Merge "FINAL ATTEMPT: HTTP services are now provided from JAVA and made available to media code"
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index b1143b9..b1ed7b0 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -171,30 +171,6 @@
////////////////////////////////////////////////////////////////////////////////
int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
-////////////////////////////////////////////////////////////////////////////////
-//
-// Function: EffectGetSubEffects
-//
-// Description: Returns the descriptors of the sub effects of the effect
-// whose uuid is pointed to by first argument.
-//
-// Input:
-// pEffectUuid: pointer to the effect uuid.
-// size: size of the buffer pointed by pDescriptor.
-//
-// Input/Output:
-// pDescriptor: address where to return the sub effect descriptors.
-//
-// Output:
-// returned value: 0 successful operation.
-// -ENODEV factory failed to initialize
-// -EINVAL invalid pEffectUuid or pDescriptor
-// -ENOENT no effect with this uuid found
-// *pDescriptor: updated with the sub effect descriptors.
-//
-////////////////////////////////////////////////////////////////////////////////
-int EffectGetSubEffects(const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptors, size_t size);
-
#if __cplusplus
} // extern "C"
#endif
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 9c8451c..6643736 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -142,6 +142,7 @@
enum InternalOptionType {
INTERNAL_OPTION_SUSPEND, // data is a bool
INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY, // data is an int64_t
+ INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t
};
virtual status_t setInternalOption(
node_id node,
diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
index 2c4aaff..b16e20a 100644
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ b/include/media/nbaio/AudioBufferProviderSource.h
@@ -27,7 +27,7 @@
class AudioBufferProviderSource : public NBAIO_Source {
public:
- AudioBufferProviderSource(AudioBufferProvider *provider, NBAIO_Format format);
+ AudioBufferProviderSource(AudioBufferProvider *provider, const NBAIO_Format& format);
virtual ~AudioBufferProviderSource();
// NBAIO_Port interface
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
index d3802fe..b09b35f 100644
--- a/include/media/nbaio/MonoPipe.h
+++ b/include/media/nbaio/MonoPipe.h
@@ -41,7 +41,7 @@
// Note: whatever shares this object with another thread needs to do so in an SMP-safe way (like
// creating it the object before creating the other thread, or storing the object with a
// release_store). Otherwise the other thread could see a partially-constructed object.
- MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock = false);
+ MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock = false);
virtual ~MonoPipe();
// NBAIO_Port interface
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index 1da0c73..9c9721a 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -53,24 +53,23 @@
// too large, then this decision should be re-visited.
// Sample rate and channel count are explicit, PCM interleaved 16-bit is assumed.
typedef unsigned NBAIO_Format;
-enum {
- Format_Invalid
-};
+
+extern const NBAIO_Format Format_Invalid;
// Return the frame size of an NBAIO_Format in bytes
-size_t Format_frameSize(NBAIO_Format format);
+size_t Format_frameSize(const NBAIO_Format& format);
// Return the frame size of an NBAIO_Format as a bit shift
-size_t Format_frameBitShift(NBAIO_Format format);
+size_t Format_frameBitShift(const NBAIO_Format& format);
// Convert a sample rate in Hz and channel count to an NBAIO_Format
NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount);
// Return the sample rate in Hz of an NBAIO_Format
-unsigned Format_sampleRate(NBAIO_Format format);
+unsigned Format_sampleRate(const NBAIO_Format& format);
// Return the channel count of an NBAIO_Format
-unsigned Format_channelCount(NBAIO_Format format);
+unsigned Format_channelCount(const NBAIO_Format& format);
// Callbacks used by NBAIO_Sink::writeVia() and NBAIO_Source::readVia() below.
typedef ssize_t (*writeVia_t)(void *user, void *buffer, size_t count);
@@ -115,8 +114,8 @@
virtual NBAIO_Format format() const { return mNegotiated ? mFormat : Format_Invalid; }
protected:
- NBAIO_Port(NBAIO_Format format) : mNegotiated(false), mFormat(format),
- mBitShift(Format_frameBitShift(format)) { }
+ NBAIO_Port(const NBAIO_Format& format) : mNegotiated(false), mFormat(format),
+ mBitShift(Format_frameBitShift(format)) { }
virtual ~NBAIO_Port() { }
// Implementations are free to ignore these if they don't need them
@@ -220,7 +219,7 @@
virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
protected:
- NBAIO_Sink(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
+ NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
virtual ~NBAIO_Sink() { }
// Implementations are free to ignore these if they don't need them
@@ -311,7 +310,7 @@
virtual void onTimestamp(const AudioTimestamp& timestamp) { }
protected:
- NBAIO_Source(NBAIO_Format format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
+ NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
virtual ~NBAIO_Source() { }
// Implementations are free to ignore these if they don't need them
diff --git a/include/media/nbaio/Pipe.h b/include/media/nbaio/Pipe.h
index 79a4eee..c784129 100644
--- a/include/media/nbaio/Pipe.h
+++ b/include/media/nbaio/Pipe.h
@@ -30,7 +30,7 @@
public:
// maxFrames will be rounded up to a power of 2, and all slots are available. Must be >= 2.
- Pipe(size_t maxFrames, NBAIO_Format format);
+ Pipe(size_t maxFrames, const NBAIO_Format& format);
virtual ~Pipe();
// NBAIO_Port interface
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index f1636e6..7ba5acc 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -67,6 +67,8 @@
void signalRequestIDRFrame();
+ bool isConfiguredForAdaptivePlayback() { return mIsConfiguredForAdaptivePlayback; }
+
struct PortDescription : public RefBase {
size_t countBuffers();
IOMX::buffer_id bufferIDAt(size_t index) const;
@@ -187,6 +189,7 @@
bool mIsEncoder;
bool mUseMetadataOnEncoderOutput;
bool mShutdownInProgress;
+ bool mIsConfiguredForAdaptivePlayback;
// If "mKeepComponentAllocated" we only transition back to Loaded state
// and do not release the component instance.
@@ -202,6 +205,7 @@
int32_t mMetaDataBuffersToSubmit;
int64_t mRepeatFrameDelayUs;
+ int64_t mMaxPtsGapUs;
status_t setCyclicIntraMacroblockRefresh(const sp<AMessage> &msg, int32_t mode);
status_t allocateBuffersOnPort(OMX_U32 portIndex);
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index b5a4c0b..3901e79 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -65,7 +65,9 @@
struct AudioTrackSharedStatic {
StaticAudioTrackSingleStateQueue::Shared
mSingleStateQueue;
- size_t mBufferPosition; // updated asynchronously by server,
+ // This field should be a size_t, but since it is located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mBufferPosition; // updated asynchronously by server,
// "for entertainment purposes only"
};
@@ -104,7 +106,9 @@
private:
- size_t mMinimum; // server wakes up client if available >= mMinimum
+ // This field should be a size_t, but since it is located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mMinimum; // server wakes up client if available >= mMinimum
// Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
// Left channel is in [0:15], right channel is in [16:31].
@@ -241,7 +245,11 @@
}
void setMinimum(size_t minimum) {
- mCblk->mMinimum = minimum;
+ // This can only happen on a 64-bit client
+ if (minimum > UINT32_MAX) {
+ minimum = UINT32_MAX;
+ }
+ mCblk->mMinimum = (uint32_t) minimum;
}
// Return the number of frames that would need to be obtained and released
diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
index 46a5946..d483061 100644
--- a/include/private/media/StaticAudioTrackState.h
+++ b/include/private/media/StaticAudioTrackState.h
@@ -25,9 +25,13 @@
// state is wrapped by a SingleStateQueue.
struct StaticAudioTrackState {
// do not define constructors, destructors, or virtual methods
- size_t mLoopStart;
- size_t mLoopEnd;
- int mLoopCount;
+
+ // These fields should both be size_t, but since they are located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+ uint32_t mLoopStart;
+ uint32_t mLoopEnd;
+
+ int mLoopCount;
};
} // namespace android
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index f8d6041..6d30d64 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -368,27 +368,21 @@
}
if (e1 == NULL) {
ret = -ENOENT;
- pthread_mutex_unlock(&gLibLock);
goto exit;
}
// release effect in library
if (fx->lib == NULL) {
ALOGW("EffectRelease() fx %p library already unloaded", handle);
- pthread_mutex_unlock(&gLibLock);
} else {
pthread_mutex_lock(&fx->lib->lock);
- // Releasing the gLibLock here as the list access is over as the
- // effect is removed from the list.
