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-rw-r--r--services/audioflinger/AudioFlinger.cpp33
-rw-r--r--services/audioflinger/AudioFlinger.h3
-rw-r--r--services/audioflinger/AudioMixer.h4
-rw-r--r--services/audioflinger/AudioWatchdog.cpp2
-rw-r--r--services/audioflinger/Effects.h1
-rw-r--r--services/audioflinger/FastMixer.cpp75
-rw-r--r--services/audioflinger/FastMixer.h8
-rw-r--r--services/audioflinger/FastMixerState.h6
-rw-r--r--services/audioflinger/FastThread.cpp7
-rw-r--r--services/audioflinger/PlaybackTracks.h2
-rw-r--r--services/audioflinger/StateQueue.cpp4
-rw-r--r--services/audioflinger/Threads.cpp127
-rw-r--r--services/audioflinger/Threads.h2
-rw-r--r--services/audioflinger/TrackBase.h9
-rw-r--r--services/audioflinger/Tracks.cpp56
15 files changed, 193 insertions, 146 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 45e17f8..9bd0e9b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -143,7 +143,7 @@ static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
if (rc) {
goto out;
}
- if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) {
+ if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
rc = BAD_VALUE;
goto out;
@@ -427,7 +427,7 @@ status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
if (mLogMemoryDealer != 0) {
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
if (binder != 0) {
- fdprintf(fd, "\nmedia.log:\n");
+ dprintf(fd, "\nmedia.log:\n");
Vector<String16> args;
binder->dump(fd, args);
}
@@ -635,8 +635,12 @@ sp<IAudioTrack> AudioFlinger::createTrack(
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the Track so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
- Mutex::Autolock _cl(mClientLock);
- client.clear();
+ // Don't hold mClientLock when releasing the reference on the track as the
+ // destructor will acquire it.
+ {
+ Mutex::Autolock _cl(mClientLock);
+ client.clear();
+ }
track.clear();
goto Exit;
}
@@ -1173,7 +1177,7 @@ void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
}
// mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
- // ThreadBase mutex and teh locknig order is ThreadBase::mLock then AudioFlinger::mClientLock.
+ // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
if (clientAdded) {
// the config change is always sent from playback or record threads to avoid deadlock
// with AudioSystem::gLock
@@ -1419,8 +1423,12 @@ sp<IAudioRecord> AudioFlinger::openRecord(
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the RecordTrack so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
- Mutex::Autolock _cl(mClientLock);
- client.clear();
+ // Don't hold mClientLock when releasing the reference on the track as the
+ // destructor will acquire it.
+ {
+ Mutex::Autolock _cl(mClientLock);
+ client.clear();
+ }
recordTrack.clear();
goto Exit;
}
@@ -2380,6 +2388,11 @@ sp<IEffect> AudioFlinger::createEffect(
if (handle != 0 && id != NULL) {
*id = handle->id();
}
+ if (handle == 0) {
+ // remove local strong reference to Client with mClientLock held
+ Mutex::Autolock _cl(mClientLock);
+ client.clear();
+ }
}
Exit:
@@ -2590,7 +2603,7 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
}
} else {
if (fd >= 0) {
- fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
+ dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno));
}
}
char teeTime[16];
@@ -2644,11 +2657,11 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
write(teeFd, &temp, sizeof(temp));
close(teeFd);
if (fd >= 0) {
- fdprintf(fd, "tee copied to %s\n", teePath);
+ dprintf(fd, "tee copied to %s\n", teePath);
}
} else {
if (fd >= 0) {
- fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
+ dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
}
}
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index d69d6a2..d2ded9a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -82,9 +82,6 @@ class ServerProxy;
static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
-#define MAX_GAIN 4096.0f
-#define MAX_GAIN_INT 0x1000
-
#define INCLUDING_FROM_AUDIOFLINGER_H
class AudioFlinger :
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index eca9848..573ba96 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -30,6 +30,9 @@
#include <system/audio.h>
#include <media/nbaio/NBLog.h>
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixer::UNITY_GAIN
+
namespace android {
// ----------------------------------------------------------------------------
@@ -91,6 +94,7 @@ public:
REMOVE = 0x4102, // Remove the sample rate converter on this track name;
// the track is restored to the mix sample rate.
