Merge "audioflinger: Forward stream parameters to downstream patches"
diff --git a/camera/VendorTagDescriptor.cpp b/camera/VendorTagDescriptor.cpp
index 4c28789..a86cc87 100644
--- a/camera/VendorTagDescriptor.cpp
+++ b/camera/VendorTagDescriptor.cpp
@@ -566,7 +566,7 @@
for (size_t i = 0; i < static_cast<size_t>(tagCount); ++i) {
uint32_t tag = tagArray[i];
- String8 sectionString = tagToSectionMap.valueFor(tag);
+ const String8& sectionString = tagToSectionMap.valueFor(tag);
// Set up tag to section index map
ssize_t index = sections.indexOf(sectionString);
diff --git a/drm/common/Android.bp b/drm/common/Android.bp
index 1552c3f..272684c 100644
--- a/drm/common/Android.bp
+++ b/drm/common/Android.bp
@@ -35,7 +35,7 @@
cflags: ["-Wall", "-Werror"],
- static_libs: ["libbinder"],
+ shared_libs: ["libbinder"],
export_include_dirs: ["include"],
}
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index c072901..cb4bcfc 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -878,31 +878,25 @@
flags, selectedDeviceId, portId);
}
-status_t AudioSystem::startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+status_t AudioSystem::startOutput(audio_port_handle_t portId)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->startOutput(output, stream, session);
+ return aps->startOutput(portId);
}
-status_t AudioSystem::stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+status_t AudioSystem::stopOutput(audio_port_handle_t portId)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->stopOutput(output, stream, session);
+ return aps->stopOutput(portId);
}
-void AudioSystem::releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+void AudioSystem::releaseOutput(audio_port_handle_t portId)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return;
- aps->releaseOutput(output, stream, session);
+ aps->releaseOutput(portId);
}
status_t AudioSystem::getInputForAttr(const audio_attributes_t *attr,
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 316105c..9d77376 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -244,41 +244,29 @@
return status;
}
- virtual status_t startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+ virtual status_t startOutput(audio_port_handle_t portId)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(output);
- data.writeInt32((int32_t) stream);
- data.writeInt32((int32_t) session);
+ data.writeInt32((int32_t)portId);
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,
- audio_session_t session)
+ virtual status_t stopOutput(audio_port_handle_t portId)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(output);
- data.writeInt32((int32_t) stream);
- data.writeInt32((int32_t) session);
+ data.writeInt32((int32_t)portId);
remote()->transact(STOP_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
- virtual void releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+ virtual void releaseOutput(audio_port_handle_t portId)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(output);
- data.writeInt32((int32_t)stream);
- data.writeInt32((int32_t)session);
+ data.writeInt32((int32_t)portId);
remote()->transact(RELEASE_OUTPUT, data, &reply);
}
@@ -1074,34 +1062,22 @@
case START_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- 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());
- audio_session_t session = (audio_session_t)data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(startOutput(output,
- stream,
- session)));
+ const audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+ reply->writeInt32(static_cast <uint32_t>(startOutput(portId)));
return NO_ERROR;
} break;
case STOP_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- 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());
- audio_session_t session = (audio_session_t)data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(stopOutput(output,
- stream,
- session)));
+ const audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+ reply->writeInt32(static_cast <uint32_t>(stopOutput(portId)));
return NO_ERROR;
} break;
case RELEASE_OUTPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
- audio_stream_type_t stream = (audio_stream_type_t)data.readInt32();
- audio_session_t session = (audio_session_t)data.readInt32();
- releaseOutput(output, stream, session);
+ const audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+ releaseOutput(portId);
return NO_ERROR;
} break;
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 4c0f796..10d6e92 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -224,15 +224,9 @@
audio_output_flags_t flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId);
- static status_t startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- static status_t stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- static void releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
+ static status_t startOutput(audio_port_handle_t portId);
+ static status_t stopOutput(audio_port_handle_t portId);
+ static void releaseOutput(audio_port_handle_t portId);
// Client must successfully hand off the handle reference to AudioFlinger via createRecord(),
// or release it with releaseInput().
