resolved conflicts for merge of 566be7c3 to master
Change-Id: I7b1cc71057b2bd4f771e7bcf508a8c3abd6017ce
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 7c9c3ec..21bd2f4 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1112,7 +1112,7 @@
return ret;
}
-status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames,
+status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const
{
status_t status;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 00063c7..459d2ec 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -186,7 +186,7 @@
virtual status_t setVoiceVolume(float volume);
- virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames,
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const;
virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index f2af9b2..67d83b1 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -398,15 +398,16 @@
ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
track_t& track = mState.tracks[name];
- int valueInt = (int)value;
- int32_t *valueBuf = (int32_t *)value;
+ int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+ int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
switch (target) {
case TRACK:
switch (param) {
case CHANNEL_MASK: {
- audio_channel_mask_t mask = (audio_channel_mask_t) value;
+ audio_channel_mask_t mask =
+ static_cast<audio_channel_mask_t>(reinterpret_cast<uintptr_t>(value));
if (track.channelMask != mask) {
uint32_t channelCount = popcount(mask);
ALOG_ASSERT((channelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX) && channelCount);
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index a7d22e1..29b56db 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -926,8 +926,8 @@
}
result.append("\t\tSession Status State Engine:\n");
- snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n",
- mSessionId, mStatus, mState, (uint32_t)mEffectInterface);
+ snprintf(buffer, SIZE, "\t\t%05d %03d %03d %p\n",
+ mSessionId, mStatus, mState, mEffectInterface);
result.append(buffer);
result.append("\t\tDescriptor:\n");
@@ -957,20 +957,20 @@
result.append(buffer);
result.append("\t\t- Input configuration:\n");
- result.append("\t\t\tBuffer Frames SRate Channels Format\n");
- snprintf(buffer, SIZE, "\t\t\t0x%08x %5d %5d %08x %6x (%s)\n",
- (uint32_t)mConfig.inputCfg.buffer.raw,
+ result.append("\t\t\tFrames Smp rate Channels Format Buffer\n");
+ snprintf(buffer, SIZE, "\t\t\t%05zu %05d %08x %6d (%s) %p\n",
mConfig.inputCfg.buffer.frameCount,
mConfig.inputCfg.samplingRate,
mConfig.inputCfg.channels,
mConfig.inputCfg.format,
- formatToString((audio_format_t)mConfig.inputCfg.format));
+ formatToString((audio_format_t)mConfig.inputCfg.format),
+ mConfig.inputCfg.buffer.raw);
result.append(buffer);
result.append("\t\t- Output configuration:\n");
- result.append("\t\t\tBuffer Frames SRate Channels Format\n");
- snprintf(buffer, SIZE, "\t\t\t0x%08x %5d %5d %08x %6x (%s)\n",
- (uint32_t)mConfig.outputCfg.buffer.raw,
+ result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
+ snprintf(buffer, SIZE, "\t\t\t%p %05zu %05d %08x %d (%s)\n",
+ mConfig.outputCfg.buffer.raw,
mConfig.outputCfg.buffer.frameCount,
mConfig.outputCfg.samplingRate,
mConfig.outputCfg.channels,
@@ -978,7 +978,7 @@
formatToString((audio_format_t)mConfig.outputCfg.format));
result.append(buffer);
- snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size());
+ snprintf(buffer, SIZE, "\t\t%zu Clients:\n", mHandles.size());
result.append(buffer);
result.append("\t\t\t Pid Priority Ctrl Locked client server\n");
for (size_t i = 0; i < mHandles.size(); ++i) {
@@ -1689,9 +1689,9 @@
}
result.append("\tIn buffer Out buffer Active tracks:\n");
- snprintf(buffer, SIZE, "\t0x%08x 0x%08x %d\n",
- (uint32_t)mInBuffer,
- (uint32_t)mOutBuffer,
+ snprintf(buffer, SIZE, "\t%p %p %d\n",
+ mInBuffer,
+ mOutBuffer,
mActiveTrackCnt);
result.append(buffer);
write(fd, result.string(), result.size());
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 03f95e2..90122e0 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -321,7 +321,7 @@
(void *) mixBuffer);
// newly allocated track names default to full scale volume
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
- (void *) fastTrack->mChannelMask);
+ (void *)(uintptr_t)fastTrack->mChannelMask);
mixer->enable(name);
}
generations[i] = fastTrack->mGeneration;
@@ -351,7 +351,7 @@
mixer->setParameter(name, AudioMixer::RESAMPLE,
AudioMixer::REMOVE, NULL);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
- (void *) fastTrack->mChannelMask);
+ (void *)(uintptr_t) fastTrack->mChannelMask);
// already enabled
}
generations[i] = fastTrack->mGeneration;
@@ -400,9 +400,9 @@
if (fastTrack->mVolumeProvider != NULL) {
uint32_t vlr = fastTrack->mVolumeProvider->getVolumeLR();
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0,
- (void *)(vlr & 0xFFFF));
+ (void *)(uintptr_t)(vlr & 0xFFFF));
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
- (void *)(vlr >> 16));
+ (void *)(uintptr_t)(vlr >> 16));
}
// FIXME The current implementation of framesReady() for fast tracks
// takes a tryLock, which can block
@@ -732,7 +732,7 @@
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=%u measuredWarmup=%.3g ms, warmupCycles=%u\n"
+ " sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
" mixPeriod=%.2f ms\n",
string, mWriteSequence, mFramesWritten,
mNumTracks, mWriteErrors, mUnderruns, mOverruns,
@@ -852,7 +852,7 @@
mostRecent = "?";
break;
}
- fdprintf(fd, " %5u %6s %4u %7u %5u %7s %5u\n", i, isActive ? "yes" : "no",