// for target RAMP_VOLUME and VOLUME (8 channels max)
+ // FIXME use float for these 3 to improve the dynamic range
VOLUME0 = 0x4200,
VOLUME1 = 0x4201,
AUXLEVEL = 0x4210,
diff --git a/services/audioflinger/AudioWatchdog.cpp b/services/audioflinger/AudioWatchdog.cpp
index 93d185e..877e776 100644
--- a/services/audioflinger/AudioWatchdog.cpp
+++ b/services/audioflinger/AudioWatchdog.cpp
@@ -34,7 +34,7 @@ void AudioWatchdogDump::dump(int fd)
} else {
strcpy(buf, "N/A\n");
}
- fdprintf(fd, "Watchdog: underruns=%u, logs=%u, most recent underrun log at %s",
+ dprintf(fd, "Watchdog: underruns=%u, logs=%u, most recent underrun log at %s",
mUnderruns, mLogs, buf);
}
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index ccc4825..4170fd4 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -270,6 +270,7 @@ public:
sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
sp<EffectModule> getEffectFromId_l(int id);
sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
+ // FIXME use float to improve the dynamic range
bool setVolume_l(uint32_t *left, uint32_t *right);
void setDevice_l(audio_devices_t device);
void setMode_l(audio_mode_t mode);
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index f7cc8aa..f0ce902 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -26,7 +26,6 @@
#define ATRACE_TAG ATRACE_TAG_AUDIO
#include "Configuration.h"
-#include <sys/atomics.h>
#include <time.h>
#include <utils/Log.h>
#include <utils/Trace.h>
@@ -256,9 +255,9 @@ void FastMixer::onStateChange()
mixer->setBufferProvider(name, bufferProvider);
if (fastTrack->mVolumeProvider == NULL) {
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
- (void *)0x1000);
+ (void *) MAX_GAIN_INT);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
- (void *)0x1000);
+ (void *) MAX_GAIN_INT);
}
mixer->setParameter(name, AudioMixer::RESAMPLE,
AudioMixer::REMOVE, NULL);
@@ -313,11 +312,13 @@ void FastMixer::onWork()
int name = fastTrackNames[i];
ALOG_ASSERT(name >= 0);
if (fastTrack->mVolumeProvider != NULL) {
- uint32_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
+ gain_minifloat_packed_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
- (void *)(uintptr_t)(vlr & 0xFFFF));
+ (void *) (uintptr_t)
+ (float_from_gain(gain_minifloat_unpack_left(vlr)) * MAX_GAIN_INT));
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
- (void *)(uintptr_t)(vlr >> 16));
+ (void *) (uintptr_t)
+ (float_from_gain(gain_minifloat_unpack_right(vlr)) * MAX_GAIN_INT));
}
// FIXME The current implementation of framesReady() for fast tracks
// takes a tryLock, which can block
@@ -464,7 +465,7 @@ static int compare_uint32_t(const void *pa, const void *pb)
void FastMixerDumpState::dump(int fd) const
{
if (mCommand == FastMixerState::INITIAL) {
- fdprintf(fd, " FastMixer not initialized\n");
+ dprintf(fd, " FastMixer not initialized\n");
return;
}
#define COMMAND_MAX 32
@@ -498,10 +499,10 @@ void FastMixerDumpState::dump(int fd) const
double measuredWarmupMs = (mMeasuredWarmupTs.tv_sec * 1000.0) +
(mMeasuredWarmupTs.tv_nsec / 1000000.0);
double mixPeriodSec = (double) mFrameCount / (double) mSampleRate;
- fdprintf(fd, " FastMixer command=%s writeSequence=%u framesWritten=%u\n"
- " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
- " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
- " mixPeriod=%.2f ms\n",
+ dprintf(fd, " FastMixer command=%s writeSequence=%u framesWritten=%u\n"
+ " numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
+ " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
+ " mixPeriod=%.2f ms\n",
string, mWriteSequence, mFramesWritten,
mNumTracks, mWriteErrors, mUnderruns, mOverruns,
mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
@@ -553,26 +554,26 @@ void FastMixerDumpState::dump(int fd) const
#endif
}
if (n) {
- fdprintf(fd, " Simple moving statistics over last %.1f seconds:\n",
- wall.n() * mixPeriodSec);
- fdprintf(fd, " wall clock time in ms per mix cycle:\n"
- " mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
- wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
- wall.stddev()*1e-6);
- fdprintf(fd, " raw CPU load in us per mix cycle:\n"
- " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
- loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
- loadNs.stddev()*1e-3);
+ dprintf(fd, " Simple moving statistics over last %.1f seconds:\n",
+ wall.n() * mixPeriodSec);