- // If the gLibLock is not released, we will have a deadlock situation
- // since we call the sub effect release inside the EffectRelease of Proxy
- pthread_mutex_unlock(&gLibLock);
fx->lib->desc->release_effect(fx->subItfe);
pthread_mutex_unlock(&fx->lib->lock);
}
free(fx);
exit:
+ pthread_mutex_unlock(&gLibLock);
return ret;
}
@@ -404,8 +398,8 @@
// is pointed by the first argument. It searches the gSubEffectList for the
// matching uuid and then copies the corresponding sub effect descriptors
// to the inout param
-int EffectGetSubEffects(const effect_uuid_t *uuid,
- effect_descriptor_t *pDescriptors, size_t size)
+int EffectGetSubEffects(const effect_uuid_t *uuid, sub_effect_entry_t **pSube,
+ size_t size)
{
ALOGV("EffectGetSubEffects() UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X"
"%02X\n",uuid->timeLow, uuid->timeMid, uuid->timeHiAndVersion,
@@ -413,8 +407,7 @@
uuid->node[3],uuid->node[4],uuid->node[5]);
// Check if the size of the desc buffer is large enough for 2 subeffects
- if ((uuid == NULL) || (pDescriptors == NULL) ||
- (size < 2*sizeof(effect_descriptor_t))) {
+ if ((uuid == NULL) || (pSube == NULL) || (size < 2)) {
ALOGW("NULL pointer or insufficient memory. Cannot query subeffects");
return -EINVAL;
}
@@ -432,11 +425,10 @@
list_elem_t *subefx = e->sub_elem;
while (subefx != NULL) {
subeffect = (sub_effect_entry_t*)subefx->object;
- d = (effect_descriptor_t*)(subeffect->object);
- pDescriptors[count++] = *d;
+ pSube[count++] = subeffect;
subefx = subefx->next;
}
- ALOGV("EffectGetSubEffects end - copied the sub effect descriptors");
+ ALOGV("EffectGetSubEffects end - copied the sub effect structures");
return count;
}
e = e->next;
diff --git a/media/libeffects/factory/EffectsFactory.h b/media/libeffects/factory/EffectsFactory.h
index 147ff18..560b485 100644
--- a/media/libeffects/factory/EffectsFactory.h
+++ b/media/libeffects/factory/EffectsFactory.h
@@ -20,7 +20,7 @@
#include <cutils/log.h>
#include <pthread.h>
#include <dirent.h>
-#include <media/EffectsFactoryApi.h>
+#include <hardware/audio_effect.h>
#if __cplusplus
extern "C" {
@@ -66,6 +66,32 @@
void *object;
} sub_effect_entry_t;
+
+////////////////////////////////////////////////////////////////////////////////
+//
+// Function: EffectGetSubEffects
+//
+// Description: Returns the descriptors of the sub effects of the effect
+// whose uuid is pointed to by first argument.
+//
+// Input:
+// pEffectUuid: pointer to the effect uuid.
+// size: max number of sub_effect_entry_t * in pSube.
+//
+// Input/Output:
+// pSube: address where to return the sub effect structures.
+// Output:
+// returned value: 0 successful operation.
+// -ENODEV factory failed to initialize
+// -EINVAL invalid pEffectUuid or pDescriptor
+// -ENOENT no effect with this uuid found
+// *pDescriptor: updated with the sub effect descriptors.
+//
+////////////////////////////////////////////////////////////////////////////////
+int EffectGetSubEffects(const effect_uuid_t *pEffectUuid,
+ sub_effect_entry_t **pSube,
+ size_t size);
+
#if __cplusplus
} // extern "C"
#endif
diff --git a/media/libeffects/proxy/Android.mk b/media/libeffects/proxy/Android.mk
index d6d6c1e..b438796 100644
--- a/media/libeffects/proxy/Android.mk
+++ b/media/libeffects/proxy/Android.mk
@@ -28,7 +28,8 @@
LOCAL_C_INCLUDES := \
system/media/audio_effects/include \
- bionic/libc/include
+ bionic/libc/include \
+ frameworks/av/media/libeffects/factory
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libeffects/proxy/EffectProxy.cpp b/media/libeffects/proxy/EffectProxy.cpp
index e6faaaf..62d3fd3 100644
--- a/media/libeffects/proxy/EffectProxy.cpp
+++ b/media/libeffects/proxy/EffectProxy.cpp
@@ -56,6 +56,8 @@
effect_handle_t *pHandle) {
effect_descriptor_t* desc;
+ audio_effect_library_t** aeli;
+ sub_effect_entry_t** sube;
EffectContext* pContext;
if (pHandle == NULL || uuid == NULL) {
ALOGE("EffectProxyCreate() called with NULL pointer");
@@ -74,31 +76,52 @@
// Get the HW and SW sub effect descriptors from the effects factory
desc = new effect_descriptor_t[SUB_FX_COUNT];
+ aeli = new audio_effect_library_t*[SUB_FX_COUNT];
+ sube = new sub_effect_entry_t*[SUB_FX_COUNT];
+ pContext->sube = new sub_effect_entry_t*[SUB_FX_COUNT];
pContext->desc = new effect_descriptor_t[SUB_FX_COUNT];
- int retValue = EffectGetSubEffects(uuid, desc,
- sizeof(effect_descriptor_t) * SUB_FX_COUNT);
+ pContext->aeli = new audio_effect_library_t*[SUB_FX_COUNT];
+ int retValue = EffectGetSubEffects(uuid, sube, SUB_FX_COUNT);
// EffectGetSubEffects returns the number of sub-effects copied.
if (retValue != SUB_FX_COUNT) {
ALOGE("EffectCreate() could not get the sub effects");
- delete desc;
- delete pContext->desc;
+ delete[] sube;
+ delete[] desc;
+ delete[] aeli;
+ delete[] pContext->sube;
+ delete[] pContext->desc;
+ delete[] pContext->aeli;
return -EINVAL;
}
// Check which is the HW descriptor and copy the descriptors
// to the Context desc array
// Also check if there is only one HW and one SW descriptor.
// HW descriptor alone has the HW_TUNNEL flag.
