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-rw-r--r--media/libmedia/Android.mk3
-rw-r--r--media/libmedia/AudioPolicy.cpp115
-rw-r--r--media/libmedia/AudioRecord.cpp31
-rw-r--r--media/libmedia/AudioSystem.cpp310
-rw-r--r--media/libmedia/AudioTrack.cpp240
-rw-r--r--media/libmedia/AudioTrackShared.cpp99
-rw-r--r--media/libmedia/IAudioPolicyService.cpp225
-rw-r--r--media/libmedia/ICrypto.cpp22
-rw-r--r--media/libmedia/IDrm.cpp43
-rw-r--r--media/libmedia/Visualizer.cpp19
10 files changed, 684 insertions, 423 deletions
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index e012116..a2e0909 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -59,7 +59,8 @@ LOCAL_SRC_FILES:= \
MemoryLeakTrackUtil.cpp \
SoundPool.cpp \
SoundPoolThread.cpp \
- StringArray.cpp
+ StringArray.cpp \
+ AudioPolicy.cpp
LOCAL_SRC_FILES += ../libnbaio/roundup.c
diff --git a/media/libmedia/AudioPolicy.cpp b/media/libmedia/AudioPolicy.cpp
new file mode 100644
index 0000000..d2d0971
--- /dev/null
+++ b/media/libmedia/AudioPolicy.cpp
@@ -0,0 +1,115 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicy"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+#include <media/AudioPolicy.h>
+
+namespace android {
+
+//
+// AttributeMatchCriterion implementation
+//
+AttributeMatchCriterion::AttributeMatchCriterion(audio_usage_t usage,
+ audio_source_t source,
+ uint32_t rule)
+: mRule(rule)
+{
+ if (mRule == RULE_MATCH_ATTRIBUTE_USAGE ||
+ mRule == RULE_EXCLUDE_ATTRIBUTE_USAGE) {
+ mAttr.mUsage = usage;
+ } else {
+ mAttr.mSource = source;
+ }
+}
+
+status_t AttributeMatchCriterion::readFromParcel(Parcel *parcel)
+{
+ mRule = parcel->readInt32();
+ if (mRule == RULE_MATCH_ATTRIBUTE_USAGE ||
+ mRule == RULE_EXCLUDE_ATTRIBUTE_USAGE) {
+ mAttr.mUsage = (audio_usage_t)parcel->readInt32();
+ } else {
+ mAttr.mSource = (audio_source_t)parcel->readInt32();
+ }
+ return NO_ERROR;
+}
+
+status_t AttributeMatchCriterion::writeToParcel(Parcel *parcel) const
+{
+ parcel->writeInt32(mRule);
+ parcel->writeInt32(mAttr.mUsage);
+ return NO_ERROR;
+}
+
+//
+// AudioMix implementation
+//
+
+status_t AudioMix::readFromParcel(Parcel *parcel)
+{
+ mMixType = parcel->readInt32();
+ mFormat.sample_rate = (uint32_t)parcel->readInt32();
+ mFormat.channel_mask = (audio_channel_mask_t)parcel->readInt32();
+ mFormat.format = (audio_format_t)parcel->readInt32();
+ mRouteFlags = parcel->readInt32();
+ mRegistrationId = parcel->readString8();
+ size_t size = (size_t)parcel->readInt32();
+ if (size > MAX_CRITERIA_PER_MIX) {
+ size = MAX_CRITERIA_PER_MIX;
+ }
+ for (size_t i = 0; i < size; i++) {
+ AttributeMatchCriterion criterion;
+ if (criterion.readFromParcel(parcel) == NO_ERROR) {
+ mCriteria.add(criterion);
+ }
+ }
+ return NO_ERROR;
+}
+
+status_t AudioMix::writeToParcel(Parcel *parcel) const
+{
+ parcel->writeInt32(mMixType);
+ parcel->writeInt32(mFormat.sample_rate);
+ parcel->writeInt32(mFormat.channel_mask);
+ parcel->writeInt32(mFormat.format);
+ parcel->writeInt32(mRouteFlags);
+ parcel->writeString8(mRegistrationId);
+ size_t size = mCriteria.size();
+ if (size > MAX_CRITERIA_PER_MIX) {
+ size = MAX_CRITERIA_PER_MIX;
+ }
+ size_t sizePosition = parcel->dataPosition();
+ parcel->writeInt32(size);
+ size_t finalSize = size;
+ for (size_t i = 0; i < size; i++) {
+ size_t position = parcel->dataPosition();
+ if (mCriteria[i].writeToParcel(parcel) != NO_ERROR) {
+ parcel->setDataPosition(position);
+ finalSize--;
+ }
+ }
+ if (size != finalSize) {
+ size_t position = parcel->dataPosition();
+ parcel->setDataPosition(sizePosition);
+ parcel->writeInt32(finalSize);
+ parcel->setDataPosition(position);
+ }
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 9e7ba88..ca3832d 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -82,14 +82,16 @@ AudioRecord::AudioRecord(
uint32_t notificationFrames,
int sessionId,
transfer_type transferType,
- audio_input_flags_t flags)
+ audio_input_flags_t flags,
+ const audio_attributes_t* pAttributes)
: mStatus(NO_INIT), mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL),
mPreviousSchedulingGroup(SP_DEFAULT),
mProxy(NULL)
{
mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
- notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags);
+ notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
+ pAttributes);
}
AudioRecord::~AudioRecord()
@@ -126,7 +128,8 @@ status_t AudioRecord::set(
bool threadCanCallJava,
int sessionId,
transfer_type transferType,
- audio_input_flags_t flags)
+ audio_input_flags_t flags,
+ const audio_attributes_t* pAttributes)
{
ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
"notificationFrames %u, sessionId %d, transferType %d, flags %#x",
@@ -164,11 +167,15 @@ status_t AudioRecord::set(
return INVALID_OPERATION;
}
- // handle default values first.
