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-rw-r--r--services/audioflinger/AudioFlinger.cpp24
-rw-r--r--services/audioflinger/FastMixer.cpp2
-rw-r--r--services/audioflinger/Threads.cpp4
-rw-r--r--services/audioflinger/Tracks.cpp2
-rw-r--r--services/audiopolicy/AudioPolicyManager.cpp20
-rw-r--r--services/audiopolicy/AudioPolicyService.cpp36
-rw-r--r--services/camera/libcameraservice/api1/CameraClient.cpp15
-rw-r--r--services/soundtrigger/SoundTriggerHwService.cpp58
8 files changed, 125 insertions, 36 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1f77b2f..1843722 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1941,9 +1941,8 @@ sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t m
TEE_SINK_NEW, // copy input using a new pipe
TEE_SINK_OLD, // copy input using an existing pipe
} kind;
- NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common),
- audio_channel_count_from_in_mask(
- inStream->common.get_channels(&inStream->common)));
+ NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
+ audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
} else if (!Format_isValid(format)) {
@@ -2700,24 +2699,26 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
// if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
if (teeFd >= 0) {
+ // FIXME use libsndfile
char wavHeader[44];
memcpy(wavHeader,
"RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
sizeof(wavHeader));
NBAIO_Format format = teeSource->format();
unsigned channelCount = Format_channelCount(format);
- ALOG_ASSERT(channelCount <= FCC_2);
uint32_t sampleRate = Format_sampleRate(format);
+ size_t frameSize = Format_frameSize(format);
wavHeader[22] = channelCount; // number of channels
wavHeader[24] = sampleRate; // sample rate
wavHeader[25] = sampleRate >> 8;
- wavHeader[32] = channelCount * 2; // block alignment
+ wavHeader[32] = frameSize; // block alignment
+ wavHeader[33] = frameSize >> 8;
write(teeFd, wavHeader, sizeof(wavHeader));
size_t total = 0;
bool firstRead = true;
+#define TEE_SINK_READ 1024 // frames per I/O operation
+ void *buffer = malloc(TEE_SINK_READ * frameSize);
for (;;) {
-#define TEE_SINK_READ 1024
- short buffer[TEE_SINK_READ * FCC_2];
size_t count = TEE_SINK_READ;
ssize_t actual = teeSource->read(buffer, count,
AudioBufferProvider::kInvalidPTS);
@@ -2730,14 +2731,17 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
break;
}
ALOG_ASSERT(actual <= (ssize_t)count);
- write(teeFd, buffer, actual * channelCount * sizeof(short));
+ write(teeFd, buffer, actual * frameSize);
total += actual;
}
+ free(buffer);
lseek(teeFd, (off_t) 4, SEEK_SET);
- uint32_t temp = 44 + total * channelCount * sizeof(short) - 8;
+ uint32_t temp = 44 + total * frameSize - 8;
+ // FIXME not big-endian safe
write(teeFd, &temp, sizeof(temp));
lseek(teeFd, (off_t) 40, SEEK_SET);
- temp = total * channelCount * sizeof(short);
+ temp = total * frameSize;
+ // FIXME not big-endian safe
write(teeFd, &temp, sizeof(temp));
close(teeFd);
if (fd >= 0) {
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 9e15293..2678cbf 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -420,7 +420,7 @@ void FastMixer::onWork()
// if non-NULL, then duplicate write() to this non-blocking sink
NBAIO_Sink* teeSink;
if ((teeSink = current->mTeeSink) != NULL) {
- (void) teeSink->write(mMixerBuffer, frameCount);
+ (void) teeSink->write(buffer, frameCount);
}
// FIXME write() is non-blocking and lock-free for a properly implemented NBAIO sink,
// but this code should be modified to handle both non-blocking and blocking sinks
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index f721d5c..942bff6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -3662,6 +3662,10 @@ track_is_ready: ;
// remove all the tracks that need to be...