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index c3876af..6c017a3 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -66,15 +66,9 @@
audio_output_flags_t flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId) = 0;
- virtual status_t startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session) = 0;
- virtual status_t stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session) = 0;
- virtual void releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session) = 0;
+ virtual status_t startOutput(audio_port_handle_t portId) = 0;
+ virtual status_t stopOutput(audio_port_handle_t portId) = 0;
+ virtual void releaseOutput(audio_port_handle_t portId) = 0;
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 7f39d10..3526047 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -5556,6 +5556,11 @@
break;
}
+ case kWhatCheckIfStuck: {
+ ALOGV("No-op by default");
+ break;
+ }
+
default:
return false;
}
@@ -7873,6 +7878,18 @@
break;
}
+ case kWhatCheckIfStuck:
+ {
+ int32_t generation = 0;
+ CHECK(msg->findInt32("generation", &generation));
+ if (generation == mCodec->mStateGeneration) {
+ mCodec->signalError(OMX_ErrorUndefined, TIMED_OUT);
+ }
+
+ handled = true;
+ break;
+ }
+
default:
handled = BaseState::onMessageReceived(msg);
break;
@@ -7884,6 +7901,11 @@
void ACodec::OutputPortSettingsChangedState::stateEntered() {
ALOGV("[%s] Now handling output port settings change",
mCodec->mComponentName.c_str());
+
+ // If we haven't transitioned after 3 seconds, we're probably stuck.
+ sp<AMessage> msg = new AMessage(ACodec::kWhatCheckIfStuck, mCodec);
+ msg->setInt32("generation", mCodec->mStateGeneration);
+ msg->post(3000000);
}
bool ACodec::OutputPortSettingsChangedState::onOMXFrameRendered(
@@ -8146,6 +8168,11 @@
ALOGV("[%s] Now Flushing", mCodec->mComponentName.c_str());
mFlushComplete[kPortIndexInput] = mFlushComplete[kPortIndexOutput] = false;
+
+ // If we haven't transitioned after 3 seconds, we're probably stuck.
+ sp<AMessage> msg = new AMessage(ACodec::kWhatCheckIfStuck, mCodec);
+ msg->setInt32("generation", mCodec->mStateGeneration);
+ msg->post(3000000);
}
bool ACodec::FlushingState::onMessageReceived(const sp<AMessage> &msg) {
@@ -8160,6 +8187,7 @@
msg->setInt32("generation", mCodec->mStateGeneration);
msg->post(3000000);
}
+ handled = true;
break;
}
@@ -8180,6 +8208,18 @@
break;
}
+ case kWhatCheckIfStuck:
+ {
+ int32_t generation = 0;
+ CHECK(msg->findInt32("generation", &generation));
+ if (generation == mCodec->mStateGeneration) {
+ mCodec->signalError(OMX_ErrorUndefined, TIMED_OUT);
+ }
+
+ handled = true;
+ break;
+ }
+
default:
handled = BaseState::onMessageReceived(msg);
break;
diff --git a/media/libstagefright/ACodecBufferChannel.cpp b/media/libstagefright/ACodecBufferChannel.cpp
index 710ae68..266a240 100644
--- a/media/libstagefright/ACodecBufferChannel.cpp
+++ b/media/libstagefright/ACodecBufferChannel.cpp
@@ -129,6 +129,7 @@
secureHandle = static_cast<native_handle_t *>(secureData->getDestinationPointer());
}
ssize_t result = -1;
+ ssize_t codecDataOffset = 0;
if (mCrypto != NULL) {
ICrypto::DestinationBuffer destination;
if (secure) {
@@ -180,9 +181,16 @@
Status status = Status::OK;
hidl_string detailedError;
+ ScramblingControl sctrl = ScramblingControl::UNSCRAMBLED;
+
+ if (key != NULL) {
+ sctrl = (ScramblingControl)key[0];
+ // Adjust for the PES offset
+ codecDataOffset = key[2] | (key[3] << 8);
+ }
auto returnVoid = mDescrambler->descramble(
- key != NULL ? (ScramblingControl)key[0] : ScramblingControl::UNSCRAMBLED,
+ sctrl,
hidlSubSamples,
srcBuffer,
0,
@@ -202,6 +210,11 @@
return UNKNOWN_ERROR;
}
+ if (result < codecDataOffset) {
+ ALOGD("invalid codec data offset: %zd, result %zd", codecDataOffset, result);
+ return BAD_VALUE;
+ }
+
ALOGV("descramble succeeded, %zd bytes", result);
if (dstBuffer.type == BufferType::SHARED_MEMORY) {
@@ -210,7 +223,7 @@
}
}
- it->mCodecBuffer->setRange(0, result);
+ it->mCodecBuffer->setRange(codecDataOffset, result - codecDataOffset);
// Copy metadata from client to codec buffer.
