Merge "Remove warnings in Audio mixer resampler compilation."
diff --git a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
index e251f82..48b0afe 100644
--- a/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
+++ b/drm/libdrmframework/plugins/forward-lock/FwdLockEngine/Android.mk
@@ -61,7 +61,7 @@
$(LOCAL_PATH)/include \
external/openssl/include
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/drm
+LOCAL_MODULE_RELATIVE_PATH := drm
LOCAL_MODULE_TAGS := optional
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 45134c4..3d839fc 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -486,12 +486,11 @@
int mSessionId;
transfer_type mTransfer;
- audio_io_handle_t mInput; // returned by AudioSystem::getInput()
-
- // Next 3 fields may be changed if IAudioRecord is re-created, but always != 0
+ // Next 4 fields may be changed if IAudioRecord is re-created, but always != 0
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mInput; // returned by AudioSystem::getInput()
int mPreviousPriority; // before start()
SchedPolicy mPreviousSchedulingGroup;
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 644e55c..3a60217 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -454,7 +454,7 @@
* Returned value:
* handle on audio hardware output
*/
- audio_io_handle_t getOutput();
+ audio_io_handle_t getOutput() const;
/* Returns the unique session ID associated with this track.
*
@@ -634,20 +634,12 @@
// caller must hold lock on mLock for all _l methods
- status_t createTrack_l(audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- size_t frameCount,
- audio_output_flags_t flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- size_t epoch);
+ status_t createTrack_l(size_t epoch);
// can only be called when mState != STATE_ACTIVE
void flush_l();
void setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
- audio_io_handle_t getOutput_l();
// FIXME enum is faster than strcmp() for parameter 'from'
status_t restoreTrack_l(const char *from);
@@ -655,10 +647,11 @@
bool isOffloaded_l() const
{ return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
- // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
+ // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk; // re-load after mLock.unlock()
+ audio_io_handle_t mOutput; // returned by AudioSystem::getOutput()
sp<AudioTrackThread> mAudioTrackThread;
@@ -763,7 +756,6 @@
sp<DeathNotifier> mDeathNotifier;
uint32_t mSequence; // incremented for each new IAudioTrack attempt
- audio_io_handle_t mOutput; // cached output io handle
int mClientUid;
};
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index a794e87..4bd111a 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -64,7 +64,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
// On successful return, AudioFlinger takes over the handle
@@ -88,7 +88,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
pid_t tid, // -1 means unused, otherwise must be valid non-0
int *sessionId,
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 85862a8..b5a4c0b 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -96,11 +96,7 @@
// The value should be used "for entertainment purposes only",
// which means don't make important decisions based on it.
- size_t frameCount_; // used during creation to pass actual track buffer size
- // from AudioFlinger to client, and not referenced again
- // FIXME remove here and replace by createTrack() in/out
- // parameter
- // renamed to "_" to detect incorrect use
+ uint32_t mPad1; // unused
volatile int32_t mFutex; // event flag: down (P) by client,
// up (V) by server or binderDied() or interrupt()
diff --git a/media/libeffects/downmix/Android.mk b/media/libeffects/downmix/Android.mk
index 5d0a87c..2bb6dbe 100644
--- a/media/libeffects/downmix/Android.mk
+++ b/media/libeffects/downmix/Android.mk
@@ -13,7 +13,7 @@
LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
LOCAL_LDLIBS += -ldl
diff --git a/media/libeffects/loudness/Android.mk b/media/libeffects/loudness/Android.mk
index dcb7b27..edf964e 100644
--- a/media/libeffects/loudness/Android.mk
+++ b/media/libeffects/loudness/Android.mk
@@ -14,7 +14,7 @@
liblog \
libstlport
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_MODULE:= libldnhncr
LOCAL_C_INCLUDES := \
diff --git a/media/libeffects/lvm/wrapper/Android.mk b/media/libeffects/lvm/wrapper/Android.mk
index f1af389..68ba34c 100644
--- a/media/libeffects/lvm/wrapper/Android.mk