removeTracks_l(*tracksToRemove);
+ if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
+ mEffectBufferValid = true;
+ }
+
// sink or mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to the sink or mix buffer
// and track effects will accumulate into it
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index c5ab832..6cbab04 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -182,7 +182,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
#ifdef TEE_SINK
if (mTeeSinkTrackEnabled) {
- NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount);
+ NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat);
if (Format_isValid(pipeFormat)) {
Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
size_t numCounterOffers = 0;
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index 06dd22c..a805923 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -1297,21 +1297,23 @@ audio_io_handle_t AudioPolicyManager::getInput(audio_source_t inputSource,
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
bool isSoundTrigger = false;
+ audio_source_t halInputSource = inputSource;
if (inputSource == AUDIO_SOURCE_HOTWORD) {
ssize_t index = mSoundTriggerSessions.indexOfKey(session);
if (index >= 0) {
input = mSoundTriggerSessions.valueFor(session);
isSoundTrigger = true;
ALOGV("SoundTrigger capture on session %d input %d", session, input);
+ } else {
+ halInputSource = AUDIO_SOURCE_VOICE_RECOGNITION;
}
}
-
status_t status = mpClientInterface->openInput(profile->mModule->mHandle,
&input,
&config,
&device,
String8(""),
- inputSource,
+ halInputSource,
flags);
// only accept input with the exact requested set of parameters
@@ -4317,6 +4319,20 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output,
mpClientInterface->onAudioPatchListUpdate();
}
}
+
+ // inform all input as well
+ for (size_t i = 0; i < mInputs.size(); i++) {
+ const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i);
+ if (!isVirtualInputDevice(inputDescriptor->mDevice)) {
+ AudioParameter inputCmd = AudioParameter();
+ ALOGV("%s: inform input %d of device:%d", __func__,
+ inputDescriptor->mIoHandle, device);
+ inputCmd.addInt(String8(AudioParameter::keyRouting),device);
+ mpClientInterface->setParameters(inputDescriptor->mIoHandle,
+ inputCmd.toString(),
+ delayMs);
+ }
+ }
}
// update stream volumes according to new device
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/AudioPolicyService.cpp
index 7f14960..50bb8c7 100644
--- a/services/audiopolicy/AudioPolicyService.cpp
+++ b/services/audiopolicy/AudioPolicyService.cpp
@@ -765,7 +765,16 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(sp<AudioCommand>& c
sp<AudioCommand> command2 = mAudioCommands[i];
// commands are sorted by increasing time stamp: no need to scan the rest of mAudioCommands
if (command2->mTime <= command->mTime) break;
- if (command2->mCommand != command->mCommand) continue;
+
+ // create audio patch or release audio patch commands are equivalent
+ // with regard to filtering
+ if ((command->mCommand == CREATE_AUDIO_PATCH) ||
+ (command->mCommand == RELEASE_AUDIO_PATCH)) {
+ if ((command2->mCommand != CREATE_AUDIO_PATCH) &&
+ (command2->mCommand != RELEASE_AUDIO_PATCH)) {
+ continue;
+ }
+ } else if (command2->mCommand != command->mCommand) continue;
switch (command->mCommand) {
case SET_PARAMETERS: {
@@ -817,6 +826,31 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(sp<AudioCommand>& c
// command status as the command is now delayed
delayMs = 1;
} break;
+
+ case CREATE_AUDIO_PATCH:
+ case RELEASE_AUDIO_PATCH: {
+ audio_patch_handle_t handle;
+ if (command->mCommand == CREATE_AUDIO_PATCH) {
+ handle = ((CreateAudioPatchData *)command->mParam.get())->mHandle;
+ } else {
+ handle = ((ReleaseAudioPatchData *)command->mParam.get())->mHandle;
+ }
+ audio_patch_handle_t handle2;
+ if (command2->mCommand == CREATE_AUDIO_PATCH) {
+ handle2 = ((CreateAudioPatchData *)command2->mParam.get())->mHandle;
+ } else {
+ handle2 = ((ReleaseAudioPatchData *)command2->mParam.get())->mHandle;
+ }
+ if (handle != handle2) break;
+ ALOGV("Filtering out %s audio patch command for handle %d",