+ dprintf(fd, " wall clock time in ms per mix cycle:\n"
+ " mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+ wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
+ wall.stddev()*1e-6);
+ dprintf(fd, " raw CPU load in us per mix cycle:\n"
+ " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+ loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
+ loadNs.stddev()*1e-3);
} else {
- fdprintf(fd, " No FastMixer statistics available currently\n");
+ dprintf(fd, " No FastMixer statistics available currently\n");
}
#ifdef CPU_FREQUENCY_STATISTICS
- fdprintf(fd, " CPU clock frequency in MHz:\n"
- " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
- kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
- fdprintf(fd, " adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
- " mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
- loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
+ dprintf(fd, " CPU clock frequency in MHz:\n"
+ " mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
+ kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
+ dprintf(fd, " adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
+ " mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
+ loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
#endif
if (tail != NULL) {
qsort(tail, n, sizeof(uint32_t), compare_uint32_t);
@@ -583,12 +584,12 @@ void FastMixerDumpState::dump(int fd) const
left.sample(tail[i]);
right.sample(tail[n - (i + 1)]);
}
- fdprintf(fd, " Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
- " left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
- " right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
- left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
- right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
- right.stddev()*1e-6);
+ dprintf(fd, " Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
+ " left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
+ " right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
+ left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
+ right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
+ right.stddev()*1e-6);
delete[] tail;
}
#endif
@@ -598,9 +599,9 @@ void FastMixerDumpState::dump(int fd) const
// Instead we always display all tracks, with an indication
// of whether we think the track is active.
uint32_t trackMask = mTrackMask;
- fdprintf(fd, " Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
+ dprintf(fd, " Fast tracks: kMaxFastTracks=%u activeMask=%#x\n",
FastMixerState::kMaxFastTracks, trackMask);
- fdprintf(fd, " Index Active Full Partial Empty Recent Ready\n");
+ dprintf(fd, " Index Active Full Partial Empty Recent Ready\n");
for (uint32_t i = 0; i < FastMixerState::kMaxFastTracks; ++i, trackMask >>= 1) {
bool isActive = trackMask & 1;
const FastTrackDump *ftDump = &mTracks[i];
@@ -620,7 +621,7 @@ void FastMixerDumpState::dump(int fd) const
mostRecent = "?";
break;
}
- fdprintf(fd, " %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
+ dprintf(fd, " %5u %6s %4u %7u %5u %7s %5zu\n", i, isActive ? "yes" : "no",
(underruns.mBitFields.mFull) & UNDERRUN_MASK,
(underruns.mBitFields.mPartial) & UNDERRUN_MASK,
(underruns.mBitFields.mEmpty) & UNDERRUN_MASK,
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index 981c1a7..db89ef4 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -17,13 +17,11 @@
#ifndef ANDROID_AUDIO_FAST_MIXER_H
#define ANDROID_AUDIO_FAST_MIXER_H
+#include <linux/futex.h>
+#include <sys/syscall.h>
#include <utils/Debug.h>
-#if 1 // FIXME move to where used
-extern "C" {
-#include "../private/bionic_futex.h"
-}
-#endif
#include "FastThread.h"
+#include <utils/Thread.h>
#include "StateQueue.h"
#include "FastMixerState.h"
#include "FastMixerDumpState.h"
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index cb54ff1..661c9ca 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -17,6 +17,7 @@
#ifndef ANDROID_AUDIO_FAST_MIXER_STATE_H
#define ANDROID_AUDIO_FAST_MIXER_STATE_H
+#include <audio_utils/minifloat.h>
#include <system/audio.h>
#include <media/ExtendedAudioBufferProvider.h>
#include <media/nbaio/NBAIO.h>
@@ -29,9 +30,8 @@ struct FastMixerDumpState;
class VolumeProvider {
public:
- // Return the track volume in U4_12 format: left in lower half, right in upper half. The
- // provider implementation is responsible for validating that the return value is in range.