+++ b/media/libeffects/lvm/wrapper/Android.mk
@@ -13,7 +13,7 @@
LOCAL_MODULE:= libbundlewrapper
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_STATIC_LIBRARIES += libmusicbundle
@@ -42,7 +42,7 @@
LOCAL_MODULE:= libreverbwrapper
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_STATIC_LIBRARIES += libreverb
diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk
index c344352..9e8cb83 100644
--- a/media/libeffects/preprocessing/Android.mk
+++ b/media/libeffects/preprocessing/Android.mk
@@ -5,7 +5,7 @@
LOCAL_MODULE:= libaudiopreprocessing
LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_SRC_FILES:= \
PreProcessing.cpp
diff --git a/media/libeffects/proxy/Android.mk b/media/libeffects/proxy/Android.mk
index 01b3be1..d6d6c1e 100644
--- a/media/libeffects/proxy/Android.mk
+++ b/media/libeffects/proxy/Android.mk
@@ -15,7 +15,7 @@
LOCAL_PATH:= $(call my-dir)
include $(CLEAR_VARS)
LOCAL_MODULE:= libeffectproxy
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_MODULE_TAGS := optional
diff --git a/media/libeffects/testlibs/Android.mk_ b/media/libeffects/testlibs/Android.mk_
index 2954908..672ebba 100644
--- a/media/libeffects/testlibs/Android.mk_
+++ b/media/libeffects/testlibs/Android.mk_
@@ -11,7 +11,7 @@
LOCAL_SHARED_LIBRARIES := \
libcutils
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_MODULE:= libreverbtest
ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
@@ -47,7 +47,7 @@
LOCAL_SHARED_LIBRARIES := \
libcutils
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_MODULE:= libequalizertest
ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
diff --git a/media/libeffects/visualizer/Android.mk b/media/libeffects/visualizer/Android.mk
index e196eb2..dd2d306 100644
--- a/media/libeffects/visualizer/Android.mk
+++ b/media/libeffects/visualizer/Android.mk
@@ -13,7 +13,7 @@
liblog \
libdl
-LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/soundfx
+LOCAL_MODULE_RELATIVE_PATH := soundfx
LOCAL_MODULE:= libvisualizer
LOCAL_C_INCLUDES := \
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 6565a16..b69e3ae 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -244,7 +244,7 @@
// create the IAudioRecord
status = openRecord_l(0 /*epoch*/);
- if (status) {
+ if (status != NO_ERROR) {
return status;
}
@@ -255,9 +255,6 @@
mStatus = NO_ERROR;
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount_;
-
mActive = false;
mCbf = cbf;
mRefreshRemaining = true;
@@ -466,12 +463,17 @@
ALOGE("Could not get audio input for record source %d", mInputSource);
return BAD_VALUE;
}
+ {
+ // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+ // we must release it ourselves if anything goes wrong.
+ size_t temp = mFrameCount; // temp may be replaced by a revised value of frameCount,
+ // but we will still need the original value also
int originalSessionId = mSessionId;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
mSampleRate, mFormat,
mChannelMask,
- mFrameCount,
+ &temp,
&trackFlags,
tid,
&mSessionId,
@@ -481,9 +483,11 @@
if (record == 0 || status != NO_ERROR) {
ALOGE("AudioFlinger could not create record track, status: %d", status);
- AudioSystem::releaseInput(input);
- return status;
+ goto release;
}
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
sp<IMemory> iMem = record->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
@@ -498,11 +502,19 @@
mAudioRecord->asBinder()->unlinkToDeath(mDeathNotifier, this);
mDeathNotifier.clear();
}
+
+ // We retain a copy of the I/O handle, but don't own the reference
mInput = input;
mAudioRecord = record;
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
+ // note that temp is the (possibly revised) value of mFrameCount
+ if (temp < mFrameCount || (mFrameCount == 0 && temp == 0)) {
+ ALOGW("Requested frameCount %u but received frameCount %u", mFrameCount, temp);
+ }
+ mFrameCount = temp;
+
// FIXME missing fast track frameCount logic
mAwaitBoost = false;
if (mFlags & AUDIO_INPUT_FLAG_FAST) {
@@ -535,6 +547,14 @@
mAudioRecord->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
+ }
+
+release:
+ AudioSystem::releaseInput(input);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ return status;
}
status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -550,13 +570,13 @@
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 72be5ca..cfd79bc 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -245,8 +245,6 @@
return INVALID_OPERATION;
}
- mOutput = 0;
-
// handle default values first.