- if (inputSource == AUDIO_SOURCE_DEFAULT) {
- inputSource = AUDIO_SOURCE_MIC;
+ if (pAttributes == NULL) {
+ memset(&mAttributes, 0, sizeof(audio_attributes_t));
+ mAttributes.source = inputSource;
+ } else {
+ // stream type shouldn't be looked at, this track has audio attributes
+ memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
+ ALOGV("Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",
+ mAttributes.source, mAttributes.flags, mAttributes.tags);
}
- mInputSource = inputSource;
if (sampleRate == 0) {
ALOGE("Invalid sample rate %u", sampleRate);
@@ -444,12 +451,14 @@ status_t AudioRecord::openRecord_l(size_t epoch)
}
}
- audio_io_handle_t input = AudioSystem::getInput(mInputSource, mSampleRate, mFormat,
- mChannelMask, mSessionId, mFlags);
- if (input == AUDIO_IO_HANDLE_NONE) {
+ audio_io_handle_t input;
+ status = AudioSystem::getInputForAttr(&mAttributes, &input, (audio_session_t)mSessionId,
+ mSampleRate, mFormat, mChannelMask, mFlags);
+
+ if (status != NO_ERROR) {
ALOGE("Could not get audio input for record source %d, sample rate %u, format %#x, "
"channel mask %#x, session %d, flags %#x",
- mInputSource, mSampleRate, mFormat, mChannelMask, mSessionId, mFlags);
+ mAttributes.source, mSampleRate, mFormat, mChannelMask, mSessionId, mFlags);
return BAD_VALUE;
}
{
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index dda3657..9cae21c 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -32,6 +32,9 @@ namespace android {
// client singleton for AudioFlinger binder interface
Mutex AudioSystem::gLock;
+Mutex AudioSystem::gLockCache;
+Mutex AudioSystem::gLockAPS;
+Mutex AudioSystem::gLockAPC;
sp<IAudioFlinger> AudioSystem::gAudioFlinger;
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
@@ -48,33 +51,40 @@ size_t AudioSystem::gInBuffSize = 0; // zero indicates cache is invalid
sp<AudioSystem::AudioPortCallback> AudioSystem::gAudioPortCallback;
// establish binder interface to AudioFlinger service
-const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
-{
- Mutex::Autolock _l(gLock);
- if (gAudioFlinger == 0) {
- sp<IServiceManager> sm = defaultServiceManager();
- sp<IBinder> binder;
- do {
- binder = sm->getService(String16("media.audio_flinger"));
- if (binder != 0)
- break;
- ALOGW("AudioFlinger not published, waiting...");
- usleep(500000); // 0.5 s
- } while (true);
- if (gAudioFlingerClient == NULL) {
- gAudioFlingerClient = new AudioFlingerClient();
- } else {
- if (gAudioErrorCallback) {
- gAudioErrorCallback(NO_ERROR);
+const sp<IAudioFlinger> AudioSystem::get_audio_flinger()
+{
+ sp<IAudioFlinger> af;
+ sp<AudioFlingerClient> afc;
+ {
+ Mutex::Autolock _l(gLock);
+ if (gAudioFlinger == 0) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder;
+ do {
+ binder = sm->getService(String16("media.audio_flinger"));
+ if (binder != 0)
+ break;
+ ALOGW("AudioFlinger not published, waiting...");
+ usleep(500000); // 0.5 s
+ } while (true);
+ if (gAudioFlingerClient == NULL) {
+ gAudioFlingerClient = new AudioFlingerClient();
+ } else {
+ if (gAudioErrorCallback) {
+ gAudioErrorCallback(NO_ERROR);
+ }
}
+ binder->linkToDeath(gAudioFlingerClient);
+ gAudioFlinger = interface_cast<IAudioFlinger>(binder);
+ LOG_ALWAYS_FATAL_IF(gAudioFlinger == 0);
+ afc = gAudioFlingerClient;
}
- binder->linkToDeath(gAudioFlingerClient);
- gAudioFlinger = interface_cast<IAudioFlinger>(binder);
- gAudioFlinger->registerClient(gAudioFlingerClient);
+ af = gAudioFlinger;
}
- ALOGE_IF(gAudioFlinger==0, "no AudioFlinger!?");
-
- return gAudioFlinger;
+ if (afc != 0) {
+ af->registerClient(afc);
+ }
+ return af;
}
/* static */ status_t AudioSystem::checkAudioFlinger()
@@ -245,36 +255,23 @@ status_t AudioSystem::getOutputSamplingRate(uint32_t* samplingRate, audio_stream
return getSamplingRate(output, samplingRate);
}
-status_t AudioSystem::getOutputSamplingRateForAttr(uint32_t* samplingRate,
- const audio_attributes_t *attr)
-{
- if (attr == NULL) {
- return BAD_VALUE;
- }
- audio_io_handle_t output = getOutputForAttr(attr);
- if (output == 0) {
- return PERMISSION_DENIED;
- }
- return getSamplingRate(output, samplingRate);
-}
-
status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
uint32_t* samplingRate)
{
- OutputDescriptor *outputDesc;
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) return PERMISSION_DENIED;
- gLock.lock();
- outputDesc = AudioSystem::gOutputs.valueFor(output);
+ Mutex::Autolock _l(gLockCache);
+
+ OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);
if (outputDesc == NULL) {
ALOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
- gLock.unlock();
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) return PERMISSION_DENIED;
+ gLockCache.unlock();
*samplingRate = af->sampleRate(output);
+ gLockCache.lock();
} else {
ALOGV("getOutputSamplingRate() reading from output desc");
*samplingRate = outputDesc->samplingRate;
- gLock.unlock();
}
if (*samplingRate == 0) {
ALOGE("AudioSystem::getSamplingRate failed for output %d", output);
@@ -305,18 +302,18 @@ status_t AudioSystem::getOutputFrameCount(size_t* frameCount, audio_stream_type_
status_t AudioSystem::getFrameCount(audio_io_handle_t output,
size_t* frameCount)
{
- OutputDescriptor *outputDesc;
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) return PERMISSION_DENIED;
+
+ Mutex::Autolock _l(gLockCache);
- gLock.lock();
- outputDesc = AudioSystem::gOutputs.valueFor(output);
+ OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);
if (outputDesc == NULL) {
- gLock.unlock();
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) return PERMISSION_DENIED;
+ gLockCache.unlock();
*frameCount = af->frameCount(output);
+ gLockCache.lock();
} else {
*frameCount = outputDesc->frameCount;
- gLock.unlock();
}
if (*frameCount == 0) {
ALOGE("AudioSystem::getFrameCount failed for output %d", output);
@@ -347,18 +344,18 @@ status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t st
status_t AudioSystem::getLatency(audio_io_handle_t output,
uint32_t* latency)
{
- OutputDescriptor *outputDesc;
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) return PERMISSION_DENIED;
- gLock.lock();
- outputDesc = AudioSystem::gOutputs.valueFor(output);
+ Mutex::Autolock _l(gLockCache);
+
+ OutputDescriptor *outputDesc = AudioSystem::gOutputs.valueFor(output);
if (outputDesc == NULL) {
- gLock.unlock();
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) return PERMISSION_DENIED;
+ gLockCache.unlock();
*latency = af->latency(output);
+ gLockCache.lock();
} else {
*latency = outputDesc->latency;
- gLock.unlock();
}
ALOGV("getLatency() output %d, latency %d", output, *latency);
@@ -369,24 +366,24 @@ status_t AudioSystem::getLatency(audio_io_handle_t output,
status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask, size_t* buffSize)
{
- gLock.lock();
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+ Mutex::Autolock _l(gLockCache);
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
size_t inBuffSize = gInBuffSize;
if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
|| (channelMask != gPrevInChannelMask)) {
- gLock.unlock();
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) {
- return PERMISSION_DENIED;
- }
+ gLockCache.unlock();
inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
+ gLockCache.lock();
if (inBuffSize == 0) {
ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
// A benign race is possible here: we could overwrite a fresher cache entry
- gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
@@ -394,7 +391,6 @@ status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t for
gInBuffSize = inBuffSize;
}
- gLock.unlock();
*buffSize = inBuffSize;
return NO_ERROR;