- virtual uint32_t getVolumeLR() = 0;
+ // The provider implementation is responsible for validating that the return value is in range.
+ virtual gain_minifloat_packed_t getVolumeLR() = 0;
protected:
VolumeProvider() { }
virtual ~VolumeProvider() { }
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index 8a216b3..216dace 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -20,10 +20,9 @@
#define ATRACE_TAG ATRACE_TAG_AUDIO
#include "Configuration.h"
+#include <linux/futex.h>
+#include <sys/syscall.h>
#include <utils/Log.h>
-extern "C" {
-#include "../private/bionic_futex.h"
-}
#include <utils/Trace.h>
#include "FastThread.h"
@@ -157,7 +156,7 @@ bool FastThread::threadLoop()
ALOG_ASSERT(coldFutexAddr != NULL);
int32_t old = android_atomic_dec(coldFutexAddr);
if (old <= 0) {
- __futex_syscall4(coldFutexAddr, FUTEX_WAIT_PRIVATE, old - 1, NULL);
+ syscall(__NR_futex, coldFutexAddr, FUTEX_WAIT_PRIVATE, old - 1, NULL);
}
int policy = sched_getscheduler(0);
if (!(policy == SCHED_FIFO || policy == SCHED_RR)) {
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 08b1728..6f1f293 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -65,7 +65,7 @@ public:
void signal();
// implement FastMixerState::VolumeProvider interface
- virtual uint32_t getVolumeLR();
+ virtual gain_minifloat_packed_t getVolumeLR();
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
diff --git a/services/audioflinger/StateQueue.cpp b/services/audioflinger/StateQueue.cpp
index 48399c0..7e01c9f 100644
--- a/services/audioflinger/StateQueue.cpp
+++ b/services/audioflinger/StateQueue.cpp
@@ -28,12 +28,12 @@ namespace android {
#ifdef STATE_QUEUE_DUMP
void StateQueueObserverDump::dump(int fd)
{
- fdprintf(fd, "State queue observer: stateChanges=%u\n", mStateChanges);
+ dprintf(fd, "State queue observer: stateChanges=%u\n", mStateChanges);
}
void StateQueueMutatorDump::dump(int fd)
{
- fdprintf(fd, "State queue mutator: pushDirty=%u pushAck=%u blockedSequence=%u\n",
+ dprintf(fd, "State queue mutator: pushDirty=%u pushAck=%u blockedSequence=%u\n",
mPushDirty, mPushAck, mBlockedSequence);
}
#endif
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 0a18433..7843387 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -35,6 +35,7 @@
#include <audio_effects/effect_aec.h>
#include <audio_utils/primitives.h>
#include <audio_utils/format.h>
+#include <audio_utils/minifloat.h>
// NBAIO implementations
#include <media/nbaio/AudioStreamOutSink.h>
@@ -504,30 +505,30 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u
bool locked = AudioFlinger::dumpTryLock(mLock);
if (!locked) {
- fdprintf(fd, "thread %p maybe dead locked\n", this);
+ dprintf(fd, "thread %p maybe dead locked\n", this);
}
- fdprintf(fd, " I/O handle: %d\n", mId);
- fdprintf(fd, " TID: %d\n", getTid());
- fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
- fdprintf(fd, " Sample rate: %u\n", mSampleRate);
- fdprintf(fd, " HAL frame count: %zu\n", mFrameCount);
- fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
- fdprintf(fd, " Channel Count: %u\n", mChannelCount);
- fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask,
+ dprintf(fd, " I/O handle: %d\n", mId);
+ dprintf(fd, " TID: %d\n", getTid());
+ dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
+ dprintf(fd, " Sample rate: %u\n", mSampleRate);
+ dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
+ dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
+ dprintf(fd, " Channel Count: %u\n", mChannelCount);
+ dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask,
channelMaskToString(mChannelMask, mType != RECORD).string());
- fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
- fdprintf(fd, " Frame size: %zu\n", mFrameSize);
- fdprintf(fd, " Pending config events:");
+ dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+ dprintf(fd, " Frame size: %zu\n", mFrameSize);
+ dprintf(fd, " Pending config events:");
size_t numConfig = mConfigEvents.size();
if (numConfig) {
for (size_t i = 0; i < numConfig; i++) {
mConfigEvents[i]->dump(buffer, SIZE);
- fdprintf(fd, "\n %s", buffer);
+ dprintf(fd, "\n %s", buffer);
}
- fdprintf(fd, "\n");
+ dprintf(fd, "\n");
} else {
- fdprintf(fd, " none\n");
+ dprintf(fd, " none\n");
}
if (locked) {
@@ -1190,15 +1191,15 @@ void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& ar
// These values are "raw"; they will wrap around. See prepareTracks_l() for a better way.
FastTrackUnderruns underruns = getFastTrackUnderruns(0);
- fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
+ dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n",
underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty);
size_t numtracks = mTracks.size();
size_t numactive = mActiveTracks.size();
- fdprintf(fd, " %d Tracks", numtracks);
+ dprintf(fd, " %d Tracks", numtracks);
size_t numactiveseen = 0;
if (numtracks) {
- fdprintf(fd, " of which %d are active\n", numactive);
+ dprintf(fd, " of which %d are active\n", numactive);
Track::appendDumpHeader(result);
for (size_t i = 0; i < numtracks; ++i) {
sp<Track> track = mTracks[i];
@@ -1230,22 +1231,21 @@ void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& ar
}
write(fd, result.string(), result.size());
-
}
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
- fdprintf(fd, "\nOutput thread %p:\n", this);
- fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
- fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
- fdprintf(fd, " Total writes: %d\n", mNumWrites);
- fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
- fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
- fdprintf(fd, " Suspend count: %d\n", mSuspended);
- fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
- fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
- fdprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
- fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
+ dprintf(fd, "\nOutput thread %p:\n", this);
+ dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
+ dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+ dprintf(fd, " Total writes: %d\n", mNumWrites);
+ dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites);
+ dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no");
+ dprintf(fd, " Suspend count: %d\n", mSuspended);
+ dprintf(fd, " Sink buffer : %p\n", mSinkBuffer);
+ dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
+ dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
+ dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
dumpBase(fd, args);
}
@@ -2753,7 +2753,7 @@ AudioFlinger::MixerThread::~MixerThread()
if (state->mCommand == FastMixerState::COLD_IDLE) {
int32_t old = android_atomic_inc(&mFastMixerFutex);
if (old == -1) {
- (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+ (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
}
}
state->mCommand = FastMixerState::EXIT;
@@ -2810,7 +2810,7 @@ ssize_t AudioFlinger::MixerThread::threadLoop_write()