+ desc[0] = *(effect_descriptor_t*)(sube[0])->object;
+ desc[1] = *(effect_descriptor_t*)(sube[1])->object;
+ aeli[0] = sube[0]->lib->desc;
+ aeli[1] = sube[1]->lib->desc;
if ((desc[0].flags & EFFECT_FLAG_HW_ACC_TUNNEL) &&
!(desc[1].flags & EFFECT_FLAG_HW_ACC_TUNNEL)) {
+ pContext->sube[SUB_FX_OFFLOAD] = sube[0];
pContext->desc[SUB_FX_OFFLOAD] = desc[0];
+ pContext->aeli[SUB_FX_OFFLOAD] = aeli[0];
+ pContext->sube[SUB_FX_HOST] = sube[1];
pContext->desc[SUB_FX_HOST] = desc[1];
+ pContext->aeli[SUB_FX_HOST] = aeli[1];
}
else if ((desc[1].flags & EFFECT_FLAG_HW_ACC_TUNNEL) &&
!(desc[0].flags & EFFECT_FLAG_HW_ACC_TUNNEL)) {
+ pContext->sube[SUB_FX_HOST] = sube[0];
pContext->desc[SUB_FX_HOST] = desc[0];
+ pContext->aeli[SUB_FX_HOST] = aeli[0];
+ pContext->sube[SUB_FX_OFFLOAD] = sube[1];
pContext->desc[SUB_FX_OFFLOAD] = desc[1];
+ pContext->aeli[SUB_FX_OFFLOAD] = aeli[1];
}
- delete desc;
+ delete[] desc;
+ delete[] aeli;
+ delete[] sube;
#if (LOG_NDEBUG == 0)
effect_uuid_t uuid_print = pContext->desc[SUB_FX_HOST].uuid;
ALOGV("EffectCreate() UUID of HOST: %08X-%04X-%04X-%04X-%02X%02X%02X%02X"
@@ -128,13 +151,15 @@
return -EINVAL;
}
ALOGV("EffectRelease");
- delete pContext->desc;
+ delete[] pContext->desc;
free(pContext->replyData);
if (pContext->eHandle[SUB_FX_HOST])
- EffectRelease(pContext->eHandle[SUB_FX_HOST]);
+ pContext->aeli[SUB_FX_HOST]->release_effect(pContext->eHandle[SUB_FX_HOST]);
if (pContext->eHandle[SUB_FX_OFFLOAD])
- EffectRelease(pContext->eHandle[SUB_FX_OFFLOAD]);
+ pContext->aeli[SUB_FX_OFFLOAD]->release_effect(pContext->eHandle[SUB_FX_OFFLOAD]);
+ delete[] pContext->aeli;
+ delete[] pContext->sube;
delete pContext;
pContext = NULL;
return 0;
@@ -187,7 +212,8 @@
}
if (pContext->eHandle[SUB_FX_HOST] == NULL) {
ALOGV("Effect_command() Calling HOST EffectCreate");
- status = EffectCreate(&pContext->desc[SUB_FX_HOST].uuid,
+ status = pContext->aeli[SUB_FX_HOST]->create_effect(
+ &pContext->desc[SUB_FX_HOST].uuid,
pContext->sessionId, pContext->ioId,
&(pContext->eHandle[SUB_FX_HOST]));
if (status != NO_ERROR || (pContext->eHandle[SUB_FX_HOST] == NULL)) {
@@ -197,11 +223,13 @@
}
if (pContext->eHandle[SUB_FX_OFFLOAD] == NULL) {
ALOGV("Effect_command() Calling OFFLOAD EffectCreate");
- status = EffectCreate(&pContext->desc[SUB_FX_OFFLOAD].uuid,
+ status = pContext->aeli[SUB_FX_OFFLOAD]->create_effect(
+ &pContext->desc[SUB_FX_OFFLOAD].uuid,
pContext->sessionId, pContext->ioId,
&(pContext->eHandle[SUB_FX_OFFLOAD]));
if (status != NO_ERROR || (pContext->eHandle[SUB_FX_OFFLOAD] == NULL)) {
ALOGV("Effect_command() Error creating HW effect");
+ pContext->eHandle[SUB_FX_OFFLOAD] = NULL;
// Do not return error here as SW effect is created
// Return error if the CMD_OFFLOAD sends the index as OFFLOAD
}
@@ -233,11 +261,17 @@
// Update the DSP wrapper with the new ioHandle.
// Pass the OFFLOAD command to the wrapper.
// The DSP wrapper needs to handle this CMD
- if (pContext->eHandle[SUB_FX_OFFLOAD])
- status = (*pContext->eHandle[SUB_FX_OFFLOAD])->command(
- pContext->eHandle[SUB_FX_OFFLOAD], cmdCode, cmdSize,
- pCmdData, replySize, pReplyData);
- return status;
+ if (pContext->eHandle[SUB_FX_OFFLOAD]) {
+ ALOGV("Effect_command: Calling OFFLOAD command");
+ return (*pContext->eHandle[SUB_FX_OFFLOAD])->command(
+ pContext->eHandle[SUB_FX_OFFLOAD], cmdCode, cmdSize,
+ pCmdData, replySize, pReplyData);
+ }
+ *(int*)pReplyData = NO_ERROR;
+ ALOGV("Effect_command OFFLOAD return 0, replyData %d",
+ *(int*)pReplyData);
+
+ return NO_ERROR;
}
int index = pContext->index;
diff --git a/media/libeffects/proxy/EffectProxy.h b/media/libeffects/proxy/EffectProxy.h
index acbe17e..046b93e 100644
--- a/media/libeffects/proxy/EffectProxy.h
+++ b/media/libeffects/proxy/EffectProxy.h
@@ -16,6 +16,8 @@
#include <hardware/audio.h>
#include <hardware/audio_effect.h>
+#include "EffectsFactory.h"
+
namespace android {
enum {
SUB_FX_HOST, // Index of HOST in the descriptor and handle arrays
@@ -62,7 +64,9 @@
struct EffectContext {
const struct effect_interface_s *common_itfe; // Holds the itfe of the Proxy
+ sub_effect_entry_t** sube; // Points to the sub effects
effect_descriptor_t* desc; // Points to the sub effect descriptors
+ audio_effect_library_t** aeli; // Points to the sub effect aeli
effect_handle_t eHandle[SUB_FX_COUNT]; // The effect handles of the sub effects
int index; // The index that is currently active - HOST or OFFLOAD
int32_t sessionId; // The sessiond in which the effect is created.
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 8dfffb3..35f6557 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -380,9 +380,9 @@
}
void AudioEffect::commandExecuted(uint32_t cmdCode,
- uint32_t cmdSize,
+ uint32_t cmdSize __unused,
void *cmdData,
- uint32_t replySize,
+ uint32_t replySize __unused,
void *replyData)
{
if (cmdData == NULL || replyData == NULL) {
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index b69e3ae..286096e 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -87,7 +87,7 @@
int notificationFrames,
int sessionId,
transfer_type transferType,
- audio_input_flags_t flags)
+ audio_input_flags_t flags __unused)
: mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index dcb72f8..1ab94fa 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -333,7 +333,7 @@
}
status_t AudioSystem::getLatency(audio_io_handle_t output,
- audio_stream_type_t streamType,
+ audio_stream_type_t streamType __unused,
uint32_t* latency)
{
OutputDescriptor *outputDesc;
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 21018a0..fdd1a12 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -475,9 +475,14 @@
void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
{
+ // This can only happen on a 64-bit client
+ if (loopStart > UINT32_MAX || loopEnd > UINT32_MAX) {
+ // FIXME Should return an error status
+ return;
+ }
StaticAudioTrackState newState;
- newState.mLoopStart = loopStart;
- newState.mLoopEnd = loopEnd;
+ newState.mLoopStart = (uint32_t) loopStart;
+ newState.mLoopEnd = (uint32_t) loopEnd;