it->mCodecBuffer->meta()->clear();
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 97d15a7..1137cf1 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -137,6 +137,7 @@
kWhatOMXDied = 'OMXd',
kWhatReleaseCodecInstance = 'relC',
kWhatForceStateTransition = 'fstt',
+ kWhatCheckIfStuck = 'Cstk',
};
enum {
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 271d601..cc31815 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -1437,7 +1437,7 @@
// Perform the 1st pass descrambling if needed
if (descrambleBytes > 0) {
memcpy(mDescrambledBuffer->data(), mBuffer->data(), descrambleBytes);
- mDescrambledBuffer->setRange(0, descrambleBytes);
+ mDescrambledBuffer->setRange(0, mBuffer->size());
hidl_vec<SubSample> subSamples;
subSamples.resize(descrambleSubSamples);
@@ -1454,10 +1454,9 @@
}
}
- uint64_t srcOffset = 0, dstOffset = 0;
- // If scrambled at PES-level, PES header should be skipped
+ // If scrambled at PES-level, PES header is in the clear
if (pesScramblingControl != 0) {
- srcOffset = dstOffset = pesOffset;
+ subSamples[0].numBytesOfClearData = pesOffset;
subSamples[0].numBytesOfEncryptedData -= pesOffset;
}
@@ -1473,9 +1472,9 @@
(ScramblingControl) sctrl,
subSamples,
mDescramblerSrcBuffer,
- srcOffset,
+ 0 /*srcOffset*/,
dstBuffer,
- dstOffset,
+ 0 /*dstOffset*/,
[&status, &bytesWritten, &detailedError] (
Status _status, uint32_t _bytesWritten,
const hidl_string& _detailedError) {
@@ -1492,9 +1491,15 @@
ALOGV("[stream %d] descramble succeeded, %d bytes",
mElementaryPID, bytesWritten);
- memcpy(mBuffer->data(), mDescrambledBuffer->data(), descrambleBytes);
+
+ // Set descrambleBytes to the returned result.
+ // Note that this might be smaller than the total length of input data.
+ // (eg. when we're descrambling the PES header portion of a secure stream,
+ // the plugin might cut it off right after the PES header.)
+ descrambleBytes = bytesWritten;
}
+ sp<ABuffer> buffer;
if (mQueue->isScrambled()) {
// Queue subSample info for scrambled queue
sp<ABuffer> clearSizesBuffer = new ABuffer(mSubSamples.size() * 4);
@@ -1506,8 +1511,7 @@
for (auto it = mSubSamples.begin();
it != mSubSamples.end(); it++, i++) {
if ((it->transport_scrambling_mode == 0
- && pesScramblingControl == 0)
- || i < descrambleSubSamples) {
+ && pesScramblingControl == 0)) {
clearSizePtr[i] = it->subSampleSize;
encSizePtr[i] = 0;
} else {
@@ -1516,14 +1520,26 @@
}
isSync |= it->random_access_indicator;
}
+
+ // If scrambled at PES-level, PES header is in the clear
+ if (pesScramblingControl != 0) {
+ clearSizePtr[0] = pesOffset;
+ encSizePtr[0] -= pesOffset;
+ }
// Pass the original TS subsample size now. The PES header adjust
// will be applied when the scrambled AU is dequeued.
mQueue->appendScrambledData(
mBuffer->data(), mBuffer->size(), sctrl,
isSync, clearSizesBuffer, encSizesBuffer);
+
+ buffer = mDescrambledBuffer;
+ } else {
+ memcpy(mBuffer->data(), mDescrambledBuffer->data(), descrambleBytes);
+
+ buffer = mBuffer;
}
- ABitReader br(mBuffer->data(), mBuffer->size());
+ ABitReader br(buffer->data(), buffer->size());
status_t err = parsePES(&br, event);
if (err != OK) {
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index ece0692..9e154a3 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -226,6 +226,7 @@
int32_t cryptoMode;
if (buffer->meta()->findInt32("cryptoMode", &cryptoMode)) {
int32_t cryptoKey;
+ int32_t pesOffset;
sp<ABuffer> clearBytesBuffer, encBytesBuffer;
CHECK(buffer->meta()->findInt32("cryptoKey", &cryptoKey));
@@ -233,6 +234,8 @@
&& clearBytesBuffer != NULL);
CHECK(buffer->meta()->findBuffer("encBytes", &encBytesBuffer)
&& encBytesBuffer != NULL);
+ CHECK(buffer->meta()->findInt32("pesOffset", &pesOffset)
+ && (pesOffset >= 0) && (pesOffset < 65536));
bufmeta.setInt32(kKeyCryptoMode, cryptoMode);
@@ -240,6 +243,11 @@
bufmeta.setData(kKeyCryptoIV, 0, array, 16);
array[0] = (uint8_t) (cryptoKey & 0xff);
+ // array[1] contains PES header flag, which we don't use.
+ // array[2~3] contain the PES offset.