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
@@ -319,17 +317,6 @@
mFrameSizeAF = sizeof(uint8_t);
}
- audio_io_handle_t output = AudioSystem::getOutput(
- streamType,
- sampleRate, format, channelMask,
- flags,
- offloadInfo);
-
- if (output == 0) {
- ALOGE("Could not get audio output for stream type %d", streamType);
- return BAD_VALUE;
- }
-
// Make copy of input parameter offloadInfo so that in the future:
// (a) createTrack_l doesn't need it as an input parameter
// (b) we can support re-creation of offloaded tracks
@@ -363,14 +350,7 @@
}
// create the IAudioTrack
- status = createTrack_l(streamType,
- sampleRate,
- format,
- frameCount,
- flags,
- sharedBuffer,
- output,
- 0 /*epoch*/);
+ status = createTrack_l(0 /*epoch*/);
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
@@ -379,9 +359,15 @@
mAudioTrackThread.clear();
}
// Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0 // FIXME This should no longer be needed
+ //Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
- AudioSystem::releaseOutput(output);
+ if (mOutput != 0) {
+ AudioSystem::releaseOutput(mOutput);
+ mOutput = 0;
+ }
+#endif
return status;
}
@@ -397,7 +383,6 @@
mSequence = 1;
mObservedSequence = mSequence;
mInUnderrun = false;
- mOutput = output;
return NO_ERROR;
}
@@ -821,23 +806,12 @@
return NO_ERROR;
}
-audio_io_handle_t AudioTrack::getOutput()
+audio_io_handle_t AudioTrack::getOutput() const
{
AutoMutex lock(mLock);
return mOutput;
}
-// must be called with mLock held
-audio_io_handle_t AudioTrack::getOutput_l()
-{
- if (mOutput) {
- return mOutput;
- } else {
- return AudioSystem::getOutput(mStreamType,
- mSampleRate, mFormat, mChannelMask, mFlags);
- }
-}
-
status_t AudioTrack::attachAuxEffect(int effectId)
{
AutoMutex lock(mLock);
@@ -851,15 +825,7 @@
// -------------------------------------------------------------------------
// must be called with mLock held
-status_t AudioTrack::createTrack_l(
- audio_stream_type_t streamType,
- uint32_t sampleRate,
- audio_format_t format,
- size_t frameCount,
- audio_output_flags_t flags,
- const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output,
- size_t epoch)
+status_t AudioTrack::createTrack_l(size_t epoch)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -868,41 +834,52 @@
return NO_INIT;
}
+ audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+ mChannelMask, mFlags, mOffloadInfo);
+ if (output == 0) {
+ ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, "
+ "channel mask %#x, flags %#x",
+ mStreamType, mSampleRate, mFormat, mChannelMask, mFlags);
+ return BAD_VALUE;
+ }
+ {
+ // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
+ // we must release it ourselves if anything goes wrong.
+
// Not all of these values are needed under all conditions, but it is easier to get them all
uint32_t afLatency;
- status = AudioSystem::getLatency(output, streamType, &afLatency);
+ status = AudioSystem::getLatency(output, mStreamType, &afLatency);
if (status != NO_ERROR) {
ALOGE("getLatency(%d) failed status %d", output, status);
- return NO_INIT;
+ goto release;
}
size_t afFrameCount;
- status = AudioSystem::getFrameCount(output, streamType, &afFrameCount);
+ status = AudioSystem::getFrameCount(output, mStreamType, &afFrameCount);
if (status != NO_ERROR) {
- ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status);
- return NO_INIT;
+ ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, mStreamType, status);
+ goto release;
}
uint32_t afSampleRate;
- status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate);
+ status = AudioSystem::getSamplingRate(output, mStreamType, &afSampleRate);
if (status != NO_ERROR) {
- ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, status);
- return NO_INIT;
+ ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, mStreamType, status);
+ goto release;
}
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
- if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
+ if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(
// either of these use cases:
// use case 1: shared buffer
- (sharedBuffer != 0) ||
+ (mSharedBuffer != 0) ||
// use case 2: callback handler
(mCbf != NULL))) {
ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client");
// once denied, do not request again if IAudioTrack is re-created
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
- mFlags = flags;
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
ALOGV("createTrack_l() output %d afLatency %d", output, afLatency);
@@ -913,43 +890,45 @@
// n = 3 normal track, with sample rate conversion
// (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering)
// n > 3 very high latency or very small notification interval; nBuffering is ignored
- const uint32_t nBuffering = (sampleRate == afSampleRate) ? 2 : 3;
+ const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3;
mNotificationFramesAct = mNotificationFramesReq;
- if (!audio_is_linear_pcm(format)) {
+ size_t frameCount = mReqFrameCount;
+ if (!audio_is_linear_pcm(mFormat)) {
- if (sharedBuffer != 0) {
+ if (mSharedBuffer != 0) {
// Same comment as below about ignoring frameCount parameter for set()
- frameCount = sharedBuffer->size();
+ frameCount = mSharedBuffer->size();
} else if (frameCount == 0) {
frameCount = afFrameCount;
}
if (mNotificationFramesAct != frameCount) {
mNotificationFramesAct = frameCount;
}
- } else if (sharedBuffer != 0) {
+ } else if (mSharedBuffer != 0) {
// Ensure that buffer alignment matches channel count