@@ -461,14 +457,21 @@ audio_hw_sync_t AudioSystem::getAudioHwSyncForSession(audio_session_t sessionId)
void AudioSystem::AudioFlingerClient::binderDied(const wp<IBinder>& who __unused)
{
- Mutex::Autolock _l(AudioSystem::gLock);
+ audio_error_callback cb = NULL;
+ {
+ Mutex::Autolock _l(AudioSystem::gLock);
+ AudioSystem::gAudioFlinger.clear();
+ cb = gAudioErrorCallback;
+ }
- AudioSystem::gAudioFlinger.clear();
- // clear output handles and stream to output map caches
- AudioSystem::gOutputs.clear();
+ {
+ // clear output handles and stream to output map caches
+ Mutex::Autolock _l(gLockCache);
+ AudioSystem::gOutputs.clear();
+ }
- if (gAudioErrorCallback) {
- gAudioErrorCallback(DEAD_OBJECT);
+ if (cb) {
+ cb(DEAD_OBJECT);
}
ALOGW("AudioFlinger server died!");
}
@@ -481,7 +484,7 @@ void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle
if (ioHandle == AUDIO_IO_HANDLE_NONE) return;
- Mutex::Autolock _l(AudioSystem::gLock);
+ Mutex::Autolock _l(AudioSystem::gLockCache);
switch (event) {
case STREAM_CONFIG_CHANGED:
@@ -543,55 +546,44 @@ void AudioSystem::setErrorCallback(audio_error_callback cb)
gAudioErrorCallback = cb;
}
-
-bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType)
-{
- switch (streamType) {
- case AUDIO_STREAM_MUSIC:
- case AUDIO_STREAM_VOICE_CALL:
- case AUDIO_STREAM_BLUETOOTH_SCO:
- case AUDIO_STREAM_SYSTEM:
- return true;
- default:
- return false;
- }
-}
-
-
// client singleton for AudioPolicyService binder interface
+// protected by gLockAPS
sp<IAudioPolicyService> AudioSystem::gAudioPolicyService;
sp<AudioSystem::AudioPolicyServiceClient> AudioSystem::gAudioPolicyServiceClient;
// establish binder interface to AudioPolicy service
-const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
-{
- gLock.lock();
- if (gAudioPolicyService == 0) {
- sp<IServiceManager> sm = defaultServiceManager();
- sp<IBinder> binder;
- do {
- binder = sm->getService(String16("media.audio_policy"));
- if (binder != 0)
- break;
- ALOGW("AudioPolicyService not published, waiting...");
- usleep(500000); // 0.5 s
- } while (true);
- if (gAudioPolicyServiceClient == NULL) {
- gAudioPolicyServiceClient = new AudioPolicyServiceClient();
+const sp<IAudioPolicyService> AudioSystem::get_audio_policy_service()
+{
+ sp<IAudioPolicyService> ap;
+ sp<AudioPolicyServiceClient> apc;
+ {
+ Mutex::Autolock _l(gLockAPS);
+ if (gAudioPolicyService == 0) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder;
+ do {
+ binder = sm->getService(String16("media.audio_policy"));
+ if (binder != 0)
+ break;
+ ALOGW("AudioPolicyService not published, waiting...");
+ usleep(500000); // 0.5 s
+ } while (true);
+ if (gAudioPolicyServiceClient == NULL) {
+ gAudioPolicyServiceClient = new AudioPolicyServiceClient();
+ }
+ binder->linkToDeath(gAudioPolicyServiceClient);
+ gAudioPolicyService = interface_cast<IAudioPolicyService>(binder);
+ LOG_ALWAYS_FATAL_IF(gAudioPolicyService == 0);
+ apc = gAudioPolicyServiceClient;
}
- binder->linkToDeath(gAudioPolicyServiceClient);
- gAudioPolicyService = interface_cast<IAudioPolicyService>(binder);
- gLock.unlock();
- // Registering the client takes the AudioPolicyService lock.
- // Don't hold the AudioSystem lock at the same time.
- gAudioPolicyService->registerClient(gAudioPolicyServiceClient);
- } else {
- // There exists a benign race condition where gAudioPolicyService
- // is set, but gAudioPolicyServiceClient is not yet registered.
- gLock.unlock();
+ ap = gAudioPolicyService;
+ }
+ if (apc != 0) {
+ ap->registerClient(apc);
}
- return gAudioPolicyService;
+
+ return ap;
}
// ---------------------------------------------------------------------------
@@ -657,22 +649,26 @@ audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
return aps->getOutput(stream, samplingRate, format, channelMask, flags, offloadInfo);
}
-audio_io_handle_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
+status_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
- if (attr == NULL) return 0;
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return 0;
- return aps->getOutputForAttr(attr, samplingRate, format, channelMask, flags, offloadInfo);
+ if (aps == 0) return NO_INIT;
+ return aps->getOutputForAttr(attr, output, session, stream,
+ samplingRate, format, channelMask,
+ flags, offloadInfo);
}
status_t AudioSystem::startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session)
+ audio_session_t session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
@@ -681,30 +677,33 @@ status_t AudioSystem::startOutput(audio_io_handle_t output,
status_t AudioSystem::stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session)
+ audio_session_t session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->stopOutput(output, stream, session);
}
-void AudioSystem::releaseOutput(audio_io_handle_t output)
+void AudioSystem::releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return;
- aps->releaseOutput(output);
+ aps->releaseOutput(output, stream, session);
}
-audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- int sessionId,
- audio_input_flags_t flags)
+status_t AudioSystem::getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return 0;
- return aps->getInput(inputSource, samplingRate, format, channelMask, sessionId, flags);
+ if (aps == 0) return NO_INIT;
+ return aps->getInputForAttr(attr, input, session, samplingRate, format, channelMask, flags);
}
status_t AudioSystem::startInput(audio_io_handle_t input,
@@ -856,9 +855,21 @@ status_t AudioSystem::setLowRamDevice(bool isLowRamDevice)
void AudioSystem::clearAudioConfigCache()
{
- Mutex::Autolock _l(gLock);
+ // called by restoreTrack_l(), which needs new IAudioFlinger and IAudioPolicyService instances
ALOGV("clearAudioConfigCache()");
- gOutputs.clear();
+ {
+ Mutex::Autolock _l(gLockCache);
+ gOutputs.clear();
+ }
+ {
+ Mutex::Autolock _l(gLock);
+ gAudioFlinger.clear();
+ }
+ {
+ Mutex::Autolock _l(gLockAPS);
+ gAudioPolicyService.clear();
+ }
+ // Do not clear gAudioPortCallback
}
bool AudioSystem::isOffloadSupported(const audio_offload_info_t& info)
@@ -920,7 +931,7 @@ status_t AudioSystem::setAudioPortConfig(const struct audio_port_config *config)
void AudioSystem::setAudioPortCallback(sp<AudioPortCallback> callBack)
{
- Mutex::Autolock _l(gLock);
+ Mutex::Autolock _l(gLockAPC);
gAudioPortCallback = callBack;
}
@@ -947,23 +958,34 @@ audio_mode_t AudioSystem::getPhoneState()
return aps->getPhoneState();
}
+status_t AudioSystem::registerPolicyMixes(Vector<AudioMix> mixes, bool registration)
+{
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return PERMISSION_DENIED;
+ return aps->registerPolicyMixes(mixes, registration);
+}
// ---------------------------------------------------------------------------
void AudioSystem::AudioPolicyServiceClient::binderDied(const wp<IBinder>& who __unused)
{
- Mutex::Autolock _l(gLock);
- if (gAudioPortCallback != 0) {
- gAudioPortCallback->onServiceDied();
+ {
+ Mutex::Autolock _l(gLockAPC);
+ if (gAudioPortCallback != 0) {
+ gAudioPortCallback->onServiceDied();
+ }
+ }
+ {
+ Mutex::Autolock _l(gLockAPS);
+ AudioSystem::gAudioPolicyService.clear();
}
- AudioSystem::gAudioPolicyService.clear();
ALOGW("AudioPolicyService server died!");
}
void AudioSystem::AudioPolicyServiceClient::onAudioPortListUpdate()
{
- Mutex::Autolock _l(gLock);
+ Mutex::Autolock _l(gLockAPC);
if (gAudioPortCallback != 0) {
gAudioPortCallback->onAudioPortListUpdate();
}
@@ -971,7 +993,7 @@ void AudioSystem::AudioPolicyServiceClient::onAudioPortListUpdate()
void AudioSystem::AudioPolicyServiceClient::onAudioPatchListUpdate()
{
- Mutex::Autolock _l(gLock);
+ Mutex::Autolock _l(gLockAPC);