+ array[2] = (uint8_t) (pesOffset & 0xff);
+ array[3] = (uint8_t) ((pesOffset >> 8) & 0xff);
+
bufmeta.setData(kKeyCryptoKey, 0, array, 16);
bufmeta.setData(kKeyPlainSizes, 0,
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index 3deee7e..34d0bcc 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -691,25 +691,9 @@
return NULL;
}
- // skip the PES header, and copy the rest into scrambled access unit
+ // copy into scrambled access unit
sp<ABuffer> scrambledAccessUnit = ABuffer::CreateAsCopy(
- mScrambledBuffer->data() + pesOffset,
- scrambledLength - pesOffset);
-
- // fix up first sample size after skipping the PES header
- if (pesOffset > 0) {
- int32_t &firstClearSize = *(int32_t*)clearSizes->data();
- int32_t &firstEncSize = *(int32_t*)encSizes->data();
- // Cut away the PES header
- if (firstClearSize >= pesOffset) {
- // This is for TS-level scrambling, we descrambled the first
- // (or it was clear to begin with)
- firstClearSize -= pesOffset;
- } else if (firstEncSize >= pesOffset) {
- // This can only be PES-level scrambling
- firstEncSize -= pesOffset;
- }
- }
+ mScrambledBuffer->data(), scrambledLength);
scrambledAccessUnit->meta()->setInt64("timeUs", timeUs);
if (isSync) {
@@ -723,6 +707,7 @@
scrambledAccessUnit->meta()->setInt32("cryptoKey", keyId);
scrambledAccessUnit->meta()->setBuffer("clearBytes", clearSizes);
scrambledAccessUnit->meta()->setBuffer("encBytes", encSizes);
+ scrambledAccessUnit->meta()->setInt32("pesOffset", pesOffset);
memmove(mScrambledBuffer->data(),
mScrambledBuffer->data() + scrambledLength,
diff --git a/media/ndk/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index 6d10f1c..5597488 100644
--- a/media/ndk/NdkMediaDrm.cpp
+++ b/media/ndk/NdkMediaDrm.cpp
@@ -309,6 +309,7 @@
}
String8 defaultUrl;
DrmPlugin::KeyRequestType keyRequestType;
+ mObj->mKeyRequest.clear();
status_t status = mObj->mDrm->getKeyRequest(*iter, mdInit, String8(mimeType),
mdKeyType, mdOptionalParameters, mObj->mKeyRequest, defaultUrl,
&keyRequestType);
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 0beb86c..79501cd 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -306,7 +306,7 @@
*sessionId = actualSessionId;
} else {
if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
- AudioSystem::releaseOutput(io, streamType, actualSessionId);
+ AudioSystem::releaseOutput(portId);
} else {
AudioSystem::releaseInput(portId);
}
@@ -777,7 +777,7 @@
Exit:
if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
- AudioSystem::releaseOutput(output.outputId, streamType, sessionId);
+ AudioSystem::releaseOutput(portId);
}
*status = lStatus;
return trackHandle;
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 7f46747..15a60c2 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -235,9 +235,19 @@
audio_devices_t device = patch->sinks[0].ext.device.type;
String8 address = String8(patch->sinks[0].ext.device.address);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- audio_output_flags_t flags =
- patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
- patch->sinks[0].flags.output : AUDIO_OUTPUT_FLAG_NONE;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
+ if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ config.sample_rate = patch->sinks[0].sample_rate;
+ }
+ if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ config.channel_mask = patch->sinks[0].channel_mask;
+ }
+ if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ config.format = patch->sinks[0].format;
+ }
+ if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS) {
+ flags = patch->sinks[0].flags.output;
+ }
sp<ThreadBase> thread = mAudioFlinger.openOutput_l(
patch->sinks[0].ext.device.hw_module,
&output,
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index f68bfee..e3b83f9 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2324,15 +2324,13 @@
if (track->isExternalTrack()) {
TrackBase::track_state state = track->mState;
mLock.unlock();
- status = AudioSystem::startOutput(mId, track->streamType(),
- track->sessionId());
+ status = AudioSystem::startOutput(track->portId());
mLock.lock();
// abort track was stopped/paused while we released the lock
if (state != track->mState) {
if (status == NO_ERROR) {
mLock.unlock();
- AudioSystem::stopOutput(mId, track->streamType(),
- track->sessionId());
+ AudioSystem::stopOutput(track->portId());
mLock.lock();
}
return INVALID_OPERATION;
@@ -2812,15 +2810,13 @@
for (size_t i = 0 ; i < count ; i++) {
const sp<Track>& track = tracksToRemove.itemAt(i);
if (track->isExternalTrack()) {
- AudioSystem::stopOutput(mId, track->streamType(),
- track->sessionId());
+ AudioSystem::stopOutput(track->portId());
#ifdef ADD_BATTERY_DATA
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
#endif
if (track->isTerminated()) {
- AudioSystem::releaseOutput(mId, track->streamType(),
- track->sessionId());
+ AudioSystem::releaseOutput(track->portId());
}
}
}
@@ -7065,6 +7061,12 @@
goto Exit;
}
+ if (!audio_is_linear_pcm(mFormat) && (*flags & AUDIO_INPUT_FLAG_DIRECT) == 0) {
+ ALOGE("createRecordTrack_l() on an encoded stream requires AUDIO_INPUT_FLAG_DIRECT");
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
if (*pSampleRate == 0) {
*pSampleRate = mSampleRate;
}
@@ -7779,10 +7781,15 @@
{
status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
- mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
- LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_8, "HAL channel count %d > %d", mChannelCount, FCC_8);
mFormat = mHALFormat;
- LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat);
+ mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
+ if (audio_is_linear_pcm(mFormat)) {
+ LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_8, "HAL channel count %d > %d",
+ mChannelCount, FCC_8);
+ } else {
+ // Can have more that FCC_8 channels in encoded streams.