// 8-bit data in shared memory is not currently supported by AudioFlinger
- size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
+ size_t alignment = /* mFormat == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2;
if (mChannelCount > 1) {
// More than 2 channels does not require stronger alignment than stereo
alignment <<= 1;
}
- if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) {
+ if (((size_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) {
ALOGE("Invalid buffer alignment: address %p, channel count %u",
- sharedBuffer->pointer(), mChannelCount);
- return BAD_VALUE;
+ mSharedBuffer->pointer(), mChannelCount);
+ status = BAD_VALUE;
+ goto release;
}
// When initializing a shared buffer AudioTrack via constructors,
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
- frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t);
+ frameCount = mSharedBuffer->size()/mChannelCount/sizeof(int16_t);
- } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
+ } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
// FIXME move these calculations and associated checks to server
@@ -961,10 +940,10 @@
minBufCount = nBuffering;
}
- size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+ size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
", afLatency=%d",
- minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency);
+ minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
if (frameCount == 0) {
frameCount = minFrameCount;
@@ -989,26 +968,28 @@
}
pid_t tid = -1;
- if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
trackFlags |= IAudioFlinger::TRACK_FAST;
if (mAudioTrackThread != 0) {
tid = mAudioTrackThread->getTid();
}
}
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
}
- sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
- sampleRate,
+ size_t temp = frameCount; // temp may be replaced by a revised value of frameCount,
+ // but we will still need the original value also
+ sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
+ mSampleRate,
// AudioFlinger only sees 16-bit PCM
- format == AUDIO_FORMAT_PCM_8_BIT ?
- AUDIO_FORMAT_PCM_16_BIT : format,
+ mFormat == AUDIO_FORMAT_PCM_8_BIT ?
+ AUDIO_FORMAT_PCM_16_BIT : mFormat,
mChannelMask,
- frameCount,
+ &temp,
&trackFlags,
- sharedBuffer,
+ mSharedBuffer,
output,
tid,
&mSessionId,
@@ -1018,8 +999,11 @@
if (track == 0) {
ALOGE("AudioFlinger could not create track, status: %d", status);
- return status;
+ goto release;
}
+ // AudioFlinger now owns the reference to the I/O handle,
+ // so we are no longer responsible for releasing it.
+
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
@@ -1039,7 +1023,7 @@
mCblkMemory = iMem;
audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
mCblk = cblk;
- size_t temp = cblk->frameCount_;
+ // note that temp is the (possibly revised) value of frameCount
if (temp < frameCount || (frameCount == 0 && temp == 0)) {
// In current design, AudioTrack client checks and ensures frame count validity before
// passing it to AudioFlinger so AudioFlinger should not return a different value except
@@ -1048,11 +1032,11 @@
}
frameCount = temp;
mAwaitBoost = false;
- if (flags & AUDIO_OUTPUT_FLAG_FAST) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
if (trackFlags & IAudioFlinger::TRACK_FAST) {
ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
mAwaitBoost = true;
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
// Theoretically double-buffering is not required for fast tracks,
// due to tighter scheduling. But in practice, to accommodate kernels with
// scheduling jitter, and apps with computation jitter, we use double-buffering.
@@ -1063,26 +1047,27 @@
} else {
ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
// once denied, do not request again if IAudioTrack is re-created
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
- mFlags = flags;
- if (sharedBuffer == 0) {
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
+ if (mSharedBuffer == 0) {
if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) {
mNotificationFramesAct = frameCount/nBuffering;
}
}
}
}
- if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) {
ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful");
} else {
ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server");
- flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
- mFlags = flags;
- return NO_INIT;
+ mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
+ // FIXME This is a warning, not an error, so don't return error status
+ //return NO_INIT;
}
}
+ // We retain a copy of the I/O handle, but don't own the reference
+ mOutput = output;
mRefreshRemaining = true;
// Starting address of buffers in shared memory. If there is a shared buffer, buffers
@@ -1090,15 +1075,15 @@
// immediately after the control block. This address is for the mapping within client
// address space. AudioFlinger::TrackBase::mBuffer is for the server address space.
void* buffers;
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
buffers = (char*)cblk + sizeof(audio_track_cblk_t);
} else {
- buffers = sharedBuffer->pointer();
+ buffers = mSharedBuffer->pointer();
}
mAudioTrack->attachAuxEffect(mAuxEffectId);
// FIXME don't believe this lie
- mLatency = afLatency + (1000*frameCount) / sampleRate;
+ mLatency = afLatency + (1000*frameCount) / mSampleRate;
mFrameCount = frameCount;
// If IAudioTrack is re-created, don't let the requested frameCount
// decrease. This can confuse clients that cache frameCount().