+ (command->mCommand == CREATE_AUDIO_PATCH) ? "create" : "release", handle);
+ removedCommands.add(command2);
+ command->mTime = command2->mTime;
+ // force delayMs to non 0 so that code below does not request to wait for
+ // command status as the command is now delayed
+ delayMs = 1;
+ } break;
+
case START_TONE:
case STOP_TONE:
default:
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index abe1235..33bdaa3 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -938,7 +938,20 @@ void CameraClient::copyFrameAndPostCopiedFrame(
}
previewBuffer = mPreviewBuffer;
- memcpy(previewBuffer->base(), (uint8_t *)heap->base() + offset, size);
+ void* previewBufferBase = previewBuffer->base();
+ void* heapBase = heap->base();
+
+ if (heapBase == MAP_FAILED) {
+ ALOGE("%s: Failed to mmap heap for preview frame.", __FUNCTION__);
+ mLock.unlock();
+ return;
+ } else if (previewBufferBase == MAP_FAILED) {
+ ALOGE("%s: Failed to mmap preview buffer for preview frame.", __FUNCTION__);
+ mLock.unlock();
+ return;
+ }
+
+ memcpy(previewBufferBase, (uint8_t *) heapBase + offset, size);
sp<MemoryBase> frame = new MemoryBase(previewBuffer, 0, size);
if (frame == 0) {
diff --git a/services/soundtrigger/SoundTriggerHwService.cpp b/services/soundtrigger/SoundTriggerHwService.cpp
index 2502e0d..b5aaee3 100644
--- a/services/soundtrigger/SoundTriggerHwService.cpp
+++ b/services/soundtrigger/SoundTriggerHwService.cpp
@@ -249,7 +249,7 @@ sp<IMemory> SoundTriggerHwService::prepareRecognitionEvent_l(
event->data_offset = sizeof(struct sound_trigger_recognition_event);
break;
default:
- return eventMemory;
+ return eventMemory;
}
size_t size = event->data_offset + event->data_size;
@@ -653,7 +653,6 @@ void SoundTriggerHwService::Module::onCallbackEvent(const sp<CallbackEvent>& eve
{
ALOGV("onCallbackEvent type %d", event->mType);
- AutoMutex lock(mLock);
sp<IMemory> eventMemory = event->mMemory;
if (eventMemory == 0 || eventMemory->pointer() == NULL) {
@@ -668,34 +667,53 @@ void SoundTriggerHwService::Module::onCallbackEvent(const sp<CallbackEvent>& eve
case CallbackEvent::TYPE_RECOGNITION: {
struct sound_trigger_recognition_event *recognitionEvent =
(struct sound_trigger_recognition_event *)eventMemory->pointer();
+ sp<ISoundTriggerClient> client;
+ {
+ AutoMutex lock(mLock);
+ sp<Model> model = getModel(recognitionEvent->model);
+ if (model == 0) {
+ ALOGW("%s model == 0", __func__);
+ return;
+ }
+ if (model->mState != Model::STATE_ACTIVE) {
+ ALOGV("onCallbackEvent model->mState %d != Model::STATE_ACTIVE", model->mState);
+ return;
+ }
- sp<Model> model = getModel(recognitionEvent->model);
- if (model == 0) {
- ALOGW("%s model == 0", __func__);
- return;
+ recognitionEvent->capture_session = model->mCaptureSession;
+ model->mState = Model::STATE_IDLE;
+ client = mClient;
}
- if (model->mState != Model::STATE_ACTIVE) {
- ALOGV("onCallbackEvent model->mState %d != Model::STATE_ACTIVE", model->mState);
- return;
+ if (client != 0) {
+ client->onRecognitionEvent(eventMemory);
}
-
- recognitionEvent->capture_session = model->mCaptureSession;
- mClient->onRecognitionEvent(eventMemory);
- model->mState = Model::STATE_IDLE;
} break;
case CallbackEvent::TYPE_SOUNDMODEL: {
struct sound_trigger_model_event *soundmodelEvent =
(struct sound_trigger_model_event *)eventMemory->pointer();
-
- sp<Model> model = getModel(soundmodelEvent->model);
- if (model == 0) {
- ALOGW("%s model == 0", __func__);
- return;
+ sp<ISoundTriggerClient> client;
+ {
+ AutoMutex lock(mLock);
+ sp<Model> model = getModel(soundmodelEvent->model);
+ if (model == 0) {
+ ALOGW("%s model == 0", __func__);
+ return;
+ }
+ client = mClient;
+ }
+ if (client != 0) {
+ client->onSoundModelEvent(eventMemory);
}
- mClient->onSoundModelEvent(eventMemory);
} break;
case CallbackEvent::TYPE_SERVICE_STATE: {
- mClient->onServiceStateChange(eventMemory);
+ sp<ISoundTriggerClient> client;
+ {
+ AutoMutex lock(mLock);
+ client = mClient;
+ }
+ if (client != 0) {
+ client->onServiceStateChange(eventMemory);
+ }
} break;
default:
LOG_ALWAYS_FATAL("onCallbackEvent unknown event type %d", event->mType);
generated by cgit v1.1 at 2024-05-29 05:32:52 +0000