+ ALOGI("HAL format %#x is not linear pcm", mFormat);
+ }
result = mInput->stream->getFrameSize(&mFrameSize);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
result = mInput->stream->getBufferSize(&mBufferSize);
@@ -8100,7 +8107,7 @@
}
// This will decrement references and may cause the destruction of this thread.
if (isOutput()) {
- AudioSystem::releaseOutput(mId, streamType(), mSessionId);
+ AudioSystem::releaseOutput(mPortId);
} else {
AudioSystem::releaseInput(mPortId);
}
@@ -8214,7 +8221,7 @@
bool silenced = false;
if (isOutput()) {
- ret = AudioSystem::startOutput(mId, streamType(), mSessionId);
+ ret = AudioSystem::startOutput(portId);
} else {
ret = AudioSystem::startInput(portId, &silenced);
}
@@ -8226,7 +8233,7 @@
if (mActiveTracks.size() != 0) {
mLock.unlock();
if (isOutput()) {
- AudioSystem::releaseOutput(mId, streamType(), mSessionId);
+ AudioSystem::releaseOutput(portId);
} else {
AudioSystem::releaseInput(portId);
}
@@ -8298,8 +8305,8 @@
mLock.unlock();
if (isOutput()) {
- AudioSystem::stopOutput(mId, streamType(), track->sessionId());
- AudioSystem::releaseOutput(mId, streamType(), track->sessionId());
+ AudioSystem::stopOutput(track->portId());
+ AudioSystem::releaseOutput(track->portId());
} else {
AudioSystem::stopInput(track->portId());
AudioSystem::releaseInput(track->portId());
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 22e610e..8b9485f 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -486,7 +486,7 @@
wasActive = playbackThread->destroyTrack_l(this);
}
if (isExternalTrack() && !wasActive) {
- AudioSystem::releaseOutput(mThreadIoHandle, mStreamType, mSessionId);
+ AudioSystem::releaseOutput(mPortId);
}
}
}
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index d2bc40d..859072b 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -17,12 +17,12 @@
#define LOG_TAG "AudioPolicyIntefaceImpl"
//#define LOG_NDEBUG 0
-#include <utils/Log.h>
-#include <media/MediaAnalyticsItem.h>
-
#include "AudioPolicyService.h"
-#include <mediautils/ServiceUtilities.h>
#include "TypeConverter.h"
+#include <media/AudioPolicyHelper.h>
+#include <media/MediaAnalyticsItem.h>
+#include <mediautils/ServiceUtilities.h>
+#include <utils/Log.h>
namespace android {
@@ -208,93 +208,128 @@
config,
&flags, selectedDeviceId, portId);
}
+
+ if (result == NO_ERROR) {
+ sp <AudioPlaybackClient> client =
+ new AudioPlaybackClient(*attr, *output, uid, pid, session, *selectedDeviceId, *stream);
+ mAudioPlaybackClients.add(*portId, client);
+ }
return result;
}
-status_t AudioPolicyService::startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+status_t AudioPolicyService::startOutput(audio_port_handle_t portId)
{
- if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
- return BAD_VALUE;
- }
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
ALOGV("startOutput()");
+ sp<AudioPlaybackClient> client;
sp<AudioPolicyEffects>audioPolicyEffects;
{
Mutex::Autolock _l(mLock);
+ const ssize_t index = mAudioPlaybackClients.indexOfKey(portId);
+ if (index < 0) {
+ ALOGE("%s AudioTrack client not found for portId %d", __FUNCTION__, portId);
+ return INVALID_OPERATION;
+ }
+ client = mAudioPlaybackClients.valueAt(index);
audioPolicyEffects = mAudioPolicyEffects;
}
if (audioPolicyEffects != 0) {
// create audio processors according to stream
- status_t status = audioPolicyEffects->addOutputSessionEffects(output, stream, session);
+ status_t status = audioPolicyEffects->addOutputSessionEffects(
+ client->io, client->stream, client->session);
if (status != NO_ERROR && status != ALREADY_EXISTS) {
- ALOGW("Failed to add effects on session %d", session);
+ ALOGW("Failed to add effects on session %d", client->session);
}
}
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
- return mAudioPolicyManager->startOutput(output, stream, session);
+ status_t status = mAudioPolicyManager->startOutput(
+ client->io, client->stream, client->session);
+ if (status == NO_ERROR) {
+ client->active = true;
+ }
+ return status;
}
-status_t AudioPolicyService::stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+status_t AudioPolicyService::stopOutput(audio_port_handle_t portId)
{
- if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
- return BAD_VALUE;
+ {
+ Mutex::Autolock _l(mLock);
+
+ const ssize_t index = mAudioPlaybackClients.indexOfKey(portId);
+ if (index < 0) {
+ ALOGE("%s AudioTrack client not found for portId %d", __FUNCTION__, portId);
+ return INVALID_OPERATION;
+ }
}
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
ALOGV("stopOutput()");
- mOutputCommandThread->stopOutputCommand(output, stream, session);