if (gAudioPortCallback != 0) {
gAudioPortCallback->onAudioPatchListUpdate();
}
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 0a89fbb..389aacc 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -28,6 +28,7 @@
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
#include <media/IAudioFlinger.h>
+#include <media/AudioPolicyHelper.h>
#include <media/AudioResamplerPublic.h>
#define WAIT_PERIOD_MS 10
@@ -281,38 +282,21 @@ status_t AudioTrack::set(
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
-
if (pAttributes == NULL) {
- if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+ if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
ALOGE("Invalid stream type %d", streamType);
return BAD_VALUE;
}
- setAttributesFromStreamType(streamType);
mStreamType = streamType;
+
} else {
- if (!isValidAttributes(pAttributes)) {
- ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
- pAttributes->usage, pAttributes->content_type, pAttributes->flags,
- pAttributes->tags);
- }
// stream type shouldn't be looked at, this track has audio attributes
memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
- setStreamTypeFromAttributes(mAttributes);
ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
+ mStreamType = AUDIO_STREAM_DEFAULT;
}
- status_t status;
- if (sampleRate == 0) {
- status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes);
- if (status != NO_ERROR) {
- ALOGE("Could not get output sample rate for stream type %d; status %d",
- mStreamType, status);
- return status;
- }
- }
- mSampleRate = sampleRate;
-
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
@@ -350,9 +334,10 @@ status_t AudioTrack::set(
// FIXME why can't we allow direct AND fast?
((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
}
- // only allow deep buffering for music stream type
- if (mStreamType != AUDIO_STREAM_MUSIC) {
- flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+
+ // force direct flag if HW A/V sync requested
+ if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
+ flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
}
if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
@@ -371,6 +356,12 @@ status_t AudioTrack::set(
// so no need to check for specific PCM formats here
}
+ // sampling rate must be specified for direct outputs
+ if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+ return BAD_VALUE;
+ }
+ mSampleRate = sampleRate;
+
// 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
@@ -388,7 +379,11 @@ status_t AudioTrack::set(
mReqFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mNotificationFramesAct = 0;
- mSessionId = sessionId;
+ if (sessionId == AUDIO_SESSION_ALLOCATE) {
+ mSessionId = AudioSystem::newAudioUniqueId();
+ } else {
+ mSessionId = sessionId;
+ }
int callingpid = IPCThreadState::self()->getCallingPid();
int mypid = getpid();
if (uid == -1 || (callingpid != mypid)) {
@@ -411,7 +406,7 @@ status_t AudioTrack::set(
}
// create the IAudioTrack
- status = createTrack_l();
+ status_t status = createTrack_l();
if (status != NO_ERROR) {
if (mAudioTrackThread != 0) {
@@ -678,15 +673,18 @@ status_t AudioTrack::setSampleRate(uint32_t rate)
return INVALID_OPERATION;
}
+ AutoMutex lock(mLock);
+ if (mOutput == AUDIO_IO_HANDLE_NONE) {
+ return NO_INIT;
+ }
uint32_t afSamplingRate;
- if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {
+ if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) {
return NO_INIT;
}
if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
return BAD_VALUE;
}
- AutoMutex lock(mLock);
mSampleRate = rate;
mProxy->setSampleRate(rate);
@@ -742,8 +740,7 @@ status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount
void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
- // FIXME If setting a loop also sets position to start of loop, then
- // this is correct. Otherwise it should be removed.
+ // Setting the loop will reset next notification update period (like setPosition).
mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0;
mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
@@ -859,6 +856,10 @@ status_t AudioTrack::getPosition(uint32_t *position)
// due to hardware latency. We leave this behavior for now.
*position = dspFrames;
} else {
+ if (mCblk->mFlags & CBLK_INVALID) {
+ restoreTrack_l("getPosition");
+ }
+
// IAudioTrack::stop() isn't synchronous; we don't know when presentation completes
*position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ?
0 : updateAndGetPosition_l();
@@ -915,24 +916,38 @@ status_t AudioTrack::attachAuxEffect(int effectId)
return status;
}
+audio_stream_type_t AudioTrack::streamType() const
+{
+ if (mStreamType == AUDIO_STREAM_DEFAULT) {
+ return audio_attributes_to_stream_type(&mAttributes);
+ }
+ return mStreamType;
+}
+
// -------------------------------------------------------------------------
// must be called with mLock held
status_t AudioTrack::createTrack_l()
{
- status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("Could not get audioflinger");
return NO_INIT;
}
- audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat,
- mChannelMask, mFlags, mOffloadInfo);
- if (output == AUDIO_IO_HANDLE_NONE) {
+ audio_io_handle_t output;
+ audio_stream_type_t streamType = mStreamType;
+ audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL;
+ status_t status = AudioSystem::getOutputForAttr(attr, &output,
+ (audio_session_t)mSessionId, &streamType,
+ mSampleRate, mFormat, mChannelMask,
+ mFlags, mOffloadInfo);
+
+
+ if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x,"
" channel mask %#x, flags %#x",
- mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
+ streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
return BAD_VALUE;
}
{
@@ -961,7 +976,9 @@ status_t AudioTrack::createTrack_l()
ALOGE("getSamplingRate(output=%d) status %d", output, status);
goto release;
}
-
+ if (mSampleRate == 0) {
+ mSampleRate = afSampleRate;
+ }
// Client decides whether the track is TIMED (see below), but can only express a preference
// for FAST. Server will perform additional tests.
if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !((
@@ -1084,7 +1101,7 @@ status_t AudioTrack::createTrack_l()
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,
+ sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
mSampleRate,
// AudioFlinger only sees 16-bit PCM
mFormat == AUDIO_FORMAT_PCM_8_BIT &&
@@ -1218,7 +1235,11 @@ status_t AudioTrack::createTrack_l()
mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
mProxy = mStaticProxy;
}
- mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY);
+
+ mProxy->setVolumeLR(gain_minifloat_pack(
+ gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
+ gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));
+
mProxy->setSendLevel(mSendLevel);
mProxy->setSampleRate(mSampleRate);
mProxy->setMinimum(mNotificationFramesAct);
@@ -1230,7 +1251,7 @@ status_t AudioTrack::createTrack_l()
}
release:
- AudioSystem::releaseOutput(output);
+ AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId);
if (status == NO_ERROR) {
status = NO_INIT;
}
@@ -1826,7 +1847,7 @@ status_t AudioTrack::restoreTrack_l(const char *from)
status_t result;
// refresh the audio configuration cache in this process to make sure we get new
- // output parameters in createTrack_l()
+ // output parameters and new IAudioFlinger in createTrack_l()
AudioSystem::clearAudioConfigCache();
if (isOffloadedOrDirect_l()) {
@@ -1934,6 +1955,10 @@ status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
break;
}
+ if (mCblk->mFlags & CBLK_INVALID) {
+ restoreTrack_l("getTimestamp");
+ }
+
// The presented frame count must always lag behind the consumed frame count.