newState.mLoopCount = loopCount;
mBufferPosition = loopStart;
(void) mMutator.push(newState);
@@ -487,7 +492,7 @@
{
size_t bufferPosition;
if (mMutator.ack()) {
- bufferPosition = mCblk->u.mStatic.mBufferPosition;
+ bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
if (bufferPosition > mFrameCount) {
bufferPosition = mFrameCount;
}
@@ -622,7 +627,7 @@
if (half == 0) {
half = 1;
}
- size_t minimum = cblk->mMinimum;
+ size_t minimum = (size_t) cblk->mMinimum;
if (minimum == 0) {
minimum = mIsOut ? half : 1;
} else if (minimum > half) {
@@ -760,7 +765,8 @@
mIsShutdown = true;
return (ssize_t) NO_INIT;
}
- mCblk->u.mStatic.mBufferPosition = position;
+ // This may overflow, but client is not supposed to rely on it
+ mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
}
return (ssize_t) position;
}
@@ -836,7 +842,8 @@
mPosition = newPosition;
cblk->mServer += stepCount;
- cblk->u.mStatic.mBufferPosition = newPosition;
+ // This may overflow, but client is not supposed to rely on it
+ cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition;
if (setFlags != 0) {
(void) android_atomic_or(setFlags, &cblk->mFlags);
// this would be a good place to wake a futex
diff --git a/media/libnbaio/AudioBufferProviderSource.cpp b/media/libnbaio/AudioBufferProviderSource.cpp
index 74a6fdb..e952a00 100644
--- a/media/libnbaio/AudioBufferProviderSource.cpp
+++ b/media/libnbaio/AudioBufferProviderSource.cpp
@@ -24,7 +24,7 @@
namespace android {
AudioBufferProviderSource::AudioBufferProviderSource(AudioBufferProvider *provider,
- NBAIO_Format format) :
+ const NBAIO_Format& format) :
NBAIO_Source(format), mProvider(provider), mConsumed(0)
{
ALOG_ASSERT(provider != NULL);
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 3c61b60..b23967b 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -30,7 +30,7 @@
namespace android {
-MonoPipe::MonoPipe(size_t reqFrames, NBAIO_Format format, bool writeCanBlock) :
+MonoPipe::MonoPipe(size_t reqFrames, const NBAIO_Format& format, bool writeCanBlock) :
NBAIO_Sink(format),
mUpdateSeq(0),
mReqFrames(reqFrames),
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index e0d2c21..8a8155d 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -22,17 +22,19 @@
namespace android {
-size_t Format_frameSize(NBAIO_Format format)
+size_t Format_frameSize(const NBAIO_Format& format)
{
return Format_channelCount(format) * sizeof(short);
}
-size_t Format_frameBitShift(NBAIO_Format format)
+size_t Format_frameBitShift(const NBAIO_Format& format)
{
// sizeof(short) == 2, so frame size == 1 << channels
return Format_channelCount(format);
}
+const NBAIO_Format Format_Invalid = { 0 };
+
enum {
Format_SR_8000,
Format_SR_11025,
@@ -51,7 +53,7 @@
Format_C_Mask = 0x18
};
-unsigned Format_sampleRate(NBAIO_Format format)
+unsigned Format_sampleRate(const NBAIO_Format& format)
{
if (format == Format_Invalid) {
return 0;
@@ -78,7 +80,7 @@
}
}
-unsigned Format_channelCount(NBAIO_Format format)
+unsigned Format_channelCount(const NBAIO_Format& format)
{
if (format == Format_Invalid) {
return 0;
diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp
index 1c21f9c..115f311 100644
--- a/media/libnbaio/Pipe.cpp
+++ b/media/libnbaio/Pipe.cpp
@@ -25,7 +25,7 @@
namespace android {
-Pipe::Pipe(size_t maxFrames, NBAIO_Format format) :
+Pipe::Pipe(size_t maxFrames, const NBAIO_Format& format) :
NBAIO_Sink(format),
mMaxFrames(roundup(maxFrames)),
mBuffer(malloc(mMaxFrames * Format_frameSize(format))),
diff --git a/media/libnbaio/PipeReader.cpp b/media/libnbaio/PipeReader.cpp
index d786b84..24da1bd 100644
--- a/media/libnbaio/PipeReader.cpp
+++ b/media/libnbaio/PipeReader.cpp
@@ -59,7 +59,7 @@
return avail;
}
-ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS)
+ssize_t PipeReader::read(void *buffer, size_t count, int64_t readPTS __unused)
{
ssize_t avail = availableToRead();
if (CC_UNLIKELY(avail <= 0)) {
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index eb274a8..c9f89fc 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -365,6 +365,7 @@
mIsEncoder(false),
mUseMetadataOnEncoderOutput(false),
mShutdownInProgress(false),
+ mIsConfiguredForAdaptivePlayback(false),
mEncoderDelay(0),
mEncoderPadding(0),
mChannelMaskPresent(false),
@@ -372,7 +373,8 @@
mDequeueCounter(0),
mStoreMetaDataInOutputBuffers(false),
mMetaDataBuffersToSubmit(0),
- mRepeatFrameDelayUs(-1ll) {
+ mRepeatFrameDelayUs(-1ll),
+ mMaxPtsGapUs(-1l) {
mUninitializedState = new UninitializedState(this);
mLoadedState = new LoadedState(this);
mLoadedToIdleState = new LoadedToIdleState(this);
@@ -1114,6 +1116,10 @@
&mRepeatFrameDelayUs)) {
mRepeatFrameDelayUs = -1ll;
}
+
+ if (!msg->findInt64("max-pts-gap-to-encoder", &mMaxPtsGapUs)) {
+ mMaxPtsGapUs = -1l;
+ }
}
// Always try to enable dynamic output buffers on native surface
@@ -1121,6 +1127,7 @@
int32_t haveNativeWindow = msg->findObject("native-window", &obj) &&
obj != NULL;
mStoreMetaDataInOutputBuffers = false;
+ mIsConfiguredForAdaptivePlayback = false;
if (!encoder && video && haveNativeWindow) {
err = mOMX->storeMetaDataInBuffers(mNode, kPortIndexOutput, OMX_TRUE);
if (err != OK) {
@@ -1165,12 +1172,14 @@
ALOGW_IF(err != OK,
"[%s] prepareForAdaptivePlayback failed w/ err %d",
mComponentName.c_str(), err);
+ mIsConfiguredForAdaptivePlayback = (err == OK);
}
// allow failure
err = OK;
} else {
ALOGV("[%s] storeMetaDataInBuffers succeeded", mComponentName.c_str());
mStoreMetaDataInOutputBuffers = true;
+ mIsConfiguredForAdaptivePlayback = true;
}
int32_t push;
@@ -3318,11 +3327,11 @@
mCodec->mInputEOSResult = err;
}
break;
-
- default:
- CHECK_EQ((int)mode, (int)FREE_BUFFERS);
- break;
}
+
+ default:
+ CHECK_EQ((int)mode, (int)FREE_BUFFERS);
+ break;
}
}
@@ -3498,7 +3507,7 @@
int32_t render;
if (mCodec->mNativeWindow != NULL
&& msg->findInt32("render", &render) && render != 0
- && (info->mData == NULL || info->mData->size() != 0)) {
+ && info->mData != NULL && info->mData->size() != 0) {
// The client wants this buffer to be rendered.