+ fdprintf(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/StateQueue.cpp b/services/audioflinger/StateQueue.cpp
index c2d3bbd..48399c0 100644
--- a/services/audioflinger/StateQueue.cpp
+++ b/services/audioflinger/StateQueue.cpp
@@ -58,7 +58,11 @@
template<typename T> const T* StateQueue<T>::poll()
{
+#ifdef __LP64__
+ const T *next = (const T *) android_atomic_acquire_load64((volatile int64_t *) &mNext);
+#else
const T *next = (const T *) android_atomic_acquire_load((volatile int32_t *) &mNext);
+#endif
if (next != mCurrent) {
mAck = next; // no additional barrier needed
mCurrent = next;
@@ -140,7 +144,11 @@
}
// publish
+#ifdef __LP64__
+ android_atomic_release_store64((int64_t) mMutating, (volatile int64_t *) &mNext);
+#else
android_atomic_release_store((int32_t) mMutating, (volatile int32_t *) &mNext);
+#endif
mExpecting = mMutating;
// copy with circular wraparound
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index fc98867..b064e89 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -487,19 +487,19 @@
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: %d\n", mFrameCount);
+ 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,
channelMaskToString(mChannelMask, mType != RECORD).string());
fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
- fdprintf(fd, " Frame size: %u\n", mFrameSize);
+ fdprintf(fd, " Frame size: %zu\n", mFrameSize);
fdprintf(fd, " Pending setParameters commands:");
size_t numParams = mNewParameters.size();
if (numParams) {
fdprintf(fd, "\n Index Command");
for (size_t i = 0; i < numParams; ++i) {
- fdprintf(fd, "\n %02d ", i);
+ fdprintf(fd, "\n %02zu ", i);
fdprintf(fd, mNewParameters[i]);
}
fdprintf(fd, "\n");
@@ -530,7 +530,7 @@
String8 result;
size_t numEffectChains = mEffectChains.size();
- snprintf(buffer, SIZE, " %d Effect Chains\n", numEffectChains);
+ snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < numEffectChains; ++i) {
@@ -1203,7 +1203,7 @@
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
fdprintf(fd, "\nOutput thread %p:\n", this);
- fdprintf(fd, " Normal frame count: %d\n", mNormalFrameCount);
+ 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);
@@ -1776,7 +1776,7 @@
}
-status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames)
+status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
{
if (halFrames == NULL || dspFrames == NULL) {
return BAD_VALUE;
@@ -1794,7 +1794,11 @@
*dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0;
return NO_ERROR;
} else {
- return mOutput->stream->get_render_position(mOutput->stream, dspFrames);
+ status_t status;
+ uint32_t frames;
+ status = mOutput->stream->get_render_position(mOutput->stream, &frames);
+ *dspFrames = (size_t)frames;
+ return status;
}
}
@@ -3209,9 +3213,9 @@
mAudioMixer->setBufferProvider(name, track);
mAudioMixer->enable(name);
- mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
- mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
- mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr);
+ mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
@@ -3219,7 +3223,7 @@
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
- AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
+ AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
// limit track sample rate to 2 x output sample rate, which changes at re-configuration
uint32_t maxSampleRate = mSampleRate * 2;
uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate();
@@ -3232,7 +3236,7 @@
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
- (void *)reqSampleRate);
+ (void *)(uintptr_t)reqSampleRate);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
@@ -5147,8 +5151,8 @@
fdprintf(fd, "\nInput thread %p:\n", this);
if (mActiveTracks.size() > 0) {
- fdprintf(fd, " In index: %d\n", mRsmpInIndex);
- fdprintf(fd, " Buffer size: %u bytes\n", mBufferSize);
+ fdprintf(fd, " In index: %zu\n", mRsmpInIndex);
+ fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize);
fdprintf(fd, " Resampling: %d\n", (mResampler != NULL));
fdprintf(fd, " Out channel count: %u\n", mReqChannelCount);
fdprintf(fd, " Out sample rate: %u\n", mReqSampleRate);
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index f5b5efd..999fea3 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -449,7 +449,7 @@
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
- status_t getRenderPosition(size_t *halFrames, size_t *dspFrames);
+ status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
int16_t *mixBuffer() const { return mMixBuffer; };
virtual void detachAuxEffect_l(int effectId);
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index faaa6da..e5152b8 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -502,8 +502,8 @@
nowInUnderrun = '?';
break;
}
- snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g "
- "%08X %08X %08X 0x%03X %9u%c\n",
+ snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g "
+ "%08X %p %p 0x%03X %9u%c\n",
active ? "yes" : "no",
(mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
@@ -517,8 +517,8 @@
20.0 * log10((vlr & 0xFFFF) / 4096.0),
20.0 * log10((vlr >> 16) / 4096.0),
mCblk->mServer,
- (int)mMainBuffer,
- (int)mAuxBuffer,
+ mMainBuffer,
+ mAuxBuffer,
mCblk->mFlags,
mAudioTrackServerProxy->getUnderrunFrames(),
nowInUnderrun);
@@ -1873,7 +1873,7 @@
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active)
{
- snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6u\n",
+ snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu\n",
active ? "yes" : "no",
(mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,