// To avoid a race, read the presented frames first. This ensures that presented <= consumed.
status_t status = mAudioTrack->getTimestamp(timestamp);
@@ -2064,143 +2089,6 @@ uint32_t AudioTrack::getUnderrunFrames() const
return mProxy->getUnderrunFrames();
}
-void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) {
- mAttributes.flags = 0x0;
-
- switch(streamType) {
- case AUDIO_STREAM_DEFAULT:
- case AUDIO_STREAM_MUSIC:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
- mAttributes.usage = AUDIO_USAGE_MEDIA;
- break;
- case AUDIO_STREAM_VOICE_CALL:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
- mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
- break;
- case AUDIO_STREAM_ENFORCED_AUDIBLE:
- mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED;
- // intended fall through, attributes in common with STREAM_SYSTEM
- case AUDIO_STREAM_SYSTEM:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
- break;
- case AUDIO_STREAM_RING:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
- break;
- case AUDIO_STREAM_ALARM:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- mAttributes.usage = AUDIO_USAGE_ALARM;
- break;
- case AUDIO_STREAM_NOTIFICATION:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- mAttributes.usage = AUDIO_USAGE_NOTIFICATION;
- break;
- case AUDIO_STREAM_BLUETOOTH_SCO:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
- mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
- mAttributes.flags |= AUDIO_FLAG_SCO;
- break;
- case AUDIO_STREAM_DTMF:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
- mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
- break;
- case AUDIO_STREAM_TTS:
- mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
- mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
- break;
- default:
- ALOGE("invalid stream type %d when converting to attributes", streamType);
- }
-}
-
-void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) {
- // flags to stream type mapping
- if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
- mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE;
- return;
- }
- if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
- mStreamType = AUDIO_STREAM_BLUETOOTH_SCO;
- return;
- }
-
- // usage to stream type mapping
- switch (aa.usage) {
- case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
- // TODO once AudioPolicyManager fully supports audio_attributes_t,
- // remove stream change based on phone state
- if (AudioSystem::getPhoneState() == AUDIO_MODE_RINGTONE) {
- mStreamType = AUDIO_STREAM_RING;
- break;
- }
- /// FALL THROUGH
- case AUDIO_USAGE_MEDIA:
- case AUDIO_USAGE_GAME:
- case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
- mStreamType = AUDIO_STREAM_MUSIC;
- return;
- case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
- mStreamType = AUDIO_STREAM_SYSTEM;
- return;
- case AUDIO_USAGE_VOICE_COMMUNICATION:
- mStreamType = AUDIO_STREAM_VOICE_CALL;
- return;
-
- case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
- mStreamType = AUDIO_STREAM_DTMF;
- return;
-
- case AUDIO_USAGE_ALARM:
- mStreamType = AUDIO_STREAM_ALARM;
- return;
- case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
- mStreamType = AUDIO_STREAM_RING;
- return;
-
- case AUDIO_USAGE_NOTIFICATION:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
- case AUDIO_USAGE_NOTIFICATION_EVENT:
- mStreamType = AUDIO_STREAM_NOTIFICATION;
- return;
-
- case AUDIO_USAGE_UNKNOWN:
- default:
- mStreamType = AUDIO_STREAM_MUSIC;
- }
-}
-
-bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) {
- // has flags that map to a strategy?
- if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) {
- return true;
- }
-
- // has known usage?
- switch (paa->usage) {
- case AUDIO_USAGE_UNKNOWN:
- case AUDIO_USAGE_MEDIA:
- case AUDIO_USAGE_VOICE_COMMUNICATION:
- case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
- case AUDIO_USAGE_ALARM:
- case AUDIO_USAGE_NOTIFICATION:
- case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
- case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
- case AUDIO_USAGE_NOTIFICATION_EVENT:
- case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
- case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
- case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
- case AUDIO_USAGE_GAME:
- break;
- default:
- return false;
- }
- return true;
-}
// =========================================================================
void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index eec025e..ff24475 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -25,6 +25,12 @@
namespace android {
+// used to clamp a value to size_t. TODO: move to another file.
+template <typename T>
+size_t clampToSize(T x) {
+ return x > SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
+}
+
audio_track_cblk_t::audio_track_cblk_t()
: mServer(0), mFutex(0), mMinimum(0),
mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0), mFlags(0)
@@ -301,6 +307,7 @@ void ClientProxy::binderDied()
{
audio_track_cblk_t* cblk = mCblk;
if (!(android_atomic_or(CBLK_INVALID, &cblk->mFlags) & CBLK_INVALID)) {
+ android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
// it seems that a FUTEX_WAKE_PRIVATE will not wake a FUTEX_WAIT, even within same process
(void) syscall(__NR_futex, &cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE,
1);
@@ -311,6 +318,7 @@ void ClientProxy::interrupt()
{
audio_track_cblk_t* cblk = mCblk;
if (!(android_atomic_or(CBLK_INTERRUPT, &cblk->mFlags) & CBLK_INTERRUPT)) {
+ android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
(void) syscall(__NR_futex, &cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE,
1);
}
@@ -348,7 +356,13 @@ size_t ClientProxy::getFramesFilled() {
void AudioTrackClientProxy::flush()
{
- mCblk->u.mStreaming.mFlush++;
+ // This works for mFrameCountP2 <= 2^30
+ size_t increment = mFrameCountP2 << 1;
+ size_t mask = increment - 1;
+ audio_track_cblk_t* cblk = mCblk;
+ int32_t newFlush = (cblk->u.mStreaming.mRear & mask) |
+ ((cblk->u.mStreaming.mFlush & ~mask) + increment);
+ android_atomic_release_store(newFlush, &cblk->u.mStreaming.mFlush);
}
bool AudioTrackClientProxy::clearStreamEndDone() {
@@ -491,7 +505,11 @@ void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int
newState.mLoopStart = (uint32_t) loopStart;
newState.mLoopEnd = (uint32_t) loopEnd;
newState.mLoopCount = loopCount;
- mBufferPosition = loopStart;
+ size_t bufferPosition;
+ if (loopCount == 0 || (bufferPosition = getBufferPosition()) >= loopEnd) {
+ bufferPosition = loopStart;
+ }
+ mBufferPosition = bufferPosition; // snapshot buffer position until loop is acknowledged.