status_t err;
@@ -3773,6 +3782,7 @@
mCodec->mDequeueCounter = 0;
mCodec->mMetaDataBuffersToSubmit = 0;
mCodec->mRepeatFrameDelayUs = -1ll;
+ mCodec->mIsConfiguredForAdaptivePlayback = false;
if (mCodec->mShutdownInProgress) {
bool keepComponentAllocated = mCodec->mKeepComponentAllocated;
@@ -3921,6 +3931,21 @@
}
}
+ if (err == OK && mCodec->mMaxPtsGapUs > 0l) {
+ err = mCodec->mOMX->setInternalOption(
+ mCodec->mNode,
+ kPortIndexInput,
+ IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP,
+ &mCodec->mMaxPtsGapUs,
+ sizeof(mCodec->mMaxPtsGapUs));
+
+ if (err != OK) {
+ ALOGE("[%s] Unable to configure max timestamp gap (err %d)",
+ mCodec->mComponentName.c_str(),
+ err);
+ }
+ }
+
if (err == OK) {
notify->setObject("input-surface",
new BufferProducerWrapper(bufferProducer));
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index a0f17b5..b7a4b75 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -972,13 +972,16 @@
if (param && param->findInt32(kKeyFileType, &fileType) &&
fileType != OUTPUT_FORMAT_MPEG_4) {
writeFourcc("3gp4");
- } else {
+ writeInt32(0);
writeFourcc("isom");
+ writeFourcc("3gp4");
+ } else {
+ writeFourcc("mp42");
+ writeInt32(0);
+ writeFourcc("isom");
+ writeFourcc("mp42");
}
- writeInt32(0);
- writeFourcc("isom");
- writeFourcc("3gp4");
endBox();
}
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 8af1aaf..c4c47b3 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -1680,7 +1680,7 @@
return -EACCES;
}
- if (render && (info->mData == NULL || info->mData->size() != 0)) {
+ if (render && info->mData != NULL && info->mData->size() != 0) {
info->mNotify->setInt32("render", true);
if (mSoftRenderer != NULL) {
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index d260d0f..dcb1cda 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -716,41 +716,61 @@
return mIsLiveStreaming;
}
+static int bytesForSize(size_t size) {
+ // use at most 28 bits (4 times 7)
+ CHECK(size <= 0xfffffff);
+
+ if (size > 0x1fffff) {
+ return 4;
+ } else if (size > 0x3fff) {
+ return 3;
+ } else if (size > 0x7f) {
+ return 2;
+ }
+ return 1;
+}
+
+static void storeSize(uint8_t *data, size_t &idx, size_t size) {
+ int numBytes = bytesForSize(size);
+ idx += numBytes;
+
+ data += idx;
+ size_t next = 0;
+ while (numBytes--) {
+ *--data = (size & 0x7f) | next;
+ size >>= 7;
+ next = 0x80;
+ }
+}
+
static void addESDSFromCodecPrivate(
const sp<MetaData> &meta,
bool isAudio, const void *priv, size_t privSize) {
- static const uint8_t kStaticESDS[] = {
- 0x03, 22,
- 0x00, 0x00, // ES_ID
- 0x00, // streamDependenceFlag, URL_Flag, OCRstreamFlag
- 0x04, 17,
- 0x40, // ObjectTypeIndication
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
-
- 0x05,
- // CodecSpecificInfo (with size prefix) follows
- };
-
- // Make sure all sizes can be coded in a single byte.
- CHECK(privSize + 22 - 2 < 128);
- size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
+ int privSizeBytesRequired = bytesForSize(privSize);
+ int esdsSize2 = 14 + privSizeBytesRequired + privSize;
+ int esdsSize2BytesRequired = bytesForSize(esdsSize2);
+ int esdsSize1 = 4 + esdsSize2BytesRequired + esdsSize2;
+ int esdsSize1BytesRequired = bytesForSize(esdsSize1);
+ size_t esdsSize = 1 + esdsSize1BytesRequired + esdsSize1;
uint8_t *esds = new uint8_t[esdsSize];
- memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
- uint8_t *ptr = esds + sizeof(kStaticESDS);
- *ptr++ = privSize;
- memcpy(ptr, priv, privSize);
- // Increment by codecPrivateSize less 2 bytes that are accounted for
- // already in lengths of 22/17
- esds[1] += privSize - 2;
- esds[6] += privSize - 2;
-
- // Set ObjectTypeIndication.
- esds[7] = isAudio ? 0x40 // Audio ISO/IEC 14496-3
- : 0x20; // Visual ISO/IEC 14496-2
+ size_t idx = 0;
+ esds[idx++] = 0x03;
+ storeSize(esds, idx, esdsSize1);
+ esds[idx++] = 0x00; // ES_ID
+ esds[idx++] = 0x00; // ES_ID
+ esds[idx++] = 0x00; // streamDependenceFlag, URL_Flag, OCRstreamFlag
+ esds[idx++] = 0x04;
+ storeSize(esds, idx, esdsSize2);
+ esds[idx++] = isAudio ? 0x40 // Audio ISO/IEC 14496-3
+ : 0x20; // Visual ISO/IEC 14496-2
+ for (int i = 0; i < 12; i++) {
+ esds[idx++] = 0x00;
+ }
+ esds[idx++] = 0x05;
+ storeSize(esds, idx, privSize);
+ memcpy(esds + idx, priv, privSize);
meta->setData(kKeyESDS, 0, esds, esdsSize);
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index 7492577..20fa7ce 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -42,7 +42,11 @@
mEndOfStream(false),
mEndOfStreamSent(false),
mRepeatAfterUs(-1ll),
+ mMaxTimestampGapUs(-1ll),
+ mPrevOriginalTimeUs(-1ll),
+ mPrevModifiedTimeUs(-1ll),
mRepeatLastFrameGeneration(0),
+ mRepeatLastFrameTimestamp(-1ll),
mLatestSubmittedBufferId(-1),
mLatestSubmittedBufferFrameNum(0),
mLatestSubmittedBufferUseCount(0),
@@ -299,6 +303,32 @@
return;
}
+void GraphicBufferSource::codecBufferFilled(OMX_BUFFERHEADERTYPE* header) {
+ Mutex::Autolock autoLock(mMutex);
+
+ if (mMaxTimestampGapUs > 0ll
+ && !(header->nFlags & OMX_BUFFERFLAG_CODECCONFIG)) {
+ ssize_t index = mOriginalTimeUs.indexOfKey(header->nTimeStamp);
+ if (index >= 0) {
+ ALOGV("OUT timestamp: %lld -> %lld",
+ header->nTimeStamp, mOriginalTimeUs[index]);
+ header->nTimeStamp = mOriginalTimeUs[index];
+ mOriginalTimeUs.removeItemsAt(index);
+ } else {
+ // giving up the effort as encoder doesn't appear to preserve pts
+ ALOGW("giving up limiting timestamp gap (pts = %lld)",
+ header->nTimeStamp);
+ mMaxTimestampGapUs = -1ll;
+ }
+ if (mOriginalTimeUs.size() > BufferQueue::NUM_BUFFER_SLOTS) {
+ // something terribly wrong must have happened, giving up...
+ ALOGE("mOriginalTimeUs has too many entries (%d)",
+ mOriginalTimeUs.size());
+ mMaxTimestampGapUs = -1ll;
+ }
+ }
+}
+
void GraphicBufferSource::suspend(bool suspend) {
Mutex::Autolock autoLock(mMutex);
@@ -431,6 +461,7 @@
BufferQueue::BufferItem item;
item.mBuf = mLatestSubmittedBufferId;
item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+ item.mTimestamp = mRepeatLastFrameTimestamp;
status_t err = submitBuffer_l(item, cbi);
@@ -440,6 +471,20 @@
++mLatestSubmittedBufferUseCount;
+ /* repeat last frame up to kRepeatLastFrameCount times.
+ * in case of static scene, a single repeat might not get rid of encoder
+ * ghosting completely, refresh a couple more times to get better quality
+ */
+ if (--mRepeatLastFrameCount > 0) {
+ mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
+ if (mReflector != NULL) {
+ sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
+ msg->setInt32("generation", ++mRepeatLastFrameGeneration);
+ msg->post(mRepeatAfterUs);
+ }
+ }
+
return true;
}
@@ -460,8 +505,11 @@
mLatestSubmittedBufferId = item.mBuf;
mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+ mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
+
mLatestSubmittedBufferUseCount = 1;
mRepeatBufferDeferred = false;
+ mRepeatLastFrameCount = kRepeatLastFrameCount;
if (mReflector != NULL) {
sp<AMessage> msg = new AMessage(kWhatRepeatLastFrame, mReflector->id());
@@ -497,6 +545,39 @@
return OK;
}
+int64_t GraphicBufferSource::getTimestamp(const BufferQueue::BufferItem &item) {
+ int64_t timeUs = item.mTimestamp / 1000;
+
+ if (mMaxTimestampGapUs > 0ll) {
+ /* Cap timestamp gap between adjacent frames to specified max
+ *
+ * In the scenario of cast mirroring, encoding could be suspended for
+ * prolonged periods. Limiting the pts gap to workaround the problem
+ * where encoder's rate control logic produces huge frames after a
+ * long period of suspension.