@@ -1107,7 +1092,7 @@
}
// update proxy
- if (sharedBuffer == 0) {
+ if (mSharedBuffer == 0) {
mStaticProxy.clear();
mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
} else {
@@ -1125,6 +1110,14 @@
mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this);
return NO_ERROR;
+ }
+
+release:
+ AudioSystem::releaseOutput(output);
+ if (status == NO_ERROR) {
+ status = NO_INIT;
+ }
+ return status;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
@@ -1140,13 +1133,13 @@
}
const struct timespec *requested;
+ struct timespec timeout;
if (waitCount == -1) {
requested = &ClientProxy::kForever;
} else if (waitCount == 0) {
requested = &ClientProxy::kNonBlocking;
} else if (waitCount > 0) {
long long ms = WAIT_PERIOD_MS * (long long) waitCount;
- struct timespec timeout;
timeout.tv_sec = ms / 1000;
timeout.tv_nsec = (int) (ms % 1000) * 1000000;
requested = &timeout;
@@ -1706,7 +1699,7 @@
status_t result;
// refresh the audio configuration cache in this process to make sure we get new
- // output parameters in getOutput_l() and createTrack_l()
+ // output parameters in createTrack_l()
AudioSystem::clearAudioConfigCache();
if (isOffloaded_l()) {
@@ -1714,10 +1707,6 @@
return DEAD_OBJECT;
}
- // force new output query from audio policy manager;
- mOutput = 0;
- audio_io_handle_t output = getOutput_l();
-
// if the new IAudioTrack is created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory
@@ -1725,14 +1714,7 @@
// take the frames that will be lost by track recreation into account in saved position
size_t position = mProxy->getPosition() + mProxy->getFramesFilled();
size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
- result = createTrack_l(mStreamType,
- mSampleRate,
- mFormat,
- mReqFrameCount, // so that frame count never goes down
- mFlags,
- mSharedBuffer,
- output,
- position /*epoch*/);
+ result = createTrack_l(position /*epoch*/);
if (result == NO_ERROR) {
// continue playback from last known position, but
@@ -1761,9 +1743,15 @@
}
if (result != NO_ERROR) {
// Use of direct and offloaded output streams is ref counted by audio policy manager.
+#if 0 // FIXME This should no longer be needed
+ //Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
- AudioSystem::releaseOutput(output);
+ if (mOutput != 0) {
+ AudioSystem::releaseOutput(mOutput);
+ mOutput = 0;
+ }
+#endif
ALOGW("restoreTrack_l() failed status %d", result);
mState = STATE_STOPPED;
}
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 7a1e207..21018a0 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -26,7 +26,7 @@
namespace android {
audio_track_cblk_t::audio_track_cblk_t()
- : mServer(0), frameCount_(0), mFutex(0), mMinimum(0),
+ : mServer(0), mFutex(0), mMinimum(0),
mVolumeLR(0x10001000), mSampleRate(0), mSendLevel(0), mFlags(0)
{
memset(&u, 0, sizeof(u));
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 86a4d74..f3f3e15 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,7 +89,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -106,6 +106,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
+ size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
data.writeInt32(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
@@ -127,6 +128,10 @@
if (lStatus != NO_ERROR) {
ALOGE("createTrack error: %s", strerror(-lStatus));
} else {
+ frameCount = reply.readInt32();
+ if (pFrameCount != NULL) {
+ *pFrameCount = frameCount;
+ }
lFlags = reply.readInt32();
if (flags != NULL) {
*flags = lFlags;
@@ -161,7 +166,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
track_flags_t *flags,
pid_t tid,
int *sessionId,
@@ -174,6 +179,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
+ size_t frameCount = pFrameCount != NULL ? *pFrameCount : 0;
data.writeInt32(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
@@ -187,6 +193,10 @@
if (lStatus != NO_ERROR) {
ALOGE("openRecord error: %s", strerror(-lStatus));
} else {
+ frameCount = reply.readInt32();
+ if (pFrameCount != NULL) {
+ *pFrameCount = frameCount;
+ }
lFlags = reply.readInt32();
if (flags != NULL) {
*flags = lFlags;
@@ -807,10 +817,11 @@
} else {
track = createTrack(
(audio_stream_type_t) streamType, sampleRate, format,
- channelMask, frameCount, &flags, buffer, output, tid,
+ channelMask, &frameCount, &flags, buffer, output, tid,
&sessionId, name, clientUid, &status);