(void) mMutator.push(newState);
}
@@ -536,17 +554,27 @@ status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
front = cblk->u.mStreaming.mFront;
if (flush != mFlush) {
- mFlush = flush;
// effectively obtain then release whatever is in the buffer
- android_atomic_release_store(rear, &cblk->u.mStreaming.mFront);
- if (front != rear) {
+ size_t mask = (mFrameCountP2 << 1) - 1;
+ int32_t newFront = (front & ~mask) | (flush & mask);
+ ssize_t filled = rear - newFront;
+ // Rather than shutting down on a corrupt flush, just treat it as a full flush
+ if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
+ ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, filled %d=%#x",
+ mFlush, flush, front, rear, mask, newFront, filled, filled);
+ newFront = rear;
+ }
+ mFlush = flush;
+ android_atomic_release_store(newFront, &cblk->u.mStreaming.mFront);
+ // There is no danger from a false positive, so err on the side of caution
+ if (true /*front != newFront*/) {
int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1);
}
}
- front = rear;
+ front = newFront;
}
} else {
front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
@@ -668,6 +696,7 @@ size_t AudioTrackServerProxy::framesReady()
int32_t flush = cblk->u.mStreaming.mFlush;
if (flush != mFlush) {
+ // FIXME should return an accurate value, but over-estimate is better than under-estimate
return mFrameCount;
}
// the acquire might not be necessary since not doing a subsequent read
@@ -711,7 +740,8 @@ StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cbl
size_t frameCount, size_t frameSize)
: AudioTrackServerProxy(cblk, buffers, frameCount, frameSize),
mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0),
- mEnd(frameCount), mFramesReadyIsCalledByMultipleThreads(false)
+ mFramesReadySafe(frameCount), mFramesReady(frameCount),
+ mFramesReadyIsCalledByMultipleThreads(false)
{
mState.mLoopStart = 0;
mState.mLoopEnd = 0;
@@ -725,20 +755,11 @@ void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads()
size_t StaticAudioTrackServerProxy::framesReady()
{
- // FIXME
- // This is racy if called by normal mixer thread,
- // as we're reading 2 independent variables without a lock.
- // Can't call mObserver.poll(), as we might be called from wrong thread.
- // If looping is enabled, should return a higher number (since includes non-contiguous).
- size_t position = mPosition;
+ // Can't call pollPosition() from multiple threads.
if (!mFramesReadyIsCalledByMultipleThreads) {
- ssize_t positionOrStatus = pollPosition();
- if (positionOrStatus >= 0) {
- position = (size_t) positionOrStatus;
- }
+ (void) pollPosition();
}
- size_t end = mEnd;
- return position < end ? end - position : 0;
+ return mFramesReadySafe;
}
ssize_t StaticAudioTrackServerProxy::pollPosition()
@@ -755,25 +776,37 @@ ssize_t StaticAudioTrackServerProxy::pollPosition()
}
// ignore loopEnd
mPosition = position = loopStart;
- mEnd = mFrameCount;
+ mFramesReady = mFrameCount - mPosition;
mState.mLoopCount = 0;
valid = true;
- } else {
+ } else if (state.mLoopCount >= -1) {
if (loopStart < loopEnd && loopEnd <= mFrameCount &&
loopEnd - loopStart >= MIN_LOOP) {
- if (!(loopStart <= position && position < loopEnd)) {
+ // If the current position is greater than the end of the loop
+ // we "wrap" to the loop start. This might cause an audible pop.
+ if (position >= loopEnd) {
mPosition = position = loopStart;
}
- mEnd = loopEnd;
+ if (state.mLoopCount == -1) {
+ mFramesReady = INT64_MAX;
+ } else {
+ // mFramesReady is 64 bits to handle the effective number of frames
+ // that the static audio track contains, including loops.
+ // TODO: Later consider fixing overflow, but does not seem needed now
+ // as will not overflow if loopStart and loopEnd are Java "ints".
+ mFramesReady = int64_t(state.mLoopCount) * (loopEnd - loopStart)
+ + mFrameCount - mPosition;
+ }
mState = state;
valid = true;
}
}
- if (!valid) {
+ if (!valid || mPosition > mFrameCount) {
ALOGE("%s client pushed an invalid state, shutting down", __func__);
mIsShutdown = true;
return (ssize_t) NO_INIT;
}
+ mFramesReadySafe = clampToSize(mFramesReady);
// This may overflow, but client is not supposed to rely on it
mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
}
@@ -798,9 +831,10 @@ status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush
return (status_t) positionOrStatus;
}
size_t position = (size_t) positionOrStatus;
+ size_t end = mState.mLoopCount != 0 ? mState.mLoopEnd : mFrameCount;
size_t avail;
- if (position < mEnd) {
- avail = mEnd - position;
+ if (position < end) {
+ avail = end - position;
size_t wanted = buffer->mFrameCount;
if (avail < wanted) {
buffer->mFrameCount = avail;
@@ -813,7 +847,10 @@ status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
}
- buffer->mNonContig = 0; // FIXME should be > 0 for looping
+ // As mFramesReady is the total remaining frames in the static audio track,
+ // it is always larger or equal to avail.
+ LOG_ALWAYS_FATAL_IF(mFramesReady < avail);
+ buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail);
mUnreleased = avail;
return NO_ERROR;
}
@@ -821,6 +858,7 @@ status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush
void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
{
size_t stepCount = buffer->mFrameCount;
+ LOG_ALWAYS_FATAL_IF(!(stepCount <= mFramesReady));
LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased));
if (stepCount == 0) {
// prevent accidental re-use of buffer
@@ -837,11 +875,10 @@ void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount);
newPosition = mFrameCount;
} else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
+ newPosition = mState.mLoopStart;
if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) {
- newPosition = mState.mLoopStart;
setFlags = CBLK_LOOP_CYCLE;
} else {
- mEnd = mFrameCount; // this is what allows playback to continue after the loop
setFlags = CBLK_LOOP_FINAL;
}
}
@@ -849,6 +886,10 @@ void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
setFlags |= CBLK_BUFFER_END;
}
mPosition = newPosition;
+ if (mFramesReady != INT64_MAX) {
+ mFramesReady -= stepCount;
+ }
+ mFramesReadySafe = clampToSize(mFramesReady);
cblk->mServer += stepCount;
// This may overflow, but client is not supposed to rely on it
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 4e47c79..4cdabaf 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -41,7 +41,7 @@ enum {
START_OUTPUT,
STOP_OUTPUT,
RELEASE_OUTPUT,
- GET_INPUT,
+ GET_INPUT_FOR_ATTR,
START_INPUT,
STOP_INPUT,
RELEASE_INPUT,
@@ -69,7 +69,8 @@ enum {