if (state->mCommand == FastMixerState::COLD_IDLE) {
int32_t old = android_atomic_inc(&mFastMixerFutex);
if (old == -1) {
- (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
+ (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1);
}
#ifdef AUDIO_WATCHDOG
if (mAudioWatchdog != 0) {
@@ -3258,21 +3258,23 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
- uint32_t vlr = proxy->getVolumeLR();
- vl = vlr & 0xFFFF;
- vr = vlr >> 16;
+ gain_minifloat_packed_t vlr = proxy->getVolumeLR();
+ float vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
+ float vrf = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
- if (vl > MAX_GAIN_INT) {
- ALOGV("Track left volume out of range: %04X", vl);
- vl = MAX_GAIN_INT;
+ if (vlf > GAIN_FLOAT_UNITY) {
+ ALOGV("Track left volume out of range: %.3g", vlf);
+ vlf = GAIN_FLOAT_UNITY;
}
- if (vr > MAX_GAIN_INT) {
- ALOGV("Track right volume out of range: %04X", vr);
- vr = MAX_GAIN_INT;
+ if (vrf > GAIN_FLOAT_UNITY) {
+ ALOGV("Track right volume out of range: %.3g", vrf);
+ vrf = GAIN_FLOAT_UNITY;
}
// now apply the master volume and stream type volume
- vl = (uint32_t)(v * vl) << 12;
- vr = (uint32_t)(v * vr) << 12;
+ // FIXME we're losing the wonderful dynamic range in the minifloat representation
+ float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT);
+ vl = (uint32_t) (v8_24 * vlf);
+ vr = (uint32_t) (v8_24 * vrf);
// assuming master volume and stream type volume each go up to 1.0,
// vl and vr are now in 8.24 format
@@ -3299,6 +3301,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
track->mHasVolumeController = false;
}
+ // FIXME Use float
// Convert volumes from 8.24 to 4.12 format
// This additional clamping is needed in case chain->setVolume_l() overshot
vl = (vl + (1 << 11)) >> 12;
@@ -3674,7 +3677,7 @@ void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& ar
PlaybackThread::dumpInternals(fd, args);
- fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
+ dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames());
// Make a non-atomic copy of fast mixer dump state so it won't change underneath us
const FastMixerDumpState copy(mFastMixerDumpState);
@@ -3755,13 +3758,17 @@ void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTr
float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy;
- uint32_t vlr = proxy->getVolumeLR();
- float v_clamped = v * (vlr & 0xFFFF);
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- left = v_clamped/MAX_GAIN;
- v_clamped = v * (vlr >> 16);
- if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
- right = v_clamped/MAX_GAIN;
+ gain_minifloat_packed_t vlr = proxy->getVolumeLR();
+ left = float_from_gain(gain_minifloat_unpack_left(vlr));
+ if (left > GAIN_FLOAT_UNITY) {
+ left = GAIN_FLOAT_UNITY;
+ }
+ left *= v;
+ right = float_from_gain(gain_minifloat_unpack_right(vlr));
+ if (right > GAIN_FLOAT_UNITY) {
+ right = GAIN_FLOAT_UNITY;
+ }
+ right *= v;
}
if (lastTrack) {
@@ -4157,7 +4164,10 @@ void AudioFlinger::OffloadThread::threadLoop_exit()
mMixerStatus = MIXER_DRAIN_ALL;
threadLoop_drain();
}
- mCallbackThread->exit();
+ if (mUseAsyncWrite) {
+ ALOG_ASSERT(mCallbackThread != 0);
+ mCallbackThread->exit();
+ }
PlaybackThread::threadLoop_exit();
}
@@ -5122,6 +5132,7 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
// to be at least 2 x the record thread frame count and cover audio hardware latency.
// This is probably too conservative, but legacy application code may depend on it.