LOG_ALWAYS_FATAL_IF((track != 0) != (status == NO_ERROR));
}
+ reply->writeInt32(frameCount);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
reply->writeString8(name);
@@ -830,8 +841,9 @@
int sessionId = data.readInt32();
status_t status;
sp<IAudioRecord> record = openRecord(input,
- sampleRate, format, channelMask, frameCount, &flags, tid, &sessionId, &status);
+ sampleRate, format, channelMask, &frameCount, &flags, tid, &sessionId, &status);
LOG_ALWAYS_FATAL_IF((record != 0) != (status == NO_ERROR));
+ reply->writeInt32(frameCount);
reply->writeInt32(flags);
reply->writeInt32(sessionId);
reply->writeInt32(status);
diff --git a/media/libmedia/MemoryLeakTrackUtil.cpp b/media/libmedia/MemoryLeakTrackUtil.cpp
index 6a108ae..f004ca4 100644
--- a/media/libmedia/MemoryLeakTrackUtil.cpp
+++ b/media/libmedia/MemoryLeakTrackUtil.cpp
@@ -49,7 +49,7 @@
}
void append(const char *s) {
- strcat(mPtr, s);
+ strncat(mPtr, s, MAX_SIZE - size() - 1);
}
const char *string() const {
@@ -60,6 +60,10 @@
return strlen(mPtr);
}
+ void clear() {
+ *mPtr = '\0';
+ }
+
private:
char *mPtr;
@@ -139,6 +143,9 @@
}
} while (moved);
+ write(fd, result.string(), result.size());
+ result.clear();
+
for (size_t i = 0; i < count; i++) {
AllocEntry *e = &entries[i];
@@ -152,13 +159,14 @@
result.append(buffer);
}
result.append("\n");
+
+ write(fd, result.string(), result.size());
+ result.clear();
}
delete[] entries;
free_malloc_leak_info(info);
}
-
- write(fd, result.string(), result.size());
}
#else
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index dedd186..0afac69 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -318,7 +318,7 @@
void TimedEventQueue::acquireWakeLock_l()
{
- if (mWakeLockCount++ == 0) {
+ if (mWakeLockCount == 0) {
CHECK(mWakeLockToken == 0);
if (mPowerManager == 0) {
// use checkService() to avoid blocking if power service is not up yet
@@ -341,21 +341,23 @@
IPCThreadState::self()->restoreCallingIdentity(token);
if (status == NO_ERROR) {
mWakeLockToken = binder;
+ mWakeLockCount++;
}
}
+ } else {
+ mWakeLockCount++;
}
}
void TimedEventQueue::releaseWakeLock_l(bool force)
{
+ if (mWakeLockCount == 0) {
+ return;
+ }
if (force) {
- if (mWakeLockCount == 0) {
- return;
- }
// Force wakelock release below by setting reference count to 1.
mWakeLockCount = 1;
}
- CHECK(mWakeLockCount != 0);
if (--mWakeLockCount == 0) {
CHECK(mWakeLockToken != 0);
if (mPowerManager != 0) {
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1257161..f9cc17b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -481,7 +481,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -1277,7 +1277,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 0ab43e0..e0d1404 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -102,7 +102,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
@@ -117,7 +117,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int *sessionId,
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 45ef03d..59d1a59 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1100,11 +1100,6 @@
t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
- // t.in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t.in == NULL) {
- enabledTracks &= ~(1<<i);
- }
}
e0 = enabledTracks;
@@ -1140,6 +1135,13 @@
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
+ // t.in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t.in == NULL) {
+ enabledTracks &= ~(1<<i);
+ e1 &= ~(1<<i);
+ break;
+ }
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
if (inFrames > 0) {
t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames,
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d5a0e21..515368c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1190,7 +1190,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
@@ -1198,6 +1198,7 @@
int uid,
status_t *status)
{
+ size_t frameCount = *pFrameCount;
sp<Track> track;
status_t lStatus;
@@ -1266,6 +1267,7 @@
}
}
}
+ *pFrameCount = frameCount;
if (mType == DIRECT) {
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
@@ -3038,10 +3040,12 @@
// add frames already consumed but not yet released by the resampler
// because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
+#if 0
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
// of unreleased frames after each pass, but just in case...