GET_OUTPUT_FOR_ATTR,
ACQUIRE_SOUNDTRIGGER_SESSION,
RELEASE_SOUNDTRIGGER_SESSION,
- GET_PHONE_STATE
+ GET_PHONE_STATE,
+ REGISTER_POLICY_MIXES,
};
#define MAX_ITEMS_PER_LIST 1024
@@ -162,21 +163,45 @@ public:
return static_cast <audio_io_handle_t> (reply.readInt32());
}
- virtual audio_io_handle_t getOutputForAttr(
- const audio_attributes_t *attr,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo)
+ virtual status_t getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
if (attr == NULL) {
- ALOGE("Writing NULL audio attributes - shouldn't happen");
- return (audio_io_handle_t) 0;
+ if (stream == NULL) {
+ ALOGE("getOutputForAttr(): NULL audio attributes and stream type");
+ return BAD_VALUE;
+ }
+ if (*stream == AUDIO_STREAM_DEFAULT) {
+ ALOGE("getOutputForAttr unspecified stream type");
+ return BAD_VALUE;
+ }
+ }
+ if (output == NULL) {
+ ALOGE("getOutputForAttr NULL output - shouldn't happen");
+ return BAD_VALUE;
+ }
+ if (attr == NULL) {
+ data.writeInt32(0);
+ } else {
+ data.writeInt32(1);
+ data.write(attr, sizeof(audio_attributes_t));
+ }
+ data.writeInt32(session);
+ if (stream == NULL) {
+ data.writeInt32(0);
+ } else {
+ data.writeInt32(1);
+ data.writeInt32(*stream);
}
- data.write(attr, sizeof(audio_attributes_t));
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channelMask);
@@ -188,62 +213,93 @@ public:
data.writeInt32(1);
data.write(offloadInfo, sizeof(audio_offload_info_t));
}
- remote()->transact(GET_OUTPUT_FOR_ATTR, data, &reply);
- return static_cast <audio_io_handle_t> (reply.readInt32());
+ status_t status = remote()->transact(GET_OUTPUT_FOR_ATTR, data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = (status_t)reply.readInt32();
+ if (status != NO_ERROR) {
+ return status;
+ }
+ *output = (audio_io_handle_t)reply.readInt32();
+ if (stream != NULL) {
+ *stream = (audio_stream_type_t)reply.readInt32();
+ }
+ return status;
}
virtual status_t startOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session)
+ audio_session_t session)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
data.writeInt32((int32_t) stream);
- data.writeInt32(session);
+ data.writeInt32((int32_t)session);
remote()->transact(START_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
virtual status_t stopOutput(audio_io_handle_t output,
audio_stream_type_t stream,
- int session)
+ audio_session_t session)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
data.writeInt32((int32_t) stream);
- data.writeInt32(session);
+ data.writeInt32((int32_t)session);
remote()->transact(STOP_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
- virtual void releaseOutput(audio_io_handle_t output)
+ virtual void releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
+ data.writeInt32((int32_t)stream);
+ data.writeInt32((int32_t)session);
remote()->transact(RELEASE_OUTPUT, data, &reply);
}
- virtual audio_io_handle_t getInput(
- audio_source_t inputSource,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- int audioSession,
- audio_input_flags_t flags)
+ virtual status_t getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32((int32_t) inputSource);
+ if (attr == NULL) {
+ ALOGE("getInputForAttr NULL attr - shouldn't happen");
+ return BAD_VALUE;
+ }
+ if (input == NULL) {
+ ALOGE("getInputForAttr NULL input - shouldn't happen");
+ return BAD_VALUE;
+ }
+ data.write(attr, sizeof(audio_attributes_t));
+ data.writeInt32(session);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channelMask);
- data.writeInt32(audioSession);
data.writeInt32(flags);
- remote()->transact(GET_INPUT, data, &reply);
- return static_cast <audio_io_handle_t> (reply.readInt32());
+ status_t status = remote()->transact(GET_INPUT_FOR_ATTR, data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = reply.readInt32();
+ if (status != NO_ERROR) {
+ return status;
+ }
+ *input = (audio_io_handle_t)reply.readInt32();
+ return NO_ERROR;
}
virtual status_t startInput(audio_io_handle_t input,
@@ -622,6 +678,38 @@ public:
}
return (audio_mode_t)reply.readInt32();
}
+
+ virtual status_t registerPolicyMixes(Vector<AudioMix> mixes, bool registration)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ data.writeInt32(registration ? 1 : 0);
+ size_t size = mixes.size();
+ if (size > MAX_MIXES_PER_POLICY) {
+ size = MAX_MIXES_PER_POLICY;
+ }
+ size_t sizePosition = data.dataPosition();
+ data.writeInt32(size);
+ size_t finalSize = size;
+ for (size_t i = 0; i < size; i++) {
+ size_t position = data.dataPosition();
+ if (mixes[i].writeToParcel(&data) != NO_ERROR) {
+ data.setDataPosition(position);
+ finalSize--;
+ }
+ }
+ if (size != finalSize) {
+ size_t position = data.dataPosition();
+ data.setDataPosition(sizePosition);
+ data.writeInt32(finalSize);
+ data.setDataPosition(position);
+ }
+ status_t status = remote()->transact(REGISTER_POLICY_MIXES, data, &reply);
+ if (status == NO_ERROR) {
+ status = (status_t)reply.readInt32();
+ }
+ return status;
+ }
};
IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
@@ -707,8 +795,17 @@ status_t BnAudioPolicyService::onTransact(
case GET_OUTPUT_FOR_ATTR: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- audio_attributes_t *attr = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
- data.read(attr, sizeof(audio_attributes_t));
+ audio_attributes_t attr;
+ bool hasAttributes = data.readInt32() != 0;
+ if (hasAttributes) {
+ data.read(&attr, sizeof(audio_attributes_t));
+ }
+ audio_session_t session = (audio_session_t)data.readInt32();
+ audio_stream_type_t stream = AUDIO_STREAM_DEFAULT;
+ bool hasStream = data.readInt32() != 0;
+ if (hasStream) {
+ stream = (audio_stream_type_t)data.readInt32();
+ }
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
@@ -719,13 +816,14 @@ status_t BnAudioPolicyService::onTransact(
if (hasOffloadInfo) {
data.read(&offloadInfo, sizeof(audio_offload_info_t));
}
- audio_io_handle_t output = getOutputForAttr(attr,
- samplingRate,
- format,
- channelMask,
- flags,
- hasOffloadInfo ? &offloadInfo : NULL);
- reply->writeInt32(static_cast <int>(output));
+ audio_io_handle_t output = 0;
+ status_t status = getOutputForAttr(hasAttributes ? &attr : NULL,
+ &output, session, &stream,
+ samplingRate, format, channelMask,
+ flags, hasOffloadInfo ? &offloadInfo : NULL);
+ reply->writeInt32(status);
+ reply->writeInt32(output);
+ reply->writeInt32(stream);
return NO_ERROR;
} break;
@@ -734,7 +832,7 @@ status_t BnAudioPolicyService::onTransact(
audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
- int session = data.readInt32();
+ audio_session_t session = (audio_session_t)data.readInt32();
