Merge "Fix build warnings" into nyc-dev (311fab52) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioFlinger.cpp

+7 −7

Original line number	Diff line number	Diff line
		@@ -1302,7 +1302,7 @@ void AudioFlinger::removeNotificationClient(pid_t pid)
		bool removed = false;
		for (size_t i = 0; i< num; ) {
		AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
		ALOGV(" pid %d @ %d", ref->mPid, i);
		ALOGV(" pid %d @ %zu", ref->mPid, i);
		if (ref->mPid == pid) {
		ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
		mAudioSessionRefs.removeAt(i);
		@@ -1563,7 +1563,7 @@ audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)

		int rc = load_audio_interface(name, &dev);
		if (rc) {
		ALOGI("loadHwModule() error %d loading module %s ", rc, name);
		ALOGE("loadHwModule() error %d loading module %s", rc, name);
		return 0;
		}

		@@ -1571,7 +1571,7 @@ audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
		rc = dev->init_check(dev);
		mHardwareStatus = AUDIO_HW_IDLE;
		if (rc) {
		ALOGI("loadHwModule() init check error %d for module %s ", rc, name);
		ALOGE("loadHwModule() init check error %d for module %s", rc, name);
		return 0;
		}

		@@ -2857,7 +2857,7 @@ status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain
		{
		audio_session_t session = chain->sessionId();
		ssize_t index = mOrphanEffectChains.indexOfKey(session);
		ALOGV("putOrphanEffectChain_l session %d index %d", session, index);
		ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
		if (index >= 0) {
		ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
		return ALREADY_EXISTS;
		@@ -2870,7 +2870,7 @@ sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session
		{
		sp<EffectChain> chain;
		ssize_t index = mOrphanEffectChains.indexOfKey(session);
		ALOGV("getOrphanEffectChain_l session %d index %d", session, index);
		ALOGV("getOrphanEffectChain_l session %d index %zd", session, index);
		if (index >= 0) {
		chain = mOrphanEffectChains.valueAt(index);
		mOrphanEffectChains.removeItemsAt(index);
		@@ -2883,11 +2883,11 @@ bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>
		Mutex::Autolock _l(mLock);
		audio_session_t session = effect->sessionId();
		ssize_t index = mOrphanEffectChains.indexOfKey(session);
		ALOGV("updateOrphanEffectChains session %d index %d", session, index);
		ALOGV("updateOrphanEffectChains session %d index %zd", session, index);
		if (index >= 0) {
		sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
		if (chain->removeEffect_l(effect) == 0) {
		ALOGV("updateOrphanEffectChains removing effect chain at index %d", index);
		ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index);
		mOrphanEffectChains.removeItemsAt(index);
		}
		return true;

services/audioflinger/Effects.cpp

+7 −7

Original line number	Diff line number	Diff line
		@@ -138,7 +138,7 @@ status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle)
		} else {
		status = ALREADY_EXISTS;
		}
		ALOGV("addHandle() %p added handle %p in position %d", this, handle, i);
		ALOGV("addHandle() %p added handle %p in position %zu", this, handle, i);
		mHandles.insertAt(handle, i);
		return status;
		}
		@@ -156,7 +156,7 @@ size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle)
		if (i == size) {
		return size;
		}
		ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i);
		ALOGV("removeHandle() %p removed handle %p in position %zu", this, handle, i);

		mHandles.removeAt(i);
		// if removed from first place, move effect control from this handle to next in line
		@@ -380,7 +380,7 @@ status_t AudioFlinger::EffectModule::configure()
		mConfig.inputCfg.buffer.frameCount = thread->frameCount();
		mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;

		ALOGV("configure() %p thread %p buffer %p framecount %d",
		ALOGV("configure() %p thread %p buffer %p framecount %zu",
		this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);

		status_t cmdStatus;
		@@ -1050,7 +1050,7 @@ AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
		mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset);
		if (mCblkMemory == 0 \|\|
		(mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer())) == NULL) {
		ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE +
		ALOGE("not enough memory for Effect size=%zu", EFFECT_PARAM_BUFFER_SIZE +
		sizeof(effect_param_cblk_t));
		mCblkMemory.clear();
		return;
		@@ -1579,7 +1579,7 @@ status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect)
		}
		mEffects.insertAt(effect, idx_insert);

		ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this,
		ALOGV("addEffect_l() effect %p, added in chain %p at rank %zu", effect.get(), this,
		idx_insert);
		}
		effect->configure();
		@@ -1611,7 +1611,7 @@ size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
		}
		}
		mEffects.removeAt(i);
		ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(),
		ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %zu", effect.get(),
		this, i);
		break;
		}
		@@ -1726,7 +1726,7 @@ void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
		String8 result;

		size_t numEffects = mEffects.size();
		snprintf(buffer, SIZE, " %d effects for session %d\n", numEffects, mSessionId);
		snprintf(buffer, SIZE, " %zu effects for session %d\n", numEffects, mSessionId);
		result.append(buffer);

		if (numEffects) {

services/audioflinger/FastMixer.cpp

+2 −1

Original line number	Diff line number	Diff line
		@@ -433,7 +433,8 @@ void FastMixer::onWork()
		}

		if (mMasterMono.load()) { // memory_order_seq_cst
		mono_blend(mMixerBuffer, mMixerBufferFormat, Format_channelCount(mFormat), frameCount, true /limit/);
		mono_blend(mMixerBuffer, mMixerBufferFormat, Format_channelCount(mFormat), frameCount,
		true /limit/);
		}
		// prepare the buffer used to write to sink
		void *buffer = mSinkBuffer != NULL ? mSinkBuffer : mMixerBuffer;

services/audioflinger/PatchPanel.cpp

+1 −1

Original line number	Diff line number	Diff line
		@@ -401,7 +401,7 @@ status_t AudioFlinger::PatchPanel::createPatchConnections(Patch *patch,
		shift = playbackShift;
		}
		size_t frameCount = (playbackFrameCount * recordFramecount) >> shift;
		ALOGV("createPatchConnections() playframeCount %d recordFramecount %d frameCount %d ",
		ALOGV("createPatchConnections() playframeCount %zu recordFramecount %zu frameCount %zu",
		playbackFrameCount, recordFramecount, frameCount);

		// create a special record track to capture from record thread

services/audioflinger/Threads.cpp

+24 −23

Original line number	Diff line number	Diff line
		@@ -725,7 +725,7 @@ status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
		return status;
		}
		mConfigEvents.add(event);
		ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType);
		ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType);
		mWaitWorkCV.signal();
		mLock.unlock();
		{
		@@ -817,7 +817,7 @@ void AudioFlinger::ThreadBase::processConfigEvents_l()
		bool configChanged = false;

		while (!mConfigEvents.isEmpty()) {
		ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size());
		ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size());
		sp<ConfigEvent> event = mConfigEvents[0];
		mConfigEvents.removeAt(0);
		switch (event->mType) {
		@@ -949,7 +949,7 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __u
		dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
		dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
		dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
		dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
		dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize);
		dprintf(fd, " Channel count: %u\n", mChannelCount);
		dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
		channelMaskToString(mChannelMask, mType != RECORD).string());
		@@ -1685,10 +1685,10 @@ void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& ar

		size_t numtracks = mTracks.size();
		size_t numactive = mActiveTracks.size();
		dprintf(fd, " %d Tracks", numtracks);
		dprintf(fd, " %zu Tracks", numtracks);
		size_t numactiveseen = 0;
		if (numtracks) {
		dprintf(fd, " of which %d are active\n", numactive);
		dprintf(fd, " of which %zu are active\n", numactive);
		Track::appendDumpHeader(result);
		for (size_t i = 0; i < numtracks; ++i) {
		sp<Track> track = mTracks[i];
		@@ -1729,7 +1729,8 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>&
		dumpBase(fd, args);

		dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
		dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
		dprintf(fd, " Last write occurred (msecs): %llu\n",
		(unsigned long long) 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");
		@@ -1823,11 +1824,11 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
		}
		frameCount = mFrameCount * sFastTrackMultiplier;
		}
		ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d",
		ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
		frameCount, mFrameCount);
		} else {
		ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%d "
		"mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
		ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu "
		"mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x "
		"sampleRate=%u mSampleRate=%u "
		"hasFastMixer=%d tid=%d fastTrackAvailMask=%#x",
		sharedBuffer.get(), frameCount, mFrameCount, format, mFormat,
		@@ -2259,7 +2260,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
		mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
		mFrameCount = mBufferSize / mFrameSize;
		if (mFrameCount & 15) {
		ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
		ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
		mFrameCount);
		}