+ mOutputCommandThread->stopOutputCommand(portId);
return NO_ERROR;
}
-status_t AudioPolicyService::doStopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+status_t AudioPolicyService::doStopOutput(audio_port_handle_t portId)
{
- ALOGV("doStopOutput from tid %d", gettid());
+ ALOGV("doStopOutput");
+ sp<AudioPlaybackClient> client;
sp<AudioPolicyEffects>audioPolicyEffects;
{
Mutex::Autolock _l(mLock);
+
+ const ssize_t index = mAudioPlaybackClients.indexOfKey(portId);
+ if (index < 0) {
+ ALOGE("%s AudioTrack client not found for portId %d", __FUNCTION__, portId);
+ return INVALID_OPERATION;
+ }
+ client = mAudioPlaybackClients.valueAt(index);
audioPolicyEffects = mAudioPolicyEffects;
}
if (audioPolicyEffects != 0) {
// release audio processors from the stream
- status_t status = audioPolicyEffects->releaseOutputSessionEffects(output, stream, session);
+ status_t status = audioPolicyEffects->releaseOutputSessionEffects(
+ client->io, client->stream, client->session);
if (status != NO_ERROR && status != ALREADY_EXISTS) {
- ALOGW("Failed to release effects on session %d", session);
+ ALOGW("Failed to release effects on session %d", client->session);
}
}
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
- return mAudioPolicyManager->stopOutput(output, stream, session);
+ status_t status = mAudioPolicyManager->stopOutput(
+ client->io, client->stream, client->session);
+ if (status == NO_ERROR) {
+ client->active = false;
+ }
+ return status;
}
-void AudioPolicyService::releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+void AudioPolicyService::releaseOutput(audio_port_handle_t portId)
{
if (mAudioPolicyManager == NULL) {
return;
}
ALOGV("releaseOutput()");
- mOutputCommandThread->releaseOutputCommand(output, stream, session);
+ mOutputCommandThread->releaseOutputCommand(portId);
}
-void AudioPolicyService::doReleaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+void AudioPolicyService::doReleaseOutput(audio_port_handle_t portId)
{
ALOGV("doReleaseOutput from tid %d", gettid());
Mutex::Autolock _l(mLock);
+ const ssize_t index = mAudioPlaybackClients.indexOfKey(portId);
+ if (index < 0) {
+ ALOGE("%s AudioTrack client not found for portId %d", __FUNCTION__, portId);
+ return;
+ }
+ sp<AudioPlaybackClient> client = mAudioPlaybackClients.valueAt(index);
+ mAudioRecordClients.removeItem(portId);
+
// called from internal thread: no need to clear caller identity
- mAudioPolicyManager->releaseOutput(output, stream, session);
+ mAudioPolicyManager->releaseOutput(
+ client->io, client->stream, client->session);
}
status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr,
@@ -403,12 +438,8 @@
return status;
}
- sp<AudioRecordClient> client =
- new AudioRecordClient(*attr, *input, uid, pid, opPackageName, session);
- client->active = false;
- client->isConcurrent = false;
- client->isVirtualDevice = false; //TODO : update from APM->getInputForAttr()
- client->deviceId = *selectedDeviceId;
+ sp<AudioRecordClient> client = new AudioRecordClient(*attr, *input, uid, pid, session,
+ *selectedDeviceId, opPackageName);
mAudioRecordClients.add(*portId, client);
}
@@ -497,7 +528,7 @@
{
AutoCallerClear acc;
status = mAudioPolicyManager->startInput(
- client->input, client->session, *silenced, &concurrency);
+ client->io, client->session, *silenced, &concurrency);
}
@@ -610,7 +641,7 @@
// finish the recording app op
finishRecording(client->opPackageName, client->uid);
AutoCallerClear acc;
- return mAudioPolicyManager->stopInput(client->input, client->session);
+ return mAudioPolicyManager->stopInput(client->io, client->session);
}
void AudioPolicyService::releaseInput(audio_port_handle_t portId)
@@ -635,15 +666,15 @@
}
if (audioPolicyEffects != 0) {
// release audio processors from the input
- status_t status = audioPolicyEffects->releaseInputEffects(client->input, client->session);
+ status_t status = audioPolicyEffects->releaseInputEffects(client->io, client->session);
if(status != NO_ERROR) {
- ALOGW("Failed to release effects on input %d", client->input);
+ ALOGW("Failed to release effects on input %d", client->io);
}
}
{
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
- mAudioPolicyManager->releaseInput(client->input, client->session);
+ mAudioPolicyManager->releaseInput(client->io, client->session);
}
}
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index ca3b6b6..8bca221 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -695,26 +695,26 @@
}break;
case STOP_OUTPUT: {
StopOutputData *data = (StopOutputData *)command->mParam.get();
- ALOGV("AudioCommandThread() processing stop output %d",