+ */
+
+ int64_t originalTimeUs = timeUs;
+ if (mPrevOriginalTimeUs >= 0ll) {
+ if (originalTimeUs < mPrevOriginalTimeUs) {
+ // Drop the frame if it's going backward in time. Bad timestamp
+ // could disrupt encoder's rate control completely.
+ ALOGV("Dropping frame that's going backward in time");
+ return -1;
+ }
+ int64_t timestampGapUs = originalTimeUs - mPrevOriginalTimeUs;
+ timeUs = (timestampGapUs < mMaxTimestampGapUs ?
+ timestampGapUs : mMaxTimestampGapUs) + mPrevModifiedTimeUs;
+ }
+ mPrevOriginalTimeUs = originalTimeUs;
+ mPrevModifiedTimeUs = timeUs;
+ mOriginalTimeUs.add(timeUs, originalTimeUs);
+ ALOGV("IN timestamp: %lld -> %lld", originalTimeUs, timeUs);
+ }
+
+ return timeUs;
+}
+
status_t GraphicBufferSource::submitBuffer_l(
const BufferQueue::BufferItem &item, int cbi) {
ALOGV("submitBuffer_l cbi=%d", cbi);
@@ -513,9 +594,15 @@
memcpy(data, &type, 4);
memcpy(data + 4, &handle, sizeof(buffer_handle_t));
+ int64_t timeUs = getTimestamp(item);
+ if (timeUs < 0ll) {
+ ALOGE("Dropping frame with bad timestamp");
+ return UNKNOWN_ERROR;
+ }
+
status_t err = mNodeInstance->emptyDirectBuffer(header, 0,
4 + sizeof(buffer_handle_t), OMX_BUFFERFLAG_ENDOFFRAME,
- item.mTimestamp / 1000);
+ timeUs);
if (err != OK) {
ALOGW("WARNING: emptyDirectBuffer failed: 0x%x", err);
codecBuffer.mGraphicBuffer = NULL;
@@ -664,6 +751,17 @@
return OK;
}
+status_t GraphicBufferSource::setMaxTimestampGapUs(int64_t maxGapUs) {
+ Mutex::Autolock autoLock(mMutex);
+
+ if (mExecuting || maxGapUs <= 0ll) {
+ return INVALID_OPERATION;
+ }
+
+ mMaxTimestampGapUs = maxGapUs;
+
+ return OK;
+}
void GraphicBufferSource::onMessageReceived(const sp<AMessage> &msg) {
switch (msg->what()) {
case kWhatRepeatLastFrame:
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index 9e5eee6..3b0e454 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -87,6 +87,10 @@
// fill it with a new frame of data; otherwise, just mark it as available.
void codecBufferEmptied(OMX_BUFFERHEADERTYPE* header);
+ // Called when omx_message::FILL_BUFFER_DONE is received. (Currently the
+ // buffer source will fix timestamp in the header if needed.)
+ void codecBufferFilled(OMX_BUFFERHEADERTYPE* header);
+
// This is called after the last input frame has been submitted. We
// need to submit an empty buffer with the EOS flag set. If we don't
// have a codec buffer ready, we just set the mEndOfStream flag.
@@ -105,6 +109,15 @@
// state and once this behaviour is specified it cannot be reset.
status_t setRepeatPreviousFrameDelayUs(int64_t repeatAfterUs);
+ // When set, the timestamp fed to the encoder will be modified such that
+ // the gap between two adjacent frames is capped at maxGapUs. Timestamp
+ // will be restored to the original when the encoded frame is returned to
+ // the client.
+ // This is to solve a problem in certain real-time streaming case, where
+ // encoder's rate control logic produces huge frames after a long period
+ // of suspension on input.
+ status_t setMaxTimestampGapUs(int64_t maxGapUs);
+
protected:
// BufferQueue::ConsumerListener interface, called when a new frame of
// data is available. If we're executing and a codec buffer is
@@ -165,6 +178,7 @@
void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item);
bool repeatLatestSubmittedBuffer_l();
+ int64_t getTimestamp(const BufferQueue::BufferItem &item);
// Lock, covers all member variables.
mutable Mutex mMutex;
@@ -206,13 +220,22 @@
enum {
kWhatRepeatLastFrame,
};
-
+ enum {
+ kRepeatLastFrameCount = 10,
+ };
int64_t mRepeatAfterUs;
+ int64_t mMaxTimestampGapUs;
+
+ KeyedVector<int64_t, int64_t> mOriginalTimeUs;
+ int64_t mPrevOriginalTimeUs;
+ int64_t mPrevModifiedTimeUs;
sp<ALooper> mLooper;
sp<AHandlerReflector<GraphicBufferSource> > mReflector;
int32_t mRepeatLastFrameGeneration;
+ int64_t mRepeatLastFrameTimestamp;
+ int32_t mRepeatLastFrameCount;
int mLatestSubmittedBufferId;
uint64_t mLatestSubmittedBufferFrameNum;
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 5f104fc..6c5c857 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -849,6 +849,7 @@
switch (type) {
case IOMX::INTERNAL_OPTION_SUSPEND:
case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
+ case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
{
const sp<GraphicBufferSource> &bufferSource =
getGraphicBufferSource();
@@ -864,7 +865,8 @@
bool suspend = *(bool *)data;
bufferSource->suspend(suspend);
- } else {
+ } else if (type ==
+ IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY){
if (size != sizeof(int64_t)) {
return INVALID_OPERATION;
}
@@ -872,6 +874,14 @@
int64_t delayUs = *(int64_t *)data;
return bufferSource->setRepeatPreviousFrameDelayUs(delayUs);
+ } else {
+ if (size != sizeof(int64_t)) {
+ return INVALID_OPERATION;
+ }
+
+ int64_t maxGapUs = *(int64_t *)data;
+
+ return bufferSource->setMaxTimestampGapUs(maxGapUs);
}
return OK;
@@ -883,6 +893,8 @@
}
void OMXNodeInstance::onMessage(const omx_message &msg) {
+ const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
+
if (msg.type == omx_message::FILL_BUFFER_DONE) {
OMX_BUFFERHEADERTYPE *buffer =
static_cast<OMX_BUFFERHEADERTYPE *>(
@@ -892,10 +904,18 @@
static_cast<BufferMeta *>(buffer->pAppPrivate);
buffer_meta->CopyFromOMX(buffer);
- } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
- const sp<GraphicBufferSource>& bufferSource(getGraphicBufferSource());
if (bufferSource != NULL) {
+ // fix up the buffer info (especially timestamp) if needed
+ bufferSource->codecBufferFilled(buffer);
+
+ omx_message newMsg = msg;
+ newMsg.u.extended_buffer_data.timestamp = buffer->nTimeStamp;
+ mObserver->onMessage(newMsg);
+ return;
+ }
+ } else if (msg.type == omx_message::EMPTY_BUFFER_DONE) {
+ if (bufferSource != NULL) {
// This is one of the buffers used exclusively by
// GraphicBufferSource.