		@@ -2345,7 +2346,7 @@ void AudioFlinger::PlaybackThread::readOutputParameters_l()
		if (mType == MIXER \|\| mType == DUPLICATING) {
		mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
		}
		ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
		ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
		mNormalFrameCount);

		// Check if we want to throttle the processing to no more than 2x normal rate
		@@ -2682,7 +2683,7 @@ void AudioFlinger::PlaybackThread::cacheParameters_l()

		void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
		{
		ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
		ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
		this, streamType, mTracks.size());
		Mutex::Autolock _l(mLock);

		@@ -3126,7 +3127,7 @@ bool AudioFlinger::PlaybackThread::threadLoop()
		if ((now - lastWarning) > kWarningThrottleNs) {
		ATRACE_NAME("underrun");
		ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
		ns2ms(delta), mNumDelayedWrites, this);
		(unsigned long long) ns2ms(delta), mNumDelayedWrites, this);
		lastWarning = now;
		}
		}
		@@ -3471,8 +3472,8 @@ AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, Aud
		// mNormalSink below
		{
		ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
		ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, "
		"mFrameCount=%d, mNormalFrameCount=%d",
		ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, "
		"mFrameCount=%zu, mNormalFrameCount=%zu",
		mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
		mNormalFrameCount);
		mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
		@@ -4965,7 +4966,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
		// For compressed offload, use faster sleep time when underruning until more than an
		// entire buffer was written to the audio HAL
		if (!audio_has_proportional_frames(mFormat) &&
		(mType == OFFLOAD) && (mBytesWritten < mBufferSize)) {
		(mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) {
		mSleepTimeUs = kDirectMinSleepTimeUs;
		} else {
		mSleepTimeUs = mActiveSleepTimeUs;
		@@ -5276,7 +5277,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTr
		bool doHwPause = false;
		bool doHwResume = false;

		ALOGV("OffloadThread::prepareTracks_l active tracks %d", count);
		ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count);

		// find out which tracks need to be processed
		for (size_t i = 0; i < count; i++) {
		@@ -6129,7 +6130,7 @@ reacquire_wakelock:
		// ALOGD("%s", mTimestamp.toString().c_str());

		if (framesRead < 0 \|\| (framesRead == 0 && mPipeSource == 0)) {
		ALOGE("read failed: framesRead=%d", framesRead);
		ALOGE("read failed: framesRead=%zd", framesRead);
		// Force input into standby so that it tries to recover at next read attempt
		inputStandBy();
		sleepUs = kRecordThreadSleepUs;
		@@ -6365,10 +6366,10 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe
		// there are sufficient fast track slots available
		mFastTrackAvail
		) {
		ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u",
		ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
		frameCount, mFrameCount);
		} else {
		ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u "
		ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "
		"format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u "
		"hasFastCapture=%d tid=%d mFastTrackAvail=%d",
		frameCount, mFrameCount, mPipeFramesP2,
		@@ -6661,9 +6662,9 @@ void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args
		size_t numtracks = mTracks.size();
		size_t numactive = mActiveTracks.size();
		size_t numactiveseen = 0;
		dprintf(fd, " %d Tracks", numtracks);
		dprintf(fd, " %zu Tracks", numtracks);
		if (numtracks) {
		dprintf(fd, " of which %d are active\n", numactive);
		dprintf(fd, " of which %zu are active\n", numactive);
		RecordTrack::appendDumpHeader(result);
		for (size_t i = 0; i < numtracks ; ++i) {
		sp<RecordTrack> track = mTracks[i];
		@@ -7327,7 +7328,7 @@ size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& ch
		{
		ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
		ALOGW_IF(mEffectChains.size() != 1,
		"removeEffectChain_l() %p invalid chain size %d on thread %p",
		"removeEffectChain_l() %p invalid chain size %zu on thread %p",
		chain.get(), mEffectChains.size(), this);
		if (mEffectChains.size() == 1) {
		mEffectChains.removeAt(0);