ALOG_ASSERT(desiredFrames <= cblk->frameCount_);
+#endif
}
uint32_t minFrames = 1;
if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() &&
@@ -4470,13 +4474,17 @@
{ // scope for mLock
Mutex::Autolock _l(mLock);
- if (exitPending()) {
- break;
- }
+
processConfigEvents_l();
// return value 'reconfig' is currently unused
bool reconfig = checkForNewParameters_l();
+ // check exitPending here because checkForNewParameters_l() and
+ // checkForNewParameters_l() can temporarily release mLock
+ if (exitPending()) {
+ break;
+ }
+
// if no active track(s), then standby and release wakelock
size_t size = mActiveTracks.size();
if (size == 0) {
@@ -4772,13 +4780,14 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status)
{
+ size_t frameCount = *pFrameCount;
sp<RecordTrack> track;
status_t lStatus;
@@ -4837,6 +4846,7 @@
}
}
}
+ *pFrameCount = frameCount;
// FIXME use flags and tid similar to createTrack_l()
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 2b749fa..8df6f94 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -422,7 +422,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
@@ -880,7 +880,7 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index a5b9ac5..d8d7790 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -133,7 +133,6 @@
if (mCblk != NULL) {
new(mCblk) audio_track_cblk_t();
// clear all buffers
- mCblk->frameCount_ = frameCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, bufferSize);
@@ -1516,9 +1515,9 @@
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, "
- "mCblk->frameCount_ %u, mChannelMask 0x%08x",
+ "frameCount %u, mChannelMask 0x%08x",
mCblk, mBuffer,
- mCblk->frameCount_, mChannelMask);
+ frameCount, mChannelMask);
// 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);
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index c370536..2bb3ff8 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -731,6 +731,7 @@
return OK;
}
params.state = Parameters::STOPPED;
+ int lastPreviewStreamId = mStreamingProcessor->getPreviewStreamId();
res = mStreamingProcessor->updatePreviewStream(params);
if (res != OK) {
@@ -739,6 +740,8 @@
return res;
}
+ bool previewStreamChanged = mStreamingProcessor->getPreviewStreamId() != lastPreviewStreamId;
+
// We could wait to create the JPEG output stream until first actual use
// (first takePicture call). However, this would substantially increase the
// first capture latency on HAL3 devices, and potentially on some HAL2
@@ -788,6 +791,19 @@
return res;
}
outputStreams.push(getCallbackStreamId());
+ } else if (previewStreamChanged && mCallbackProcessor->getStreamId() != NO_STREAM) {
+ /**
+ * Delete the unused callback stream when preview stream is changed and
+ * preview is not enabled. Don't need stop preview stream as preview is in
+ * STOPPED state now.
+ */
+ ALOGV("%s: Camera %d: Delete unused preview callback stream.", __FUNCTION__, mCameraId);
+ res = mCallbackProcessor->deleteStream();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Unable to delete callback stream %s (%d)",
+ __FUNCTION__, mCameraId, strerror(-res), res);
+ return res;
+ }
}
if (params.zslMode && !params.recordingHint) {
res = updateProcessorStream(mZslProcessor, params);
@@ -797,6 +813,8 @@
return res;
}
outputStreams.push(getZslStreamId());
+ } else {
+ mZslProcessor->deleteStream();
}
outputStreams.push(getPreviewStreamId());
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index 6fae399..f84f019 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -1858,23 +1858,23 @@
size_t reqFocusingAreasSize = focusingAreas.size() * 5;
int32_t *reqFocusingAreas = new int32_t[reqFocusingAreasSize];
- for (size_t i = 0; i < reqFocusingAreasSize; i += 5) {
- if (focusingAreas[i].weight != 0) {
+ for (size_t i = 0, j = 0; i < reqFocusingAreasSize; i += 5, j++) {
