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-rw-r--r--services/audioflinger/AudioFlinger.cpp4
-rw-r--r--services/audioflinger/AudioFlinger.h4
-rw-r--r--services/audioflinger/AudioMixer.cpp7
-rw-r--r--services/audioflinger/AudioResampler.cpp4
-rw-r--r--services/audioflinger/Configuration.h3
-rw-r--r--services/audioflinger/Effects.cpp24
-rw-r--r--services/audioflinger/FastMixer.cpp39
-rw-r--r--services/audioflinger/FastMixerState.cpp2
-rw-r--r--services/audioflinger/FastMixerState.h1
-rw-r--r--services/audioflinger/StateQueue.cpp8
-rw-r--r--services/audioflinger/Threads.cpp46
-rw-r--r--services/audioflinger/Threads.h2
-rw-r--r--services/audioflinger/Tracks.cpp10
13 files changed, 67 insertions, 87 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index acbd19a..26dac95 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1012,7 +1012,7 @@ size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t form
return size;
}
-unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
+uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
@@ -1044,7 +1044,7 @@ status_t AudioFlinger::setVoiceVolume(float value)
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 53e238e..7320144 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -186,10 +186,10 @@ public:
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 unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
+ virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const;
virtual int newAudioSessionId();
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 07dc6dd..f92421e 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -421,15 +421,16 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
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/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 2c3c719..e5cceb1 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -526,7 +526,7 @@ void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t
" ldr r8, [sp, #" MO_PARAM5 " + 4]\n" // out
" ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
" ldr r0, [r0]\n" // outputIndex
- " add r8, r0, asl #2\n" // curOut
+ " add r8, r8, r0, asl #2\n" // curOut
" ldr r9, [sp, #" MO_PARAM5 " + 24]\n" // phaseIncrement
" ldr r10, [sp, #" MO_PARAM5 " + 12]\n" // vl
" ldr r11, [sp, #" MO_PARAM5 " + 16]\n" // vr
@@ -636,7 +636,7 @@ void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32
" ldr r8, [sp, #" ST_PARAM5 " + 4]\n" // out
" ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex
" ldr r0, [r0]\n" // outputIndex
- " add r8, r0, asl #2\n" // curOut
+ " add r8, r8, r0, asl #2\n" // curOut
" ldr r9, [sp, #" ST_PARAM5 " + 24]\n" // phaseIncrement
" ldr r10, [sp, #" ST_PARAM5 " + 12]\n" // vl
" ldr r11, [sp, #" ST_PARAM5 " + 16]\n" // vr
diff --git a/services/audioflinger/Configuration.h b/services/audioflinger/Configuration.h
index bc2038a..0754d9d 100644
--- a/services/audioflinger/Configuration.h
+++ b/services/audioflinger/Configuration.h
@@ -32,9 +32,6 @@
// uncomment to enable fast mixer to take performance samples for later statistical analysis
#define FAST_MIXER_STATISTICS
-// uncomment to allow fast tracks at non-native sample rate
-//#define FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
-
// uncomment for debugging timing problems related to StateQueue::push()
//#define STATE_QUEUE_DUMP
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index a8a5169..010e233 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -820,8 +820,8 @@ void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
}
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");
@@ -850,26 +850,26 @@ void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
result.append(buffer);
result.append("\t\t- Input configuration:\n");
- result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
- snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\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 %p\n",
mConfig.inputCfg.buffer.frameCount,
mConfig.inputCfg.samplingRate,
mConfig.inputCfg.channels,
- mConfig.inputCfg.format);
+ mConfig.inputCfg.format,
+ mConfig.inputCfg.buffer.raw);
result.append(buffer);
result.append("\t\t- Output configuration:\n");
result.append("\t\t\tBuffer Frames Smp rate Channels Format\n");
- snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n",
- (uint32_t)mConfig.outputCfg.buffer.raw,
+ snprintf(buffer, SIZE, "\t\t\t%p %05zu %05d %08x %d\n",
+ mConfig.outputCfg.buffer.raw,
mConfig.outputCfg.buffer.frameCount,
mConfig.outputCfg.samplingRate,
mConfig.outputCfg.channels,
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\tPid Priority Ctrl Locked client server\n");