- data->mIO);
+ ALOGV("AudioCommandThread() processing stop output portId %d",
+ data->mPortId);
svc = mService.promote();
if (svc == 0) {
break;
}
mLock.unlock();
- svc->doStopOutput(data->mIO, data->mStream, data->mSession);
+ svc->doStopOutput(data->mPortId);
mLock.lock();
}break;
case RELEASE_OUTPUT: {
ReleaseOutputData *data = (ReleaseOutputData *)command->mParam.get();
- ALOGV("AudioCommandThread() processing release output %d",
- data->mIO);
+ ALOGV("AudioCommandThread() processing release output portId %d",
+ data->mPortId);
svc = mService.promote();
if (svc == 0) {
break;
}
mLock.unlock();
- svc->doReleaseOutput(data->mIO, data->mStream, data->mSession);
+ svc->doReleaseOutput(data->mPortId);
mLock.lock();
}break;
case CREATE_AUDIO_PATCH: {
@@ -925,33 +925,25 @@
return sendCommand(command, delayMs);
}
-void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_port_handle_t portId)
{
sp<AudioCommand> command = new AudioCommand();
command->mCommand = STOP_OUTPUT;
sp<StopOutputData> data = new StopOutputData();
- data->mIO = output;
- data->mStream = stream;
- data->mSession = session;
+ data->mPortId = portId;
command->mParam = data;
- ALOGV("AudioCommandThread() adding stop output %d", output);
+ ALOGV("AudioCommandThread() adding stop output portId %d", portId);
sendCommand(command);
}
-void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session)
+void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_port_handle_t portId)
{
sp<AudioCommand> command = new AudioCommand();
command->mCommand = RELEASE_OUTPUT;
sp<ReleaseOutputData> data = new ReleaseOutputData();
- data->mIO = output;
- data->mStream = stream;
- data->mSession = session;
+ data->mPortId = portId;
command->mParam = data;
- ALOGV("AudioCommandThread() adding release output %d", output);
+ ALOGV("AudioCommandThread() adding release output portId %d", portId);
sendCommand(command);
}
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index a1366bb..d41069e 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -81,15 +81,9 @@
audio_output_flags_t flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId);
- virtual status_t startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual status_t stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual void releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
+ virtual status_t startOutput(audio_port_handle_t portId);
+ virtual status_t stopOutput(audio_port_handle_t portId);
+ virtual void releaseOutput(audio_port_handle_t portId);
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_session_t session,
@@ -205,12 +199,8 @@
bool reported);
virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled);
- status_t doStopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- void doReleaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
+ status_t doStopOutput(audio_port_handle_t portId);
+ void doReleaseOutput(audio_port_handle_t portId);
status_t clientCreateAudioPatch(const struct audio_patch *patch,
audio_patch_handle_t *handle,
@@ -340,12 +330,8 @@
status_t parametersCommand(audio_io_handle_t ioHandle,
const char *keyValuePairs, int delayMs = 0);
status_t voiceVolumeCommand(float volume, int delayMs = 0);
- void stopOutputCommand(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- void releaseOutputCommand(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
+ void stopOutputCommand(audio_port_handle_t portId);
+ void releaseOutputCommand(audio_port_handle_t portId);
status_t sendCommand(sp<AudioCommand>& command, int delayMs = 0);
void insertCommand_l(sp<AudioCommand>& command, int delayMs = 0);
status_t createAudioPatchCommand(const struct audio_patch *patch,
@@ -413,16 +399,12 @@
class StopOutputData : public AudioCommandData {
public:
- audio_io_handle_t mIO;
- audio_stream_type_t mStream;
- audio_session_t mSession;
+ audio_port_handle_t mPortId;
};
class ReleaseOutputData : public AudioCommandData {
public:
- audio_io_handle_t mIO;
- audio_stream_type_t mStream;
- audio_session_t mSession;
+ audio_port_handle_t mPortId;
};
class CreateAudioPatchData : public AudioCommandData {
@@ -603,30 +585,56 @@
bool mAudioPortCallbacksEnabled;
};
+ class AudioClient : public virtual RefBase {
+ public:
+ AudioClient(const audio_attributes_t attributes,
+ const audio_io_handle_t io, uid_t uid, pid_t pid,
+ const audio_session_t session, const audio_port_handle_t deviceId) :
+ attributes(attributes), io(io), uid(uid), pid(pid),
+ session(session), deviceId(deviceId), active(false) {}
+ ~AudioClient() override = default;
+
+
+ const audio_attributes_t attributes; // source, flags ...
