ public:
AudioStreamOut* output,
audio_io_handle_t id,
audio_devices_t device,
+ bool systemReady,
type_t type = MIXER);
virtual ~MixerThread();
@@ -903,7 +913,7 @@ class DirectOutputThread : public PlaybackThread {
public:
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device);
+ audio_io_handle_t id, audio_devices_t device, bool systemReady);
virtual ~DirectOutputThread();
// Thread virtuals
@@ -933,7 +943,8 @@ protected:
float mRightVolFloat;
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, uint32_t device, ThreadBase::type_t type);
+ audio_io_handle_t id, uint32_t device, ThreadBase::type_t type,
+ bool systemReady);
void processVolume_l(Track *track, bool lastTrack);
// prepareTracks_l() tells threadLoop_mix() the name of the single active track
@@ -946,7 +957,7 @@ class OffloadThread : public DirectOutputThread {
public:
OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, uint32_t device);
+ audio_io_handle_t id, uint32_t device, bool systemReady);
virtual ~OffloadThread() {};
virtual void flushHw_l();
@@ -1001,7 +1012,7 @@ private:
class DuplicatingThread : public MixerThread {
public:
DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
- audio_io_handle_t id);
+ audio_io_handle_t id, bool systemReady);
virtual ~DuplicatingThread();
// Thread virtuals
@@ -1177,7 +1188,8 @@ public:
AudioStreamIn *input,
audio_io_handle_t id,
audio_devices_t outDevice,
- audio_devices_t inDevice
+ audio_devices_t inDevice,
+ bool systemReady
#ifdef TEE_SINK
, const sp<NBAIO_Sink>& teeSink
#endif
@@ -1294,6 +1306,7 @@ private:
// one-time initialization, no locks required
sp<FastCapture> mFastCapture; // non-0 if there is also
// a fast capture
+
// FIXME audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
generated by cgit v1.1 at 2024-05-20 13:49:13 +0000