for (size_t i = 0; i < mHandles.size(); ++i) {
@@ -1578,10 +1578,10 @@ void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
}
result.append("\tNum fx In buffer Out buffer Active tracks:\n");
- snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n",
+ snprintf(buffer, SIZE, "\t%02zu %p %p %d\n",
mEffects.size(),
- (uint32_t)mInBuffer,
- (uint32_t)mOutBuffer,
+ 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 f27ea17..85d637e 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -236,7 +236,6 @@ bool FastMixer::threadLoop()
sampleRate = Format_sampleRate(format);
ALOG_ASSERT(Format_channelCount(format) == FCC_2);
}
- dumpState->mSampleRate = sampleRate;
}
if ((format != previousFormat) || (frameCount != previous->mFrameCount)) {
@@ -321,12 +320,8 @@ bool FastMixer::threadLoop()
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(void *) mixBuffer);
// newly allocated track names default to full scale volume
- if (fastTrack->mSampleRate != 0 && fastTrack->mSampleRate != sampleRate) {
- mixer->setParameter(name, AudioMixer::RESAMPLE,
- AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate);
- }
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::CHANNEL_MASK,
- (void *) fastTrack->mChannelMask);
+ (void *)(uintptr_t)fastTrack->mChannelMask);
mixer->enable(name);
}
generations[i] = fastTrack->mGeneration;
@@ -353,16 +348,10 @@ bool FastMixer::threadLoop()
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1,
(void *)0x1000);
}
- if (fastTrack->mSampleRate != 0 &&
- fastTrack->mSampleRate != sampleRate) {
- mixer->setParameter(name, AudioMixer::RESAMPLE,
- AudioMixer::SAMPLE_RATE, (void*) fastTrack->mSampleRate);
- } else {
- mixer->setParameter(name, AudioMixer::RESAMPLE,
- AudioMixer::REMOVE, NULL);
- }
+ 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;
@@ -392,16 +381,8 @@ bool FastMixer::threadLoop()
// Refresh the per-track timestamp
if (timestampStatus == NO_ERROR) {
- uint32_t trackFramesWrittenButNotPresented;
- uint32_t trackSampleRate = fastTrack->mSampleRate;
- // There is currently no sample rate conversion for fast tracks currently
- if (trackSampleRate != 0 && trackSampleRate != sampleRate) {
- trackFramesWrittenButNotPresented =
- ((int64_t) nativeFramesWrittenButNotPresented * trackSampleRate) /
- sampleRate;
- } else {
- trackFramesWrittenButNotPresented = nativeFramesWrittenButNotPresented;
- }
+ uint32_t trackFramesWrittenButNotPresented =
+ nativeFramesWrittenButNotPresented;
uint32_t trackFramesWritten = fastTrack->mBufferProvider->framesReleased();
// Can't provide an AudioTimestamp before first frame presented,
// or during the brief 32-bit wraparound window
@@ -419,9 +400,9 @@ bool FastMixer::threadLoop()
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
@@ -750,7 +731,7 @@ void FastMixerDumpState::dump(int fd) const
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,
@@ -864,7 +845,7 @@ void FastMixerDumpState::dump(int fd) const
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/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp
index 737de97..43ff233 100644
--- a/services/audioflinger/FastMixerState.cpp
+++ b/services/audioflinger/FastMixerState.cpp
@@ -20,7 +20,7 @@
namespace android {
FastTrack::FastTrack() :
- mBufferProvider(NULL), mVolumeProvider(NULL), mSampleRate(0),
+ mBufferProvider(NULL), mVolumeProvider(NULL),
mChannelMask(AUDIO_CHANNEL_OUT_STEREO), mGeneration(0)
{
}
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index f6e7903..9739fe9 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -43,7 +43,6 @@ struct FastTrack {
ExtendedAudioBufferProvider* mBufferProvider; // must be NULL if inactive, or non-NULL if active
VolumeProvider* mVolumeProvider; // optional; if NULL then full-scale
- unsigned mSampleRate; // optional; if zero then use mixer sample rate
audio_channel_mask_t mChannelMask; // AUDIO_CHANNEL_OUT_MONO or AUDIO_CHANNEL_OUT_STEREO
int mGeneration; // increment when any field is assigned
};
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> StateQueue<T>::~StateQueue()
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 @@ template<typename T> bool StateQueue<T>::push(StateQueue<T>::block_t block)
}
// 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 14629de..cac785a 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -425,7 +425,7 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate);
result.append(buffer);
- snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
+ snprintf(buffer, SIZE, "HAL frame count: %zu\n", mFrameCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);
result.append(buffer);