// Don't dispatch a message back to ACodec, since it doesn't
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index d5207d5..a347951 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -37,7 +37,7 @@
using namespace android;
-int main(int argc, char** argv)
+int main(int argc __unused, char** argv)
{
signal(SIGPIPE, SIG_IGN);
char value[PROPERTY_VALUE_MAX];
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f9cc17b..7c73b19 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -264,7 +264,7 @@
return NULL;
}
-void AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -296,7 +296,7 @@
}
-void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -311,7 +311,7 @@
write(fd, result.string(), result.size());
}
-void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -559,7 +559,7 @@
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
- LOG_ALWAYS_FATAL_IF((track != 0) != (lStatus == NO_ERROR));
+ LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
// move effect chain to this output thread if an effect on same session was waiting
@@ -1259,7 +1259,7 @@
{
}
-void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
{
sp<NotificationClient> keep(this);
mAudioFlinger->removeNotificationClient(mPid);
@@ -1340,7 +1340,7 @@
frameCount, lSessionId,
IPCThreadState::self()->getCallingUid(),
flags, tid, &lStatus);
- LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
+ LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
}
if (lStatus != NO_ERROR) {
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 59d1a59..f2af9b2 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -536,7 +536,7 @@
// Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
if (!((value == 44100 && devSampleRate == 48000) ||
(value == 48000 && devSampleRate == 44100))) {
- quality = AudioResampler::LOW_QUALITY;
+ quality = AudioResampler::DYN_LOW_QUALITY;
} else {
quality = AudioResampler::DEFAULT_QUALITY;
}
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 415f696..6a115ff 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -613,7 +613,7 @@
return NO_ERROR;
}
-status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
+status_t AudioPolicyService::dump(int fd, const Vector<String16>& args __unused)
{
if (!dumpAllowed()) {
dumpPermissionDenial(fd);
@@ -1461,7 +1461,7 @@
extern "C" {
-static audio_module_handle_t aps_load_hw_module(void *service,
+static audio_module_handle_t aps_load_hw_module(void *service __unused,
const char *name)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -1474,7 +1474,7 @@
}
// deprecated: replaced by aps_open_output_on_module()
-static audio_io_handle_t aps_open_output(void *service,
+static audio_io_handle_t aps_open_output(void *service __unused,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
@@ -1492,7 +1492,7 @@
pLatencyMs, flags);
}
-static audio_io_handle_t aps_open_output_on_module(void *service,
+static audio_io_handle_t aps_open_output_on_module(void *service __unused,
audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
@@ -1511,7 +1511,7 @@
pLatencyMs, flags, offloadInfo);
}
-static audio_io_handle_t aps_open_dup_output(void *service,
+static audio_io_handle_t aps_open_dup_output(void *service __unused,
audio_io_handle_t output1,
audio_io_handle_t output2)
{
@@ -1523,7 +1523,7 @@
return af->openDuplicateOutput(output1, output2);
}
-static int aps_close_output(void *service, audio_io_handle_t output)
+static int aps_close_output(void *service __unused, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1533,7 +1533,7 @@
return af->closeOutput(output);
}
-static int aps_suspend_output(void *service, audio_io_handle_t output)
+static int aps_suspend_output(void *service __unused, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1544,7 +1544,7 @@
return af->suspendOutput(output);
}
-static int aps_restore_output(void *service, audio_io_handle_t output)
+static int aps_restore_output(void *service __unused, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1556,12 +1556,12 @@
}
// deprecated: replaced by aps_open_input_on_module(), and acoustics parameter is ignored
-static audio_io_handle_t aps_open_input(void *service,
+static audio_io_handle_t aps_open_input(void *service __unused,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
audio_channel_mask_t *pChannelMask,
- audio_in_acoustics_t acoustics)
+ audio_in_acoustics_t acoustics __unused)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1572,7 +1572,7 @@
return af->openInput((audio_module_handle_t)0, pDevices, pSamplingRate, pFormat, pChannelMask);
}
-static audio_io_handle_t aps_open_input_on_module(void *service,
+static audio_io_handle_t aps_open_input_on_module(void *service __unused,
audio_module_handle_t module,
audio_devices_t *pDevices,
uint32_t *pSamplingRate,
@@ -1588,7 +1588,7 @@
return af->openInput(module, pDevices, pSamplingRate, pFormat, pChannelMask);
}
-static int aps_close_input(void *service, audio_io_handle_t input)
+static int aps_close_input(void *service __unused, audio_io_handle_t input)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
if (af == 0) {
@@ -1598,7 +1598,7 @@
return af->closeInput(input);
}
-static int aps_set_stream_output(void *service, audio_stream_type_t stream,
+static int aps_set_stream_output(void *service __unused, audio_stream_type_t stream,
audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
@@ -1609,7 +1609,7 @@
return af->setStreamOutput(stream, output);
}
-static int aps_move_effects(void *service, int session,
+static int aps_move_effects(void *service __unused, int session,
audio_io_handle_t src_output,
audio_io_handle_t dst_output)
{
@@ -1621,7 +1621,7 @@
return af->moveEffects(session, src_output, dst_output);
}
-static char * aps_get_parameters(void *service, audio_io_handle_t io_handle,
+static char * aps_get_parameters(void *service __unused, audio_io_handle_t io_handle,
const char *keys)
{
String8 result = AudioSystem::getParameters(io_handle, String8(keys));
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 24ff55d..07574ab 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -158,6 +158,16 @@
atFinalQuality = true;
}
+ /* if the caller requests DEFAULT_QUALITY and af.resampler.property
+ * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
+ * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
+ * due to estimated CPU load of having too many active resamplers
+ * (the code below the if).
+ */
+ if (quality == DEFAULT_QUALITY) {
+ quality = DYN_MED_QUALITY;
+ }
+
// naive implementation of CPU load throttling doesn't account for whether resampler is active
pthread_mutex_lock(&mutex);
for (;;) {
@@ -172,7 +182,6 @@
// not enough CPU available for proposed quality level, so try next lowest level
switch (quality) {
default:
- case DEFAULT_QUALITY:
case LOW_QUALITY:
atFinalQuality = true;
break;
@@ -202,7 +211,6 @@
switch (quality) {
default:
- case DEFAULT_QUALITY:
case LOW_QUALITY:
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
@@ -563,7 +571,7 @@
" 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
@@ -683,7 +691,7 @@
" 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/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 984548d..cd67df5 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -372,11 +372,7 @@
if (locked) {
mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
}
- if (!USE_NEON) {
- stride = 2; // C version only
- }
- // TODO: Remove this for testing
- //stride = 2;
+
mResampleType = RESAMPLETYPE(mChannelCount, locked, stride, !!useS32);
#ifdef DEBUG_RESAMPLER
printf("channels:%d %s stride:%d %s coef:%d shift:%d\n",
@@ -392,7 +388,7 @@
// 24 cases - this perhaps can be reduced later, as testing might take too long
switch (mResampleType) {
- // stride 16 (stride 2 for machines that do not support NEON)
+ // stride 16 (falls back to stride 2 for machines that do not support NEON)
case RESAMPLETYPE(1, true, 16, 0):
return resample<1, true, 16>(out, outFrameCount, mConstants.mFirCoefsS16, provider);
case RESAMPLETYPE(2, true, 16, 0):
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 59b4770..537c076 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -805,7 +805,7 @@
return mOffloaded;
}
-void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 515368c..24df473 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -185,7 +185,11 @@
{
}
-void CpuStats::sample(const String8 &title) {
+void CpuStats::sample(const String8 &title
+#ifndef DEBUG_CPU_USAGE
+ __unused
+#endif
+ ) {
#ifdef DEBUG_CPU_USAGE
// get current thread's delta CPU time in wall clock ns
double wcNs;
@@ -421,7 +425,7 @@
}
}
-void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
+void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -604,7 +608,7 @@
mPowerManager.clear();
}
-void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who)