reply->writeInt32(static_cast <uint32_t>(startOutput(output,
stream,
session)));
@@ -746,7 +844,7 @@ status_t BnAudioPolicyService::onTransact(
audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
- int session = data.readInt32();
+ audio_session_t session = (audio_session_t)data.readInt32();
reply->writeInt32(static_cast <uint32_t>(stopOutput(output,
stream,
session)));
@@ -756,25 +854,29 @@ status_t BnAudioPolicyService::onTransact(
case RELEASE_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
- releaseOutput(output);
+ audio_stream_type_t stream = (audio_stream_type_t)data.readInt32();
+ audio_session_t session = (audio_session_t)data.readInt32();
+ releaseOutput(output, stream, session);
return NO_ERROR;
} break;
- case GET_INPUT: {
+ case GET_INPUT_FOR_ATTR: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- audio_source_t inputSource = (audio_source_t) data.readInt32();
+ audio_attributes_t attr;
+ data.read(&attr, sizeof(audio_attributes_t));
+ audio_session_t session = (audio_session_t)data.readInt32();
uint32_t samplingRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
- int audioSession = data.readInt32();
audio_input_flags_t flags = (audio_input_flags_t) data.readInt32();
- audio_io_handle_t input = getInput(inputSource,
- samplingRate,
- format,
- channelMask,
- audioSession,
- flags);
- reply->writeInt32(static_cast <int>(input));
+ audio_io_handle_t input;
+ status_t status = getInputForAttr(&attr, &input, session,
+ samplingRate, format, channelMask,
+ flags);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->writeInt32(input);
+ }
return NO_ERROR;
} break;
@@ -1096,6 +1198,25 @@ status_t BnAudioPolicyService::onTransact(
return NO_ERROR;
} break;
+ case REGISTER_POLICY_MIXES: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ bool registration = data.readInt32() == 1;
+ Vector<AudioMix> mixes;
+ size_t size = (size_t)data.readInt32();
+ if (size > MAX_MIXES_PER_POLICY) {
+ size = MAX_MIXES_PER_POLICY;
+ }
+ for (size_t i = 0; i < size; i++) {
+ AudioMix mix;
+ if (mix.readFromParcel((Parcel*)&data) == NO_ERROR) {
+ mixes.add(mix);
+ }
+ }
+ status_t status = registerPolicyMixes(mixes, registration);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ } break;
+
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libmedia/ICrypto.cpp b/media/libmedia/ICrypto.cpp
index 7bd120e..f7d8bc6 100644
--- a/media/libmedia/ICrypto.cpp
+++ b/media/libmedia/ICrypto.cpp
@@ -33,6 +33,7 @@ enum {
DESTROY_PLUGIN,
REQUIRES_SECURE_COMPONENT,
DECRYPT,
+ NOTIFY_RESOLUTION,
};
struct BpCrypto : public BpInterface<ICrypto> {
@@ -149,6 +150,15 @@ struct BpCrypto : public BpInterface<ICrypto> {
return result;
}
+ virtual void notifyResolution(
+ uint32_t width, uint32_t height) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICrypto::getInterfaceDescriptor());
+ data.writeInt32(width);
+ data.writeInt32(height);
+ remote()->transact(NOTIFY_RESOLUTION, data, &reply);
+ }
+
private:
DISALLOW_EVIL_CONSTRUCTORS(BpCrypto);
};
@@ -290,10 +300,20 @@ status_t BnCrypto::onTransact(
return OK;
}
+ case NOTIFY_RESOLUTION:
+ {
+ CHECK_INTERFACE(ICrypto, data, reply);
+
+ int32_t width = data.readInt32();
+ int32_t height = data.readInt32();
+ notifyResolution(width, height);
+
+ return OK;
+ }
+
default:
return BBinder::onTransact(code, data, reply, flags);
}
}
} // namespace android
-
diff --git a/media/libmedia/IDrm.cpp b/media/libmedia/IDrm.cpp
index 1904839..7e74de9 100644
--- a/media/libmedia/IDrm.cpp
+++ b/media/libmedia/IDrm.cpp
@@ -54,7 +54,9 @@ enum {
SIGN_RSA,
VERIFY,
SET_LISTENER,
- UNPROVISION_DEVICE
+ UNPROVISION_DEVICE,
+ GET_SECURE_STOP,
+ RELEASE_ALL_SECURE_STOPS
};
struct BpDrm : public BpInterface<IDrm> {
@@ -255,6 +257,17 @@ struct BpDrm : public BpInterface<IDrm> {
return reply.readInt32();
}
+ virtual status_t getSecureStop(Vector<uint8_t> const &ssid, Vector<uint8_t> &secureStop) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+ writeVector(data, ssid);
+ remote()->transact(GET_SECURE_STOP, data, &reply);
+
+ readVector(reply, secureStop);
+ return reply.readInt32();
+ }
+
virtual status_t releaseSecureStops(Vector<uint8_t> const &ssRelease) {
Parcel data, reply;
data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
@@ -265,6 +278,15 @@ struct BpDrm : public BpInterface<IDrm> {
return reply.readInt32();
}
+ virtual status_t releaseAllSecureStops() {
+ Parcel data, reply;
+ data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
+
+ remote()->transact(RELEASE_ALL_SECURE_STOPS, data, &reply);
+
+ return reply.readInt32();
+ }
+
virtual status_t getPropertyString(String8 const &name, String8 &value) const {
Parcel data, reply;
data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
@@ -655,6 +677,17 @@ status_t BnDrm::onTransact(
return OK;
}
+ case GET_SECURE_STOP:
+ {
+ CHECK_INTERFACE(IDrm, data, reply);
+ Vector<uint8_t> ssid, secureStop;
+ readVector(data, ssid);
+ status_t result = getSecureStop(ssid, secureStop);
+ writeVector(reply, secureStop);
+ reply->writeInt32(result);
+ return OK;
+ }
+
case RELEASE_SECURE_STOPS:
{
CHECK_INTERFACE(IDrm, data, reply);
@@ -664,6 +697,13 @@ status_t BnDrm::onTransact(
return OK;
}
+ case RELEASE_ALL_SECURE_STOPS:
+ {
+ CHECK_INTERFACE(IDrm, data, reply);
+ reply->writeInt32(releaseAllSecureStops());
+ return OK;
+ }
+
case GET_PROPERTY_STRING:
{
CHECK_INTERFACE(IDrm, data, reply);
@@ -809,4 +849,3 @@ status_t BnDrm::onTransact(
}
} // namespace android
-
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index c146b8d..f91e3e4 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -52,6 +52,13 @@ Visualizer::Visualizer (int32_t priority,
Visualizer::~Visualizer()
{
+ ALOGV("Visualizer::~Visualizer()");
+ if (mCaptureThread != NULL) {
+ mCaptureThread->requestExitAndWait();
+ mCaptureThread.clear();
+ }
+ mCaptureCallBack = NULL;
+ mCaptureFlags = 0;
}
status_t Visualizer::setEnabled(bool enabled)
@@ -102,20 +109,18 @@ status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t
return INVALID_OPERATION;
}
- sp<CaptureThread> t = mCaptureThread;
- if (t != 0) {
- t->mLock.lock();
+ if (mCaptureThread != 0) {
+ mCaptureLock.unlock();
+ mCaptureThread->requestExitAndWait();
+ mCaptureLock.lock();
}
+
mCaptureThread.clear();
mCaptureCallBack = cbk;
mCaptureCbkUser = user;
mCaptureFlags = flags;
mCaptureRate = rate;
- if (t != 0) {
- t->mLock.unlock();
- }
-
if (cbk != NULL) {
mCaptureThread = new CaptureThread(*this, rate, ((flags & CAPTURE_CALL_JAVA) != 0));
}
generated by cgit v1.1 at 2024-05-17 22:38:08 +0000