// If you change this calculation, also review the start threshold which is related.
+ // FIXME It's not clear how input latency actually matters. Perhaps this should be 0.
uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream);
size_t mNormalFrameCount = 2048; // FIXME
uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
@@ -5354,12 +5365,12 @@ void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
{
- fdprintf(fd, "\nInput thread %p:\n", this);
+ dprintf(fd, "\nInput thread %p:\n", this);
if (mActiveTracks.size() > 0) {
- fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
+ dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
} else {
- fdprintf(fd, " No active record clients\n");
+ dprintf(fd, " No active record clients\n");
}
dumpBase(fd, args);
@@ -5374,9 +5385,9 @@ void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args
size_t numtracks = mTracks.size();
size_t numactive = mActiveTracks.size();
size_t numactiveseen = 0;
- fdprintf(fd, " %d Tracks", numtracks);
+ dprintf(fd, " %d Tracks", numtracks);
if (numtracks) {
- fdprintf(fd, " of which %d are active\n", numactive);
+ dprintf(fd, " of which %d are active\n", numactive);
RecordTrack::appendDumpHeader(result);
for (size_t i = 0; i < numtracks ; ++i) {
sp<RecordTrack> track = mTracks[i];
@@ -5390,7 +5401,7 @@ void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args
}
}
} else {
- fdprintf(fd, "\n");
+ dprintf(fd, "\n");
}
if (numactiveseen != numactive) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 1b01512..44008e5 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -301,6 +301,8 @@ public:
// If a thread does not have such a heap, this method returns 0.
virtual sp<MemoryDealer> readOnlyHeap() const { return 0; }
+ virtual sp<IMemory> pipeMemory() const { return 0; }
+
mutable Mutex mLock;
protected:
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 5f13be3..4cba3fd 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -39,6 +39,13 @@ public:
STARTING_2, // for RecordTrack only
};
+ // where to allocate the data buffer
+ enum alloc_type {
+ ALLOC_CBLK, // allocate immediately after control block
+ ALLOC_READONLY, // allocate from a separate read-only heap per thread
+ ALLOC_PIPE, // do not allocate; use the pipe buffer
+ };
+
TrackBase(ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
@@ -50,7 +57,7 @@ public:
int uid,
IAudioFlinger::track_flags_t flags,
bool isOut,
- bool useReadOnlyHeap = false);
+ alloc_type alloc = ALLOC_CBLK);
virtual ~TrackBase();
virtual status_t initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 222ec53..7ddc71c 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -21,6 +21,7 @@
#include "Configuration.h"
#include <math.h>
+#include <sys/syscall.h>
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
@@ -34,6 +35,7 @@
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
+#include <audio_utils/minifloat.h>
// ----------------------------------------------------------------------------
@@ -71,7 +73,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
int clientUid,
IAudioFlinger::track_flags_t flags,
bool isOut,
- bool useReadOnlyHeap)
+ alloc_type alloc)
: RefBase(),
mThread(thread),
mClient(client),
@@ -115,7 +117,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
size_t size = sizeof(audio_track_cblk_t);
size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
- if (sharedBuffer == 0 && !useReadOnlyHeap) {
+ if (sharedBuffer == 0 && alloc == ALLOC_CBLK) {
size += bufferSize;
}
@@ -137,7 +139,8 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
// construct the shared structure in-place.
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
- if (useReadOnlyHeap) {
+ switch (alloc) {
+ case ALLOC_READONLY: {
const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
if (roHeap == 0 ||
(mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
@@ -151,7 +154,17 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
return;
}
memset(mBuffer, 0, bufferSize);
- } else {
+ } break;
+ case ALLOC_PIPE:
+ mBufferMemory = thread->pipeMemory();
+ // mBuffer is the virtual address as seen from current process (mediaserver),
+ // and should normally be coming from mBufferMemory->pointer().
+ // However in this case the TrackBase does not reference the buffer directly.