+ const audio_io_handle_t io; // audio HAL stream IO handle
+ const uid_t uid; // client UID
+ const pid_t pid; // client PID
+ const audio_session_t session; // audio session ID
+ const audio_port_handle_t deviceId; // selected input device port ID
+ bool active; // Playback/Capture is active or inactive
+ };
+
// --- AudioRecordClient ---
// Information about each registered AudioRecord client
// (between calls to getInputForAttr() and releaseInput())
- class AudioRecordClient : public RefBase {
+ class AudioRecordClient : public AudioClient {
public:
AudioRecordClient(const audio_attributes_t attributes,
- const audio_io_handle_t input, uid_t uid, pid_t pid,
- const String16& opPackageName, const audio_session_t session) :
- attributes(attributes),
- input(input), uid(uid), pid(pid),
- opPackageName(opPackageName), session(session),
- active(false), isConcurrent(false), isVirtualDevice(false) {}
- virtual ~AudioRecordClient() {}
+ const audio_io_handle_t io, uid_t uid, pid_t pid,
+ const audio_session_t session, const audio_port_handle_t deviceId,
+ const String16& opPackageName) :
+ AudioClient(attributes, io, uid, pid, session, deviceId),
+ opPackageName(opPackageName), isConcurrent(false), isVirtualDevice(false) {}
+ ~AudioRecordClient() override = default;
- const audio_attributes_t attributes; // source, flags ...
- const audio_io_handle_t input; // audio HAL input IO handle
- const uid_t uid; // client UID
- const pid_t pid; // client PID
const String16 opPackageName; // client package name
- const audio_session_t session; // audio session ID
- bool active; // Capture is active or inactive
bool isConcurrent; // is allowed to concurrent capture
bool isVirtualDevice; // uses virtual device: updated by APM::getInputForAttr()
- audio_port_handle_t deviceId; // selected input device port ID
+ };
+
+ // --- AudioPlaybackClient ---
+ // Information about each registered AudioTrack client
+ // (between calls to getOutputForAttr() and releaseOutput())
+ class AudioPlaybackClient : public AudioClient {
+ public:
+ AudioPlaybackClient(const audio_attributes_t attributes,
+ const audio_io_handle_t io, uid_t uid, pid_t pid,
+ const audio_session_t session, audio_port_handle_t deviceId,
+ audio_stream_type_t stream) :
+ AudioClient(attributes, io, uid, pid, session, deviceId), stream(stream) {}
+ ~AudioPlaybackClient() override = default;
+
+ const audio_stream_type_t stream;
};
// A class automatically clearing and restoring binder caller identity inside
@@ -670,6 +678,7 @@
sp<UidPolicy> mUidPolicy;
DefaultKeyedVector< audio_port_handle_t, sp<AudioRecordClient> > mAudioRecordClients;
+ DefaultKeyedVector< audio_port_handle_t, sp<AudioPlaybackClient> > mAudioPlaybackClients;
};
} // namespace android
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index b3c3717..2c020a2 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -564,7 +564,7 @@
// Only transition to STATE_ABANDONED from STATE_CONFIGURED. (If it is STATE_PREPARING,
// let prepareNextBuffer handle the error.)
- if (res == NO_INIT && mState == STATE_CONFIGURED) {
+ if ((res == NO_INIT || res == DEAD_OBJECT) && mState == STATE_CONFIGURED) {
mState = STATE_ABANDONED;
}