@@ -433,14 +433,14 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
- snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
+ snprintf(buffer, SIZE, "Frame size: %zu\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);
+ snprintf(buffer, SIZE, "\n %02zu ", i);
result.append(buffer);
result.append(mNewParameters[i]);
}
@@ -466,7 +466,7 @@ void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>&
char buffer[SIZE];
String8 result;
- snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size());
+ snprintf(buffer, SIZE, "\n- %zu Effect Chains:\n", mEffectChains.size());
write(fd, buffer, strlen(buffer));
for (size_t i = 0; i < mEffectChains.size(); ++i) {
@@ -1128,7 +1128,7 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>&
snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
- snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
+ snprintf(buffer, SIZE, "Normal frame count: %zu\n", mNormalFrameCount);
result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
ns2ms(systemTime() - mLastWriteTime));
@@ -1218,10 +1218,8 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
// mono or stereo
( (channelMask == AUDIO_CHANNEL_OUT_MONO) ||
(channelMask == AUDIO_CHANNEL_OUT_STEREO) ) &&
-#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
// hardware sample rate
(sampleRate == mSampleRate) &&
-#endif
// normal mixer has an associated fast mixer
hasFastMixer() &&
// there are sufficient fast track slots available
@@ -1720,7 +1718,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters()
}
-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;
@@ -1738,7 +1736,11 @@ status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size
*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;
}
}
@@ -2975,7 +2977,6 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
VolumeProvider *vp = track;
fastTrack->mBufferProvider = eabp;
fastTrack->mVolumeProvider = vp;
- fastTrack->mSampleRate = track->mSampleRate;
fastTrack->mChannelMask = track->mChannelMask;
fastTrack->mGeneration++;
state->mTrackMask |= 1 << j;
@@ -3038,15 +3039,8 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
(mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) {
minFrames = desiredFrames;
}
- // It's not safe to call framesReady() for a static buffer track, so assume it's ready
- size_t framesReady;
- if (track->sharedBuffer() == 0) {
- framesReady = track->framesReady();
- } else if (track->isStopped()) {
- framesReady = 0;
- } else {
- framesReady = 1;
- }
+
+ size_t framesReady = track->framesReady();
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
@@ -3159,9 +3153,9 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
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,
@@ -3169,7 +3163,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
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();
@@ -3182,7 +3176,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
- (void *)reqSampleRate);
+ (void *)(uintptr_t)reqSampleRate);
mAudioMixer->setParameter(
name,
AudioMixer::TRACK,
@@ -4966,9 +4960,9 @@ void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& a
result.append(buffer);
if (mActiveTrack != 0) {
- snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex);
+ snprintf(buffer, SIZE, "In index: %zu\n", mRsmpInIndex);
result.append(buffer);
- snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize);
+ snprintf(buffer, SIZE, "Buffer size: %zu bytes\n", mBufferSize);
result.append(buffer);
snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
result.append(buffer);
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 207f1eb..a2fb874 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -446,7 +446,7 @@ public:
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 d6b9908..813ec8a 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -487,8 +487,8 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
nowInUnderrun = '?';
break;
}
- snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g "
- "%08X %08X %08X 0x%03X %9u%c\n",
+ snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g "
+ "%08X %p %p 0x%03X %9u%c\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
@@ -501,8 +501,8 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
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);
@@ -1850,7 +1850,7 @@ void AudioFlinger::RecordThread::RecordTrack::invalidate()
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
- snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n",
+ snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6zu\n",
(mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
generated by cgit v1.1 at 2024-06-11 15:22:46 +0000