+ if (focusingAreas[j].weight != 0) {
reqFocusingAreas[i + 0] =
- normalizedXToArray(focusingAreas[i].left);
+ normalizedXToArray(focusingAreas[j].left);
reqFocusingAreas[i + 1] =
- normalizedYToArray(focusingAreas[i].top);
+ normalizedYToArray(focusingAreas[j].top);
reqFocusingAreas[i + 2] =
- normalizedXToArray(focusingAreas[i].right);
+ normalizedXToArray(focusingAreas[j].right);
reqFocusingAreas[i + 3] =
- normalizedYToArray(focusingAreas[i].bottom);
+ normalizedYToArray(focusingAreas[j].bottom);
} else {
reqFocusingAreas[i + 0] = 0;
reqFocusingAreas[i + 1] = 0;
reqFocusingAreas[i + 2] = 0;
reqFocusingAreas[i + 3] = 0;
}
- reqFocusingAreas[i + 4] = focusingAreas[i].weight;
+ reqFocusingAreas[i + 4] = focusingAreas[j].weight;
}
res = request->update(ANDROID_CONTROL_AF_REGIONS,
reqFocusingAreas, reqFocusingAreasSize);
@@ -1887,23 +1887,23 @@
size_t reqMeteringAreasSize = meteringAreas.size() * 5;
int32_t *reqMeteringAreas = new int32_t[reqMeteringAreasSize];
- for (size_t i = 0; i < reqMeteringAreasSize; i += 5) {
- if (meteringAreas[i].weight != 0) {
+ for (size_t i = 0, j = 0; i < reqMeteringAreasSize; i += 5, j++) {
+ if (meteringAreas[j].weight != 0) {
reqMeteringAreas[i + 0] =
- normalizedXToArray(meteringAreas[i].left);
+ normalizedXToArray(meteringAreas[j].left);
reqMeteringAreas[i + 1] =
- normalizedYToArray(meteringAreas[i].top);
+ normalizedYToArray(meteringAreas[j].top);
reqMeteringAreas[i + 2] =
- normalizedXToArray(meteringAreas[i].right);
+ normalizedXToArray(meteringAreas[j].right);
reqMeteringAreas[i + 3] =
- normalizedYToArray(meteringAreas[i].bottom);
+ normalizedYToArray(meteringAreas[j].bottom);
} else {
reqMeteringAreas[i + 0] = 0;
reqMeteringAreas[i + 1] = 0;
reqMeteringAreas[i + 2] = 0;
reqMeteringAreas[i + 3] = 0;
}
- reqMeteringAreas[i + 4] = meteringAreas[i].weight;
+ reqMeteringAreas[i + 4] = meteringAreas[j].weight;
}
res = request->update(ANDROID_CONTROL_AE_REGIONS,
reqMeteringAreas, reqMeteringAreasSize);
diff --git a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
index 776ebe2..e17d05d 100644
--- a/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
+++ b/services/camera/libcameraservice/api1/client2/ZslProcessor3.cpp
@@ -273,6 +273,15 @@
return INVALID_OPERATION;
}
+ // Flush device to clear out all in-flight requests pending in HAL.
+ res = client->getCameraDevice()->flush();
+ if (res != OK) {
+ ALOGE("%s: Camera %d: Failed to flush device: "
+ "%s (%d)",
+ __FUNCTION__, client->getCameraId(), strerror(-res), res);
+ return res;
+ }
+
// Update JPEG settings
{
SharedParameters::Lock l(client->getParameters());
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index edb77aa..da3e121 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -840,16 +840,20 @@
}
sp<Camera3StreamInterface> deletedStream;
+ ssize_t outputStreamIdx = mOutputStreams.indexOfKey(id);
if (mInputStream != NULL && id == mInputStream->getId()) {
deletedStream = mInputStream;
mInputStream.clear();
} else {
- ssize_t idx = mOutputStreams.indexOfKey(id);
- if (idx == NAME_NOT_FOUND) {
+ if (outputStreamIdx == NAME_NOT_FOUND) {
CLOGE("Stream %d does not exist", id);
return BAD_VALUE;
}
- deletedStream = mOutputStreams.editValueAt(idx);
+ }
+
+ // Delete output stream or the output part of a bi-directional stream.
+ if (outputStreamIdx != NAME_NOT_FOUND) {
+ deletedStream = mOutputStreams.editValueAt(outputStreamIdx);
mOutputStreams.removeItem(id);
}
@@ -1124,7 +1128,14 @@
Mutex::Autolock l(mLock);
mRequestThread->clear();
- return mHal3Device->ops->flush(mHal3Device);
+ status_t res;
+ if (mHal3Device->common.version >= CAMERA_DEVICE_API_VERSION_3_1) {
+ res = mHal3Device->ops->flush(mHal3Device);
+ } else {
+ res = waitUntilDrained();
+ }
+
+ return res;
}
/**