+ // It should references the buffer via the pipe.
+ // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
+ mBuffer = NULL;
+ break;
+ case ALLOC_CBLK:
// clear all buffers
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
@@ -162,6 +175,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic
#endif
}
+ break;
}
#ifdef TEE_SINK
@@ -461,7 +475,7 @@ void AudioFlinger::PlaybackThread::Track::destroy()
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active)
{
- uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
+ gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
if (isFastTrack()) {
sprintf(buffer, " F %2d", mFastIndex);
} else if (mName >= AudioMixer::TRACK0) {
@@ -534,8 +548,8 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool a
stateChar,
mFillingUpStatus,
mAudioTrackServerProxy->getSampleRate(),
- 20.0 * log10((vlr & 0xFFFF) / 4096.0),
- 20.0 * log10((vlr >> 16) / 4096.0),
+ 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
+ 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
mCblk->mServer,
mMainBuffer,
mAuxBuffer,
@@ -961,27 +975,27 @@ void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_
// implement VolumeBufferProvider interface
-uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
+gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR()
{
// called by FastMixer, so not allowed to take any locks, block, or do I/O including logs
ALOG_ASSERT(isFastTrack() && (mCblk != NULL));
- uint32_t vlr = mAudioTrackServerProxy->getVolumeLR();
- uint32_t vl = vlr & 0xFFFF;
- uint32_t vr = vlr >> 16;
+ gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR();
+ float vl = float_from_gain(gain_minifloat_unpack_left(vlr));
+ float vr = float_from_gain(gain_minifloat_unpack_right(vlr));
// track volumes come from shared memory, so can't be trusted and must be clamped
- if (vl > MAX_GAIN_INT) {
- vl = MAX_GAIN_INT;
+ if (vl > GAIN_FLOAT_UNITY) {
+ vl = GAIN_FLOAT_UNITY;
}
- if (vr > MAX_GAIN_INT) {
- vr = MAX_GAIN_INT;
+ if (vr > GAIN_FLOAT_UNITY) {
+ vr = GAIN_FLOAT_UNITY;
}
// now apply the cached master volume and stream type volume;
// this is trusted but lacks any synchronization or barrier so may be stale
float v = mCachedVolume;
vl *= v;
vr *= v;
- // re-combine into U4.16
- vlr = (vr << 16) | (vl & 0xFFFF);
+ // re-combine into packed minifloat
+ vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr));
// FIXME look at mute, pause, and stop flags
return vlr;
}
@@ -1007,7 +1021,7 @@ void AudioFlinger::PlaybackThread::Track::invalidate()
android_atomic_or(CBLK_INVALID, &cblk->mFlags);
android_atomic_release_store(0x40000000, &cblk->mFutex);
// client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
- (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
+ (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
mIsInvalid = true;
}
@@ -1592,7 +1606,7 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
// since client and server are in the same process,
// the buffer has the same virtual address on both sides
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize);
- mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000));
+ mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
mClientProxy->setSendLevel(0.0);
mClientProxy->setSampleRate(sampleRate);
mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize,
@@ -1840,7 +1854,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
: TrackBase(thread, client, sampleRate, format,
channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid,
flags, false /*isOut*/,
- (flags & IAudioFlinger::TRACK_FAST) != 0 /*useReadOnlyHeap*/),
+ (flags & IAudioFlinger::TRACK_FAST) != 0 ? ALLOC_READONLY : ALLOC_CBLK),
mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0),
// See real initialization of mRsmpInFront at RecordThread::start()
mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL)
@@ -1936,7 +1950,7 @@ void AudioFlinger::RecordThread::RecordTrack::invalidate()
android_atomic_or(CBLK_INVALID, &cblk->mFlags);
android_atomic_release_store(0x40000000, &cblk->mFutex);
// client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE
- (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX);
+ (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX);
}
generated by cgit v1.1 at 2024-05-28 19:05:34 +0000