+void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
@@ -1088,7 +1092,7 @@
dumpEffectChains(fd, args);
}
-void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -1342,7 +1346,7 @@
lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
if (lStatus != NO_ERROR) {
ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus);
- track.clear();
+ // track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
@@ -1614,7 +1618,7 @@
// static
int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event,
- void *param,
+ void *param __unused,
void *cookie)
{
AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie;
@@ -3739,14 +3743,14 @@
}
// getTrackName_l() must be called with ThreadBase::mLock held
-int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask,
- int sessionId)
+int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused,
+ int sessionId __unused)
{
return 0;
}
// deleteTrackName_l() must be called with ThreadBase::mLock held
-void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name)
+void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused)
{
}
@@ -4859,7 +4863,7 @@
lStatus = track->initCheck();
if (lStatus != NO_ERROR) {
ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
- track.clear();
+ // track must be cleared from the caller as the caller has the AF lock
goto Exit;
}
mTracks.add(track);
@@ -5026,12 +5030,12 @@
return false;
}
-bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const
+bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const
{
return false;
}
-status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event)
+status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused)
{
#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
if (!isValidSyncEvent(event)) {
@@ -5109,7 +5113,7 @@
dumpBase(fd, args);
}
-void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args)
+void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -5142,7 +5146,7 @@
}
// AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
{
int32_t rear = mRsmpInRear;
int32_t front = mRsmpInFront;
@@ -5321,7 +5325,7 @@
return out_s8;
}
-void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) {
AudioSystem::OutputDescriptor desc;
const void *param2 = NULL;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 8df6f94..1745ea1 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -201,13 +201,13 @@
// effect
void removeEffect_l(const sp< EffectModule>& effect);
// detach all tracks connected to an auxiliary effect
- virtual void detachAuxEffect_l(int effectId) {}
+ virtual void detachAuxEffect_l(int effectId __unused) {}
// returns either EFFECT_SESSION if effects on this audio session exist in one
// chain, or TRACK_SESSION if tracks on this audio session exist, or both
virtual uint32_t hasAudioSession(int sessionId) const = 0;
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
- virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
+ virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; }
// suspend or restore effect according to the type of effect passed. a NULL
// type pointer means suspend all effects in the session
@@ -625,7 +625,7 @@
sp<NBLog::Writer> mFastMixerNBLogWriter;
public:
virtual bool hasFastMixer() const = 0;
- virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
+ virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
{ FastTrackUnderruns dummy; return dummy; }
protected:
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index d8d7790..ecedca9 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -526,7 +526,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts)
+ AudioBufferProvider::Buffer* buffer, int64_t pts __unused)
{
ServerProxy::Buffer buf;
size_t desiredFrames = buffer->frameCount;
@@ -576,8 +576,8 @@
return false;
}
-status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event,
- int triggerSession)
+status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused,
+ int triggerSession __unused)
{
status_t status = NO_ERROR;
ALOGV("start(%d), calling pid %d session %d",
@@ -1163,7 +1163,7 @@
void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l(
const TimedBuffer& buf,
- const char* logTag) {
+ const char* logTag __unused) {
uint32_t bufBytes = buf.buffer()->size();
uint32_t consumedAlready = buf.position();
@@ -1786,7 +1786,7 @@
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
- int64_t pts)
+ int64_t pts __unused)
{
ServerProxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.cpp b/services/camera/libcameraservice/api1/client2/Parameters.cpp
index f84f019..2cf0d29 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.cpp
+++ b/services/camera/libcameraservice/api1/client2/Parameters.cpp
@@ -664,13 +664,13 @@
focusState = ANDROID_CONTROL_AF_STATE_INACTIVE;
shadowFocusMode = FOCUS_MODE_INVALID;
- camera_metadata_ro_entry_t max3aRegions =
- staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1);
- if (!max3aRegions.count) return NO_INIT;
+ camera_metadata_ro_entry_t max3aRegions = staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION);
+ if (max3aRegions.count != Parameters::NUM_REGION) return NO_INIT;
int32_t maxNumFocusAreas = 0;
if (focusMode != Parameters::FOCUS_MODE_FIXED) {
- maxNumFocusAreas = max3aRegions.data.i32[0];
+ maxNumFocusAreas = max3aRegions.data.i32[Parameters::REGION_AF];
}
params.set(CameraParameters::KEY_MAX_NUM_FOCUS_AREAS, maxNumFocusAreas);
params.set(CameraParameters::KEY_FOCUS_AREAS,
@@ -730,7 +730,7 @@
meteringAreas.add(Parameters::Area(0, 0, 0, 0, 0));
params.set(CameraParameters::KEY_MAX_NUM_METERING_AREAS,
- max3aRegions.data.i32[0]);
+ max3aRegions.data.i32[Parameters::REGION_AE]);
params.set(CameraParameters::KEY_METERING_AREAS,
"(0,0,0,0,0)");
@@ -1591,10 +1591,11 @@
// FOCUS_AREAS
res = parseAreas(newParams.get(CameraParameters::KEY_FOCUS_AREAS),
&validatedParams.focusingAreas);
- size_t max3aRegions =
- (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS, 1, 1).data.i32[0];
+ size_t maxAfRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION).
+ data.i32[Parameters::REGION_AF];
if (res == OK) res = validateAreas(validatedParams.focusingAreas,
- max3aRegions, AREA_KIND_FOCUS);
+ maxAfRegions, AREA_KIND_FOCUS);
if (res != OK) {
ALOGE("%s: Requested focus areas are malformed: %s",
__FUNCTION__, newParams.get(CameraParameters::KEY_FOCUS_AREAS));
@@ -1624,10 +1625,13 @@
newParams.get(CameraParameters::KEY_AUTO_WHITEBALANCE_LOCK));
// METERING_AREAS
+ size_t maxAeRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION).
+ data.i32[Parameters::REGION_AE];
res = parseAreas(newParams.get(CameraParameters::KEY_METERING_AREAS),
&validatedParams.meteringAreas);
if (res == OK) {
- res = validateAreas(validatedParams.meteringAreas, max3aRegions,
+ res = validateAreas(validatedParams.meteringAreas, maxAeRegions,
AREA_KIND_METERING);
}
if (res != OK) {
@@ -1909,6 +1913,23 @@
reqMeteringAreas, reqMeteringAreasSize);
if (res != OK) return res;
+ // Set awb regions to be the same as the metering regions if allowed
+ size_t maxAwbRegions = (size_t)staticInfo(ANDROID_CONTROL_MAX_REGIONS,
+ Parameters::NUM_REGION, Parameters::NUM_REGION).
+ data.i32[Parameters::REGION_AWB];
+ if (maxAwbRegions > 0) {
+ if (maxAwbRegions >= meteringAreas.size()) {
+ res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+ reqMeteringAreas, reqMeteringAreasSize);
+ } else {
+ // Ensure the awb regions are zeroed if the region count is too high.
+ int32_t zeroedAwbAreas[5] = {0, 0, 0, 0, 0};
+ res = request->update(ANDROID_CONTROL_AWB_REGIONS,
+ zeroedAwbAreas, sizeof(zeroedAwbAreas)/sizeof(int32_t));
+ }
+ if (res != OK) return res;
+ }
+
delete[] reqMeteringAreas;
/* don't include jpeg thumbnail size - it's valid for
diff --git a/services/camera/libcameraservice/api1/client2/Parameters.h b/services/camera/libcameraservice/api1/client2/Parameters.h
index 32dbd42..60c4687 100644
--- a/services/camera/libcameraservice/api1/client2/Parameters.h
+++ b/services/camera/libcameraservice/api1/client2/Parameters.h
@@ -113,6 +113,14 @@
bool autoExposureLock;
bool autoWhiteBalanceLock;
+ // 3A region types, for use with ANDROID_CONTROL_MAX_REGIONS
+ enum region_t {
+ REGION_AE = 0,
+ REGION_AWB,
+ REGION_AF,
+ NUM_REGION // Number of region types
+ } region;
+
Vector<Area> meteringAreas;
int zoom;
diff --git a/services/medialog/MediaLogService.cpp b/services/medialog/MediaLogService.cpp
index 683fdf3..0c7fbbd 100644
--- a/services/medialog/MediaLogService.cpp
+++ b/services/medialog/MediaLogService.cpp
@@ -54,7 +54,7 @@
}
}
-status_t MediaLogService::dump(int fd, const Vector<String16>& args)
+status_t MediaLogService::dump(int fd, const Vector<String16>& args __unused)
{
// FIXME merge with similar but not identical code at services/audioflinger/ServiceUtilities.cpp
static const String16 sDump("android.permission.DUMP");